- Alsa-devel - mailman.alsa-project.org

[alsa-devel] [ANNOUNCE] ALSA scenario manager library
by Liam Girdwood 11 Jun '08

11 Jun '08

I've now released the first version (v0.1) of an ALSA scenario (or use case) manager. It's currently a stand alone library that can either be linked against alsa-lib or salsa (not tested - afaik I've not used any alsa-lib only symbols). This library is designed to provide userspace applications (e.g. Pulse Audio) with a generic API to change the audio *hardware* scenario or use case. Features:- * scenario get() and set() * scenario list() * scenario dump() (dumps all sound card control settings) * mixer aliasing (master hardware volumes and switches can be found in each scenario - they may be different kcontrols) API header is here :- http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=alsa-scenario;a=blo… The library saves each scenario in a simple file i.e. 1:'Master Playback Switch':1:1 2:'Master Playback Volume':2:0,0 3:'Headphone Playback Switch':1:1 4:'Headphone Playback Volume':2:17,17 This differs from the format used by alsactl in order to support salsa-lib. There is a utility called scndump (in the examples directory) to dump the sound cards current state in this format. Each sound card also needs a master file describing supported scenario information i.e. # Test scenario configuration for Machine X # Author: Joe Bloggs <joe(a)bloggs.com> # lines beginning with # are comments Section "Scenario" # scenario unique name (mandatory) Identifier = "playback speaker" # scenario file name (mandatory) File = "playback_spk" # scenario audio quality of service (optional) - HIFI, VOICE or SYSTEM QoS = HIFI # alias for master playback volume (optional) MasterPlaybackVolume = 8 # alias for master playback switch (optional) MasterPlaybackSwitch = 2 # alias for master capture volume (optional) MasterCaptureVolume = 3 # alias for master capture switch (optional) MasterCaptureSwitch = 4 EndSection Section "Scenario" # scenario unique name (mandatory) Identifier = "playback headphones" # scenario file name (mandatory) File = "playback_hp" # scenario audio quality of service (optional) - HIFI, VOICE or SYSTEM QoS = HIFI # alias for master playback volume (optional) MasterPlaybackVolume = 10 # alias for master playback switch (optional) MasterPlaybackSwitch = 20 # alias for master capture volume (optional) MasterCaptureVolume = 30 # alias for master capture switch (optional) MasterCaptureSwitch = 40 EndSection This file provides information to the library about each supported scenario. The library needs both a master file and scenario kcontrol files in order to change between scenarios. There is still some work todo (i.e QoS, locking, auto* cleanup, sequencer, etc) and it obviously needs more testing and cleanup. I'm not really in the position to continue working on this in the near future. Ideally, it would be good to have another developer(s) with spare time involved. I can provide git access. Fwiw, I'd ideally like to merge this into alsa-lib and salsa when things are mature. It useful code for embedded devices and laptops (maybe PC's too). Source tarball is here :- http://opensource.wolfsonmicro.com/~lg/alsa-scenario/scenario-lib-0.1.0.tar… Git here :- http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=alsa-scenario;a=sum… More info here :- http://opensource.wolfsonmicro.com/node/14 Feedback and patches welcome. Liam

2 1

[alsa-devel] period time in async handler
by Tim Pushor 11 Jun '08

11 Jun '08

Hi Friends, I have a need to have an audio player kick off things at various points of time through playback of an audio file. I'm trying to hack aplay as a proof of concept. If I understand correctly, the asynchronous handler seems to do what I want. I hacked in a handler to aplay and it works, but not quite like how I expected. I counted the number of times that the handler got called, and thought that if I multiplied that by the period time that I'd end up with the total audio length but that doesn't seem to be the case. I was under the assumption that the handler was called once per period. I must not understand the relationship of the asynchronous handler and period_time. Does this make sense? Am I going about this all wrong? Thanks, Tim

2 1

[alsa-devel] [PATCH 01/10] ASoC: Fix register cache sizes for Wolfson codecs
by Mark Brown 11 Jun '08

11 Jun '08

The register cache size is used by the codec_reg sysfs file which works in terms of the register cache access functions rather than in terms of raw access to the cache so the size specified needs to be in terms of the number of elements. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 5acf43a..7788053 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -534,7 +534,7 @@ static int wm8731_init(struct snd_soc_device *socdev) codec->set_bias_level = wm8731_set_bias_level; codec->dai = &wm8731_dai; codec->num_dai = 1; - codec->reg_cache_size = sizeof(wm8731_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8731_reg); codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 1f11ad2..1ae670a 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -798,7 +798,7 @@ static int wm8750_init(struct snd_soc_device *socdev) codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; - codec->reg_cache_size = sizeof(wm8750_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8750_reg); codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index c32e632..285c5ea 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1557,7 +1557,7 @@ static int wm8753_init(struct snd_soc_device *socdev) codec->set_bias_level = wm8753_set_bias_level; codec->dai = wm8753_dai; codec->num_dai = 2; - codec->reg_cache_size = sizeof(wm8753_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8753_reg); codec->reg_cache = kmemdup(wm8753_reg, sizeof(wm8753_reg), GFP_KERNEL); if (codec->reg_cache == NULL) -- 1.5.5.3

3 11

[alsa-devel] GIT tree usage.
by Rene Herman 11 Jun '08

11 Jun '08

Good day. I have been using Takashi's "sound-2.6" GIT repo: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for pulling current ALSA (branch master) into the stock kernel tree and while rebuilding stuff thought I'd try switching to the official ALSA GIT repo(s). I'd like to make sure that I am now using the right trees and branches... I start out locally with a fresh linus clone: $ git clone git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6 local and then want to pull latest ALSA into it: $ git remote add alsa git://git.alsa-project.org/alsa-kernel.git $ git remote show alsa * remote alsa URL: git://git.alsa-project.org/alsa-kernel.git Tracked remote branches for-linus linux-2.6 master $ git checkout -b current $ git merge alsa/master Am I using the right tree and branch for this? And is this branch being rebased? Rene.

2 2

[alsa-devel] [PATCH 2/2] opti93x: use cs4231 lib
by Krzysztof Helt 11 Jun '08

11 Jun '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch converts the Opti93x driver to use the cs4231 library instead of duplicating the code. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch requires the patch which adds opti93x support to the cs4231 library. It saves few hundreds LOCs. This patch was sent some time ago. It should work now on Opti93x. The patch was prepared against the 2.6.26-rc5 kernel. Tested on Opti 930, 931 and 933. The previous repost: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-January/005680.ht… diff -urp linux-rc2/sound/isa/Kconfig linux-rc5/sound/isa/Kconfig --- linux-rc2/sound/isa/Kconfig 2008-04-17 04:49:44.000000000 +0200 +++ linux-rc5/sound/isa/Kconfig 2008-06-08 00:43:27.704186772 +0200 @@ -321,7 +321,7 @@ config SND_OPTI93X depends on SND select SND_OPL3_LIB select SND_MPU401_UART - select SND_PCM + select SND_CS4231_LIB help Say Y here to include support for soundcards based on Opti 82C93x chips. diff -urp linux-rc2/sound/isa/opti9xx/opti92x-ad1848.c linux-rc5/sound/isa/opti9xx/opti92x-ad1848.c --- linux-rc2/sound/isa/opti9xx/opti92x-ad1848.c 2008-06-08 01:08:48.186548901 +0200 +++ linux-rc5/sound/isa/opti9xx/opti92x-ad1848.c 2008-06-08 10:42:36.915336562 +0200 @@ -33,15 +33,10 @@ #include <asm/io.h> #include <asm/dma.h> #include <sound/core.h> -#ifdef CS4231 +#if defined(CS4231) || defined(OPTi93X) #include <sound/cs4231.h> #else -#ifndef OPTi93X #include <sound/ad1848.h> -#else -#include <sound/control.h> -#include <sound/pcm.h> -#endif /* OPTi93X */ #endif /* CS4231 */ #include <sound/mpu401.h> #include <sound/opl3.h> @@ -109,7 +104,6 @@ module_param(dma2, int, 0444); MODULE_PARM_DESC(dma2, "2nd dma # for opti9xx driver."); #endif /* CS4231 || OPTi93X */ -#define OPTi9XX_HW_DETECT 0 #define OPTi9XX_HW_82C928 1 #define OPTi9XX_HW_82C929 2 #define OPTi9XX_HW_82C924 3 @@ -123,105 +117,12 @@ MODULE_PARM_DESC(dma2, "2nd dma # for op #ifdef OPTi93X -#define OPTi93X_INDEX 0x00 -#define OPTi93X_DATA 0x01 #define OPTi93X_STATUS 0x02 -#define OPTi93X_DDATA 0x03 #define OPTi93X_PORT(chip, r) ((chip)->port + OPTi93X_##r) -#define OPTi93X_MIXOUT_LEFT 0x00 -#define OPTi93X_MIXOUT_RIGHT 0x01 -#define OPTi93X_CD_LEFT_INPUT 0x02 -#define OPTi93X_CD_RIGHT_INPUT 0x03 -#define OPTi930_AUX_LEFT_INPUT 0x04 -#define OPTi930_AUX_RIGHT_INPUT 0x05 -#define OPTi931_FM_LEFT_INPUT 0x04 -#define OPTi931_FM_RIGHT_INPUT 0x05 -#define OPTi93X_DAC_LEFT 0x06 -#define OPTi93X_DAC_RIGHT 0x07 -#define OPTi93X_PLAY_FORMAT 0x08 -#define OPTi93X_IFACE_CONF 0x09 -#define OPTi93X_PIN_CTRL 0x0a -#define OPTi93X_ERR_INIT 0x0b -#define OPTi93X_ID 0x0c -#define OPTi93X_PLAY_UPR_CNT 0x0e -#define OPTi93X_PLAY_LWR_CNT 0x0f -#define OPTi931_AUX_LEFT_INPUT 0x10 -#define OPTi931_AUX_RIGHT_INPUT 0x11 -#define OPTi93X_LINE_LEFT_INPUT 0x12 -#define OPTi93X_LINE_RIGHT_INPUT 0x13 -#define OPTi93X_MIC_LEFT_INPUT 0x14 -#define OPTi93X_MIC_RIGHT_INPUT 0x15 -#define OPTi93X_OUT_LEFT 0x16 -#define OPTi93X_OUT_RIGHT 0x17 -#define OPTi93X_CAPT_FORMAT 0x1c -#define OPTi93X_CAPT_UPR_CNT 0x1e -#define OPTi93X_CAPT_LWR_CNT 0x1f - -#define OPTi93X_TRD 0x20 -#define OPTi93X_MCE 0x40 -#define OPTi93X_INIT 0x80 - -#define OPTi93X_MIXOUT_MIC_GAIN 0x20 -#define OPTi93X_MIXOUT_LINE 0x00 -#define OPTi93X_MIXOUT_CD 0x40 -#define OPTi93X_MIXOUT_MIC 0x80 -#define OPTi93X_MIXOUT_MIXER 0xc0 - -#define OPTi93X_STEREO 0x10 -#define OPTi93X_LINEAR_8 0x00 -#define OPTi93X_ULAW_8 0x20 -#define OPTi93X_LINEAR_16_LIT 0x40 -#define OPTi93X_ALAW_8 0x60 -#define OPTi93X_ADPCM_16 0xa0 -#define OPTi93X_LINEAR_16_BIG 0xc0 - -#define OPTi93X_CAPTURE_PIO 0x80 -#define OPTi93X_PLAYBACK_PIO 0x40 -#define OPTi93X_AUTOCALIB 0x08 -#define OPTi93X_SINGLE_DMA 0x04 -#define OPTi93X_CAPTURE_ENABLE 0x02 -#define OPTi93X_PLAYBACK_ENABLE 0x01 - -#define OPTi93X_IRQ_ENABLE 0x02 - -#define OPTi93X_DMA_REQUEST 0x10 -#define OPTi93X_CALIB_IN_PROGRESS 0x20 - #define OPTi93X_IRQ_PLAYBACK 0x04 #define OPTi93X_IRQ_CAPTURE 0x08 - -struct snd_opti93x { - unsigned long port; - struct resource *res_port; - int irq; - int dma1; - int dma2; - - struct snd_opti9xx *chip; - unsigned short hardware; - unsigned char image[32]; - - unsigned char mce_bit; - unsigned short mode; - int mute; - - spinlock_t lock; - - struct snd_card *card; - struct snd_pcm *pcm; - struct snd_pcm_substream *playback_substream; - struct snd_pcm_substream *capture_substream; - unsigned int p_dma_size; - unsigned int c_dma_size; -}; - -#define OPTi93X_MODE_NONE 0x00 -#define OPTi93X_MODE_PLAY 0x01 -#define OPTi93X_MODE_CAPTURE 0x02 -#define OPTi93X_MODE_OPEN (OPTi93X_MODE_PLAY | OPTi93X_MODE_CAPTURE) - #endif /* OPTi93X */ struct snd_opti9xx { @@ -234,6 +135,7 @@ struct snd_opti9xx { unsigned long mc_base_size; #ifdef OPTi93X unsigned long mc_indir_index; + struct snd_cs4231 *codec; #endif /* OPTi93X */ unsigned long pwd_reg; @@ -654,979 +556,23 @@ __skip_mpu: #ifdef OPTi93X -static unsigned char snd_opti93x_default_image[32] = -{ - 0x00, /* 00/00 - l_mixout_outctrl */ - 0x00, /* 01/01 - r_mixout_outctrl */ - 0x88, /* 02/02 - l_cd_inctrl */ - 0x88, /* 03/03 - r_cd_inctrl */ - 0x88, /* 04/04 - l_a1/fm_inctrl */ - 0x88, /* 05/05 - r_a1/fm_inctrl */ - 0x80, /* 06/06 - l_dac_inctrl */ - 0x80, /* 07/07 - r_dac_inctrl */ - 0x00, /* 08/08 - ply_dataform_reg */ - 0x00, /* 09/09 - if_conf */ - 0x00, /* 0a/10 - pin_ctrl */ - 0x00, /* 0b/11 - err_init_reg */ - 0x0a, /* 0c/12 - id_reg */ - 0x00, /* 0d/13 - reserved */ - 0x00, /* 0e/14 - ply_upcount_reg */ - 0x00, /* 0f/15 - ply_lowcount_reg */ - 0x88, /* 10/16 - reserved/l_a1_inctrl */ - 0x88, /* 11/17 - reserved/r_a1_inctrl */ - 0x88, /* 12/18 - l_line_inctrl */ - 0x88, /* 13/19 - r_line_inctrl */ - 0x88, /* 14/20 - l_mic_inctrl */ - 0x88, /* 15/21 - r_mic_inctrl */ - 0x80, /* 16/22 - l_out_outctrl */ - 0x80, /* 17/23 - r_out_outctrl */ - 0x00, /* 18/24 - reserved */ - 0x00, /* 19/25 - reserved */ - 0x00, /* 1a/26 - reserved */ - 0x00, /* 1b/27 - reserved */ - 0x00, /* 1c/28 - cap_dataform_reg */ - 0x00, /* 1d/29 - reserved */ - 0x00, /* 1e/30 - cap_upcount_reg */ - 0x00 /* 1f/31 - cap_lowcount_reg */ -}; - - -static int snd_opti93x_busy_wait(struct snd_opti93x *chip) -{ - int timeout; - - for (timeout = 250; timeout-- > 0; udelay(10)) - if (!(inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_INIT)) - return 0; - - snd_printk("chip still busy.\n"); - return -EBUSY; -} - -static unsigned char snd_opti93x_in(struct snd_opti93x *chip, unsigned char reg) -{ - snd_opti93x_busy_wait(chip); - outb(chip->mce_bit | (reg & 0x1f), OPTi93X_PORT(chip, INDEX)); - return inb(OPTi93X_PORT(chip, DATA)); -} - -static void snd_opti93x_out(struct snd_opti93x *chip, unsigned char reg, - unsigned char value) -{ - snd_opti93x_busy_wait(chip); - outb(chip->mce_bit | (reg & 0x1f), OPTi93X_PORT(chip, INDEX)); - outb(value, OPTi93X_PORT(chip, DATA)); -} - -static void snd_opti93x_out_image(struct snd_opti93x *chip, unsigned char reg, - unsigned char value) -{ - snd_opti93x_out(chip, reg, chip->image[reg] = value); -} - -static void snd_opti93x_out_mask(struct snd_opti93x *chip, unsigned char reg, - unsigned char mask, unsigned char value) -{ - snd_opti93x_out_image(chip, reg, - (chip->image[reg] & ~mask) | (value & mask)); -} - - -static void snd_opti93x_mce_up(struct snd_opti93x *chip) -{ - snd_opti93x_busy_wait(chip); - - chip->mce_bit = OPTi93X_MCE; - if (!(inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_MCE)) - outb(chip->mce_bit, OPTi93X_PORT(chip, INDEX)); -} - -static void snd_opti93x_mce_down(struct snd_opti93x *chip) -{ - snd_opti93x_busy_wait(chip); - - chip->mce_bit = 0; - if (inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_MCE) - outb(chip->mce_bit, OPTi93X_PORT(chip, INDEX)); -} - -#define snd_opti93x_mute_reg(chip, reg, mute) \ - snd_opti93x_out(chip, reg, mute ? 0x80 : chip->image[reg]); - -static void snd_opti93x_mute(struct snd_opti93x *chip, int mute) -{ - mute = mute ? 1 : 0; - if (chip->mute == mute) - return; - - chip->mute = mute; - - snd_opti93x_mute_reg(chip, OPTi93X_CD_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_CD_RIGHT_INPUT, mute); - switch (chip->hardware) { - case OPTi9XX_HW_82C930: - snd_opti93x_mute_reg(chip, OPTi930_AUX_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi930_AUX_RIGHT_INPUT, mute); - break; - case OPTi9XX_HW_82C931: - case OPTi9XX_HW_82C933: - snd_opti93x_mute_reg(chip, OPTi931_FM_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_FM_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_AUX_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_AUX_RIGHT_INPUT, mute); - } - snd_opti93x_mute_reg(chip, OPTi93X_DAC_LEFT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_DAC_RIGHT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_LINE_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_LINE_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_MIC_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_MIC_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_OUT_LEFT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_OUT_RIGHT, mute); -} - - -static unsigned int snd_opti93x_get_count(unsigned char format, - unsigned int size) -{ - switch (format & 0xe0) { - case OPTi93X_LINEAR_16_LIT: - case OPTi93X_LINEAR_16_BIG: - size >>= 1; - break; - case OPTi93X_ADPCM_16: - return size >> 2; - } - return (format & OPTi93X_STEREO) ? (size >> 1) : size; -} - -static unsigned int rates[] = { 5512, 6615, 8000, 9600, 11025, 16000, - 18900, 22050, 27428, 32000, 33075, 37800, - 44100, 48000 }; -#define RATES ARRAY_SIZE(rates) - -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = RATES, - .list = rates, - .mask = 0, -}; - -static unsigned char bits[] = { 0x01, 0x0f, 0x00, 0x0e, 0x03, 0x02, - 0x05, 0x07, 0x04, 0x06, 0x0d, 0x09, - 0x0b, 0x0c}; - -static unsigned char snd_opti93x_get_freq(unsigned int rate) -{ - unsigned int i; - - for (i = 0; i < RATES; i++) { - if (rate == rates[i]) - return bits[i]; - } - snd_BUG(); - return bits[RATES-1]; -} - -static unsigned char snd_opti93x_get_format(struct snd_opti93x *chip, - unsigned int format, int channels) -{ - unsigned char retval = OPTi93X_LINEAR_8; - - switch (format) { - case SNDRV_PCM_FORMAT_MU_LAW: - retval = OPTi93X_ULAW_8; - break; - case SNDRV_PCM_FORMAT_A_LAW: - retval = OPTi93X_ALAW_8; - break; - case SNDRV_PCM_FORMAT_S16_LE: - retval = OPTi93X_LINEAR_16_LIT; - break; - case SNDRV_PCM_FORMAT_S16_BE: - retval = OPTi93X_LINEAR_16_BIG; - break; - case SNDRV_PCM_FORMAT_IMA_ADPCM: - retval = OPTi93X_ADPCM_16; - } - return (channels > 1) ? (retval | OPTi93X_STEREO) : retval; -} - - -static void snd_opti93x_playback_format(struct snd_opti93x *chip, unsigned char fmt) -{ - unsigned char mask; - - snd_opti93x_mute(chip, 1); - - snd_opti93x_mce_up(chip); - mask = (chip->mode & OPTi93X_MODE_CAPTURE) ? 0xf0 : 0xff; - snd_opti93x_out_mask(chip, OPTi93X_PLAY_FORMAT, mask, fmt); - snd_opti93x_mce_down(chip); - - snd_opti93x_mute(chip, 0); -} - -static void snd_opti93x_capture_format(struct snd_opti93x *chip, unsigned char fmt) -{ - snd_opti93x_mute(chip, 1); - - snd_opti93x_mce_up(chip); - if (!(chip->mode & OPTi93X_MODE_PLAY)) - snd_opti93x_out_mask(chip, OPTi93X_PLAY_FORMAT, 0x0f, fmt); - else - fmt = chip->image[OPTi93X_PLAY_FORMAT] & 0xf0; - snd_opti93x_out_image(chip, OPTi93X_CAPT_FORMAT, fmt); - snd_opti93x_mce_down(chip); - - snd_opti93x_mute(chip, 0); -} - - -static int snd_opti93x_open(struct snd_opti93x *chip, unsigned int mode) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - if (chip->mode & mode) { - spin_unlock_irqrestore(&chip->lock, flags); - return -EAGAIN; - } - - if (!(chip->mode & OPTi93X_MODE_OPEN)) { - outb(0x00, OPTi93X_PORT(chip, STATUS)); - snd_opti93x_out_mask(chip, OPTi93X_PIN_CTRL, - OPTi93X_IRQ_ENABLE, OPTi93X_IRQ_ENABLE); - chip->mode = mode; - } - else - chip->mode |= mode; - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static void snd_opti93x_close(struct snd_opti93x *chip, unsigned int mode) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - chip->mode &= ~mode; - if (chip->mode & OPTi93X_MODE_OPEN) { - spin_unlock_irqrestore(&chip->lock, flags); - return; - } - - snd_opti93x_mute(chip, 1); - - outb(0, OPTi93X_PORT(chip, STATUS)); - snd_opti93x_out_mask(chip, OPTi93X_PIN_CTRL, OPTi93X_IRQ_ENABLE, - ~OPTi93X_IRQ_ENABLE); - - snd_opti93x_mce_up(chip); - snd_opti93x_out_image(chip, OPTi93X_IFACE_CONF, 0x00); - snd_opti93x_mce_down(chip); - chip->mode = 0; - - snd_opti93x_mute(chip, 0); - spin_unlock_irqrestore(&chip->lock, flags); -} - -static int snd_opti93x_trigger(struct snd_pcm_substream *substream, - unsigned char what, int cmd) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_STOP: - { - unsigned int what = 0; - struct snd_pcm_substream *s; - snd_pcm_group_for_each_entry(s, substream) { - if (s == chip->playback_substream) { - what |= OPTi93X_PLAYBACK_ENABLE; - snd_pcm_trigger_done(s, substream); - } else if (s == chip->capture_substream) { - what |= OPTi93X_CAPTURE_ENABLE; - snd_pcm_trigger_done(s, substream); - } - } - spin_lock(&chip->lock); - if (cmd == SNDRV_PCM_TRIGGER_START) { - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, what, what); - if (what & OPTi93X_CAPTURE_ENABLE) - udelay(50); - } else - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, what, 0x00); - spin_unlock(&chip->lock); - break; - } - default: - return -EINVAL; - } - return 0; -} - -static int snd_opti93x_playback_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_opti93x_trigger(substream, - OPTi93X_PLAYBACK_ENABLE, cmd); -} - -static int snd_opti93x_capture_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_opti93x_trigger(substream, - OPTi93X_CAPTURE_ENABLE, cmd); -} - -static int snd_opti93x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); -} - - -static int snd_opti93x_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_lib_free_pages(substream); - return 0; -} - - -static int snd_opti93x_playback_prepare(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned char format; - unsigned int count = snd_pcm_lib_period_bytes(substream); - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - - spin_lock_irqsave(&chip->lock, flags); - - chip->p_dma_size = size; - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, - OPTi93X_PLAYBACK_ENABLE | OPTi93X_PLAYBACK_PIO, - ~(OPTi93X_PLAYBACK_ENABLE | OPTi93X_PLAYBACK_PIO)); - - snd_dma_program(chip->dma1, runtime->dma_addr, size, - DMA_MODE_WRITE | DMA_AUTOINIT); - - format = snd_opti93x_get_freq(runtime->rate); - format |= snd_opti93x_get_format(chip, runtime->format, - runtime->channels); - snd_opti93x_playback_format(chip, format); - format = chip->image[OPTi93X_PLAY_FORMAT]; - - count = snd_opti93x_get_count(format, count) - 1; - snd_opti93x_out_image(chip, OPTi93X_PLAY_LWR_CNT, count); - snd_opti93x_out_image(chip, OPTi93X_PLAY_UPR_CNT, count >> 8); - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static int snd_opti93x_capture_prepare(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned char format; - unsigned int count = snd_pcm_lib_period_bytes(substream); - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - - spin_lock_irqsave(&chip->lock, flags); - - chip->c_dma_size = size; - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, - OPTi93X_CAPTURE_ENABLE | OPTi93X_CAPTURE_PIO, 0); - - snd_dma_program(chip->dma2, runtime->dma_addr, size, - DMA_MODE_READ | DMA_AUTOINIT); - - format = snd_opti93x_get_freq(runtime->rate); - format |= snd_opti93x_get_format(chip, runtime->format, - runtime->channels); - snd_opti93x_capture_format(chip, format); - format = chip->image[OPTi93X_CAPT_FORMAT]; - - count = snd_opti93x_get_count(format, count) - 1; - snd_opti93x_out_image(chip, OPTi93X_CAPT_LWR_CNT, count); - snd_opti93x_out_image(chip, OPTi93X_CAPT_UPR_CNT, count >> 8); - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static snd_pcm_uframes_t snd_opti93x_playback_pointer(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[OPTi93X_IFACE_CONF] & OPTi93X_PLAYBACK_ENABLE)) - return 0; - - ptr = snd_dma_pointer(chip->dma1, chip->p_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - -static snd_pcm_uframes_t snd_opti93x_capture_pointer(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[OPTi93X_IFACE_CONF] & OPTi93X_CAPTURE_ENABLE)) - return 0; - - ptr = snd_dma_pointer(chip->dma2, chip->c_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - - -static void snd_opti93x_overrange(struct snd_opti93x *chip) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - if (snd_opti93x_in(chip, OPTi93X_ERR_INIT) & (0x08 | 0x02)) - chip->capture_substream->runtime->overrange++; - - spin_unlock_irqrestore(&chip->lock, flags); -} - static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) { - struct snd_opti93x *codec = dev_id; + struct snd_cs4231 *codec = dev_id; + struct snd_opti9xx *chip = codec->card->private_data; unsigned char status; - status = snd_opti9xx_read(codec->chip, OPTi9XX_MC_REG(11)); + status = snd_opti9xx_read(chip, OPTi9XX_MC_REG(11)); if ((status & OPTi93X_IRQ_PLAYBACK) && codec->playback_substream) snd_pcm_period_elapsed(codec->playback_substream); if ((status & OPTi93X_IRQ_CAPTURE) && codec->capture_substream) { - snd_opti93x_overrange(codec); + snd_cs4231_overrange(codec); snd_pcm_period_elapsed(codec->capture_substream); } outb(0x00, OPTi93X_PORT(codec, STATUS)); return IRQ_HANDLED; } - -static struct snd_pcm_hardware snd_opti93x_playback = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), - .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, - .rate_min = 5512, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware snd_opti93x_capture = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), - .rates = SNDRV_PCM_RATE_8000_48000, - .rate_min = 5512, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -static int snd_opti93x_playback_open(struct snd_pcm_substream *substream) -{ - int error; - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - if ((error = snd_opti93x_open(chip, OPTi93X_MODE_PLAY)) < 0) - return error; - snd_pcm_set_sync(substream); - chip->playback_substream = substream; - runtime->hw = snd_opti93x_playback; - snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - return error; -} - -static int snd_opti93x_capture_open(struct snd_pcm_substream *substream) -{ - int error; - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - if ((error = snd_opti93x_open(chip, OPTi93X_MODE_CAPTURE)) < 0) - return error; - runtime->hw = snd_opti93x_capture; - snd_pcm_set_sync(substream); - chip->capture_substream = substream; - snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - return error; -} - -static int snd_opti93x_playback_close(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - chip->playback_substream = NULL; - snd_opti93x_close(chip, OPTi93X_MODE_PLAY); - return 0; -} - -static int snd_opti93x_capture_close(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - chip->capture_substream = NULL; - snd_opti93x_close(chip, OPTi93X_MODE_CAPTURE); - return 0; -} - - -static void snd_opti93x_init(struct snd_opti93x *chip) -{ - unsigned long flags; - int i; - - spin_lock_irqsave(&chip->lock, flags); - snd_opti93x_mce_up(chip); - - for (i = 0; i < 32; i++) - snd_opti93x_out_image(chip, i, snd_opti93x_default_image[i]); - - snd_opti93x_mce_down(chip); - spin_unlock_irqrestore(&chip->lock, flags); -} - -static int snd_opti93x_probe(struct snd_opti93x *chip) -{ - unsigned long flags; - unsigned char val; - - spin_lock_irqsave(&chip->lock, flags); - val = snd_opti93x_in(chip, OPTi93X_ID) & 0x0f; - spin_unlock_irqrestore(&chip->lock, flags); - - return (val == 0x0a) ? 0 : -ENODEV; -} - -static int snd_opti93x_free(struct snd_opti93x *chip) -{ - release_and_free_resource(chip->res_port); - if (chip->dma1 >= 0) { - disable_dma(chip->dma1); - free_dma(chip->dma1); - } - if (chip->dma2 >= 0) { - disable_dma(chip->dma2); - free_dma(chip->dma2); - } - if (chip->irq >= 0) { - free_irq(chip->irq, chip); - } - kfree(chip); - return 0; -} - -static int snd_opti93x_dev_free(struct snd_device *device) -{ - struct snd_opti93x *chip = device->device_data; - return snd_opti93x_free(chip); -} - -static const char *snd_opti93x_chip_id(struct snd_opti93x *codec) -{ - switch (codec->hardware) { - case OPTi9XX_HW_82C930: return "82C930"; - case OPTi9XX_HW_82C931: return "82C931"; - case OPTi9XX_HW_82C933: return "82C933"; - default: return "???"; - } -} - -static int snd_opti93x_create(struct snd_card *card, struct snd_opti9xx *chip, - int dma1, int dma2, - struct snd_opti93x **rcodec) -{ - static struct snd_device_ops ops = { - .dev_free = snd_opti93x_dev_free, - }; - int error; - struct snd_opti93x *codec; - - *rcodec = NULL; - codec = kzalloc(sizeof(*codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - codec->irq = -1; - codec->dma1 = -1; - codec->dma2 = -1; - - if ((codec->res_port = request_region(chip->wss_base + 4, 4, "OPTI93x CODEC")) == NULL) { - snd_printk(KERN_ERR "opti9xx: can't grab port 0x%lx\n", chip->wss_base + 4); - snd_opti93x_free(codec); - return -EBUSY; - } - if (request_dma(dma1, "OPTI93x - 1")) { - snd_printk(KERN_ERR "opti9xx: can't grab DMA1 %d\n", dma1); - snd_opti93x_free(codec); - return -EBUSY; - } - codec->dma1 = chip->dma1; - if (request_dma(dma2, "OPTI93x - 2")) { - snd_printk(KERN_ERR "opti9xx: can't grab DMA2 %d\n", dma2); - snd_opti93x_free(codec); - return -EBUSY; - } - codec->dma2 = chip->dma2; - - if (request_irq(chip->irq, snd_opti93x_interrupt, IRQF_DISABLED, DEV_NAME" - WSS", codec)) { - snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); - snd_opti93x_free(codec); - return -EBUSY; - } - - codec->card = card; - codec->port = chip->wss_base + 4; - codec->irq = chip->irq; - - spin_lock_init(&codec->lock); - codec->hardware = chip->hardware; - codec->chip = chip; - - if ((error = snd_opti93x_probe(codec))) { - snd_opti93x_free(codec); - return error; - } - - snd_opti93x_init(codec); - - /* Register device */ - if ((error = snd_device_new(card, SNDRV_DEV_LOWLEVEL, codec, &ops)) < 0) { - snd_opti93x_free(codec); - return error; - } - - *rcodec = codec; - return 0; -} - -static struct snd_pcm_ops snd_opti93x_playback_ops = { - .open = snd_opti93x_playback_open, - .close = snd_opti93x_playback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_opti93x_hw_params, - .hw_free = snd_opti93x_hw_free, - .prepare = snd_opti93x_playback_prepare, - .trigger = snd_opti93x_playback_trigger, - .pointer = snd_opti93x_playback_pointer, -}; - -static struct snd_pcm_ops snd_opti93x_capture_ops = { - .open = snd_opti93x_capture_open, - .close = snd_opti93x_capture_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_opti93x_hw_params, - .hw_free = snd_opti93x_hw_free, - .prepare = snd_opti93x_capture_prepare, - .trigger = snd_opti93x_capture_trigger, - .pointer = snd_opti93x_capture_pointer, -}; - -static int snd_opti93x_pcm(struct snd_opti93x *codec, int device, struct snd_pcm **rpcm) -{ - int error; - struct snd_pcm *pcm; - - if ((error = snd_pcm_new(codec->card, "OPTi 82C93X", device, 1, 1, &pcm)) < 0) - return error; - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_opti93x_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_opti93x_capture_ops); - - pcm->private_data = codec; - pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; - - strcpy(pcm->name, snd_opti93x_chip_id(codec)); - - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_isa_data(), - 64*1024, codec->dma1 > 3 || codec->dma2 > 3 ? 128*1024 : 64*1024); - - codec->pcm = pcm; - if (rpcm) - *rpcm = pcm; - return 0; -} - -/* - * MIXER part - */ - -static int snd_opti93x_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - static char *texts[4] = { - "Line1", "Aux", "Mic", "Mix" - }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 2; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int snd_opti93x_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.enumerated.item[0] = (chip->image[OPTi93X_MIXOUT_LEFT] & OPTi93X_MIXOUT_MIXER) >> 6; - ucontrol->value.enumerated.item[1] = (chip->image[OPTi93X_MIXOUT_RIGHT] & OPTi93X_MIXOUT_MIXER) >> 6; - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static int snd_opti93x_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - unsigned short left, right; - int change; - - if (ucontrol->value.enumerated.item[0] > 3 || - ucontrol->value.enumerated.item[1] > 3) - return -EINVAL; - left = ucontrol->value.enumerated.item[0] << 6; - right = ucontrol->value.enumerated.item[1] << 6; - spin_lock_irqsave(&chip->lock, flags); - left = (chip->image[OPTi93X_MIXOUT_LEFT] & ~OPTi93X_MIXOUT_MIXER) | left; - right = (chip->image[OPTi93X_MIXOUT_RIGHT] & ~OPTi93X_MIXOUT_MIXER) | right; - change = left != chip->image[OPTi93X_MIXOUT_LEFT] || - right != chip->image[OPTi93X_MIXOUT_RIGHT]; - snd_opti93x_out_image(chip, OPTi93X_MIXOUT_LEFT, left); - snd_opti93x_out_image(chip, OPTi93X_MIXOUT_RIGHT, right); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -#if 0 - -#define OPTi93X_SINGLE(xname, xindex, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_opti93x_info_single, \ - .get = snd_opti93x_get_single, .put = snd_opti93x_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } - -static int snd_opti93x_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 16) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_opti93x_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.integer.value[0] = (chip->image[reg] >> shift) & mask; - spin_unlock_irqrestore(&chip->lock, flags); - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - return 0; -} - -static int snd_opti93x_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - int change; - unsigned short val; - - val = (ucontrol->value.integer.value[0] & mask); - if (invert) - val = mask - val; - val <<= shift; - spin_lock_irqsave(&chip->lock, flags); - val = (chip->image[reg] & ~(mask << shift)) | val; - change = val != chip->image[reg]; - snd_opti93x_out(chip, reg, val); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -#endif /* single */ - -#define OPTi93X_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_opti93x_info_double, \ - .get = snd_opti93x_get_double, .put = snd_opti93x_put_double, \ - .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } - -#define OPTi93X_DOUBLE_INVERT_INVERT(xctl) \ - do { xctl.private_value ^= 22; } while (0) -#define OPTi93X_DOUBLE_CHANGE_REGS(xctl, left_reg, right_reg) \ - do { xctl.private_value &= ~0x0000ffff; \ - xctl.private_value |= left_reg | (right_reg << 8); } while (0) - -static int snd_opti93x_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 24) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_opti93x_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.integer.value[0] = (chip->image[left_reg] >> shift_left) & mask; - ucontrol->value.integer.value[1] = (chip->image[right_reg] >> shift_right) & mask; - spin_unlock_irqrestore(&chip->lock, flags); - if (invert) { - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1]; - } - return 0; -} - -static int snd_opti93x_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - int change; - unsigned short val1, val2; - - val1 = ucontrol->value.integer.value[0] & mask; - val2 = ucontrol->value.integer.value[1] & mask; - if (invert) { - val1 = mask - val1; - val2 = mask - val2; - } - val1 <<= shift_left; - val2 <<= shift_right; - spin_lock_irqsave(&chip->lock, flags); - val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; - val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; - change = val1 != chip->image[left_reg] || val2 != chip->image[right_reg]; - snd_opti93x_out_image(chip, left_reg, val1); - snd_opti93x_out_image(chip, right_reg, val2); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -static struct snd_kcontrol_new snd_opti93x_controls[] __devinitdata = { -OPTi93X_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Master Playback Volume", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), -OPTi93X_DOUBLE("PCM Playback Switch", 0, OPTi93X_DAC_LEFT, OPTi93X_DAC_RIGHT, 7, 7, 1, 1), -OPTi93X_DOUBLE("PCM Playback Volume", 0, OPTi93X_DAC_LEFT, OPTi93X_DAC_RIGHT, 0, 0, 31, 1), -OPTi93X_DOUBLE("FM Playback Switch", 0, OPTi931_FM_LEFT_INPUT, OPTi931_FM_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("FM Playback Volume", 0, OPTi931_FM_LEFT_INPUT, OPTi931_FM_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Line Playback Switch", 0, OPTi93X_LINE_LEFT_INPUT, OPTi93X_LINE_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Line Playback Volume", 0, OPTi93X_LINE_LEFT_INPUT, OPTi93X_LINE_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Mic Playback Switch", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Mic Playback Volume", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Mic Boost", 0, OPTi93X_MIXOUT_LEFT, OPTi93X_MIXOUT_RIGHT, 5, 5, 1, 1), -OPTi93X_DOUBLE("CD Playback Switch", 0, OPTi93X_CD_LEFT_INPUT, OPTi93X_CD_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("CD Playback Volume", 0, OPTi93X_CD_LEFT_INPUT, OPTi93X_CD_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Aux Playback Switch", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Aux Playback Volume", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Capture Volume", 0, OPTi93X_MIXOUT_LEFT, OPTi93X_MIXOUT_RIGHT, 0, 0, 15, 0), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = snd_opti93x_info_mux, - .get = snd_opti93x_get_mux, - .put = snd_opti93x_put_mux, -} -}; - -static int __devinit snd_opti93x_mixer(struct snd_opti93x *chip) -{ - struct snd_card *card; - struct snd_kcontrol_new knew; - int err; - unsigned int idx; - - snd_assert(chip != NULL && chip->card != NULL, return -EINVAL); - - card = chip->card; - - strcpy(card->mixername, snd_opti93x_chip_id(chip)); - - for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { - knew = snd_opti93x_controls[idx]; - if (chip->hardware == OPTi9XX_HW_82C930) { - if (strstr(knew.name, "FM")) /* skip FM controls */ - continue; - else if (strcmp(knew.name, "Mic Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - else if (strstr(knew.name, "Aux")) - OPTi93X_DOUBLE_CHANGE_REGS(knew, OPTi930_AUX_LEFT_INPUT, OPTi930_AUX_RIGHT_INPUT); - else if (strcmp(knew.name, "PCM Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - else if (strcmp(knew.name, "Master Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - } - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_opti93x_controls[idx], chip))) < 0) - return err; - } - return 0; -} - #endif /* OPTi93X */ static int __devinit snd_card_opti9xx_detect(struct snd_card *card, @@ -1739,8 +685,16 @@ static void snd_card_opti9xx_free(struct { struct snd_opti9xx *chip = card->private_data; - if (chip) + if (chip) { +#ifdef OPTi93X + struct snd_cs4231 *codec = chip->codec; + if (codec->irq > 0) { + disable_irq(codec->irq); + free_irq(codec->irq, codec); + } +#endif release_and_free_resource(chip->res_mc_base); + } } static int __devinit snd_opti9xx_probe(struct snd_card *card) @@ -1748,11 +702,11 @@ static int __devinit snd_opti9xx_probe(s static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; int error; struct snd_opti9xx *chip = card->private_data; -#if defined(OPTi93X) - struct snd_opti93x *codec; -#elif defined(CS4231) +#if defined(CS4231) || defined(OPTi93X) struct snd_cs4231 *codec; +#ifdef CS4231 struct snd_timer *timer; +#endif #else struct snd_ad1848 *codec; #endif @@ -1784,26 +738,34 @@ static int __devinit snd_opti9xx_probe(s if ((error = snd_opti9xx_configure(chip))) return error; -#if defined(OPTi93X) - if ((error = snd_opti93x_create(card, chip, chip->dma1, chip->dma2, &codec))) - return error; - if ((error = snd_opti93x_pcm(codec, 0, &pcm)) < 0) - return error; - if ((error = snd_opti93x_mixer(codec)) < 0) - return error; -#elif defined(CS4231) +#if defined(CS4231) || defined(OPTi93X) if ((error = snd_cs4231_create(card, chip->wss_base + 4, -1, chip->irq, chip->dma1, chip->dma2, - CS4231_HW_DETECT, - 0, +#ifdef CS4231 + CS4231_HW_DETECT, 0, +#else /* OPTi93x */ + CS4231_HW_OPTI93X, CS4231_HWSHARE_IRQ, +#endif &codec)) < 0) return error; +#ifdef OPTi93X + chip->codec = codec; +#endif if ((error = snd_cs4231_pcm(codec, 0, &pcm)) < 0) return error; if ((error = snd_cs4231_mixer(codec)) < 0) return error; +#ifdef CS4231 if ((error = snd_cs4231_timer(codec, 0, &timer)) < 0) return error; +#else /* OPTI93X */ + error = request_irq(chip->irq, snd_opti93x_interrupt, + IRQF_DISABLED, DEV_NAME" - WSS", codec); + if (error < 0) { + snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); + return error; + } +#endif #else if ((error = snd_ad1848_create(card, chip->wss_base + 4, chip->irq, chip->dma1, --------------------------------------------------------------- Sprawdz jak zdobyc zdrowy usmiech! Kliknij >> http://link.interia.pl/f1e26

3 4

Re: [alsa-devel] [PATCH] sound: emu10k1 - fix system hang with Audigy2 ZS Notebook PCMCIA
by Chris Rankin 11 Jun '08

11 Jun '08

Hi, I can't say whether this patch fixes the hang at boot-time with the Audigy2 ZS Notebook card, but my workstation with its Audigy 2 PCI card boots a LOT more quietly with this patch. It used to make a sound like a herd of stampeding elephants rushing past (and had done for quiet a while...), but now is far better behaved. Cheers, Chris __________________________________________________________ Sent from Yahoo! Mail. A Smarter Email http://uk.docs.yahoo.com/nowyoucan.html

1 0

[alsa-devel] [PATCH asoc-dev 1/2] ASoC: Use correct include file for gpiolib in N810 machine driver
by Jarkko Nikula 11 Jun '08

11 Jun '08

Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/n810.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 2d982b1..c32487b 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -30,7 +30,7 @@ #include <asm/mach-types.h> #include <asm/arch/hardware.h> -#include <asm/gpio.h> +#include <linux/gpio.h> #include <asm/arch/mcbsp.h> #include "omap-mcbsp.h" -- 1.5.5.3

2 4

Re: [alsa-devel] Problems regarding alsa-info.sh
by Travis Place 10 Jun '08

10 Jun '08

Sorry for the quick 2nd post, but ive just been thinking about this.. If we can patch the GIT version with the patch i just emailed, and temporarily put that same file (version 0.4.46) into the HG repo (even if it has to be entered manually) then current script users will get the new script (which contains the new URLs). That will solve half the problem. Then we just need a friendly URL to post in wikis etc. Ive setup a very temporary tinyurl page.. http://tinyurl.com/5bz98h Seems to work ok, but we want something more perm, and more professional looking, really. Travis Place.

3 5

[alsa-devel] What does 0dB refer to? (Logitech USB Speakers)
by Lennart Poettering 10 Jun '08

10 Jun '08

Heya! I think I discussed this with both James and Takashi a while back: it would be great if we could initialize the sound card mixers in a better way by default and not everything set to 0. i.e. for output only PCM should be active, at a sensible volume, no bass/treble/3d enabled or whatever. All other sources should be muted. And that for input the mic jack should be active and nothing else, again. Right now people too often have to invest a lot of time to find the right controls to toggle/change in the mixer to get something sensible out of the speakers. Some distributions have started to ship special scripts that try to initialize the volumes properly. I know for the Fedora case that this tool is very very simple, and just initializes PCM and Master to 70% and that's basically it. It would be highly preferable if we had a sane tool like this from alsa upstream, which would set the default values to 0dB, possibly as an extension to "alsactl". I started to hack something similar to this as part of PulseAudio: if PA finds that the PCM slider is not really useful for volume control (no attenuation to at least -60dB, too few steps, no seperate controls for all channels, ...) it falls back to software volume control, however tries to initialize the the slider to 0dB first. Now the thing is, this mostly works fine, however on one of my devices, a pair of external Logitech USB speakers (ID 046d:0a04 Logitech, Inc. V20 portable speakers), the dB scale seems to be totally bogus. According to ALSA volume ranges from -41 dB to +3dB. However, Every setting > -41 dB makes audio awfully loud. Really, really fucking loud that is. Disco loud. As loud that the speakers start jump around centimeters due to the massive vibrations. Setting the volume to 0dB with those speakers really makes you fear you are trashing them. I somehow expected that the magic "0dB" setting in ALSA would refer to some sane default, where the final output would have a sensible volume. In this case this is not so. I am thus wondering how ALSA actually defines 0dB? Is this a bug in the hardware that should be worked around in ALSA? Is this an ALSA bug? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

13 82

[alsa-devel] [PATCH] Fix buffer position for ATI SB4x0
by Mauro Carvalho Chehab 09 Jun '08

09 Jun '08

ATI SB4x0 doesn't need any fix at position. This patch solves the issue of receiving several clicks during capture on those devices. Tested with a Gateway Notebook MX-6453. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)infradead.org> diff -r 33d7308ade64 pci/hda/hda_intel.c --- a/pci/hda/hda_intel.c Fri May 23 17:36:59 2008 -0300 +++ b/pci/hda/hda_intel.c Thu May 29 11:04:56 2008 -0300 @@ -1860,6 +1860,7 @@ SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_NONE), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_NONE), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_NONE), + SND_PCI_QUIRK(0x1002, 0x437b, "ATI SB4x0 HDA", POS_FIX_NONE), {} };

2 8

[alsa-devel] ASOC PCM formats
by Jon Smirl 09 Jun '08

09 Jun '08

Why is the AC97 format set to S16 in all of the drivers? The link is 20b and my codec takes 20b, how do I get 20b data to the device? I tried using SNDRV_PCM_FORMAT_S20_3BE but all I get is static. If I use S24 for the format it works but volume is divided by 16. I think the bits have been right justified into the 24 bits so the bottom four bits are lost. AC97 link is Efika, MPC5200 Codec is STAC9766. -- Jon Smirl jonsmirl(a)gmail.com

3 4

[alsa-devel] [PATCH] opti93x: fix sound ouput for Opti930
by Krzysztof Helt 09 Jun '08

09 Jun '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch fixes silenced output from the Opti930. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The bug was introduced by me in the patch which added QS1000 wavetable support: http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commit;h… Simply do not change digital input bits on Opti930. diff -urp linux-rc2/sound/isa/opti9xx/opti92x-ad1848.c linux-rc5/sound/isa/opti9xx/opti92x-ad1848.c --- linux-rc2/sound/isa/opti9xx/opti92x-ad1848.c 2008-04-17 04:49:44.000000000 +0200 +++ linux-rc5/sound/isa/opti9xx/opti92x-ad1848.c 2008-06-07 23:59:27.218950875 +0200 @@ -491,16 +491,9 @@ static int __devinit snd_opti9xx_configu break; #else /* OPTi93X */ - case OPTi9XX_HW_82C930: case OPTi9XX_HW_82C931: case OPTi9XX_HW_82C933: - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 | - (chip->hardware == OPTi9XX_HW_82C930 ? 0x00 : 0x04), - 0x34); - snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x20, 0xbf); - /* + /* * The BTC 1817DW has QS1000 wavetable which is connected * to the serial digital input of the OPTI931. */ @@ -510,6 +503,13 @@ static int __devinit snd_opti9xx_configu * or digital input signal. */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01); + case OPTi9XX_HW_82C930: /* FALL THROUGH */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0x10 | + (chip->hardware == OPTi9XX_HW_82C930 ? 0x00 : 0x04), + 0x34); + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x20, 0xbf); break; #endif /* OPTi93X */ --------------------------------------------------------------- Sprawdz jak zdobyc zdrowy usmiech! Kliknij >> http://link.interia.pl/f1e26

3 2

[alsa-devel] [PATCH] Fix "alc262_sony_unsol[]" hda_verb array
by idak 09 Jun '08

09 Jun '08

I think that hda_verb array must have "terminator (empty array)". But alc262_sony_unsol[] does not have it. And it causes gcc-4.3's buggy behavior with snd_hda_sequence_write(). Signed-off-by: Akio Idehara <zbe64533(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) --- a/sound/pci/hda/patch_realtek.c 2008-06-08 19:23:57.000000000 +0900 +++ b/sound/pci/hda/patch_realtek.c 2008-06-08 19:42:19.000000000 +0900 @@ -8839,6 +8839,7 @@ static struct hda_verb alc262_sony_unsol {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} }; /* mute/unmute internal speaker according to the hp jack and mute state */

2 1

[alsa-devel] [PATCH] Add Toshiba dynabook SS RX1 support
by Akio Idehara 09 Jun '08

09 Jun '08

I have Toshiba dynabook SS RX1 and this patch adds that support. Signed-off-by: Akio Idehara <zbe64533(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) --- a/sound/pci/hda/patch_realtek.c 2008-06-08 19:23:57.000000000 +0900 +++ b/sound/pci/hda/patch_realtek.c 2008-06-08 20:55:34.000000000 +0900 @@ -9718,6 +9718,8 @@ static struct snd_pci_quirk alc262_cfg_t SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", + ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA),

1 0

[alsa-devel] A little problem with main gitweb page
by Rafał Mużyło 08 Jun '08

08 Jun '08

Well, it would be nice if http://git.alsa-project.org/ listed alsa-plugins git too.

1 0

[alsa-devel] Problems regarding alsa-info.sh
by Travis Place 08 Jun '08

08 Jun '08

Since the move from HG->GIT there is now a new download location for alsa-info.sh . There are 2 current problems with this though.. 1) The new URL is very ugly(and hard to post on wikis etc) The new link is http://git.alsa-project.org/?p=alsa-driver.git;a=blob_plain;f=utils/alsa-in… 2) The current script version at HG repo (which is where people currently get it from, is an outdated version, that checks the HG repo for updates to the script. We need to put the new version of the script into HG (if its still accessible), with the updated 'update' URL, or, somehow redirect from the old URL to the new URL. Please let me know what should be done about this, and how to go about fixing it. Attached is the updated alsa-info.sh with the new GIT URLS (update and changelog) in place.

1 0

[alsa-devel] Wrong period size interval with hda-intel makes snd_pcm_set_params() fail
by Christian Morales Vega 08 Jun '08

08 Jun '08

I looked deeper into the problem and looks like a hda-intel problem. This code fails with "ALSA lib pcm.c:7193:(snd_pcm_set_params) Unable to get period size for PLAYBACK: Invalid argument" #include <alsa/asoundlib.h> int main(){ snd_pcm_t *handle; if(snd_pcm_open(&handle, "default:1", SND_PCM_STREAM_PLAYBACK, 0) < 0) { exit(1); } if(snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED, 2, 24000, 1, 100000) < 0) { exit(2); } } The "Invalid argument" comes from pcm_params.c:145. The test "!snd_interval_single(i)" fails since the interval for SND_PCM_HW_PARAM_PERIOD_SIZE ends being min=511 and max=513. I understand that this interval is set at the driver, correct? I don't know where exactly, so some help in finding it or a patch would be welcome. I'm using alsa-driver-kmp-default-1.0.16.20080608, libasound2-1.0.16.hg20080510 and alsa-1.0.16.hg20080510 RPM packages from multimedia:audio openSUSE Build Service repository (from Takashi Iwai) and an AD1988 (hda-intel) codec. A XUbuntu user also has the same problem with an hda-intel codec (that I suppose uses stable 1.0.16). With a SB Live! in the same computer there are no problems.

1 0

[alsa-devel] Previous patch to alsa-info.sh
by Travis Place 07 Jun '08

07 Jun '08

I sent of a patch for alsa-info.sh about a week or two ago now, and Im starting to think it lost amongst all the other posts. Unforunately my main system died, and is off getting warrantied, so i cant repost. Can someone look through the list and find the patch (it added a check to see if pastebin.ca was alive before posting to it, and falling back to --no-upload if it was down. Upped the version to 0.4.45 too i think) Thanks, Travis Place

2 2

[alsa-devel] [PATCH][ASoC V2] Update MPC8610 drivers to pick the right DMA channels
by Timur Tabi 06 Jun '08

06 Jun '08

On the Freescale MPC8610 SOC, SSI1 is hard-wired to DMA channels 0 and 1, and SSI2 is hard-wired to DMA channels 2 and 3. Fix fsl_ssi.c to reflect that. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9d141ec..4f66e84 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -790,13 +790,13 @@ static int fsl_ssi_probe(struct of_device *ofdev, /* * Get the DMA information. If it's an older device tree (i.e. without * an "fsl,playback-dma" property), then we assume that SSI1 uses DMA1 - * Channels 0 and 1, and SSI2 uses DMA2 Channels 0 and 1. + * Channels 0 and 1, and SSI2 uses DMA1 Channels 2 and 3. */ dma_ph = of_get_property(np, "fsl,playback-dma", NULL); if (!dma_ph) { dev_warn(&ofdev->dev, "please update your device tree\n"); - dma_np[0] = find_dma_node(ssi_info->id, 0); - dma_np[1] = find_dma_node(ssi_info->id, 1); + dma_np[0] = find_dma_node(0, ssi_info->id ? 2 : 0); + dma_np[1] = find_dma_node(0, ssi_info->id ? 3 : 1); } else { dma_np[0] = of_find_node_by_phandle(*dma_ph); dma_ph = of_get_property(np, "fsl,capture-dma", NULL); -- 1.5.5

2 1

[alsa-devel] [PATCH][ASoC V2] Update snd_soc_card_get_codec() to take the codec number
by Timur Tabi 06 Jun '08

06 Jun '08

Update function snd_soc_card_get_codec() and all the drivers that call it to take a codec number in addition to a codec name. This is necessary for codec drivers that register multiple instances of themselves that differ only by codec number. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- I have only tested this code on the MPC8610, so please test this on other platforms before committing. include/sound/soc-card.h | 2 +- sound/soc/at91/eti_b1_wm8731.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 10 +++++----- sound/soc/imx/imx32ads-wm8350.c | 2 +- sound/soc/pxa/amesom_tlv320.c | 2 +- sound/soc/pxa/corgi.c | 2 +- sound/soc/pxa/e800_wm9712.c | 2 +- sound/soc/pxa/h5000.c | 2 +- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/mainstone_wm8753.c | 2 +- sound/soc/pxa/mainstone_wm9712.c | 2 +- sound/soc/pxa/mainstone_wm9713.c | 2 +- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/spitz.c | 2 +- sound/soc/pxa/tosa.c | 2 +- sound/soc/pxa/zylonite.c | 2 +- sound/soc/soc-core.c | 5 +++-- 17 files changed, 23 insertions(+), 22 deletions(-) diff --git a/include/sound/soc-card.h b/include/sound/soc-card.h index 9b8600d..290f649 100644 --- a/include/sound/soc-card.h +++ b/include/sound/soc-card.h @@ -44,7 +44,7 @@ struct snd_ac97_bus_ops *snd_soc_card_get_ac97_ops( struct snd_soc_card *soc_card, const char *dai_id); struct snd_soc_codec *snd_soc_card_get_codec(struct snd_soc_card *soc_card, - const char *codec_id); + const char *codec_name, int codec_num); struct snd_soc_platform * snd_soc_card_get_platform( struct snd_soc_card *soc_card, const char *platform_id); diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index bcee769..a70fffe 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -242,7 +242,7 @@ static int eti_b1_init(struct snd_soc_card *soc_card) struct at91_ssc_periph *ssc; int ret; - codec = snd_soc_card_get_codec(soc_card, wm8731_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm8731_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 45c3311..8813291 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -38,7 +38,6 @@ * MPC8610 HPCD. Some of the data is taken from the device tree. */ struct mpc8610_hpcd_data { - char codec_name[32]; struct snd_soc_card soc_card; struct ccsr_guts __iomem *guts; unsigned int num_configs; @@ -47,8 +46,6 @@ struct mpc8610_hpcd_data { char pcm_name[32]; char ssi_name[32]; char codec_name[32]; - char platform_name[32]; - char cpu_dai_name[32]; } names[MAX_SSI]; }; @@ -99,7 +96,8 @@ static int mpc8610_hpcd_audio_init(struct snd_soc_card *soc_card) for (i = 0; i < soc_card_data->num_configs; i++) { codec = snd_soc_card_get_codec(soc_card, - soc_card_data->names[i].codec_name); + soc_card_data->names[i].codec_name, + soc_card_data->configs[i].codec_num); if (!codec) { dev_err(soc_card->dev, "could not find codec\n"); @@ -111,7 +109,9 @@ static int mpc8610_hpcd_audio_init(struct snd_soc_card *soc_card) */ ret = snd_soc_card_init_codec(codec, soc_card); if (ret < 0) { - dev_err(soc_card->dev, "could not initialize codec\n"); + dev_err(soc_card->dev, + "could not initialize codec %s-%u\n", + codec->name, codec->num); continue; } } diff --git a/sound/soc/imx/imx32ads-wm8350.c b/sound/soc/imx/imx32ads-wm8350.c index feb83cc..94de723 100644 --- a/sound/soc/imx/imx32ads-wm8350.c +++ b/sound/soc/imx/imx32ads-wm8350.c @@ -407,7 +407,7 @@ int imx32_audio_init(struct snd_soc_card *soc_card) if (pcm_runtime == NULL) return -ENODEV; - codec = snd_soc_card_get_codec(soc_card, wm8350_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm8350_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/amesom_tlv320.c b/sound/soc/pxa/amesom_tlv320.c index 9be4db0..c2ba22a 100644 --- a/sound/soc/pxa/amesom_tlv320.c +++ b/sound/soc/pxa/amesom_tlv320.c @@ -134,7 +134,7 @@ static int amesom_tlv320_init(struct snd_soc_card *soc_card) { struct snd_soc_codec *codec; - codec = snd_soc_card_get_codec(soc_card, tlv320_codec_id); + codec = snd_soc_card_get_codec(soc_card, tlv320_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index c24d76b..9f30f12 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -313,7 +313,7 @@ static int corgi_init(struct snd_soc_card *soc_card) struct snd_soc_codec *codec; int ret; - codec = snd_soc_card_get_codec(soc_card, wm8731_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm8731_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 987106a..9159d03 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -50,7 +50,7 @@ static int e800_init(struct snd_soc_card *soc_card) struct snd_ac97_bus_ops *ac97_ops; int ret; - codec = snd_soc_card_get_codec(soc_card, wm9712_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm9712_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/h5000.c b/sound/soc/pxa/h5000.c index 54ff23c..fa340f4 100644 --- a/sound/soc/pxa/h5000.c +++ b/sound/soc/pxa/h5000.c @@ -272,7 +272,7 @@ static int h5000_ak4535_init (struct snd_soc_card *soc_card) struct snd_soc_codec *codec; int ret; - codec = snd_soc_card_get_codec(soc_card, ak4535_codec_id); + codec = snd_soc_card_get_codec(soc_card, ak4535_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index ad6dea7..0d7b145 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -436,7 +436,7 @@ static int magician_uda1380_init(struct snd_soc_card *soc_card) struct snd_soc_codec *codec; int ret; - codec = snd_soc_card_get_codec(soc_card, uda1380_codec_id); + codec = snd_soc_card_get_codec(soc_card, uda1380_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/mainstone_wm8753.c b/sound/soc/pxa/mainstone_wm8753.c index f7ab6fb..2bef9e9 100644 --- a/sound/soc/pxa/mainstone_wm8753.c +++ b/sound/soc/pxa/mainstone_wm8753.c @@ -328,7 +328,7 @@ static int mainstone_wm8753_init(struct snd_soc_card *soc_card) struct snd_soc_codec *codec; int ret; - codec = snd_soc_card_get_codec(soc_card, wm8753_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm8753_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/mainstone_wm9712.c b/sound/soc/pxa/mainstone_wm9712.c index b87362b..f2ded1d 100644 --- a/sound/soc/pxa/mainstone_wm9712.c +++ b/sound/soc/pxa/mainstone_wm9712.c @@ -81,7 +81,7 @@ static int mainstone_wm9712_init(struct snd_soc_card *soc_card) struct snd_ac97_bus_ops *ac97_ops; int ret; - codec = snd_soc_card_get_codec(soc_card, wm9712_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm9712_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/mainstone_wm9713.c b/sound/soc/pxa/mainstone_wm9713.c index 9492b3e..f9723e1 100644 --- a/sound/soc/pxa/mainstone_wm9713.c +++ b/sound/soc/pxa/mainstone_wm9713.c @@ -163,7 +163,7 @@ static int mainstone_wm9713_init(struct snd_soc_card *soc_card) struct snd_ac97_bus_ops *ac97_ops; int ret; - codec = snd_soc_card_get_codec(soc_card, wm9713_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm9713_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 4ce0ce4..126ba5b 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -264,7 +264,7 @@ static int poodle_init(struct snd_soc_card *soc_card) struct snd_soc_codec *codec; int ret; - codec = snd_soc_card_get_codec(soc_card, wm8731_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm8731_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 497b3b2..393adda 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -315,7 +315,7 @@ static int spitz_init(struct snd_soc_card *soc_card) struct snd_soc_codec *codec; int ret; - codec = snd_soc_card_get_codec(soc_card, wm8750_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm8750_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 77688e2..18e8e4b 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -211,7 +211,7 @@ static int tosa_init(struct snd_soc_card *soc_card) struct snd_ac97_bus_ops *ac97_ops; int ret; - codec = snd_soc_card_get_codec(soc_card, wm9712_codec_id); + codec = snd_soc_card_get_codec(soc_card, wm9712_codec_id, 0); if (codec == NULL) return -ENODEV; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index a7e8ab7..4209fc2 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -93,7 +93,7 @@ static int zylonite_init(struct snd_soc_card *card) struct snd_ac97_bus_ops *ac97_ops; int ret; - codec = snd_soc_card_get_codec(card, wm9713_codec_id); + codec = snd_soc_card_get_codec(card, wm9713_codec_id, 0); if (codec == NULL) { printk(KERN_ERR "Unable to obtain WM9713 codec\n"); return -ENODEV; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 520469b..a88e3a0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2172,12 +2172,13 @@ EXPORT_SYMBOL_GPL(snd_soc_card_free); * Get a codec from a codec ID. */ struct snd_soc_codec *snd_soc_card_get_codec(struct snd_soc_card *soc_card, - const char *codec_id) + const char *codec_name, int codec_num) { struct soc_pcm_config *config; list_for_each_entry(config, &soc_card->config_list, list) { - if (config->codec && !strcmp(config->codec->name, codec_id)) + if (config->codec && !strcmp(config->codec->name, codec_name) + && config->codec->num == codec_num) return config->codec; } return NULL; -- 1.5.5

2 1

[alsa-devel] [PATCH] ASoC: Fix default mono mixer configuration for WM8510
by Mark Brown 06 Jun '08

06 Jun '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8510.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 6f54cb0..152e6f2 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -71,7 +71,7 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { 0x0003, 0x0010, 0x0000, 0x0000, 0x0000, 0x0002, 0x0001, 0x0000, 0x0000, 0x0000, 0x0039, 0x0000, - 0x0000, + 0x0001, }; /* -- 1.5.5.3

1 0

[alsa-devel] [PATCH 1/3] ASoC: Add WM8510 driver
by Mark Brown 06 Jun '08

06 Jun '08

The WM8510 is a mono CODEC with speaker driver optimised for telephony applications, featuring: - 16/20/24/32 bit audio at data rates between 8kHz and 48kHz - On-chip PLL - Dual microphone inputs This driver was originally written by Liam Girdwood with updates from Brett Saunders, Geoffrey Wossum and myself. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Signed-off-by: Brett Saunders <breton.saunders(a)ntlworld.com> Signed-off-by: Geoffrey Wossum <geoffrey(a)pager.net> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8510.c | 836 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8510.h | 103 ++++++ 4 files changed, 944 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm8510.c create mode 100644 sound/soc/codecs/wm8510.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7beefcc..ef400b3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -5,6 +5,9 @@ config SND_SOC_AC97_CODEC config SND_SOC_UDA1380 tristate +config SND_SOC_WM8510 + tristate + config SND_SOC_WM8731 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d5926a1..b92c665 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,5 +1,6 @@ snd-soc-ac97-objs := ac97.o snd-soc-uda1380-objs := uda1380.o +snd-soc-wm8510-objs := wm8510.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o @@ -10,6 +11,7 @@ snd-soc-tlv320aic3x-objs := tlv320aic3x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o +obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c new file mode 100644 index 0000000..6f54cb0 --- /dev/null +++ b/sound/soc/codecs/wm8510.c @@ -0,0 +1,836 @@ +/* + * wm8510.c -- WM8510 ALSA Soc Audio driver + * + * Copyright 2006 Wolfson Microelectronics PLC. + * + * Author: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "wm8510.h" + +#define AUDIO_NAME "wm8510" +#define WM8510_VERSION "0.6" + +/* + * Debug + */ + +#define WM8510_DEBUG 0 + +#ifdef WM8510_DEBUG +#define dbg(format, arg...) \ + printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) +#else +#define dbg(format, arg...) do {} while (0) +#endif +#define err(format, arg...) \ + printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) +#define info(format, arg...) \ + printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) +#define warn(format, arg...) \ + printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) + +struct snd_soc_codec_device soc_codec_dev_wm8510; + +/* + * wm8510 register cache + * We can't read the WM8510 register space when we are + * using 2 wire for device control, so we cache them instead. + */ +static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0050, 0x0000, 0x0140, 0x0000, + 0x0000, 0x0000, 0x0000, 0x00ff, + 0x0000, 0x0000, 0x0100, 0x00ff, + 0x0000, 0x0000, 0x012c, 0x002c, + 0x002c, 0x002c, 0x002c, 0x0000, + 0x0032, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0038, 0x000b, 0x0032, 0x0000, + 0x0008, 0x000c, 0x0093, 0x00e9, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0003, 0x0010, 0x0000, 0x0000, + 0x0000, 0x0002, 0x0001, 0x0000, + 0x0000, 0x0000, 0x0039, 0x0000, + 0x0000, +}; + +/* + * read wm8510 register cache + */ +static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == WM8510_RESET) + return 0; + if (reg >= WM8510_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write wm8510 register cache + */ +static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8510_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the WM8510 register space + */ +static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8510 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8510_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define wm8510_reset(c) wm8510_write(c, WM8510_RESET, 0) + +static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" }; +static const char *wm8510_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; +static const char *wm8510_alc[] = { "ALC", "Limiter" }; + +static const struct soc_enum wm8510_enum[] = { + SOC_ENUM_SINGLE(WM8510_COMP, 1, 4, wm8510_companding), /* adc */ + SOC_ENUM_SINGLE(WM8510_COMP, 3, 4, wm8510_companding), /* dac */ + SOC_ENUM_SINGLE(WM8510_DAC, 4, 4, wm8510_deemp), + SOC_ENUM_SINGLE(WM8510_ALC3, 8, 2, wm8510_alc), +}; + +static const struct snd_kcontrol_new wm8510_snd_controls[] = { + +SOC_SINGLE("Digital Loopback Switch", WM8510_COMP, 0, 1, 0), + +SOC_ENUM("DAC Companding", wm8510_enum[1]), +SOC_ENUM("ADC Companding", wm8510_enum[0]), + +SOC_ENUM("Playback De-emphasis", wm8510_enum[2]), +SOC_SINGLE("DAC Inversion Switch", WM8510_DAC, 0, 1, 0), + +SOC_SINGLE("Master Playback Volume", WM8510_DACVOL, 0, 127, 0), + +SOC_SINGLE("High Pass Filter Switch", WM8510_ADC, 8, 1, 0), +SOC_SINGLE("High Pass Cut Off", WM8510_ADC, 4, 7, 0), +SOC_SINGLE("ADC Inversion Switch", WM8510_COMP, 0, 1, 0), + +SOC_SINGLE("Capture Volume", WM8510_ADCVOL, 0, 127, 0), + +SOC_SINGLE("DAC Playback Limiter Switch", WM8510_DACLIM1, 8, 1, 0), +SOC_SINGLE("DAC Playback Limiter Decay", WM8510_DACLIM1, 4, 15, 0), +SOC_SINGLE("DAC Playback Limiter Attack", WM8510_DACLIM1, 0, 15, 0), + +SOC_SINGLE("DAC Playback Limiter Threshold", WM8510_DACLIM2, 4, 7, 0), +SOC_SINGLE("DAC Playback Limiter Boost", WM8510_DACLIM2, 0, 15, 0), + +SOC_SINGLE("ALC Enable Switch", WM8510_ALC1, 8, 1, 0), +SOC_SINGLE("ALC Capture Max Gain", WM8510_ALC1, 3, 7, 0), +SOC_SINGLE("ALC Capture Min Gain", WM8510_ALC1, 0, 7, 0), + +SOC_SINGLE("ALC Capture ZC Switch", WM8510_ALC2, 8, 1, 0), +SOC_SINGLE("ALC Capture Hold", WM8510_ALC2, 4, 7, 0), +SOC_SINGLE("ALC Capture Target", WM8510_ALC2, 0, 15, 0), + +SOC_ENUM("ALC Capture Mode", wm8510_enum[3]), +SOC_SINGLE("ALC Capture Decay", WM8510_ALC3, 4, 15, 0), +SOC_SINGLE("ALC Capture Attack", WM8510_ALC3, 0, 15, 0), + +SOC_SINGLE("ALC Capture Noise Gate Switch", WM8510_NGATE, 3, 1, 0), +SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8510_NGATE, 0, 7, 0), + +SOC_SINGLE("Capture PGA ZC Switch", WM8510_INPPGA, 7, 1, 0), +SOC_SINGLE("Capture PGA Volume", WM8510_INPPGA, 0, 63, 0), + +SOC_SINGLE("Speaker Playback ZC Switch", WM8510_SPKVOL, 7, 1, 0), +SOC_SINGLE("Speaker Playback Switch", WM8510_SPKVOL, 6, 1, 1), +SOC_SINGLE("Speaker Playback Volume", WM8510_SPKVOL, 0, 63, 0), +SOC_SINGLE("Speaker Boost", WM8510_OUTPUT, 2, 1, 0), + +SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0), +SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1), +}; + +/* add non dapm controls */ +static int wm8510_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8510_snd_controls[i], codec, + NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Speaker Output Mixer */ +static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 0), +}; + +/* Mono Output Mixer */ +static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8510_boost_controls[] = { +SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 0), +SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0), +SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0), +}; + +static const struct snd_kcontrol_new wm8510_micpga_controls[] = { +SOC_DAPM_SINGLE("MICP Switch", WM8510_INPUT, 0, 1, 0), +SOC_DAPM_SINGLE("MICN Switch", WM8510_INPUT, 1, 1, 0), +SOC_DAPM_SINGLE("AUX Switch", WM8510_INPUT, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8510_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Speaker Mixer", WM8510_POWER3, 2, 0, + &wm8510_speaker_mixer_controls[0], + ARRAY_SIZE(wm8510_speaker_mixer_controls)), +SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_POWER3, 3, 0, + &wm8510_mono_mixer_controls[0], + ARRAY_SIZE(wm8510_mono_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER2, 0, 0), +SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, + &wm8510_micpga_controls[0], + ARRAY_SIZE(wm8510_micpga_controls)), +SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, + &wm8510_boost_controls[0], + ARRAY_SIZE(wm8510_boost_controls)), + +SND_SOC_DAPM_MICBIAS("Mic Bias", WM8510_POWER1, 4, 0), + +SND_SOC_DAPM_INPUT("MICN"), +SND_SOC_DAPM_INPUT("MICP"), +SND_SOC_DAPM_INPUT("AUX"), +SND_SOC_DAPM_OUTPUT("MONOOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, + {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, + {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Microphone PGA */ + {"Mic PGA", "MICN Switch", "MICN"}, + {"Mic PGA", "MICP Switch", "MICP"}, + { "Mic PGA", "AUX Switch", "Aux Input" }, + + /* Boost Mixer */ + {"Boost Mixer", "Mic PGA Switch", "Mic PGA"}, + {"Boost Mixer", "Mic Volume", "MICP"}, + {"Boost Mixer", "Aux Volume", "Aux Input"}, + + {"ADC", NULL, "Boost Mixer"}, +}; + +static int wm8510_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8510_dapm_widgets, + ARRAY_SIZE(wm8510_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +struct pll_ { + unsigned int pre_div:4; /* prescale - 1 */ + unsigned int n:4; + unsigned int k; +}; + +static struct pll_ pll_div; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 24) * 10) + +static void pll_factors(unsigned int target, unsigned int source) +{ + unsigned long long Kpart; + unsigned int K, Ndiv, Nmod; + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div.pre_div = 1; + Ndiv = target / source; + } else + pll_div.pre_div = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8510 N value %d outwith recommended range!d\n", + Ndiv); + + pll_div.n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div.k = K; +} + +static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + if (freq_in == 0 || freq_out == 0) { + /* Clock CODEC directly from MCLK */ + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); + wm8510_write(codec, WM8510_CLOCK, reg & 0x0ff); + + /* Turn off PLL */ + reg = wm8510_read_reg_cache(codec, WM8510_POWER1); + wm8510_write(codec, WM8510_POWER1, reg & 0x1df); + return 0; + } + + pll_factors(freq_out*8, freq_in); + + wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); + wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18); + wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff); + wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff); + reg = wm8510_read_reg_cache(codec, WM8510_POWER1); + wm8510_write(codec, WM8510_POWER1, reg | 0x020); + + /* Run CODEC from PLL instead of MCLK */ + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); + wm8510_write(codec, WM8510_CLOCK, reg | 0x100); + + return 0; +} + +/* + * Configure WM8510 clock dividers. + */ +static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8510_OPCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf; + wm8510_write(codec, WM8510_GPIO, reg | div); + break; + case WM8510_MCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f; + wm8510_write(codec, WM8510_CLOCK, reg | div); + break; + case WM8510_ADCCLK: + reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7; + wm8510_write(codec, WM8510_ADC, reg | div); + break; + case WM8510_DACCLK: + reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7; + wm8510_write(codec, WM8510_DAC, reg | div); + break; + case WM8510_BCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3; + wm8510_write(codec, WM8510_CLOCK, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int wm8510_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + clk |= 0x0001; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0010; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0008; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x00018; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0180; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0100; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0080; + break; + default: + return -EINVAL; + } + + wm8510_write(codec, WM8510_IFACE, iface); + wm8510_write(codec, WM8510_CLOCK, clk); + return 0; +} + +static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f; + u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0020; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0040; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x0060; + break; + } + + /* filter coefficient */ + switch (params_rate(params)) { + case SNDRV_PCM_RATE_8000: + adn |= 0x5 << 1; + break; + case SNDRV_PCM_RATE_11025: + adn |= 0x4 << 1; + break; + case SNDRV_PCM_RATE_16000: + adn |= 0x3 << 1; + break; + case SNDRV_PCM_RATE_22050: + adn |= 0x2 << 1; + break; + case SNDRV_PCM_RATE_32000: + adn |= 0x1 << 1; + break; + case SNDRV_PCM_RATE_44100: + case SNDRV_PCM_RATE_48000: + break; + } + + wm8510_write(codec, WM8510_IFACE, iface); + wm8510_write(codec, WM8510_ADD, adn); + return 0; +} + +static int wm8510_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; + + if (mute) + wm8510_write(codec, WM8510_DAC, mute_reg | 0x40); + else + wm8510_write(codec, WM8510_DAC, mute_reg); + return 0; +} + +/* liam need to make this lower power with dapm */ +static int wm8510_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + + switch (level) { + case SND_SOC_BIAS_ON: + wm8510_write(codec, WM8510_POWER1, 0x1ff); + wm8510_write(codec, WM8510_POWER2, 0x1ff); + wm8510_write(codec, WM8510_POWER3, 0x1ff); + break; + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + wm8510_write(codec, WM8510_POWER1, 0x0); + wm8510_write(codec, WM8510_POWER2, 0x0); + wm8510_write(codec, WM8510_POWER3, 0x0); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8510_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +#define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_codec_dai wm8510_dai = { + .name = "WM8510 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8510_RATES, + .formats = WM8510_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8510_RATES, + .formats = WM8510_FORMATS,}, + .ops = { + .hw_params = wm8510_pcm_hw_params, + }, + .dai_ops = { + .digital_mute = wm8510_mute, + .set_fmt = wm8510_set_dai_fmt, + .set_clkdiv = wm8510_set_dai_clkdiv, + .set_pll = wm8510_set_dai_pll, + }, +}; +EXPORT_SYMBOL_GPL(wm8510_dai); + +static int wm8510_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8510_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8510_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the WM8510 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8510_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "WM8510"; + codec->owner = THIS_MODULE; + codec->read = wm8510_read_reg_cache; + codec->write = wm8510_write; + codec->set_bias_level = wm8510_set_bias_level; + codec->dai = &wm8510_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm8510_reg); + codec->reg_cache = kmemdup(wm8510_reg, sizeof(wm8510_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + wm8510_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8510_add_controls(codec); + wm8510_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to register card\n"); + goto card_err; + } + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *wm8510_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM8510 2 wire address is 0x1a + */ +#define I2C_DRIVERID_WM8510 0xfefe /* liam - need a proper id */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver wm8510_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ + +static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = wm8510_socdev; + struct wm8510_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + err("failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = wm8510_init(socdev); + if (ret < 0) { + err("failed to initialise WM8510\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int wm8510_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int wm8510_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, wm8510_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver wm8510_i2c_driver = { + .driver = { + .name = "WM8510 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_WM8510, + .attach_adapter = wm8510_i2c_attach, + .detach_client = wm8510_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "WM8510", + .driver = &wm8510_i2c_driver, +}; +#endif + +static int wm8510_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8510_setup_data *setup; + struct snd_soc_codec *codec; + int ret = 0; + + info("WM8510 Audio Codec %s", WM8510_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + wm8510_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&wm8510_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int wm8510_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8510_i2c_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8510 = { + .probe = wm8510_probe, + .remove = wm8510_remove, + .suspend = wm8510_suspend, + .resume = wm8510_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510); + +MODULE_DESCRIPTION("ASoC WM8510 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h new file mode 100644 index 0000000..c862e7b --- /dev/null +++ b/sound/soc/codecs/wm8510.h @@ -0,0 +1,103 @@ +/* + * wm8510.h -- WM8510 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8510_H +#define _WM8510_H + +/* WM8510 register space */ + +#define WM8510_RESET 0x0 +#define WM8510_POWER1 0x1 +#define WM8510_POWER2 0x2 +#define WM8510_POWER3 0x3 +#define WM8510_IFACE 0x4 +#define WM8510_COMP 0x5 +#define WM8510_CLOCK 0x6 +#define WM8510_ADD 0x7 +#define WM8510_GPIO 0x8 +#define WM8510_DAC 0xa +#define WM8510_DACVOL 0xb +#define WM8510_ADC 0xe +#define WM8510_ADCVOL 0xf +#define WM8510_EQ1 0x12 +#define WM8510_EQ2 0x13 +#define WM8510_EQ3 0x14 +#define WM8510_EQ4 0x15 +#define WM8510_EQ5 0x16 +#define WM8510_DACLIM1 0x18 +#define WM8510_DACLIM2 0x19 +#define WM8510_NOTCH1 0x1b +#define WM8510_NOTCH2 0x1c +#define WM8510_NOTCH3 0x1d +#define WM8510_NOTCH4 0x1e +#define WM8510_ALC1 0x20 +#define WM8510_ALC2 0x21 +#define WM8510_ALC3 0x22 +#define WM8510_NGATE 0x23 +#define WM8510_PLLN 0x24 +#define WM8510_PLLK1 0x25 +#define WM8510_PLLK2 0x26 +#define WM8510_PLLK3 0x27 +#define WM8510_ATTEN 0x28 +#define WM8510_INPUT 0x2c +#define WM8510_INPPGA 0x2d +#define WM8510_ADCBOOST 0x2f +#define WM8510_OUTPUT 0x31 +#define WM8510_SPKMIX 0x32 +#define WM8510_SPKVOL 0x36 +#define WM8510_MONOMIX 0x38 + +#define WM8510_CACHEREGNUM 57 + +/* Clock divider Id's */ +#define WM8510_OPCLKDIV 0 +#define WM8510_MCLKDIV 1 +#define WM8510_ADCCLK 2 +#define WM8510_DACCLK 3 +#define WM8510_BCLKDIV 4 + +/* DAC clock dividers */ +#define WM8510_DACCLK_F2 (1 << 3) +#define WM8510_DACCLK_F4 (0 << 3) + +/* ADC clock dividers */ +#define WM8510_ADCCLK_F2 (1 << 3) +#define WM8510_ADCCLK_F4 (0 << 3) + +/* PLL Out dividers */ +#define WM8510_OPCLKDIV_1 (0 << 4) +#define WM8510_OPCLKDIV_2 (1 << 4) +#define WM8510_OPCLKDIV_3 (2 << 4) +#define WM8510_OPCLKDIV_4 (3 << 4) + +/* BCLK clock dividers */ +#define WM8510_BCLKDIV_1 (0 << 2) +#define WM8510_BCLKDIV_2 (1 << 2) +#define WM8510_BCLKDIV_4 (2 << 2) +#define WM8510_BCLKDIV_8 (3 << 2) +#define WM8510_BCLKDIV_16 (4 << 2) +#define WM8510_BCLKDIV_32 (5 << 2) + +/* MCLK clock dividers */ +#define WM8510_MCLKDIV_1 (0 << 5) +#define WM8510_MCLKDIV_1_5 (1 << 5) +#define WM8510_MCLKDIV_2 (2 << 5) +#define WM8510_MCLKDIV_3 (3 << 5) +#define WM8510_MCLKDIV_4 (4 << 5) +#define WM8510_MCLKDIV_6 (5 << 5) +#define WM8510_MCLKDIV_8 (6 << 5) +#define WM8510_MCLKDIV_12 (7 << 5) + +struct wm8510_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_codec_dai wm8510_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8510; + +#endif -- 1.5.5.3

2 3

Re: [alsa-devel] [PATCH] ac97: add support for wm9711 master left inv switch
by Mark Brown 06 Jun '08

06 Jun '08

On Fri, Jun 06, 2008 at 05:02:57PM +0200, Guennadi Liakhovetski wrote: > From: Juergen Beisert <j.beisert(a)pengutronix.de> > > This patch adds support for Master Left Inv Switch on wm9711. > At least required to drive the mono speaker on the PXA270 platfrom > > Signed-off-by: Juergen Beisert <j.beisert(a)pengutronix.de> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Note that ALSA patches should really be CCed to the alsa-devel mailing list (CC added). > sound/pci/ac97/ac97_patch.c | 1 + > 1 file changed, 1 insertion(+) > > Index: linux-2.6/sound/pci/ac97/ac97_patch.c > =================================================================== > --- linux-2.6.orig/sound/pci/ac97/ac97_patch.c 2008-05-26 12:19:23.000000000 +0200 > +++ linux-2.6/sound/pci/ac97/ac97_patch.c 2008-05-27 13:23:11.000000000 +0200 > @@ -669,6 +669,7 @@ > AC97_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), > AC97_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0), > > +AC97_SINGLE("Master Left Inv Switch", AC97_MASTER, 6, 1, 0), > AC97_SINGLE("Master ZC Switch", AC97_MASTER, 7, 1, 0), > AC97_SINGLE("Headphone ZC Switch", AC97_HEADPHONE, 7, 1, 0), > AC97_SINGLE("Mono ZC Switch", AC97_MASTER_MONO, 7, 1, 0), > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

1 0

[alsa-devel] git workflow for casual developers
by Eliot Blennerhassett 06 Jun '08

06 Jun '08

Hi all, So. we are now using git repositories. I'd better find out how to use git... I plan to turn responses to this into a howto here http://alsa-project.org/main/index.php/GIT_Server I.e. what should someone do who is going to clone the repo, make a few changes, then email patches to a maintainer + alsa-devel? 0) clone the repository git clone git://git.alsa-project.org/alsa-driver.git alsa-driver git clone git://git.alsa-project.org/alsa-kmirror.git alsa-kmirror What should I do next if I want to make some changes? Here's what I did, or discovered I should have done :( 1) git-config --add user.email me(a)domain.com Otherwise commits get spurious email address for Author and Commiter (is there any way to fix this after the fact?) 2) should I create a branch in my cloned repo for my changes, or just go ahead and commit? (I didn't create a branch) 3) How do I generate patches for my changes to the repo? Luckily for me there was a tag that was close enough so I could git-diff v1.0.17rc1 Should I really have used this as the "revision"? git-diff remotes/origin/master Then do this to generate the patchset git-format-patch -s -n -p --subject-prefix="PATCH - my module" v1.0.17rc1 4) Check the patches /lib/modules/2.6.24-17-generic/build/scripts/checkpatch.pl --no-tree *.patch 5) Sending the patches Make sure it knows how to send git-config --add sendemail.smtpserver smtp.domain.com git-send-email --from=me --to=tiwai(a)suse.de --cc=alsa-devel(a)alsa-project.org 000*.patch 6) OK, what have I missed? regards -- -- Eliot Blennerhassett www.audioscience.com

3 2

[alsa-devel] 3779 segmentation fault rmmod snd-timer.ko
by Harsha priya gupta 06 Jun '08

06 Jun '08

Hi, When i rmmod all snd modules along with my new driver. It gives the following message when i rmmod snd-timer.ko 3779 segmentation fault rmmod snd-timer.ko...any clues as to what i am doing in my driver that is causing this? -- -Harsha

1 0

[alsa-devel] [PATCH 1/3] ASoC: Add WM8510 driver
by Mark Brown 06 Jun '08

06 Jun '08

The WM8510 is a mono CODEC with speaker driver optimised for telephony applications, featuring: - 16/20/24/32 bit audio at data rates between 8kHz and 48kHz - On-chip PLL - Dual microphone inputs This driver was originally written by Liam Girdwood with updates from Brett Saunders, Geoffrey Wossum and myself. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Signed-off-by: Brett Saunders <breton.saunders(a)ntlworld.com> Signed-off-by: Geoffrey Wossum <geoffrey(a)pager.net> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8510.c | 836 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8510.h | 103 ++++++ 4 files changed, 944 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm8510.c create mode 100644 sound/soc/codecs/wm8510.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7beefcc..ef400b3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -5,6 +5,9 @@ config SND_SOC_AC97_CODEC config SND_SOC_UDA1380 tristate +config SND_SOC_WM8510 + tristate + config SND_SOC_WM8731 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d5926a1..b92c665 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,5 +1,6 @@ snd-soc-ac97-objs := ac97.o snd-soc-uda1380-objs := uda1380.o +snd-soc-wm8510-objs := wm8510.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o @@ -10,6 +11,7 @@ snd-soc-tlv320aic3x-objs := tlv320aic3x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o +obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c new file mode 100644 index 0000000..6f54cb0 --- /dev/null +++ b/sound/soc/codecs/wm8510.c @@ -0,0 +1,836 @@ +/* + * wm8510.c -- WM8510 ALSA Soc Audio driver + * + * Copyright 2006 Wolfson Microelectronics PLC. + * + * Author: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "wm8510.h" + +#define AUDIO_NAME "wm8510" +#define WM8510_VERSION "0.6" + +/* + * Debug + */ + +#define WM8510_DEBUG 0 + +#ifdef WM8510_DEBUG +#define dbg(format, arg...) \ + printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) +#else +#define dbg(format, arg...) do {} while (0) +#endif +#define err(format, arg...) \ + printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) +#define info(format, arg...) \ + printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) +#define warn(format, arg...) \ + printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) + +struct snd_soc_codec_device soc_codec_dev_wm8510; + +/* + * wm8510 register cache + * We can't read the WM8510 register space when we are + * using 2 wire for device control, so we cache them instead. + */ +static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0050, 0x0000, 0x0140, 0x0000, + 0x0000, 0x0000, 0x0000, 0x00ff, + 0x0000, 0x0000, 0x0100, 0x00ff, + 0x0000, 0x0000, 0x012c, 0x002c, + 0x002c, 0x002c, 0x002c, 0x0000, + 0x0032, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0038, 0x000b, 0x0032, 0x0000, + 0x0008, 0x000c, 0x0093, 0x00e9, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0003, 0x0010, 0x0000, 0x0000, + 0x0000, 0x0002, 0x0001, 0x0000, + 0x0000, 0x0000, 0x0039, 0x0000, + 0x0000, +}; + +/* + * read wm8510 register cache + */ +static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == WM8510_RESET) + return 0; + if (reg >= WM8510_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write wm8510 register cache + */ +static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8510_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the WM8510 register space + */ +static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8510 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8510_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define wm8510_reset(c) wm8510_write(c, WM8510_RESET, 0) + +static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" }; +static const char *wm8510_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; +static const char *wm8510_alc[] = { "ALC", "Limiter" }; + +static const struct soc_enum wm8510_enum[] = { + SOC_ENUM_SINGLE(WM8510_COMP, 1, 4, wm8510_companding), /* adc */ + SOC_ENUM_SINGLE(WM8510_COMP, 3, 4, wm8510_companding), /* dac */ + SOC_ENUM_SINGLE(WM8510_DAC, 4, 4, wm8510_deemp), + SOC_ENUM_SINGLE(WM8510_ALC3, 8, 2, wm8510_alc), +}; + +static const struct snd_kcontrol_new wm8510_snd_controls[] = { + +SOC_SINGLE("Digital Loopback Switch", WM8510_COMP, 0, 1, 0), + +SOC_ENUM("DAC Companding", wm8510_enum[1]), +SOC_ENUM("ADC Companding", wm8510_enum[0]), + +SOC_ENUM("Playback De-emphasis", wm8510_enum[2]), +SOC_SINGLE("DAC Inversion Switch", WM8510_DAC, 0, 1, 0), + +SOC_SINGLE("Master Playback Volume", WM8510_DACVOL, 0, 127, 0), + +SOC_SINGLE("High Pass Filter Switch", WM8510_ADC, 8, 1, 0), +SOC_SINGLE("High Pass Cut Off", WM8510_ADC, 4, 7, 0), +SOC_SINGLE("ADC Inversion Switch", WM8510_COMP, 0, 1, 0), + +SOC_SINGLE("Capture Volume", WM8510_ADCVOL, 0, 127, 0), + +SOC_SINGLE("DAC Playback Limiter Switch", WM8510_DACLIM1, 8, 1, 0), +SOC_SINGLE("DAC Playback Limiter Decay", WM8510_DACLIM1, 4, 15, 0), +SOC_SINGLE("DAC Playback Limiter Attack", WM8510_DACLIM1, 0, 15, 0), + +SOC_SINGLE("DAC Playback Limiter Threshold", WM8510_DACLIM2, 4, 7, 0), +SOC_SINGLE("DAC Playback Limiter Boost", WM8510_DACLIM2, 0, 15, 0), + +SOC_SINGLE("ALC Enable Switch", WM8510_ALC1, 8, 1, 0), +SOC_SINGLE("ALC Capture Max Gain", WM8510_ALC1, 3, 7, 0), +SOC_SINGLE("ALC Capture Min Gain", WM8510_ALC1, 0, 7, 0), + +SOC_SINGLE("ALC Capture ZC Switch", WM8510_ALC2, 8, 1, 0), +SOC_SINGLE("ALC Capture Hold", WM8510_ALC2, 4, 7, 0), +SOC_SINGLE("ALC Capture Target", WM8510_ALC2, 0, 15, 0), + +SOC_ENUM("ALC Capture Mode", wm8510_enum[3]), +SOC_SINGLE("ALC Capture Decay", WM8510_ALC3, 4, 15, 0), +SOC_SINGLE("ALC Capture Attack", WM8510_ALC3, 0, 15, 0), + +SOC_SINGLE("ALC Capture Noise Gate Switch", WM8510_NGATE, 3, 1, 0), +SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8510_NGATE, 0, 7, 0), + +SOC_SINGLE("Capture PGA ZC Switch", WM8510_INPPGA, 7, 1, 0), +SOC_SINGLE("Capture PGA Volume", WM8510_INPPGA, 0, 63, 0), + +SOC_SINGLE("Speaker Playback ZC Switch", WM8510_SPKVOL, 7, 1, 0), +SOC_SINGLE("Speaker Playback Switch", WM8510_SPKVOL, 6, 1, 1), +SOC_SINGLE("Speaker Playback Volume", WM8510_SPKVOL, 0, 63, 0), +SOC_SINGLE("Speaker Boost", WM8510_OUTPUT, 2, 1, 0), + +SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0), +SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1), +}; + +/* add non dapm controls */ +static int wm8510_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8510_snd_controls[i], codec, + NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Speaker Output Mixer */ +static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 0), +}; + +/* Mono Output Mixer */ +static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8510_boost_controls[] = { +SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 0), +SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0), +SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0), +}; + +static const struct snd_kcontrol_new wm8510_micpga_controls[] = { +SOC_DAPM_SINGLE("MICP Switch", WM8510_INPUT, 0, 1, 0), +SOC_DAPM_SINGLE("MICN Switch", WM8510_INPUT, 1, 1, 0), +SOC_DAPM_SINGLE("AUX Switch", WM8510_INPUT, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8510_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Speaker Mixer", WM8510_POWER3, 2, 0, + &wm8510_speaker_mixer_controls[0], + ARRAY_SIZE(wm8510_speaker_mixer_controls)), +SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_POWER3, 3, 0, + &wm8510_mono_mixer_controls[0], + ARRAY_SIZE(wm8510_mono_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER2, 0, 0), +SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, + &wm8510_micpga_controls[0], + ARRAY_SIZE(wm8510_micpga_controls)), +SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, + &wm8510_boost_controls[0], + ARRAY_SIZE(wm8510_boost_controls)), + +SND_SOC_DAPM_MICBIAS("Mic Bias", WM8510_POWER1, 4, 0), + +SND_SOC_DAPM_INPUT("MICN"), +SND_SOC_DAPM_INPUT("MICP"), +SND_SOC_DAPM_INPUT("AUX"), +SND_SOC_DAPM_OUTPUT("MONOOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, + {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, + {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Microphone PGA */ + {"Mic PGA", "MICN Switch", "MICN"}, + {"Mic PGA", "MICP Switch", "MICP"}, + { "Mic PGA", "AUX Switch", "Aux Input" }, + + /* Boost Mixer */ + {"Boost Mixer", "Mic PGA Switch", "Mic PGA"}, + {"Boost Mixer", "Mic Volume", "MICP"}, + {"Boost Mixer", "Aux Volume", "Aux Input"}, + + {"ADC", NULL, "Boost Mixer"}, +}; + +static int wm8510_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8510_dapm_widgets, + ARRAY_SIZE(wm8510_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +struct pll_ { + unsigned int pre_div:4; /* prescale - 1 */ + unsigned int n:4; + unsigned int k; +}; + +static struct pll_ pll_div; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 24) * 10) + +static void pll_factors(unsigned int target, unsigned int source) +{ + unsigned long long Kpart; + unsigned int K, Ndiv, Nmod; + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div.pre_div = 1; + Ndiv = target / source; + } else + pll_div.pre_div = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8510 N value %d outwith recommended range!d\n", + Ndiv); + + pll_div.n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div.k = K; +} + +static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + if (freq_in == 0 || freq_out == 0) { + /* Clock CODEC directly from MCLK */ + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); + wm8510_write(codec, WM8510_CLOCK, reg & 0x0ff); + + /* Turn off PLL */ + reg = wm8510_read_reg_cache(codec, WM8510_POWER1); + wm8510_write(codec, WM8510_POWER1, reg & 0x1df); + return 0; + } + + pll_factors(freq_out*8, freq_in); + + wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); + wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18); + wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff); + wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff); + reg = wm8510_read_reg_cache(codec, WM8510_POWER1); + wm8510_write(codec, WM8510_POWER1, reg | 0x020); + + /* Run CODEC from PLL instead of MCLK */ + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); + wm8510_write(codec, WM8510_CLOCK, reg | 0x100); + + return 0; +} + +/* + * Configure WM8510 clock dividers. + */ +static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8510_OPCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf; + wm8510_write(codec, WM8510_GPIO, reg | div); + break; + case WM8510_MCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f; + wm8510_write(codec, WM8510_CLOCK, reg | div); + break; + case WM8510_ADCCLK: + reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7; + wm8510_write(codec, WM8510_ADC, reg | div); + break; + case WM8510_DACCLK: + reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7; + wm8510_write(codec, WM8510_DAC, reg | div); + break; + case WM8510_BCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3; + wm8510_write(codec, WM8510_CLOCK, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int wm8510_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + clk |= 0x0001; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0010; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0008; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x00018; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0180; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0100; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0080; + break; + default: + return -EINVAL; + } + + wm8510_write(codec, WM8510_IFACE, iface); + wm8510_write(codec, WM8510_CLOCK, clk); + return 0; +} + +static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f; + u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0020; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0040; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x0060; + break; + } + + /* filter coefficient */ + switch (params_rate(params)) { + case SNDRV_PCM_RATE_8000: + adn |= 0x5 << 1; + break; + case SNDRV_PCM_RATE_11025: + adn |= 0x4 << 1; + break; + case SNDRV_PCM_RATE_16000: + adn |= 0x3 << 1; + break; + case SNDRV_PCM_RATE_22050: + adn |= 0x2 << 1; + break; + case SNDRV_PCM_RATE_32000: + adn |= 0x1 << 1; + break; + case SNDRV_PCM_RATE_44100: + case SNDRV_PCM_RATE_48000: + break; + } + + wm8510_write(codec, WM8510_IFACE, iface); + wm8510_write(codec, WM8510_ADD, adn); + return 0; +} + +static int wm8510_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; + + if (mute) + wm8510_write(codec, WM8510_DAC, mute_reg | 0x40); + else + wm8510_write(codec, WM8510_DAC, mute_reg); + return 0; +} + +/* liam need to make this lower power with dapm */ +static int wm8510_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + + switch (level) { + case SND_SOC_BIAS_ON: + wm8510_write(codec, WM8510_POWER1, 0x1ff); + wm8510_write(codec, WM8510_POWER2, 0x1ff); + wm8510_write(codec, WM8510_POWER3, 0x1ff); + break; + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + wm8510_write(codec, WM8510_POWER1, 0x0); + wm8510_write(codec, WM8510_POWER2, 0x0); + wm8510_write(codec, WM8510_POWER3, 0x0); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8510_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +#define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_codec_dai wm8510_dai = { + .name = "WM8510 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8510_RATES, + .formats = WM8510_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8510_RATES, + .formats = WM8510_FORMATS,}, + .ops = { + .hw_params = wm8510_pcm_hw_params, + }, + .dai_ops = { + .digital_mute = wm8510_mute, + .set_fmt = wm8510_set_dai_fmt, + .set_clkdiv = wm8510_set_dai_clkdiv, + .set_pll = wm8510_set_dai_pll, + }, +}; +EXPORT_SYMBOL_GPL(wm8510_dai); + +static int wm8510_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8510_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8510_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the WM8510 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8510_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "WM8510"; + codec->owner = THIS_MODULE; + codec->read = wm8510_read_reg_cache; + codec->write = wm8510_write; + codec->set_bias_level = wm8510_set_bias_level; + codec->dai = &wm8510_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm8510_reg); + codec->reg_cache = kmemdup(wm8510_reg, sizeof(wm8510_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + wm8510_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8510_add_controls(codec); + wm8510_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to register card\n"); + goto card_err; + } + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *wm8510_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM8510 2 wire address is 0x1a + */ +#define I2C_DRIVERID_WM8510 0xfefe /* liam - need a proper id */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver wm8510_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ + +static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = wm8510_socdev; + struct wm8510_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + err("failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = wm8510_init(socdev); + if (ret < 0) { + err("failed to initialise WM8510\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int wm8510_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int wm8510_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, wm8510_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver wm8510_i2c_driver = { + .driver = { + .name = "WM8510 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_WM8510, + .attach_adapter = wm8510_i2c_attach, + .detach_client = wm8510_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "WM8510", + .driver = &wm8510_i2c_driver, +}; +#endif + +static int wm8510_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8510_setup_data *setup; + struct snd_soc_codec *codec; + int ret = 0; + + info("WM8510 Audio Codec %s", WM8510_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + wm8510_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&wm8510_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int wm8510_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8510_i2c_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8510 = { + .probe = wm8510_probe, + .remove = wm8510_remove, + .suspend = wm8510_suspend, + .resume = wm8510_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510); + +MODULE_DESCRIPTION("ASoC WM8510 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h new file mode 100644 index 0000000..c862e7b --- /dev/null +++ b/sound/soc/codecs/wm8510.h @@ -0,0 +1,103 @@ +/* + * wm8510.h -- WM8510 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8510_H +#define _WM8510_H + +/* WM8510 register space */ + +#define WM8510_RESET 0x0 +#define WM8510_POWER1 0x1 +#define WM8510_POWER2 0x2 +#define WM8510_POWER3 0x3 +#define WM8510_IFACE 0x4 +#define WM8510_COMP 0x5 +#define WM8510_CLOCK 0x6 +#define WM8510_ADD 0x7 +#define WM8510_GPIO 0x8 +#define WM8510_DAC 0xa +#define WM8510_DACVOL 0xb +#define WM8510_ADC 0xe +#define WM8510_ADCVOL 0xf +#define WM8510_EQ1 0x12 +#define WM8510_EQ2 0x13 +#define WM8510_EQ3 0x14 +#define WM8510_EQ4 0x15 +#define WM8510_EQ5 0x16 +#define WM8510_DACLIM1 0x18 +#define WM8510_DACLIM2 0x19 +#define WM8510_NOTCH1 0x1b +#define WM8510_NOTCH2 0x1c +#define WM8510_NOTCH3 0x1d +#define WM8510_NOTCH4 0x1e +#define WM8510_ALC1 0x20 +#define WM8510_ALC2 0x21 +#define WM8510_ALC3 0x22 +#define WM8510_NGATE 0x23 +#define WM8510_PLLN 0x24 +#define WM8510_PLLK1 0x25 +#define WM8510_PLLK2 0x26 +#define WM8510_PLLK3 0x27 +#define WM8510_ATTEN 0x28 +#define WM8510_INPUT 0x2c +#define WM8510_INPPGA 0x2d +#define WM8510_ADCBOOST 0x2f +#define WM8510_OUTPUT 0x31 +#define WM8510_SPKMIX 0x32 +#define WM8510_SPKVOL 0x36 +#define WM8510_MONOMIX 0x38 + +#define WM8510_CACHEREGNUM 57 + +/* Clock divider Id's */ +#define WM8510_OPCLKDIV 0 +#define WM8510_MCLKDIV 1 +#define WM8510_ADCCLK 2 +#define WM8510_DACCLK 3 +#define WM8510_BCLKDIV 4 + +/* DAC clock dividers */ +#define WM8510_DACCLK_F2 (1 << 3) +#define WM8510_DACCLK_F4 (0 << 3) + +/* ADC clock dividers */ +#define WM8510_ADCCLK_F2 (1 << 3) +#define WM8510_ADCCLK_F4 (0 << 3) + +/* PLL Out dividers */ +#define WM8510_OPCLKDIV_1 (0 << 4) +#define WM8510_OPCLKDIV_2 (1 << 4) +#define WM8510_OPCLKDIV_3 (2 << 4) +#define WM8510_OPCLKDIV_4 (3 << 4) + +/* BCLK clock dividers */ +#define WM8510_BCLKDIV_1 (0 << 2) +#define WM8510_BCLKDIV_2 (1 << 2) +#define WM8510_BCLKDIV_4 (2 << 2) +#define WM8510_BCLKDIV_8 (3 << 2) +#define WM8510_BCLKDIV_16 (4 << 2) +#define WM8510_BCLKDIV_32 (5 << 2) + +/* MCLK clock dividers */ +#define WM8510_MCLKDIV_1 (0 << 5) +#define WM8510_MCLKDIV_1_5 (1 << 5) +#define WM8510_MCLKDIV_2 (2 << 5) +#define WM8510_MCLKDIV_3 (3 << 5) +#define WM8510_MCLKDIV_4 (4 << 5) +#define WM8510_MCLKDIV_6 (5 << 5) +#define WM8510_MCLKDIV_8 (6 << 5) +#define WM8510_MCLKDIV_12 (7 << 5) + +struct wm8510_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_codec_dai wm8510_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8510; + +#endif -- 1.5.5.3

2 1

[alsa-devel] aplay works but not real player
by Harsha priya gupta 06 Jun '08

06 Jun '08

I am using .copy function in my driver. aplay playes any .wav file given. When i open real player and try playing it says device is busy. I tried the earlier advice to write the following into ~./asoundrc but does not help pcm.!default{ type hw card 0 } any clue that can help me ? -- -Harsha

2 2

[alsa-devel] git down?
by ben.stanley＠exemail.com.au 06 Jun '08

06 Jun '08

I'm having trouble with git: ben@mythtv:~/alsa-git$ git clone git://git.alsa-project.org/alsa-driver.git alsa-driver Initialized empty Git repository in /home/ben/alsa-git/alsa-driver/.git/ fatal: The remote end hung up unexpectedly fetch-pack from 'git://git.alsa-project.org/alsa-driver.git' failed. ben@mythtv:~/alsa-git$ Anyone else seeing this? Ben Stanley.

4 4

[alsa-devel] [2.6 patch] ALSA: make sparc/dbri.c:snd_dbri_proc() static
by Adrian Bunk 05 Jun '08

05 Jun '08

This patch makes the needlessly global snd_dbri_proc() static. Signed-off-by: Adrian Bunk <bunk(a)kernel.org> --- 5e6dd8dea3d11e2be46a77f8e731780f0a7b7591 diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 3d00e07..ee2e1b4 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2490,7 +2490,7 @@ static void dbri_debug_read(struct snd_info_entry *entry, } #endif -void __devinit snd_dbri_proc(struct snd_card *card) +static void __devinit snd_dbri_proc(struct snd_card *card) { struct snd_dbri *dbri = card->private_data; struct snd_info_entry *entry;

1 0

[alsa-devel] [2.6 patch] usbaudio.c: remove #ifndef CONFIG_USB_EHCI_SPLIT_ISO code
by Adrian Bunk 05 Jun '08

05 Jun '08

Since USB_EHCI_SPLIT_ISO is now unconditionally enabled the #ifndef CONFIG_USB_EHCI_SPLIT_ISO became wrong. Reported-by: Robert P. J. Day <rpjday(a)crashcourse.ca> Signed-off-by: Adrian Bunk <bunk(a)kernel.org> --- e532a1cdc3f2ece89567abd4cd0c219a8e749e3a diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 410be4a..b8cfb7c 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -819,10 +819,6 @@ static const char *usb_error_string(int err) return "device disabled"; case -EHOSTUNREACH: return "device suspended"; -#ifndef CONFIG_USB_EHCI_SPLIT_ISO - case -ENOSYS: - return "enable CONFIG_USB_EHCI_SPLIT_ISO to play through a hub"; -#endif case -EINVAL: case -EAGAIN: case -EFBIG:

1 0

[alsa-devel] [2.6 patch] remove SND_GUS_SYNTH
by Adrian Bunk 05 Jun '08

05 Jun '08

After the removal of the sequencer instrument layer SND_GUS_SYNTH was no longer used. Reported-by: Robert P. J. Day <rpjday(a)crashcourse.ca> Signed-off-by: Adrian Bunk <bunk(a)kernel.org> --- sound/isa/Kconfig | 8 -------- 1 file changed, 8 deletions(-) 9f6d64c22381f740863d4460acb037620b591ed4 diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 2639a6a..ecbf42a 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -204,15 +204,11 @@ config SND_SC6000 To compile this driver as a module, choose M here: the module will be called snd-sc6000. -config SND_GUS_SYNTH - tristate - config SND_GUSCLASSIC tristate "Gravis UltraSound Classic" depends on SND select SND_RAWMIDI select SND_PCM - select SND_GUS_SYNTH help Say Y here to include support for Gravis UltraSound Classic soundcards. @@ -226,7 +222,6 @@ config SND_GUSEXTREME select SND_HWDEP select SND_MPU401_UART select SND_PCM - select SND_GUS_SYNTH help Say Y here to include support for Gravis UltraSound Extreme soundcards. @@ -239,7 +234,6 @@ config SND_GUSMAX depends on SND select SND_RAWMIDI select SND_CS4231_LIB - select SND_GUS_SYNTH help Say Y here to include support for Gravis UltraSound MAX soundcards. @@ -252,7 +246,6 @@ config SND_INTERWAVE depends on SND && PNP && ISA select SND_RAWMIDI select SND_CS4231_LIB - select SND_GUS_SYNTH help Say Y here to include support for AMD InterWave based soundcards (Gravis UltraSound Plug & Play, STB SoundRage32, @@ -266,7 +259,6 @@ config SND_INTERWAVE_STB depends on SND && PNP && ISA select SND_RAWMIDI select SND_CS4231_LIB - select SND_GUS_SYNTH help Say Y here to include support for AMD InterWave based soundcards with a TEA6330T bass and treble regulator

1 0

[alsa-devel] timer + snd_pcm_period_elapsed
by Harsha priya gupta 05 Jun '08

05 Jun '08

Hi, In my driver, i see that first my copy function is called some 4 times. Then my timer starts (when trigger->start is reached). In my timer function, snd_pcm_period_elapsed is called after the period is considered to be processed by hardware. Everytime the period_elapsed function is called, inturn my pointer function is called but i get the following error. can anyone help as to what i am missing to do? aplay: pcm_write:1266: write error: Input/output error Can anyone hint me what i am doing wrong? -- -Harsha

2 1

[alsa-devel] is sound/arm obsolete?
by Maen Suleiman 05 Jun '08

05 Jun '08

Hi all, We are intending to develop a new ALSA driver for an audio unit on Marvell Orion SoC, should a new ALSA drivers written for ARM based SoCs be developed under sound/arm or under sound/soc/newsoc ? (is sound/arm considered as obsolete? ) Thanks in advance

2 1

[alsa-devel] [PATCH 3/3] Revised AT32 ASoC Patch
by Mark Brown 05 Jun '08

05 Jun '08

From: Geoffrey Wossum <geoffrey(a)pager.net> Attached is a revised version of my patch to add AT32 to ASoC. This cleans most of the style issues associated with the previous patch. Also fixes an issue with the playpaq_wm8510.c code depending on a non-released patch to th AT32 portmux support. Patch is against 2.6.24.3.atmel.3 kernel, the latest AVR32 kernel Atmel has released, with the linux-2.6-asoc patches from when v2.6.24 was tagged also applied. [Fixed up minor checkpatch issues and updated for current kernels -- broonie] Signed-off-by: Geoffrey Wossum <gwossum(a)acm.org> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 3 +- sound/soc/at32/Kconfig | 34 ++ sound/soc/at32/Makefile | 11 + sound/soc/at32/at32-pcm.c | 491 ++++++++++++++++++++++ sound/soc/at32/at32-pcm.h | 79 ++++ sound/soc/at32/at32-ssc.c | 849 +++++++++++++++++++++++++++++++++++++++ sound/soc/at32/at32-ssc.h | 59 +++ sound/soc/at32/playpaq_wm8510.c | 524 ++++++++++++++++++++++++ 9 files changed, 2050 insertions(+), 1 deletions(-) create mode 100644 sound/soc/at32/Kconfig create mode 100644 sound/soc/at32/Makefile create mode 100644 sound/soc/at32/at32-pcm.c create mode 100644 sound/soc/at32/at32-pcm.h create mode 100644 sound/soc/at32/at32-ssc.c create mode 100644 sound/soc/at32/at32-ssc.h create mode 100644 sound/soc/at32/playpaq_wm8510.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index fd7bc4f..b939e22 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -22,6 +22,7 @@ config SND_SOC_AC97_BUS bool # All the supported Soc's +source "sound/soc/at32/Kconfig" source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 782db21..3645f95 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/ +obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += omap/ diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig new file mode 100644 index 0000000..b0765e8 --- /dev/null +++ b/sound/soc/at32/Kconfig @@ -0,0 +1,34 @@ +config SND_AT32_SOC + tristate "SoC Audio for the Atmel AT32 System-on-a-Chip" + depends on AVR32 && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the AT32 SSC interface. You will also need to + to select the audio interfaces to support below. + + +config SND_AT32_SOC_SSC + tristate + + + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_AT32_SOC && BOARD_PLAYPAQ + select SND_AT32_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + + + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile new file mode 100644 index 0000000..c03e55e --- /dev/null +++ b/sound/soc/at32/Makefile @@ -0,0 +1,11 @@ +# AT32 Platform Support +snd-soc-at32-objs := at32-pcm.o +snd-soc-at32-ssc-objs := at32-ssc.o + +obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o +obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c new file mode 100644 index 0000000..cf76e89 --- /dev/null +++ b/sound/soc/at32/at32-pcm.c @@ -0,0 +1,491 @@ +/* sound/soc/at32/at32-pcm.c + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-pcm.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "at32-pcm.h" + + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware at32_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE), + + .formats = SNDRV_PCM_FMTBIT_S16, + .period_bytes_min = 32, + .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */ + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct at32_runtime_data { + struct at32_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of DMA buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* Save PDC registers (for power management) */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *dmabuf = &substream->dma_buffer; + size_t size = at32_pcm_hardware.buffer_bytes_max; + + dmabuf->dev.type = SNDRV_DMA_TYPE_DEV; + dmabuf->dev.dev = pcm->card->dev; + dmabuf->private_data = NULL; + dmabuf->area = dma_alloc_coherent(pcm->card->dev, size, + &dmabuf->addr, GFP_KERNEL); + pr_debug("at32_pcm: preallocate_dma_buffer: " + "area=%p, addr=%p, size=%ld\n", + (void *)dmabuf->area, (void *)dmabuf->addr, size); + + if (!dmabuf->area) + return -ENOMEM; + + dmabuf->bytes = size; + return 0; +} + + + +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + static int count; + + count++; + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "underrun" : "overrun", params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + + snd_pcm_period_elapsed(substream); +} + + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int at32_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params + */ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = at32_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("hw_params: DMA for %s initialized " + "(dma_bytes=%ld, period_size=%ld)\n", + prtd->params->name, runtime->dma_bytes, prtd->period_size); + + return 0; +} + + + +static int at32_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + + + +static int at32_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + + return 0; +} + + +static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("at32_pcm_trigger: buffer_size = %ld, " + "dma_area = %p, dma_bytes = %ld\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("trigger: period_ptr=%lx, xpr=%x, " + "xcr=%d, xnpr=%x, xncr=%d\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%x, imr=%x\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + + + +static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + + + +static int at32_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + +out: + return ret; +} + + + +static int at32_pcm_close(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + + +static int at32_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + + + +static struct snd_pcm_ops at32_pcm_ops = { + .open = at32_pcm_open, + .close = at32_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = at32_pcm_hw_params, + .hw_free = at32_pcm_hw_free, + .prepare = at32_pcm_prepare, + .trigger = at32_pcm_trigger, + .pointer = at32_pcm_pointer, + .mmap = at32_pcm_mmap, +}; + + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 at32_pcm_dmamask = 0xffffffff; + +static int at32_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &at32_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = at32_pcm_preallocate_dma_buffer( + pcm, SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n"); + ret = at32_pcm_preallocate_dma_buffer( + pcm, SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + + +out: + return ret; +} + + + +static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (substream == NULL) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + + + +#ifdef CONFIG_PM +static int at32_pcm_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) + return 0; + prtd = runtime->private_data; + params = prtd->params; + + /* Disable the PDC and save the PDC registers */ + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + + + +static int at32_pcm_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) + return 0; + prtd = runtime->private_data; + params = prtd->params; + + /* Restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else /* CONFIG_PM */ +# define at32_pcm_suspend NULL +# define at32_pcm_resume NULL +#endif /* CONFIG_PM */ + + + +struct snd_soc_platform at32_soc_platform = { + .name = "at32-audio", + .pcm_ops = &at32_pcm_ops, + .pcm_new = at32_pcm_new, + .pcm_free = at32_pcm_free_dma_buffers, + .suspend = at32_pcm_suspend, + .resume = at32_pcm_resume, +}; +EXPORT_SYMBOL_GPL(at32_soc_platform); + + + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("Atmel AT32 PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h new file mode 100644 index 0000000..2a52430 --- /dev/null +++ b/sound/soc/at32/at32-pcm.h @@ -0,0 +1,79 @@ +/* sound/soc/at32/at32-pcm.h + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_PCM_H +#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__ + +#include <linux/atmel-ssc.h> + + +/* + * Registers and status bits that are required by the PCM driver + * TODO: Is ptcr really used? + */ +struct at32_pdc_regs { + u32 xpr; /* PDC RX/TX pointer */ + u32 xcr; /* PDC RX/TX counter */ + u32 xnpr; /* PDC next RX/TX pointer */ + u32 xncr; /* PDC next RX/TX counter */ + u32 ptcr; /* PDC transfer control */ +}; + + + +/* + * SSC mask info + */ +struct at32_ssc_mask { + u32 ssc_enable; /* SSC RX/TX enable */ + u32 ssc_disable; /* SSC RX/TX disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */ + u32 pdc_enable; /* PDC RX/TX enable */ + u32 pdc_disable; /* PDC RX/TX disable */ +}; + + + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct at32_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct at32_pdc_regs *pdc; /* PDC register info */ + struct at32_ssc_mask *mask; /* SSC mask info */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler) (u32, struct snd_pcm_substream *); +}; + + + +/* + * The AT32 ASoC platform driver + */ +extern struct snd_soc_platform at32_soc_platform; + + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* __SOUND_SOC_AT32_AT32_PCM_H */ diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c new file mode 100644 index 0000000..0ca4410 --- /dev/null +++ b/sound/soc/at32/at32-ssc.c @@ -0,0 +1,849 @@ +/* sound/soc/at32/at32-ssc.c + * ASoC platform driver for AT32 using SSC as DAI + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-ssc.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +/* #define DEBUG */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "at32-pcm.h" +#include "at32-ssc.h" + + + +/*-------------------------------------------------------------------------*\ + * Constants +\*-------------------------------------------------------------------------*/ +#define NUM_SSC_DEVICES 3 + +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + + + +/*-------------------------------------------------------------------------*\ + * Module data +\*-------------------------------------------------------------------------*/ +/* + * SSC PDC registered required by the PCM DMA engine + */ +static struct at32_pdc_regs pdc_tx_reg = { + .xpr = SSC_PDC_TPR, + .xcr = SSC_PDC_TCR, + .xnpr = SSC_PDC_TNPR, + .xncr = SSC_PDC_TNCR, +}; + + + +static struct at32_pdc_regs pdc_rx_reg = { + .xpr = SSC_PDC_RPR, + .xcr = SSC_PDC_RCR, + .xnpr = SSC_PDC_RNPR, + .xncr = SSC_PDC_RNCR, +}; + + + +/* + * SSC and PDC status bits for transmit and receive + */ +static struct at32_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS), +}; + + + +static struct at32_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS), +}; + + + +/* + * DMA parameters for each SSC + */ +static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + { + { + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, +}; + + + +static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +}; + + + + +/*-------------------------------------------------------------------------*\ + * ISR +\*-------------------------------------------------------------------------*/ +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt + * handler in the PCM driver. + */ +static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id) +{ + struct at32_ssc_info *ssc_p = dev_id; + struct at32_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) & + ssc_readl(ssc_p->ssc->regs, IMR)); + + /* + * Loop through substreams attached to this SSC. If a DMA-related + * interrupt occured on that substream, call the DMA interrupt + * handler function, if one has been registered in the dma_param + * structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + + return IRQ_HANDLED; +} + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int at32_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + + + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void at32_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + int dir_mask; + + dma_params = ssc_p->dma_params[substream->stream]; + + if (dma_params != NULL) { + ssc_writel(dma_params->ssc->regs, CR, + dma_params->mask->ssc_disable); + pr_debug("%s disabled SSC_SR=0x%08x\n", + (substream->stream ? "receiver" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[substream->stream] = NULL; + } + + + dir_mask = 1 << substream->stream; + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock */ + pr_debug("at32-ssc: Stopping user %d clock\n", + ssc_p->ssc->user); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + /* clear the SSC dividers */ + ssc_p->cmr_div = 0; + ssc_p->tcmr_period = 0; + ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} + + + +/* + * Set the SSC system clock rate + */ +static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + /* TODO: What the heck do I do here? */ + return 0; +} + + + +/* + * Record DAI format for use by hw_params() + */ +static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, + unsigned int fmt) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + + + +/* + * Record SSC clock dividers for use in hw_params() + */ +static int at32_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, + int div_id, int div) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case AT32_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case AT32_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case AT32_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + + + +/* + * Configure the SSC + */ +static int at32_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int id = rtd->dai->cpu_dai->id; + struct at32_ssc_info *ssc_p = &ssc_info[id]; + struct at32_pcm_dma_params *dma_params; + int channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + + /* + * Currently, there is only one set of dma_params for each direction. + * If more are added, this code will have to be changed to select + * the proper set + */ + dma_params = &ssc_dma_params[id][substream->stream]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[substream->stream] = dma_params; + + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the PCM driver's hw_params() + * function. It should not be used for other purposes as it + * is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + + /* + * Determine sample size in bits and the PDC increment + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + + case SNDRV_PCM_FORMAT_S16: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + + case SNDRV_PCM_FORMAT_S24: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + + case SNDRV_PCM_FORMAT_S32: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + + default: + pr_warning("at32-ssc: Unsupported PCM format %d", + params_format(params)); + return -EINVAL; + } + pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n", + bits, dma_params->pdc_xfer_size, channels); + + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) + if (bits > 16) { + pr_warning("at32-ssc: " + "sample size %d is too large for I2S\n", + bits); + return -EINVAL; + } + + + /* + * Compute the SSC register settings + */ + switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRS clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, SSC_START_FALLING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(RFMR_FSLEN, bits - 1) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, SSC_START_FALLING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(TFMR_FSLEN, bits - 1) | + SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) | + SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clock. + * + * The SSC transmit clock is obtained from the BCLK signal + * on the TK line, and the SSC receive clock is generated from + * the transmit clock. + * + * For single channel data, one sample is transferred on the + * falling edge of the LRC clock. For two channel data, one + * sample is transferred on both edges of the LRC clock. + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n"); + start_event = ((channels == 1) ? + SSC_START_FALLING_RF : SSC_START_EDGE_RF); + + rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, start_event) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_CLOCK)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, start_event) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_NONE) | + SSC_BF(TCMR_CKS, SSC_CKS_PIN)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, 1) | + SSC_BF(RCMR_START, SSC_START_RISING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, 1) | + SSC_BF(TCMR_START, SSC_START_RISING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_RISING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(TFMR_DATNB, channels - 1) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + pr_warning("at32-ssc: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + + if (!ssc_p->initialized) { + /* enable peripheral clock */ + pr_debug("at32-ssc: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + pr_warning("at32-ssc: request irq failed (%d)\n", ret); + pr_debug("at32-ssc: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* Set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("at32-ssc: SSC initialized\n"); + return 0; +} + + + +static int at32_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + + dma_params = ssc_p->dma_params[substream->stream]; + + ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable); + + return 0; +} + + + +#ifdef CONFIG_PM +static int at32_ssc_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int at32_ssc_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define at32_ssc_suspend NULL +# define at32_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define AT32_SSC_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + + +#define AT32_SSC_FORMATS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \ + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) + + +struct snd_soc_cpu_dai at32_ssc_dai[NUM_SSC_DEVICES] = { + { + .name = "at32-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[0], + }, + { + .name = "at32-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[1], + }, + { + .name = "at32-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[2], + }, +}; +EXPORT_SYMBOL_GPL(at32_ssc_dai); + + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("AT32 SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h new file mode 100644 index 0000000..3c6901a --- /dev/null +++ b/sound/soc/at32/at32-ssc.h @@ -0,0 +1,59 @@ +/* sound/soc/at32/at32-ssc.h + * ASoC SSC interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_SSC_H +#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__ + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "at32-pcm.h" + + + +struct at32_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + + +struct at32_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct at32_pcm_dma_params *dma_params[2]; + struct at32_ssc_state ssc_state; +}; + + +/* SSC divider ids */ +#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ + + +extern struct snd_soc_cpu_dai at32_ssc_dai[]; + + + +#endif /* __SOUND_SOC_AT32_AT32_SSC_H */ diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c new file mode 100644 index 0000000..d6b9fd5 --- /dev/null +++ b/sound/soc/at32/playpaq_wm8510.c @@ -0,0 +1,524 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + * + * NOTE: If you don't have the AT32 enhanced portmux configured (which + * isn't currently in the mainline or Atmel patched kernel), you will + * need to set the MCLK pin (PA30) to peripheral A in your board initialization + * code. Something like: + * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); + * + */ + +/* #define DEBUG */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/arch/at32ap700x.h> +#include <asm/arch/portmux.h> + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + /* + * set CPU and CODEC DAI configuration + */ + ret = codec_dai->dai_ops.set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CODEC DAI format (%d)\n", + ret); + return ret; + } + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU DAI format (%d)\n", + ret); + return ret; + } + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", + ret); + return ret; + } + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: " + "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning + ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = codec_dai->dai_ops.set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", + ret); + return ret; + } + + + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, + WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", + ret); + return ret; + } + + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .hw_params = playpaq_wm8510_hw_params, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + {"Ext Spk", NULL, "SPKOUTP"}, + {"Ext Spk", NULL, "SPKOUTN"}, + + {"Mic Bias", NULL, "Int Mic"}, + {"MICN", NULL, "Mic Bias"}, + {"MICP", NULL, "Mic Bias"}, + + /* Terminator */ + {NULL, NULL, NULL}, +}; + + + +static int playpaq_wm8510_init(struct snd_soc_codec *codec) +{ + int i; + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], intercon[i][2]); + } + + + /* always connected endpoints */ + snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); + snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_sync_endpoints(codec); + + + + /* Make CSB show PLL rate */ + codec->dai->dai_ops.set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); +#endif + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_free_pin(MCLK_PIN); +#endif + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL"); -- 1.5.5.3

1 0

Re: [alsa-devel] [PATCH 2/3] ASoC: Add WM8990 driver
by Takashi Iwai 05 Jun '08

05 Jun '08

At Thu, 5 Jun 2008 09:54:06 +0100, Mark Brown wrote: > > The WM8990 is a highly integrated ultra-low power hi-fi codec designed > for handsets rich in multimedia features such as mobile TV, digital > audio playback and gaming. > > The bulk of this driver was written by Liam Girdwood with some > additional development and updates for new ASoC APIs by me. > > Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Hm, I obviously didn't review it in the last cycle. > +#include <linux/kernel.h> Heh, the same issue as wm8510. > +/* > + * wm8990 register cache. Note that register 0 is not included in the > + * cache. > + */ > +static const u16 wm8990_reg[] = { > + /*0x8990,*/ /* R0 - Reset */ Ditto. > +/* > + * read wm8990 register cache > + */ > +static inline unsigned int wm8990_read_reg_cache(struct snd_soc_codec *codec, > + unsigned int reg) > +{ > + u16 *cache = codec->reg_cache; > + BUG_ON(reg < 1 || reg > (ARRAY_SIZE(wm8990_reg) + 1)); Isn't it BUG_ON(reg < 1 || reg > ARRAY_SIZE(wm8990_reg)); ?? But at the first place this is bad, because codec_reg_show() in soc-core.c always loops from 0 to codec->reg_cache_size-1. At least, reg == 0 should be simply ignored. > + return cache[reg - 1]; > +} > + > +/* > + * write wm8990 register cache > + */ > +static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec, > + unsigned int reg, unsigned int value) > +{ > + u16 *cache = codec->reg_cache; > + BUG_ON(reg < 1 || reg > (ARRAY_SIZE(wm8990_reg) + 1)); Ditto. > +static int wm8990_set_bias_level(struct snd_soc_codec *codec, > + enum snd_soc_bias_level level) (snip) > + case SND_SOC_BIAS_STANDBY: > + if (codec->bias_level == SND_SOC_BIAS_OFF) { > + /* Enable all output discharge bits */ > + wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | > + WM8990_DIS_RLINE | WM8990_DIS_OUT3 | > + WM8990_DIS_OUT4 | WM8990_DIS_LOUT | > + WM8990_DIS_ROUT); > + > + /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ > + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | > + WM8990_BUFDCOPEN | WM8990_POBCTRL | > + WM8990_VMIDTOG); > + > + /* Delay to allow output caps to discharge */ > + schedule_timeout_interruptible(msecs_to_jiffies(300)); Why interruptible? Maybe msleep() is better for such a purpose? > + > + /* Disable VMIDTOG */ > + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | > + WM8990_BUFDCOPEN | WM8990_POBCTRL); > + > + /* disable all output discharge bits */ > + wm8990_write(codec, WM8990_ANTIPOP1, 0); > + > + /* Enable outputs */ > + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00); > + > + schedule_timeout_interruptible(msecs_to_jiffies(50)); Ditto. > + > + /* Enable VMID at 2x50k */ > + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02); > + > + schedule_timeout_interruptible(msecs_to_jiffies(100)); > + > + /* Enable VREF */ > + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); > + > + schedule_timeout_interruptible(msecs_to_jiffies(600)); > + > + /* Enable BUFIOEN */ > + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | > + WM8990_BUFDCOPEN | WM8990_POBCTRL | > + WM8990_BUFIOEN); > + > + /* Disable outputs */ > + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x3); > + > + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ > + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN); > + } else { > + /* ON -> standby */ > + > + } > + break; > + > + case SND_SOC_BIAS_OFF: > + /* Enable POBCTRL and SOFT_ST */ > + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | > + WM8990_POBCTRL | WM8990_BUFIOEN); > + > + /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */ > + wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | > + WM8990_BUFDCOPEN | WM8990_POBCTRL | > + WM8990_BUFIOEN); > + > + /* mute DAC */ > + val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL); > + wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); > + > + /* Enable any disabled outputs */ > + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); > + > + /* Disable VMID */ > + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01); > + > + schedule_timeout_interruptible(msecs_to_jiffies(300)); Ditto. > +static int wm8990_init(struct snd_soc_device *socdev) > +{ > + struct snd_soc_codec *codec = socdev->codec; > + u16 reg; > + int ret = 0; > + > + codec->name = "WM8990"; > + codec->owner = THIS_MODULE; > + codec->read = wm8990_read_reg_cache; > + codec->write = wm8990_write; > + codec->set_bias_level = wm8990_set_bias_level; > + codec->dai = &wm8990_dai; > + codec->num_dai = 2; > + codec->reg_cache_size = sizeof(wm8990_reg); A wrong value assignment in this file, too. But, ARRAY_SIZE() is also wrong since the max reg value is ARRAY_SIZE(). ARRAY_SIZE()+1 is the right value, but then it's really unclear what it means... thanks, Takashi

2 1

[alsa-devel] MIDI message handling
by Thorsten Edelhäußer 05 Jun '08

05 Jun '08

Hi! I'm writing a driver for the MARIAN MARC 8 Midi soundcard. Today I tested some MIDI functionality and get some errors from hydrogen ("unhandled midi message type"). I did some research and got to the point, that the MIDI alive message (0xfe) should not arrive at the application. So my question is: Do I have to process the midi messages by myself before I give it to the alsa "snd_rawmidi_receive" function? Or is there already some code, which will do it? If someone is interested here is the project page: http://sourceforge.net/projects/marc8-driver/ Regards, Thorsten

2 1

[alsa-devel] [PATCH 3/3] Revised AT32 ASoC Patch
by Mark Brown 05 Jun '08

05 Jun '08

From: Geoffrey Wossum <geoffrey(a)pager.net> Attached is a revised version of my patch to add AT32 to ASoC. This cleans most of the style issues associated with the previous patch. Also fixes an issue with the playpaq_wm8510.c code depending on a non-released patch to th AT32 portmux support. Patch is against 2.6.24.3.atmel.3 kernel, the latest AVR32 kernel Atmel has released, with the linux-2.6-asoc patches from when v2.6.24 was tagged also applied. [Fixed up minor checkpatch issues and updated for current kernels -- broonie] Signed-off-by: Geoffrey Wossum <gwossum(a)acm.org> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 3 +- sound/soc/at32/Kconfig | 34 ++ sound/soc/at32/Makefile | 11 + sound/soc/at32/at32-pcm.c | 491 ++++++++++++++++++++++ sound/soc/at32/at32-pcm.h | 79 ++++ sound/soc/at32/at32-ssc.c | 849 +++++++++++++++++++++++++++++++++++++++ sound/soc/at32/at32-ssc.h | 59 +++ sound/soc/at32/playpaq_wm8510.c | 524 ++++++++++++++++++++++++ 9 files changed, 2050 insertions(+), 1 deletions(-) create mode 100644 sound/soc/at32/Kconfig create mode 100644 sound/soc/at32/Makefile create mode 100644 sound/soc/at32/at32-pcm.c create mode 100644 sound/soc/at32/at32-pcm.h create mode 100644 sound/soc/at32/at32-ssc.c create mode 100644 sound/soc/at32/at32-ssc.h create mode 100644 sound/soc/at32/playpaq_wm8510.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index fd7bc4f..b939e22 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -22,6 +22,7 @@ config SND_SOC_AC97_BUS bool # All the supported Soc's +source "sound/soc/at32/Kconfig" source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 782db21..3645f95 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/ +obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += omap/ diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig new file mode 100644 index 0000000..b0765e8 --- /dev/null +++ b/sound/soc/at32/Kconfig @@ -0,0 +1,34 @@ +config SND_AT32_SOC + tristate "SoC Audio for the Atmel AT32 System-on-a-Chip" + depends on AVR32 && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the AT32 SSC interface. You will also need to + to select the audio interfaces to support below. + + +config SND_AT32_SOC_SSC + tristate + + + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_AT32_SOC && BOARD_PLAYPAQ + select SND_AT32_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + + + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile new file mode 100644 index 0000000..c03e55e --- /dev/null +++ b/sound/soc/at32/Makefile @@ -0,0 +1,11 @@ +# AT32 Platform Support +snd-soc-at32-objs := at32-pcm.o +snd-soc-at32-ssc-objs := at32-ssc.o + +obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o +obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c new file mode 100644 index 0000000..cf76e89 --- /dev/null +++ b/sound/soc/at32/at32-pcm.c @@ -0,0 +1,491 @@ +/* sound/soc/at32/at32-pcm.c + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-pcm.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "at32-pcm.h" + + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware at32_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE), + + .formats = SNDRV_PCM_FMTBIT_S16, + .period_bytes_min = 32, + .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */ + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct at32_runtime_data { + struct at32_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of DMA buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* Save PDC registers (for power management) */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *dmabuf = &substream->dma_buffer; + size_t size = at32_pcm_hardware.buffer_bytes_max; + + dmabuf->dev.type = SNDRV_DMA_TYPE_DEV; + dmabuf->dev.dev = pcm->card->dev; + dmabuf->private_data = NULL; + dmabuf->area = dma_alloc_coherent(pcm->card->dev, size, + &dmabuf->addr, GFP_KERNEL); + pr_debug("at32_pcm: preallocate_dma_buffer: " + "area=%p, addr=%p, size=%ld\n", + (void *)dmabuf->area, (void *)dmabuf->addr, size); + + if (!dmabuf->area) + return -ENOMEM; + + dmabuf->bytes = size; + return 0; +} + + + +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + static int count; + + count++; + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "underrun" : "overrun", params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + + snd_pcm_period_elapsed(substream); +} + + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int at32_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params + */ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = at32_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("hw_params: DMA for %s initialized " + "(dma_bytes=%ld, period_size=%ld)\n", + prtd->params->name, runtime->dma_bytes, prtd->period_size); + + return 0; +} + + + +static int at32_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + + + +static int at32_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + + return 0; +} + + +static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("at32_pcm_trigger: buffer_size = %ld, " + "dma_area = %p, dma_bytes = %ld\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("trigger: period_ptr=%lx, xpr=%x, " + "xcr=%d, xnpr=%x, xncr=%d\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%x, imr=%x\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + + + +static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + + + +static int at32_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + +out: + return ret; +} + + + +static int at32_pcm_close(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + + +static int at32_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + + + +static struct snd_pcm_ops at32_pcm_ops = { + .open = at32_pcm_open, + .close = at32_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = at32_pcm_hw_params, + .hw_free = at32_pcm_hw_free, + .prepare = at32_pcm_prepare, + .trigger = at32_pcm_trigger, + .pointer = at32_pcm_pointer, + .mmap = at32_pcm_mmap, +}; + + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 at32_pcm_dmamask = 0xffffffff; + +static int at32_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &at32_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = at32_pcm_preallocate_dma_buffer( + pcm, SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n"); + ret = at32_pcm_preallocate_dma_buffer( + pcm, SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + + +out: + return ret; +} + + + +static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (substream == NULL) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + + + +#ifdef CONFIG_PM +static int at32_pcm_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) + return 0; + prtd = runtime->private_data; + params = prtd->params; + + /* Disable the PDC and save the PDC registers */ + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + + + +static int at32_pcm_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) + return 0; + prtd = runtime->private_data; + params = prtd->params; + + /* Restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else /* CONFIG_PM */ +# define at32_pcm_suspend NULL +# define at32_pcm_resume NULL +#endif /* CONFIG_PM */ + + + +struct snd_soc_platform at32_soc_platform = { + .name = "at32-audio", + .pcm_ops = &at32_pcm_ops, + .pcm_new = at32_pcm_new, + .pcm_free = at32_pcm_free_dma_buffers, + .suspend = at32_pcm_suspend, + .resume = at32_pcm_resume, +}; +EXPORT_SYMBOL_GPL(at32_soc_platform); + + + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("Atmel AT32 PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h new file mode 100644 index 0000000..2a52430 --- /dev/null +++ b/sound/soc/at32/at32-pcm.h @@ -0,0 +1,79 @@ +/* sound/soc/at32/at32-pcm.h + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_PCM_H +#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__ + +#include <linux/atmel-ssc.h> + + +/* + * Registers and status bits that are required by the PCM driver + * TODO: Is ptcr really used? + */ +struct at32_pdc_regs { + u32 xpr; /* PDC RX/TX pointer */ + u32 xcr; /* PDC RX/TX counter */ + u32 xnpr; /* PDC next RX/TX pointer */ + u32 xncr; /* PDC next RX/TX counter */ + u32 ptcr; /* PDC transfer control */ +}; + + + +/* + * SSC mask info + */ +struct at32_ssc_mask { + u32 ssc_enable; /* SSC RX/TX enable */ + u32 ssc_disable; /* SSC RX/TX disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */ + u32 pdc_enable; /* PDC RX/TX enable */ + u32 pdc_disable; /* PDC RX/TX disable */ +}; + + + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct at32_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct at32_pdc_regs *pdc; /* PDC register info */ + struct at32_ssc_mask *mask; /* SSC mask info */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler) (u32, struct snd_pcm_substream *); +}; + + + +/* + * The AT32 ASoC platform driver + */ +extern struct snd_soc_platform at32_soc_platform; + + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* __SOUND_SOC_AT32_AT32_PCM_H */ diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c new file mode 100644 index 0000000..0ca4410 --- /dev/null +++ b/sound/soc/at32/at32-ssc.c @@ -0,0 +1,849 @@ +/* sound/soc/at32/at32-ssc.c + * ASoC platform driver for AT32 using SSC as DAI + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-ssc.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +/* #define DEBUG */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "at32-pcm.h" +#include "at32-ssc.h" + + + +/*-------------------------------------------------------------------------*\ + * Constants +\*-------------------------------------------------------------------------*/ +#define NUM_SSC_DEVICES 3 + +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + + + +/*-------------------------------------------------------------------------*\ + * Module data +\*-------------------------------------------------------------------------*/ +/* + * SSC PDC registered required by the PCM DMA engine + */ +static struct at32_pdc_regs pdc_tx_reg = { + .xpr = SSC_PDC_TPR, + .xcr = SSC_PDC_TCR, + .xnpr = SSC_PDC_TNPR, + .xncr = SSC_PDC_TNCR, +}; + + + +static struct at32_pdc_regs pdc_rx_reg = { + .xpr = SSC_PDC_RPR, + .xcr = SSC_PDC_RCR, + .xnpr = SSC_PDC_RNPR, + .xncr = SSC_PDC_RNCR, +}; + + + +/* + * SSC and PDC status bits for transmit and receive + */ +static struct at32_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS), +}; + + + +static struct at32_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS), +}; + + + +/* + * DMA parameters for each SSC + */ +static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + { + { + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, +}; + + + +static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +}; + + + + +/*-------------------------------------------------------------------------*\ + * ISR +\*-------------------------------------------------------------------------*/ +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt + * handler in the PCM driver. + */ +static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id) +{ + struct at32_ssc_info *ssc_p = dev_id; + struct at32_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) & + ssc_readl(ssc_p->ssc->regs, IMR)); + + /* + * Loop through substreams attached to this SSC. If a DMA-related + * interrupt occured on that substream, call the DMA interrupt + * handler function, if one has been registered in the dma_param + * structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + + return IRQ_HANDLED; +} + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int at32_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + + + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void at32_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + int dir_mask; + + dma_params = ssc_p->dma_params[substream->stream]; + + if (dma_params != NULL) { + ssc_writel(dma_params->ssc->regs, CR, + dma_params->mask->ssc_disable); + pr_debug("%s disabled SSC_SR=0x%08x\n", + (substream->stream ? "receiver" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[substream->stream] = NULL; + } + + + dir_mask = 1 << substream->stream; + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock */ + pr_debug("at32-ssc: Stopping user %d clock\n", + ssc_p->ssc->user); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + /* clear the SSC dividers */ + ssc_p->cmr_div = 0; + ssc_p->tcmr_period = 0; + ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} + + + +/* + * Set the SSC system clock rate + */ +static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + /* TODO: What the heck do I do here? */ + return 0; +} + + + +/* + * Record DAI format for use by hw_params() + */ +static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, + unsigned int fmt) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + + + +/* + * Record SSC clock dividers for use in hw_params() + */ +static int at32_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, + int div_id, int div) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case AT32_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case AT32_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case AT32_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + + + +/* + * Configure the SSC + */ +static int at32_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int id = rtd->dai->cpu_dai->id; + struct at32_ssc_info *ssc_p = &ssc_info[id]; + struct at32_pcm_dma_params *dma_params; + int channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + + /* + * Currently, there is only one set of dma_params for each direction. + * If more are added, this code will have to be changed to select + * the proper set + */ + dma_params = &ssc_dma_params[id][substream->stream]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[substream->stream] = dma_params; + + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the PCM driver's hw_params() + * function. It should not be used for other purposes as it + * is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + + /* + * Determine sample size in bits and the PDC increment + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + + case SNDRV_PCM_FORMAT_S16: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + + case SNDRV_PCM_FORMAT_S24: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + + case SNDRV_PCM_FORMAT_S32: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + + default: + pr_warning("at32-ssc: Unsupported PCM format %d", + params_format(params)); + return -EINVAL; + } + pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n", + bits, dma_params->pdc_xfer_size, channels); + + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) + if (bits > 16) { + pr_warning("at32-ssc: " + "sample size %d is too large for I2S\n", + bits); + return -EINVAL; + } + + + /* + * Compute the SSC register settings + */ + switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRS clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, SSC_START_FALLING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(RFMR_FSLEN, bits - 1) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, SSC_START_FALLING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(TFMR_FSLEN, bits - 1) | + SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) | + SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clock. + * + * The SSC transmit clock is obtained from the BCLK signal + * on the TK line, and the SSC receive clock is generated from + * the transmit clock. + * + * For single channel data, one sample is transferred on the + * falling edge of the LRC clock. For two channel data, one + * sample is transferred on both edges of the LRC clock. + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n"); + start_event = ((channels == 1) ? + SSC_START_FALLING_RF : SSC_START_EDGE_RF); + + rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, start_event) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_CLOCK)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, start_event) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_NONE) | + SSC_BF(TCMR_CKS, SSC_CKS_PIN)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, 1) | + SSC_BF(RCMR_START, SSC_START_RISING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, 1) | + SSC_BF(TCMR_START, SSC_START_RISING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_RISING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(TFMR_DATNB, channels - 1) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + pr_warning("at32-ssc: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + + if (!ssc_p->initialized) { + /* enable peripheral clock */ + pr_debug("at32-ssc: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + pr_warning("at32-ssc: request irq failed (%d)\n", ret); + pr_debug("at32-ssc: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* Set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("at32-ssc: SSC initialized\n"); + return 0; +} + + + +static int at32_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + + dma_params = ssc_p->dma_params[substream->stream]; + + ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable); + + return 0; +} + + + +#ifdef CONFIG_PM +static int at32_ssc_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int at32_ssc_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define at32_ssc_suspend NULL +# define at32_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define AT32_SSC_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + + +#define AT32_SSC_FORMATS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \ + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) + + +struct snd_soc_cpu_dai at32_ssc_dai[NUM_SSC_DEVICES] = { + { + .name = "at32-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[0], + }, + { + .name = "at32-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[1], + }, + { + .name = "at32-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[2], + }, +}; +EXPORT_SYMBOL_GPL(at32_ssc_dai); + + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("AT32 SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h new file mode 100644 index 0000000..3c6901a --- /dev/null +++ b/sound/soc/at32/at32-ssc.h @@ -0,0 +1,59 @@ +/* sound/soc/at32/at32-ssc.h + * ASoC SSC interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_SSC_H +#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__ + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "at32-pcm.h" + + + +struct at32_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + + +struct at32_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct at32_pcm_dma_params *dma_params[2]; + struct at32_ssc_state ssc_state; +}; + + +/* SSC divider ids */ +#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ + + +extern struct snd_soc_cpu_dai at32_ssc_dai[]; + + + +#endif /* __SOUND_SOC_AT32_AT32_SSC_H */ diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c new file mode 100644 index 0000000..d6b9fd5 --- /dev/null +++ b/sound/soc/at32/playpaq_wm8510.c @@ -0,0 +1,524 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + * + * NOTE: If you don't have the AT32 enhanced portmux configured (which + * isn't currently in the mainline or Atmel patched kernel), you will + * need to set the MCLK pin (PA30) to peripheral A in your board initialization + * code. Something like: + * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); + * + */ + +/* #define DEBUG */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/arch/at32ap700x.h> +#include <asm/arch/portmux.h> + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + /* + * set CPU and CODEC DAI configuration + */ + ret = codec_dai->dai_ops.set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CODEC DAI format (%d)\n", + ret); + return ret; + } + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU DAI format (%d)\n", + ret); + return ret; + } + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", + ret); + return ret; + } + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: " + "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning + ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = codec_dai->dai_ops.set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", + ret); + return ret; + } + + + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, + WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", + ret); + return ret; + } + + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .hw_params = playpaq_wm8510_hw_params, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + {"Ext Spk", NULL, "SPKOUTP"}, + {"Ext Spk", NULL, "SPKOUTN"}, + + {"Mic Bias", NULL, "Int Mic"}, + {"MICN", NULL, "Mic Bias"}, + {"MICP", NULL, "Mic Bias"}, + + /* Terminator */ + {NULL, NULL, NULL}, +}; + + + +static int playpaq_wm8510_init(struct snd_soc_codec *codec) +{ + int i; + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], intercon[i][2]); + } + + + /* always connected endpoints */ + snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); + snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_sync_endpoints(codec); + + + + /* Make CSB show PLL rate */ + codec->dai->dai_ops.set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); +#endif + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_free_pin(MCLK_PIN); +#endif + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL"); -- 1.5.5.3

1 0

[alsa-devel] [PATCH 1/3] ASoC: Add WM8510 driver
by Mark Brown 05 Jun '08

05 Jun '08

The WM8510 is a mono CODEC with speaker driver optimised for telephony applications, featuring: - 16/20/24/32 bit audio at data rates between 8kHz and 48kHz - On-chip PLL - Dual microphone inputs This driver was originally written by Liam Girdwood with updates from Brett Saunders, Geoffrey Wossum and myself. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Signed-off-by: Brett Saunders <breton.saunders(a)ntlworld.com> Signed-off-by: Geoffrey Wossum <geoffrey(a)pager.net> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8510.c | 836 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8510.h | 103 ++++++ 4 files changed, 944 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm8510.c create mode 100644 sound/soc/codecs/wm8510.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7beefcc..ef400b3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -5,6 +5,9 @@ config SND_SOC_AC97_CODEC config SND_SOC_UDA1380 tristate +config SND_SOC_WM8510 + tristate + config SND_SOC_WM8731 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d5926a1..b92c665 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,5 +1,6 @@ snd-soc-ac97-objs := ac97.o snd-soc-uda1380-objs := uda1380.o +snd-soc-wm8510-objs := wm8510.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o @@ -10,6 +11,7 @@ snd-soc-tlv320aic3x-objs := tlv320aic3x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o +obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c new file mode 100644 index 0000000..253bcc4 --- /dev/null +++ b/sound/soc/codecs/wm8510.c @@ -0,0 +1,836 @@ +/* + * wm8510.c -- WM8510 ALSA Soc Audio driver + * + * Copyright 2006 Wolfson Microelectronics PLC. + * + * Author: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "wm8510.h" + +#define AUDIO_NAME "wm8510" +#define WM8510_VERSION "0.6" + +/* + * Debug + */ + +#define WM8510_DEBUG 0 + +#ifdef WM8510_DEBUG +#define dbg(format, arg...) \ + printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) +#else +#define dbg(format, arg...) do {} while (0) +#endif +#define err(format, arg...) \ + printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) +#define info(format, arg...) \ + printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) +#define warn(format, arg...) \ + printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) + +struct snd_soc_codec_device soc_codec_dev_wm8510; + +/* + * wm8510 register cache + * We can't read the WM8510 register space when we are + * using 2 wire for device control, so we cache them instead. + */ +static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0050, 0x0000, 0x0140, 0x0000, + 0x0000, 0x0000, 0x0000, 0x00ff, + 0x0000, 0x0000, 0x0100, 0x00ff, + 0x0000, 0x0000, 0x012c, 0x002c, + 0x002c, 0x002c, 0x002c, 0x0000, + 0x0032, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0038, 0x000b, 0x0032, 0x0000, + 0x0008, 0x000c, 0x0093, 0x00e9, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0003, 0x0010, 0x0000, 0x0000, + 0x0000, 0x0002, 0x0001, 0x0000, + 0x0000, 0x0000, 0x0039, 0x0000, + 0x0000, +}; + +/* + * read wm8510 register cache + */ +static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == WM8510_RESET) + return 0; + if (reg >= WM8510_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write wm8510 register cache + */ +static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8510_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the WM8510 register space + */ +static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8510 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8510_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define wm8510_reset(c) wm8510_write(c, WM8510_RESET, 0) + +static const char *wm8510_companding[] = {"Off", "NC", "u-law", "A-law" }; +static const char *wm8510_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" }; +static const char *wm8510_alc[] = {"ALC", "Limiter" }; + +static const struct soc_enum wm8510_enum[] = { + SOC_ENUM_SINGLE(WM8510_COMP, 1, 4, wm8510_companding), /* adc */ + SOC_ENUM_SINGLE(WM8510_COMP, 3, 4, wm8510_companding), /* dac */ + SOC_ENUM_SINGLE(WM8510_DAC, 4, 4, wm8510_deemp), + SOC_ENUM_SINGLE(WM8510_ALC3, 8, 2, wm8510_alc), +}; + +static const struct snd_kcontrol_new wm8510_snd_controls[] = { + +SOC_SINGLE("Digital Loopback Switch", WM8510_COMP, 0, 1, 0), + +SOC_ENUM("DAC Companding", wm8510_enum[1]), +SOC_ENUM("ADC Companding", wm8510_enum[0]), + +SOC_ENUM("Playback De-emphasis", wm8510_enum[2]), +SOC_SINGLE("DAC Inversion Switch", WM8510_DAC, 0, 1, 0), + +SOC_SINGLE("Master Playback Volume", WM8510_DACVOL, 0, 127, 0), + +SOC_SINGLE("High Pass Filter Switch", WM8510_ADC, 8, 1, 0), +SOC_SINGLE("High Pass Cut Off", WM8510_ADC, 4, 7, 0), +SOC_SINGLE("ADC Inversion Switch", WM8510_COMP, 0, 1, 0), + +SOC_SINGLE("Capture Volume", WM8510_ADCVOL, 0, 127, 0), + +SOC_SINGLE("DAC Playback Limiter Switch", WM8510_DACLIM1, 8, 1, 0), +SOC_SINGLE("DAC Playback Limiter Decay", WM8510_DACLIM1, 4, 15, 0), +SOC_SINGLE("DAC Playback Limiter Attack", WM8510_DACLIM1, 0, 15, 0), + +SOC_SINGLE("DAC Playback Limiter Threshold", WM8510_DACLIM2, 4, 7, 0), +SOC_SINGLE("DAC Playback Limiter Boost", WM8510_DACLIM2, 0, 15, 0), + +SOC_SINGLE("ALC Enable Switch", WM8510_ALC1, 8, 1, 0), +SOC_SINGLE("ALC Capture Max Gain", WM8510_ALC1, 3, 7, 0), +SOC_SINGLE("ALC Capture Min Gain", WM8510_ALC1, 0, 7, 0), + +SOC_SINGLE("ALC Capture ZC Switch", WM8510_ALC2, 8, 1, 0), +SOC_SINGLE("ALC Capture Hold", WM8510_ALC2, 4, 7, 0), +SOC_SINGLE("ALC Capture Target", WM8510_ALC2, 0, 15, 0), + +SOC_ENUM("ALC Capture Mode", wm8510_enum[3]), +SOC_SINGLE("ALC Capture Decay", WM8510_ALC3, 4, 15, 0), +SOC_SINGLE("ALC Capture Attack", WM8510_ALC3, 0, 15, 0), + +SOC_SINGLE("ALC Capture Noise Gate Switch", WM8510_NGATE, 3, 1, 0), +SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8510_NGATE, 0, 7, 0), + +SOC_SINGLE("Capture PGA ZC Switch", WM8510_INPPGA, 7, 1, 0), +SOC_SINGLE("Capture PGA Volume", WM8510_INPPGA, 0, 63, 0), + +SOC_SINGLE("Speaker Playback ZC Switch", WM8510_SPKVOL, 7, 1, 0), +SOC_SINGLE("Speaker Playback Switch", WM8510_SPKVOL, 6, 1, 1), +SOC_SINGLE("Speaker Playback Volume", WM8510_SPKVOL, 0, 63, 0), +SOC_SINGLE("Speaker Boost", WM8510_OUTPUT, 2, 1, 0), + +SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0), +SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1), +}; + +/* add non dapm controls */ +static int wm8510_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8510_snd_controls[i], codec, + NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Speaker Output Mixer */ +static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 0), +}; + +/* Mono Output Mixer */ +static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8510_boost_controls[] = { +SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 0), +SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0), +SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0), +}; + +static const struct snd_kcontrol_new wm8510_micpga_controls[] = { +SOC_DAPM_SINGLE("MICP Switch", WM8510_INPUT, 0, 1, 0), +SOC_DAPM_SINGLE("MICN Switch", WM8510_INPUT, 1, 1, 0), +SOC_DAPM_SINGLE("AUX Switch", WM8510_INPUT, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8510_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Speaker Mixer", WM8510_POWER3, 2, 0, + &wm8510_speaker_mixer_controls[0], + ARRAY_SIZE(wm8510_speaker_mixer_controls)), +SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_POWER3, 3, 0, + &wm8510_mono_mixer_controls[0], + ARRAY_SIZE(wm8510_mono_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER2, 0, 0), +SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, + &wm8510_micpga_controls[0], + ARRAY_SIZE(wm8510_micpga_controls)), +SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, + &wm8510_boost_controls[0], + ARRAY_SIZE(wm8510_boost_controls)), + +SND_SOC_DAPM_MICBIAS("Mic Bias", WM8510_POWER1, 4, 0), + +SND_SOC_DAPM_INPUT("MICN"), +SND_SOC_DAPM_INPUT("MICP"), +SND_SOC_DAPM_INPUT("AUX"), +SND_SOC_DAPM_OUTPUT("MONOOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, + {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, + {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Microphone PGA */ + {"Mic PGA", "MICN Switch", "MICN"}, + {"Mic PGA", "MICP Switch", "MICP"}, + { "Mic PGA", "AUX Switch", "Aux Input" }, + + /* Boost Mixer */ + {"Boost Mixer", "Mic PGA Switch", "Mic PGA"}, + {"Boost Mixer", "Mic Volume", "MICP"}, + {"Boost Mixer", "Aux Volume", "Aux Input"}, + + {"ADC", NULL, "Boost Mixer"}, +}; + +static int wm8510_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8510_dapm_widgets, + ARRAY_SIZE(wm8510_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +struct pll_ { + unsigned int pre_div:4; /* prescale - 1 */ + unsigned int n:4; + unsigned int k; +}; + +static struct pll_ pll_div; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 24) * 10) + +static void pll_factors(unsigned int target, unsigned int source) +{ + unsigned long long Kpart; + unsigned int K, Ndiv, Nmod; + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div.pre_div = 1; + Ndiv = target / source; + } else + pll_div.pre_div = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8510 N value %d outwith recommended range!d\n", + Ndiv); + + pll_div.n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div.k = K; +} + +static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + if (freq_in == 0 || freq_out == 0) { + /* Clock CODEC directly from MCLK */ + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); + wm8510_write(codec, WM8510_CLOCK, reg & 0x0ff); + + /* Turn off PLL */ + reg = wm8510_read_reg_cache(codec, WM8510_POWER1); + wm8510_write(codec, WM8510_POWER1, reg & 0x1df); + return 0; + } + + pll_factors(freq_out*8, freq_in); + + wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); + wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18); + wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff); + wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff); + reg = wm8510_read_reg_cache(codec, WM8510_POWER1); + wm8510_write(codec, WM8510_POWER1, reg | 0x020); + + /* Run CODEC from PLL instead of MCLK */ + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); + wm8510_write(codec, WM8510_CLOCK, reg | 0x100); + + return 0; +} + +/* + * Configure WM8510 clock dividers. + */ +static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8510_OPCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf; + wm8510_write(codec, WM8510_GPIO, reg | div); + break; + case WM8510_MCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f; + wm8510_write(codec, WM8510_CLOCK, reg | div); + break; + case WM8510_ADCCLK: + reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7; + wm8510_write(codec, WM8510_ADC, reg | div); + break; + case WM8510_DACCLK: + reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7; + wm8510_write(codec, WM8510_DAC, reg | div); + break; + case WM8510_BCLKDIV: + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3; + wm8510_write(codec, WM8510_CLOCK, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int wm8510_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + clk |= 0x0001; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0010; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0008; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x00018; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0180; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0100; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0080; + break; + default: + return -EINVAL; + } + + wm8510_write(codec, WM8510_IFACE, iface); + wm8510_write(codec, WM8510_CLOCK, clk); + return 0; +} + +static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f; + u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0020; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0040; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x0060; + break; + } + + /* filter coefficient */ + switch (params_rate(params)) { + case SNDRV_PCM_RATE_8000: + adn |= 0x5 << 1; + break; + case SNDRV_PCM_RATE_11025: + adn |= 0x4 << 1; + break; + case SNDRV_PCM_RATE_16000: + adn |= 0x3 << 1; + break; + case SNDRV_PCM_RATE_22050: + adn |= 0x2 << 1; + break; + case SNDRV_PCM_RATE_32000: + adn |= 0x1 << 1; + break; + case SNDRV_PCM_RATE_44100: + break; + } + + wm8510_write(codec, WM8510_IFACE, iface); + wm8510_write(codec, WM8510_ADD, adn); + return 0; +} + +static int wm8510_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; + + if (mute) + wm8510_write(codec, WM8510_DAC, mute_reg | 0x40); + else + wm8510_write(codec, WM8510_DAC, mute_reg); + return 0; +} + +/* liam need to make this lower power with dapm */ +static int wm8510_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + + switch (level) { + case SND_SOC_BIAS_ON: + wm8510_write(codec, WM8510_POWER1, 0x1ff); + wm8510_write(codec, WM8510_POWER2, 0x1ff); + wm8510_write(codec, WM8510_POWER3, 0x1ff); + break; + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + wm8510_write(codec, WM8510_POWER1, 0x0); + wm8510_write(codec, WM8510_POWER2, 0x0); + wm8510_write(codec, WM8510_POWER3, 0x0); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8510_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +#define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +struct snd_soc_codec_dai wm8510_dai = { + .name = "WM8510 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8510_RATES, + .formats = WM8510_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8510_RATES, + .formats = WM8510_FORMATS,}, + .ops = { + .hw_params = wm8510_pcm_hw_params, + }, + .dai_ops = { + .digital_mute = wm8510_mute, + .set_fmt = wm8510_set_dai_fmt, + .set_clkdiv = wm8510_set_dai_clkdiv, + .set_pll = wm8510_set_dai_pll, + }, +}; +EXPORT_SYMBOL_GPL(wm8510_dai); + +static int wm8510_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8510_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8510_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the WM8510 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8510_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "WM8510"; + codec->owner = THIS_MODULE; + codec->read = wm8510_read_reg_cache; + codec->write = wm8510_write; + codec->set_bias_level = wm8510_set_bias_level; + codec->dai = &wm8510_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(wm8510_reg); + codec->reg_cache = kmemdup(wm8510_reg, sizeof(wm8510_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + wm8510_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8510_add_controls(codec); + wm8510_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to register card\n"); + goto card_err; + } + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *wm8510_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM8510 2 wire address is 0x1a + */ +#define I2C_DRIVERID_WM8510 0xfefe /* liam - need a proper id */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver wm8510_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ + +static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = wm8510_socdev; + struct wm8510_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + err("failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = wm8510_init(socdev); + if (ret < 0) { + err("failed to initialise WM8510\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int wm8510_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int wm8510_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, wm8510_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver wm8510_i2c_driver = { + .driver = { + .name = "WM8510 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_WM8510, + .attach_adapter = wm8510_i2c_attach, + .detach_client = wm8510_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "WM8510", + .driver = &wm8510_i2c_driver, +}; +#endif + +static int wm8510_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8510_setup_data *setup; + struct snd_soc_codec *codec; + int ret = 0; + + info("WM8510 Audio Codec %s", WM8510_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + wm8510_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&wm8510_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int wm8510_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8510_i2c_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8510 = { + .probe = wm8510_probe, + .remove = wm8510_remove, + .suspend = wm8510_suspend, + .resume = wm8510_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510); + +MODULE_DESCRIPTION("ASoC WM8510 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h new file mode 100644 index 0000000..c862e7b --- /dev/null +++ b/sound/soc/codecs/wm8510.h @@ -0,0 +1,103 @@ +/* + * wm8510.h -- WM8510 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8510_H +#define _WM8510_H + +/* WM8510 register space */ + +#define WM8510_RESET 0x0 +#define WM8510_POWER1 0x1 +#define WM8510_POWER2 0x2 +#define WM8510_POWER3 0x3 +#define WM8510_IFACE 0x4 +#define WM8510_COMP 0x5 +#define WM8510_CLOCK 0x6 +#define WM8510_ADD 0x7 +#define WM8510_GPIO 0x8 +#define WM8510_DAC 0xa +#define WM8510_DACVOL 0xb +#define WM8510_ADC 0xe +#define WM8510_ADCVOL 0xf +#define WM8510_EQ1 0x12 +#define WM8510_EQ2 0x13 +#define WM8510_EQ3 0x14 +#define WM8510_EQ4 0x15 +#define WM8510_EQ5 0x16 +#define WM8510_DACLIM1 0x18 +#define WM8510_DACLIM2 0x19 +#define WM8510_NOTCH1 0x1b +#define WM8510_NOTCH2 0x1c +#define WM8510_NOTCH3 0x1d +#define WM8510_NOTCH4 0x1e +#define WM8510_ALC1 0x20 +#define WM8510_ALC2 0x21 +#define WM8510_ALC3 0x22 +#define WM8510_NGATE 0x23 +#define WM8510_PLLN 0x24 +#define WM8510_PLLK1 0x25 +#define WM8510_PLLK2 0x26 +#define WM8510_PLLK3 0x27 +#define WM8510_ATTEN 0x28 +#define WM8510_INPUT 0x2c +#define WM8510_INPPGA 0x2d +#define WM8510_ADCBOOST 0x2f +#define WM8510_OUTPUT 0x31 +#define WM8510_SPKMIX 0x32 +#define WM8510_SPKVOL 0x36 +#define WM8510_MONOMIX 0x38 + +#define WM8510_CACHEREGNUM 57 + +/* Clock divider Id's */ +#define WM8510_OPCLKDIV 0 +#define WM8510_MCLKDIV 1 +#define WM8510_ADCCLK 2 +#define WM8510_DACCLK 3 +#define WM8510_BCLKDIV 4 + +/* DAC clock dividers */ +#define WM8510_DACCLK_F2 (1 << 3) +#define WM8510_DACCLK_F4 (0 << 3) + +/* ADC clock dividers */ +#define WM8510_ADCCLK_F2 (1 << 3) +#define WM8510_ADCCLK_F4 (0 << 3) + +/* PLL Out dividers */ +#define WM8510_OPCLKDIV_1 (0 << 4) +#define WM8510_OPCLKDIV_2 (1 << 4) +#define WM8510_OPCLKDIV_3 (2 << 4) +#define WM8510_OPCLKDIV_4 (3 << 4) + +/* BCLK clock dividers */ +#define WM8510_BCLKDIV_1 (0 << 2) +#define WM8510_BCLKDIV_2 (1 << 2) +#define WM8510_BCLKDIV_4 (2 << 2) +#define WM8510_BCLKDIV_8 (3 << 2) +#define WM8510_BCLKDIV_16 (4 << 2) +#define WM8510_BCLKDIV_32 (5 << 2) + +/* MCLK clock dividers */ +#define WM8510_MCLKDIV_1 (0 << 5) +#define WM8510_MCLKDIV_1_5 (1 << 5) +#define WM8510_MCLKDIV_2 (2 << 5) +#define WM8510_MCLKDIV_3 (3 << 5) +#define WM8510_MCLKDIV_4 (4 << 5) +#define WM8510_MCLKDIV_6 (5 << 5) +#define WM8510_MCLKDIV_8 (6 << 5) +#define WM8510_MCLKDIV_12 (7 << 5) + +struct wm8510_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_codec_dai wm8510_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8510; + +#endif -- 1.5.5.3

2 4

[alsa-devel] That "Magic Jack" thing I see on TV
by Tom Watson 05 Jun '08

05 Jun '08

Boy is this silly... I keep seeing those ads on TV for the $40 (or so) telephone interface goodie. My questions: Is this thing an audio interface? Any driver for Linux? Can the hookswitch/Ringer be read/written via some control interface? Touch tone interface? No, I haven't bought one yet (and I don't intend to right now), but it looks like a neat thing to interface to a telephone (not the line side, the set side) for various uses (stage use for one). Has anyone looked at it? While the VOIP service is nice, I'd be looking at it for a simple audio interface (admittedly low quality!). -- Tom Watson tsw(a)johana.com

2 1

[alsa-devel] alsa-drivers-HEAD does not compile against recent RHE5.1 kernels
by Ben Stanley 04 Jun '08

04 Jun '08

Hi, alsa-drivers-HEAD does not compile against recent RHE5 kernels, such as ftp://ftp.redhat.com/pub/redhat/linux/enterprise/5Client/en/os/SRPMS/kernel… The problem is mixed up with the test for init_utsname in configure.in: dnl Check for init_utsname... if test "$kversion.$kpatchlevel" = "2.6" -a $ksublevel -ge 19; then CONFIG_HAVE_INIT_UTSNAME=1 else AC_MSG_CHECKING(for init_utsname) init_utsname="0" ac_save_CFLAGS="$CFLAGS" ac_save_CC="$CC" CFLAGS="$KERNEL_CHECK_CFLAGS -Werror" CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ #include <linux/config.h> #include <linux/utsname.h> ],[ int *x = (int *)init_utsname(); *x = 0; ], AC_MSG_RESULT(yes);init_utsname="1", AC_MSG_RESULT(no);init_utsname="0", AC_MSG_RESULT(unknown);init_utsname="0" ) CFLAGS=$ac_save_CFLAGS CC=$ac_save_CC CONFIG_HAVE_INIT_UTSNAME=$init_utsname fi dnl AC_SUBST(CONFIG_HAVE_INIT_UTSNAME) if test "$CONFIG_HAVE_INIT_UTSNAME" = "1"; then AC_DEFINE(CONFIG_HAVE_INIT_UTSNAME) fi The problem is that <linux/config.h> does not exist in recent kernel.org and RH 2.6.18 kernels. The AC_TRY_COMPILE produces a negative result (because of a warning, treated as an error, about the missing linux/config.h file), which then causes CONFIG_HAVE_INIT_UTSNAME to NOT be defined. This results in a compile failure in acore/info_oss.c . I'm not sure how this should be fixed. I'm also not sure why someone else hasn't noticed it already, since it affects me with a standard kernel.org kernel. Therefore, here is my test procedure. wget http://www.kernel.org/pub/linux/kernel/v2.6/linux-2.6.25.tar.bz2 tar -xjf linux-2.6.25.tar.bz2 cd linux-2.6.25 make defconfig make modules_prepare cd .. (This next step takes *forever*) git clone git://git.alsa-project.org/alsa-kernel.git alsa-kernel git clone git://git.alsa-project.org/alsa-driver.git alsa-driver cd alsa-driver ./gitcompile --with-kernel=../linux-2.6.25 --with-build=../linux-2.6.25 This currently produces an error: No alsa-kernel is found. Please specify the correct location via ALSAKERNELDIR environment variable. So instead I run ALSAKERNELDIR=../alsa-kernel ./gitcompile --with-kernel=../linux-2.6.25 --with-build=../linux-2.6.25 and I get the same error. when I repeat the above with an alsa-kernel from hg, I do not get this problem (in fact, alsa-driver compiles completely on the Ubuntu 7.10 machine I'm using for this test). So, I've got perhaps two different problems here. I need a hand. Ben Stanley.

2 8

[alsa-devel] [PATCH 3/3] Revised AT32 ASoC Patch
by Mark Brown 04 Jun '08

04 Jun '08

From: Geoffrey Wossum <geoffrey(a)pager.net> Attached is a revised version of my patch to add AT32 to ASoC. This cleans most of the style issues associated with the previous patch. Also fixes an issue with the playpaq_wm8510.c code depending on a non-released patch to th AT32 portmux support. Patch is against 2.6.24.3.atmel.3 kernel, the latest AVR32 kernel Atmel has released, with the linux-2.6-asoc patches from when v2.6.24 was tagged also applied. [Fixed up minor checkpatch issues and updated for current kernels -- broonie] Signed-off-by: Geoffrey Wossum <gwossum(a)acm.org> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 3 +- sound/soc/at32/Kconfig | 34 ++ sound/soc/at32/Makefile | 11 + sound/soc/at32/at32-pcm.c | 491 ++++++++++++++++++++++ sound/soc/at32/at32-pcm.h | 79 ++++ sound/soc/at32/at32-ssc.c | 849 +++++++++++++++++++++++++++++++++++++++ sound/soc/at32/at32-ssc.h | 59 +++ sound/soc/at32/playpaq_wm8510.c | 524 ++++++++++++++++++++++++ 9 files changed, 2050 insertions(+), 1 deletions(-) create mode 100644 sound/soc/at32/Kconfig create mode 100644 sound/soc/at32/Makefile create mode 100644 sound/soc/at32/at32-pcm.c create mode 100644 sound/soc/at32/at32-pcm.h create mode 100644 sound/soc/at32/at32-ssc.c create mode 100644 sound/soc/at32/at32-ssc.h create mode 100644 sound/soc/at32/playpaq_wm8510.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index fd7bc4f..b939e22 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -22,6 +22,7 @@ config SND_SOC_AC97_BUS bool # All the supported Soc's +source "sound/soc/at32/Kconfig" source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 782db21..3645f95 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/ +obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += omap/ diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig new file mode 100644 index 0000000..b0765e8 --- /dev/null +++ b/sound/soc/at32/Kconfig @@ -0,0 +1,34 @@ +config SND_AT32_SOC + tristate "SoC Audio for the Atmel AT32 System-on-a-Chip" + depends on AVR32 && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the AT32 SSC interface. You will also need to + to select the audio interfaces to support below. + + +config SND_AT32_SOC_SSC + tristate + + + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_AT32_SOC && BOARD_PLAYPAQ + select SND_AT32_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + + + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile new file mode 100644 index 0000000..c03e55e --- /dev/null +++ b/sound/soc/at32/Makefile @@ -0,0 +1,11 @@ +# AT32 Platform Support +snd-soc-at32-objs := at32-pcm.o +snd-soc-at32-ssc-objs := at32-ssc.o + +obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o +obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c new file mode 100644 index 0000000..cf76e89 --- /dev/null +++ b/sound/soc/at32/at32-pcm.c @@ -0,0 +1,491 @@ +/* sound/soc/at32/at32-pcm.c + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-pcm.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "at32-pcm.h" + + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware at32_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE), + + .formats = SNDRV_PCM_FMTBIT_S16, + .period_bytes_min = 32, + .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */ + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct at32_runtime_data { + struct at32_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of DMA buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* Save PDC registers (for power management) */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *dmabuf = &substream->dma_buffer; + size_t size = at32_pcm_hardware.buffer_bytes_max; + + dmabuf->dev.type = SNDRV_DMA_TYPE_DEV; + dmabuf->dev.dev = pcm->card->dev; + dmabuf->private_data = NULL; + dmabuf->area = dma_alloc_coherent(pcm->card->dev, size, + &dmabuf->addr, GFP_KERNEL); + pr_debug("at32_pcm: preallocate_dma_buffer: " + "area=%p, addr=%p, size=%ld\n", + (void *)dmabuf->area, (void *)dmabuf->addr, size); + + if (!dmabuf->area) + return -ENOMEM; + + dmabuf->bytes = size; + return 0; +} + + + +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + static int count; + + count++; + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "underrun" : "overrun", params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + + snd_pcm_period_elapsed(substream); +} + + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int at32_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params + */ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = at32_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("hw_params: DMA for %s initialized " + "(dma_bytes=%ld, period_size=%ld)\n", + prtd->params->name, runtime->dma_bytes, prtd->period_size); + + return 0; +} + + + +static int at32_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + + + +static int at32_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + + return 0; +} + + +static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("at32_pcm_trigger: buffer_size = %ld, " + "dma_area = %p, dma_bytes = %ld\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("trigger: period_ptr=%lx, xpr=%x, " + "xcr=%d, xnpr=%x, xncr=%d\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%x, imr=%x\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + + + +static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + + + +static int at32_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + +out: + return ret; +} + + + +static int at32_pcm_close(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + + +static int at32_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + + + +static struct snd_pcm_ops at32_pcm_ops = { + .open = at32_pcm_open, + .close = at32_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = at32_pcm_hw_params, + .hw_free = at32_pcm_hw_free, + .prepare = at32_pcm_prepare, + .trigger = at32_pcm_trigger, + .pointer = at32_pcm_pointer, + .mmap = at32_pcm_mmap, +}; + + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 at32_pcm_dmamask = 0xffffffff; + +static int at32_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &at32_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = at32_pcm_preallocate_dma_buffer( + pcm, SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n"); + ret = at32_pcm_preallocate_dma_buffer( + pcm, SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + + +out: + return ret; +} + + + +static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (substream == NULL) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + + + +#ifdef CONFIG_PM +static int at32_pcm_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) + return 0; + prtd = runtime->private_data; + params = prtd->params; + + /* Disable the PDC and save the PDC registers */ + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + + + +static int at32_pcm_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) + return 0; + prtd = runtime->private_data; + params = prtd->params; + + /* Restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else /* CONFIG_PM */ +# define at32_pcm_suspend NULL +# define at32_pcm_resume NULL +#endif /* CONFIG_PM */ + + + +struct snd_soc_platform at32_soc_platform = { + .name = "at32-audio", + .pcm_ops = &at32_pcm_ops, + .pcm_new = at32_pcm_new, + .pcm_free = at32_pcm_free_dma_buffers, + .suspend = at32_pcm_suspend, + .resume = at32_pcm_resume, +}; +EXPORT_SYMBOL_GPL(at32_soc_platform); + + + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("Atmel AT32 PCM module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h new file mode 100644 index 0000000..2a52430 --- /dev/null +++ b/sound/soc/at32/at32-pcm.h @@ -0,0 +1,79 @@ +/* sound/soc/at32/at32-pcm.h + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_PCM_H +#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__ + +#include <linux/atmel-ssc.h> + + +/* + * Registers and status bits that are required by the PCM driver + * TODO: Is ptcr really used? + */ +struct at32_pdc_regs { + u32 xpr; /* PDC RX/TX pointer */ + u32 xcr; /* PDC RX/TX counter */ + u32 xnpr; /* PDC next RX/TX pointer */ + u32 xncr; /* PDC next RX/TX counter */ + u32 ptcr; /* PDC transfer control */ +}; + + + +/* + * SSC mask info + */ +struct at32_ssc_mask { + u32 ssc_enable; /* SSC RX/TX enable */ + u32 ssc_disable; /* SSC RX/TX disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */ + u32 pdc_enable; /* PDC RX/TX enable */ + u32 pdc_disable; /* PDC RX/TX disable */ +}; + + + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct at32_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct at32_pdc_regs *pdc; /* PDC register info */ + struct at32_ssc_mask *mask; /* SSC mask info */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler) (u32, struct snd_pcm_substream *); +}; + + + +/* + * The AT32 ASoC platform driver + */ +extern struct snd_soc_platform at32_soc_platform; + + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* __SOUND_SOC_AT32_AT32_PCM_H */ diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c new file mode 100644 index 0000000..0ca4410 --- /dev/null +++ b/sound/soc/at32/at32-ssc.c @@ -0,0 +1,849 @@ +/* sound/soc/at32/at32-ssc.c + * ASoC platform driver for AT32 using SSC as DAI + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-ssc.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +/* #define DEBUG */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "at32-pcm.h" +#include "at32-ssc.h" + + + +/*-------------------------------------------------------------------------*\ + * Constants +\*-------------------------------------------------------------------------*/ +#define NUM_SSC_DEVICES 3 + +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + + + +/*-------------------------------------------------------------------------*\ + * Module data +\*-------------------------------------------------------------------------*/ +/* + * SSC PDC registered required by the PCM DMA engine + */ +static struct at32_pdc_regs pdc_tx_reg = { + .xpr = SSC_PDC_TPR, + .xcr = SSC_PDC_TCR, + .xnpr = SSC_PDC_TNPR, + .xncr = SSC_PDC_TNCR, +}; + + + +static struct at32_pdc_regs pdc_rx_reg = { + .xpr = SSC_PDC_RPR, + .xcr = SSC_PDC_RCR, + .xnpr = SSC_PDC_RNPR, + .xncr = SSC_PDC_RNCR, +}; + + + +/* + * SSC and PDC status bits for transmit and receive + */ +static struct at32_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS), +}; + + + +static struct at32_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS), +}; + + + +/* + * DMA parameters for each SSC + */ +static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + { + { + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, +}; + + + +static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +}; + + + + +/*-------------------------------------------------------------------------*\ + * ISR +\*-------------------------------------------------------------------------*/ +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt + * handler in the PCM driver. + */ +static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id) +{ + struct at32_ssc_info *ssc_p = dev_id; + struct at32_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) & + ssc_readl(ssc_p->ssc->regs, IMR)); + + /* + * Loop through substreams attached to this SSC. If a DMA-related + * interrupt occured on that substream, call the DMA interrupt + * handler function, if one has been registered in the dma_param + * structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + + return IRQ_HANDLED; +} + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int at32_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + + + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void at32_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + int dir_mask; + + dma_params = ssc_p->dma_params[substream->stream]; + + if (dma_params != NULL) { + ssc_writel(dma_params->ssc->regs, CR, + dma_params->mask->ssc_disable); + pr_debug("%s disabled SSC_SR=0x%08x\n", + (substream->stream ? "receiver" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[substream->stream] = NULL; + } + + + dir_mask = 1 << substream->stream; + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock */ + pr_debug("at32-ssc: Stopping user %d clock\n", + ssc_p->ssc->user); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + /* clear the SSC dividers */ + ssc_p->cmr_div = 0; + ssc_p->tcmr_period = 0; + ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} + + + +/* + * Set the SSC system clock rate + */ +static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + /* TODO: What the heck do I do here? */ + return 0; +} + + + +/* + * Record DAI format for use by hw_params() + */ +static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, + unsigned int fmt) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + + + +/* + * Record SSC clock dividers for use in hw_params() + */ +static int at32_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, + int div_id, int div) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case AT32_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case AT32_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case AT32_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + + + +/* + * Configure the SSC + */ +static int at32_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int id = rtd->dai->cpu_dai->id; + struct at32_ssc_info *ssc_p = &ssc_info[id]; + struct at32_pcm_dma_params *dma_params; + int channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + + /* + * Currently, there is only one set of dma_params for each direction. + * If more are added, this code will have to be changed to select + * the proper set + */ + dma_params = &ssc_dma_params[id][substream->stream]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[substream->stream] = dma_params; + + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the PCM driver's hw_params() + * function. It should not be used for other purposes as it + * is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + + /* + * Determine sample size in bits and the PDC increment + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + + case SNDRV_PCM_FORMAT_S16: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + + case SNDRV_PCM_FORMAT_S24: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + + case SNDRV_PCM_FORMAT_S32: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + + default: + pr_warning("at32-ssc: Unsupported PCM format %d", + params_format(params)); + return -EINVAL; + } + pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n", + bits, dma_params->pdc_xfer_size, channels); + + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) + if (bits > 16) { + pr_warning("at32-ssc: " + "sample size %d is too large for I2S\n", + bits); + return -EINVAL; + } + + + /* + * Compute the SSC register settings + */ + switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRS clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, SSC_START_FALLING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(RFMR_FSLEN, bits - 1) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, SSC_START_FALLING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(TFMR_FSLEN, bits - 1) | + SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) | + SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clock. + * + * The SSC transmit clock is obtained from the BCLK signal + * on the TK line, and the SSC receive clock is generated from + * the transmit clock. + * + * For single channel data, one sample is transferred on the + * falling edge of the LRC clock. For two channel data, one + * sample is transferred on both edges of the LRC clock. + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n"); + start_event = ((channels == 1) ? + SSC_START_FALLING_RF : SSC_START_EDGE_RF); + + rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, start_event) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_CLOCK)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, start_event) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_NONE) | + SSC_BF(TCMR_CKS, SSC_CKS_PIN)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, 1) | + SSC_BF(RCMR_START, SSC_START_RISING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, 1) | + SSC_BF(TCMR_START, SSC_START_RISING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_RISING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(TFMR_DATNB, channels - 1) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + pr_warning("at32-ssc: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + + if (!ssc_p->initialized) { + /* enable peripheral clock */ + pr_debug("at32-ssc: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + pr_warning("at32-ssc: request irq failed (%d)\n", ret); + pr_debug("at32-ssc: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* Set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("at32-ssc: SSC initialized\n"); + return 0; +} + + + +static int at32_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + + dma_params = ssc_p->dma_params[substream->stream]; + + ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable); + + return 0; +} + + + +#ifdef CONFIG_PM +static int at32_ssc_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int at32_ssc_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define at32_ssc_suspend NULL +# define at32_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define AT32_SSC_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + + +#define AT32_SSC_FORMATS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \ + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) + + +struct snd_soc_cpu_dai at32_ssc_dai[NUM_SSC_DEVICES] = { + { + .name = "at32-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[0], + }, + { + .name = "at32-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[1], + }, + { + .name = "at32-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[2], + }, +}; +EXPORT_SYMBOL_GPL(at32_ssc_dai); + + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("AT32 SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h new file mode 100644 index 0000000..3c6901a --- /dev/null +++ b/sound/soc/at32/at32-ssc.h @@ -0,0 +1,59 @@ +/* sound/soc/at32/at32-ssc.h + * ASoC SSC interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_SSC_H +#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__ + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "at32-pcm.h" + + + +struct at32_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + + +struct at32_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct at32_pcm_dma_params *dma_params[2]; + struct at32_ssc_state ssc_state; +}; + + +/* SSC divider ids */ +#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ + + +extern struct snd_soc_cpu_dai at32_ssc_dai[]; + + + +#endif /* __SOUND_SOC_AT32_AT32_SSC_H */ diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c new file mode 100644 index 0000000..d6b9fd5 --- /dev/null +++ b/sound/soc/at32/playpaq_wm8510.c @@ -0,0 +1,524 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + * + * NOTE: If you don't have the AT32 enhanced portmux configured (which + * isn't currently in the mainline or Atmel patched kernel), you will + * need to set the MCLK pin (PA30) to peripheral A in your board initialization + * code. Something like: + * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); + * + */ + +/* #define DEBUG */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/arch/at32ap700x.h> +#include <asm/arch/portmux.h> + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + /* + * set CPU and CODEC DAI configuration + */ + ret = codec_dai->dai_ops.set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CODEC DAI format (%d)\n", + ret); + return ret; + } + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU DAI format (%d)\n", + ret); + return ret; + } + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", + ret); + return ret; + } + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: " + "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning + ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = codec_dai->dai_ops.set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", + ret); + return ret; + } + + + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, + WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", + ret); + return ret; + } + + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .hw_params = playpaq_wm8510_hw_params, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + {"Ext Spk", NULL, "SPKOUTP"}, + {"Ext Spk", NULL, "SPKOUTN"}, + + {"Mic Bias", NULL, "Int Mic"}, + {"MICN", NULL, "Mic Bias"}, + {"MICP", NULL, "Mic Bias"}, + + /* Terminator */ + {NULL, NULL, NULL}, +}; + + + +static int playpaq_wm8510_init(struct snd_soc_codec *codec) +{ + int i; + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], intercon[i][2]); + } + + + /* always connected endpoints */ + snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); + snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_sync_endpoints(codec); + + + + /* Make CSB show PLL rate */ + codec->dai->dai_ops.set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); +#endif + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_free_pin(MCLK_PIN); +#endif + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL"); -- 1.5.5.3

1 0

[alsa-devel] [PATCH] Revised AT32 ASoC Patch
by Geoffrey Wossum 04 Jun '08

04 Jun '08

Attached is a revised version of my patch to add AT32 to ASoC. This cleans up most of the style issues associated with the previous patch. Also fixes an issue with the playpaq_wm8510.c code depending on a non-released patch to the AT32 portmux support. Patch is against 2.6.24.3.atmel.3 kernel, the latest AVR32 kernel Atmel has released, with the linux-2.6-asoc patches from when v2.6.24 was tagged also applied. Will apply cleanly and compile to a vanilla 2.6.24, with the following exceptions: - The sound/soc/Makefile patch is against the linux-2.6-asoc Makefile, not the vanilla Makefile - playpaq_wm8510.c requires the WM8510 driver to compile which is only currently available in linux-2.6-asoc --- Geoffrey Signed-off-by: Geoffrey Wossum <gwossum(a)acm.org> --- /dev/null 2008-06-03 03:32:04.656004085 -0500 +++ linux-2.6.24.3.atmel.3/sound/soc/at32/at32-ssc.h 2008-05-27 17:05:14.000000000 -0500 @@ -0,0 +1,59 @@ +/* sound/soc/at32/at32-ssc.h + * ASoC SSC interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_SSC_H +#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__ + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "at32-pcm.h" + + + +struct at32_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + + +struct at32_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct at32_pcm_dma_params *dma_params[2]; + struct at32_ssc_state ssc_state; +}; + + +/* SSC divider ids */ +#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ + + +extern struct snd_soc_cpu_dai at32_ssc_dai[]; + + + +#endif /* __SOUND_SOC_AT32_AT32_SSC_H */ --- /dev/null 2008-06-03 03:32:04.656004085 -0500 +++ linux-2.6.24.3.atmel.3/sound/soc/at32/at32-ssc.c 2008-05-28 09:32:18.000000000 -0500 @@ -0,0 +1,852 @@ +/* sound/soc/at32/at32-ssc.c + * ASoC platform driver for AT32 using SSC as DAI + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-ssc.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +/* #define DEBUG */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <asm/io.h> + +#include "at32-pcm.h" +#include "at32-ssc.h" + + + +/*-------------------------------------------------------------------------*\ + * Constants +\*-------------------------------------------------------------------------*/ +#define NUM_SSC_DEVICES 3 + +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + + + +/*-------------------------------------------------------------------------*\ + * Module data +\*-------------------------------------------------------------------------*/ +/* + * SSC PDC registered required by the PCM DMA engine + */ +static struct at32_pdc_regs pdc_tx_reg = { + .xpr = SSC_PDC_TPR, + .xcr = SSC_PDC_TCR, + .xnpr = SSC_PDC_TNPR, + .xncr = SSC_PDC_TNCR, +}; + + + +static struct at32_pdc_regs pdc_rx_reg = { + .xpr = SSC_PDC_RPR, + .xcr = SSC_PDC_RCR, + .xnpr = SSC_PDC_RNPR, + .xncr = SSC_PDC_RNCR, +}; + + + +/* + * SSC and PDC status bits for transmit and receive + */ +static struct at32_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS), +}; + + + +static struct at32_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS), +}; + + + +/* + * DMA parameters for each SSC + */ +static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + { + { + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, +}; + + + +static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +}; + + + + +/*-------------------------------------------------------------------------*\ + * ISR +\*-------------------------------------------------------------------------*/ +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt + * handler in the PCM driver. + */ +static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id) +{ + struct at32_ssc_info *ssc_p = dev_id; + struct at32_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) & + ssc_readl(ssc_p->ssc->regs, IMR)); + + /* + * Loop through substreams attached to this SSC. If a DMA-related + * interrupt occured on that substream, call the DMA interrupt + * handler function, if one has been registered in the dma_param + * structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + + return IRQ_HANDLED; +} + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int at32_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + + + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void at32_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + int dir_mask; + + dma_params = ssc_p->dma_params[substream->stream]; + + if (dma_params != NULL) { + ssc_writel(dma_params->ssc->regs, CR, + dma_params->mask->ssc_disable); + pr_debug("%s disabled SSC_SR=0x%08x\n", + (substream->stream ? "receiver" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[substream->stream] = NULL; + } + + + dir_mask = 1 << substream->stream; + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock */ + pr_debug("at32-ssc: Stopping user %d clock\n", + ssc_p->ssc->user); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + /* clear the SSC dividers */ + ssc_p->cmr_div = 0; + ssc_p->tcmr_period = 0; + ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} + + + +/* + * Set the SSC system clock rate + */ +static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + /* TODO: What the heck do I do here? */ + return 0; +} + + + +/* + * Record DAI format for use by hw_params() + */ +static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, + unsigned int fmt) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + + + +/* + * Record SSC clock dividers for use in hw_params() + */ +static int at32_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, + int div_id, int div) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case AT32_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value + */ + if (ssc_p->cmr_div == 0) { + ssc_p->cmr_div = div; + } else if (div != ssc_p->cmr_div) { + return -EBUSY; + } + break; + + case AT32_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case AT32_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + + + +/* + * Configure the SSC + */ +static int at32_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int id = rtd->dai->cpu_dai->id; + struct at32_ssc_info *ssc_p = &ssc_info[id]; + struct at32_pcm_dma_params *dma_params; + int channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + + /* + * Currently, there is only one set of dma_params for each direction. + * If more are added, this code will have to be changed to select + * the proper set + */ + dma_params = &ssc_dma_params[id][substream->stream]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[substream->stream] = dma_params; + + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the PCM driver's hw_params() + * function. It should not be used for other purposes as it + * is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + + /* + * Determine sample size in bits and the PDC increment + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + + case SNDRV_PCM_FORMAT_S16: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + + case SNDRV_PCM_FORMAT_S24: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + + case SNDRV_PCM_FORMAT_S32: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + + default: + pr_warning("at32-ssc: Unsupported PCM format %d", + params_format(params)); + return -EINVAL; + } + pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n", + bits, dma_params->pdc_xfer_size, channels); + + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) + if (bits > 16) { + pr_warning("at32-ssc: " + "sample size %d is too large for I2S\n", + bits); + return -EINVAL; + } + + + /* + * Compute the SSC register settings + */ + switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRS clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, SSC_START_FALLING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(RFMR_FSLEN, bits - 1) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, SSC_START_FALLING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(TFMR_FSLEN, bits - 1) | + SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) | + SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clock. + * + * The SSC transmit clock is obtained from the BCLK signal + * on the TK line, and the SSC receive clock is generated from + * the transmit clock. + * + * For single channel data, one sample is transferred on the + * falling edge of the LRC clock. For two channel data, one + * sample is transferred on both edges of the LRC clock. + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n"); + start_event = ((channels == 1) ? + SSC_START_FALLING_RF : SSC_START_EDGE_RF); + + rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, start_event) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_CLOCK)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, start_event) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_NONE) | + SSC_BF(TCMR_CKS, SSC_CKS_PIN)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, 1) | + SSC_BF(RCMR_START, SSC_START_RISING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, 1) | + SSC_BF(TCMR_START, SSC_START_RISING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_RISING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(TFMR_DATNB, channels - 1) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + pr_warning("at32-ssc: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + + if (!ssc_p->initialized) { + /* enable peripheral clock */ + pr_debug("at32-ssc: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + pr_warning("at32-ssc: request irq failed (%d)\n", ret); + pr_debug("at32-ssc: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* Set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("at32-ssc: SSC initialized\n"); + return 0; +} + + + +static int at32_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + + dma_params = ssc_p->dma_params[substream->stream]; + + ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable); + + return 0; +} + + + +#ifdef CONFIG_PM +static int at32_ssc_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int at32_ssc_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define at32_ssc_suspend NULL +# define at32_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define AT32_SSC_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + + +#define AT32_SSC_FORMATS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \ + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) + + +struct snd_soc_cpu_dai at32_ssc_dai[NUM_SSC_DEVICES] = { + { + .name = "at32-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[0], + }, + { + .name = "at32-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[1], + }, + { + .name = "at32-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[2], + }, +}; +EXPORT_SYMBOL_GPL(at32_ssc_dai); + + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("AT32 SSC ASoC Interface"); +MODULE_LICENSE("GPL"); --- /dev/null 2008-06-03 03:32:04.656004085 -0500 +++ linux-2.6.24.3.atmel.3/sound/soc/at32/at32-pcm.h 2008-05-27 17:04:54.000000000 -0500 @@ -0,0 +1,79 @@ +/* sound/soc/at32/at32-pcm.h + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_PCM_H +#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__ + +#include <linux/atmel-ssc.h> + + +/* + * Registers and status bits that are required by the PCM driver + * TODO: Is ptcr really used? + */ +struct at32_pdc_regs { + u32 xpr; /* PDC RX/TX pointer */ + u32 xcr; /* PDC RX/TX counter */ + u32 xnpr; /* PDC next RX/TX pointer */ + u32 xncr; /* PDC next RX/TX counter */ + u32 ptcr; /* PDC transfer control */ +}; + + + +/* + * SSC mask info + */ +struct at32_ssc_mask { + u32 ssc_enable; /* SSC RX/TX enable */ + u32 ssc_disable; /* SSC RX/TX disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */ + u32 pdc_enable; /* PDC RX/TX enable */ + u32 pdc_disable; /* PDC RX/TX disable */ +}; + + + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct at32_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct at32_pdc_regs *pdc; /* PDC register info */ + struct at32_ssc_mask *mask; /* SSC mask info */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler) (u32, struct snd_pcm_substream *); +}; + + + +/* + * The AT32 ASoC platform driver + */ +extern struct snd_soc_platform at32_soc_platform; + + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* __SOUND_SOC_AT32_AT32_PCM_H */ --- /dev/null 2008-06-03 03:32:04.656004085 -0500 +++ linux-2.6.24.3.atmel.3/sound/soc/at32/at32-pcm.c 2008-05-28 09:28:48.000000000 -0500 @@ -0,0 +1,492 @@ +/* sound/soc/at32/at32-pcm.c + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-pcm.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "at32-pcm.h" + + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware at32_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE), + + .formats = SNDRV_PCM_FMTBIT_S16, + .period_bytes_min = 32, + .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */ + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct at32_runtime_data { + struct at32_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of DMA buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* Save PDC registers (for power management) */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *dmabuf = &substream->dma_buffer; + size_t size = at32_pcm_hardware.buffer_bytes_max; + + dmabuf->dev.type = SNDRV_DMA_TYPE_DEV; + dmabuf->dev.dev = pcm->card->dev; + dmabuf->private_data = NULL; + dmabuf->area = dma_alloc_coherent(pcm->card->dev, size, + &dmabuf->addr, GFP_KERNEL); + pr_debug("at32_pcm: preallocate_dma_buffer: " + "area=%p, addr=%p, size=%ld\n", + (void *)dmabuf->area, (void *)dmabuf->addr, size); + + if (!dmabuf->area) + return -ENOMEM; + + dmabuf->bytes = size; + return 0; +} + + + +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + static int count = 0; + + count++; + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "underrun" : "overrun", params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + + snd_pcm_period_elapsed(substream); +} + + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int at32_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params + */ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = at32_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("hw_params: DMA for %s initialized " + "(dma_bytes=%ld, period_size=%ld)\n", + prtd->params->name, runtime->dma_bytes, prtd->period_size); + + return 0; +} + + + +static int at32_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + + + +static int at32_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + + return 0; +} + + +static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("at32_pcm_trigger: buffer_size = %ld, " + "dma_area = %p, dma_bytes = %ld\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("trigger: period_ptr=%lx, xpr=%x, " + "xcr=%d, xnpr=%x, xncr=%d\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%x, imr=%x\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + + + +static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) + x = 0; + + return x; +} + + + +static int at32_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + +out: + return ret; +} + + + +static int at32_pcm_close(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + + +static int at32_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + + + +static struct snd_pcm_ops at32_pcm_ops = { + .open = at32_pcm_open, + .close = at32_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = at32_pcm_hw_params, + .hw_free = at32_pcm_hw_free, + .prepare = at32_pcm_prepare, + .trigger = at32_pcm_trigger, + .pointer = at32_pcm_pointer, + .mmap = at32_pcm_mmap, +}; + + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 at32_pcm_dmamask = 0xffffffff; + +static int at32_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &at32_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = at32_pcm_preallocate_dma_buffer( + pcm, SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n"); + ret = at32_pcm_preallocate_dma_buffer( + pcm, SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + + +out: + return ret; +} + + + +static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (substream == NULL) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + + + +#ifdef CONFIG_PM +static int at32_pcm_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) + return 0; + prtd = runtime->private_data; + params = prtd->params; + + /* Disable the PDC and save the PDC registers */ + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + + + +static int at32_pcm_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) + return 0; + prtd = runtime->private_data; + params = prtd->params; + + /* Restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else /* CONFIG_PM */ +# define at32_pcm_suspend NULL +# define at32_pcm_resume NULL +#endif /* CONFIG_PM */ + + + +struct snd_soc_platform at32_soc_platform = { + .name = "at32-audio", + .pcm_ops = &at32_pcm_ops, + .pcm_new = at32_pcm_new, + .pcm_free = at32_pcm_free_dma_buffers, + .suspend = at32_pcm_suspend, + .resume = at32_pcm_resume, +}; +EXPORT_SYMBOL_GPL(at32_soc_platform); + + + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("Atmel AT32 PCM module"); +MODULE_LICENSE("GPL"); --- /dev/null 2008-06-03 03:32:04.656004085 -0500 +++ linux-2.6.24.3.atmel.3/sound/soc/at32/playpaq_wm8510.c 2008-06-03 10:50:46.000000000 -0500 @@ -0,0 +1,525 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + * + * NOTE: If you don't have the AT32 enhanced portmux configured (which + * isn't currently in the mainline or Atmel patched kernel), you will + * need to set the MCLK pin (PA30) to peripheral A in your board initialization + * code. Something like: + * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); + * + */ + +/* #define DEBUG */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/arch/at32ap700x.h> +#include <asm/arch/portmux.h> + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + /* + * set CPU and CODEC DAI configuration + */ + ret = codec_dai->dai_ops.set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CODEC DAI format (%d)\n", + ret); + return ret; + } + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU DAI format (%d)\n", + ret); + return ret; + } + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", + ret); + return ret; + } + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: " + "Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: " + "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning + ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = codec_dai->dai_ops.set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", + ret); + return ret; + } + + + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, + WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", + ret); + return ret; + } + + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .hw_params = playpaq_wm8510_hw_params, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + {"Ext Spk", NULL, "SPKOUTP"}, + {"Ext Spk", NULL, "SPKOUTN"}, + + {"Mic Bias", NULL, "Int Mic"}, + {"MICN", NULL, "Mic Bias"}, + {"MICP", NULL, "Mic Bias"}, + + /* Terminator */ + {NULL, NULL, NULL}, +}; + + + +static int playpaq_wm8510_init(struct snd_soc_codec *codec) +{ + int i; + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], intercon[i][2]); + } + + + /* always connected endpoints */ + snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); + snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_sync_endpoints(codec); + + + + /* Make CSB show PLL rate */ + codec->dai->dai_ops.set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); +#endif + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + +#if defined CONFIG_AT32_ENHANCED_PORTMUX + at32_free_pin(MCLK_PIN); +#endif + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL"); --- /dev/null 2008-06-03 03:32:04.656004085 -0500 +++ linux-2.6.24.3.atmel.3/sound/soc/at32/Kconfig 2008-06-03 10:38:01.000000000 -0500 @@ -0,0 +1,34 @@ +config SND_AT32_SOC + tristate "SoC Audio for the Atmel AT32 System-on-a-Chip" + depends on AVR32 && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the AT32 SSC interface. You will also need to + to select the audio interfaces to support below. + + +config SND_AT32_SOC_SSC + tristate + + + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_AT32_SOC && BOARD_PLAYPAQ + select SND_AT32_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + + + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. Patch Makefile's and Kconfig's for ASoC with AT32 Patch is against linux-2.6-asoc repository when v.2.6.24 was tagged Signed-off-by: Geoffrey Wossum <gwossum(a)acm.org> Index: linux-2.6.24.3.atmel.3/sound/soc/Kconfig =================================================================== --- linux-2.6.24.3.atmel.3.orig/sound/soc/Kconfig +++ linux-2.6.24.3.atmel.3/sound/soc/Kconfig @@ -25,6 +25,7 @@ config SND_SOC will be called snd-soc-core. # All the supported Soc's +source "sound/soc/at32/Kconfig" source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" Index: linux-2.6.24.3.atmel.3/sound/soc/Makefile =================================================================== --- linux-2.6.24.3.atmel.3.orig/sound/soc/Makefile +++ linux-2.6.24.3.atmel.3/sound/soc/Makefile @@ -1,4 +1,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ imx/ au1x/ blackfin/ fsl/ +obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ imx/ au1x/ blackfin/ fsl/

3 6

[alsa-devel] [PATCH] alsa-info.sh - Check if pastebin.ca is up, and gracefully fallback to --no-upload if its down.
by Travis Place 04 Jun '08

04 Jun '08

Changelog: This patch will allow alsa-info.sh to check if the pastebin site (www.pastebin.ca) is down, before trying to upload to it. It will fail gracefully, with a message letting the user know what happened. The script will also keep the ALSA information it collected (same way it does with --no-upload). Patch is attached. Signed-off-by: Travis Place <wishie at wishie dot net>

2 1

[alsa-devel] [Fwd: [PATCH] Patch for alsa-info.sh to check if pastebin.ca is down, and handle with grace.]
by stan 04 Jun '08

04 Jun '08

This one?

1 0

[alsa-devel] [PATCH]model for alc888 to support intel DG33 motherboards
by Jiang zhe 04 Jun '08

04 Jun '08

Hi, Please see the attachment. These two motherboards's pin configuration are not covered by driver. I wrote a new model to support them. Thanks! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com>

2 1

[alsa-devel] RESEND: [ALC268 patch_realtek.c] COMPAL IFL90/JFL-92 laptop quirk to select ALC268_TOSHIBA
by Tony Vroon 04 Jun '08

04 Jun '08

Right then, leave it to me to send the wrong version of the patch from the wrong e-mail address. Please disregard the previous message, that diff will not apply. Apologies. Could you please consider the attached patch. The COMPAL IFL90/JFL-92 laptop defaults to an incorrect output channel on autoprobe, resulting in no audio. The user was unable to enable analog audio with the mixer controls provided. Testing confirms that the Toshiba model enables both headphone with automuting and the internal speakers. Embedded sign-off and short patch description. Tested-By: Guri <gurashka(a)gmail.com> Signed-Off-By: Tony Vroon <tony(a)linx.net> Regards, Tony V. LINX sysadmin

1 0

[alsa-devel] [ALC268 patch_realtek.c] COMPAL IFL90/JFL-92 laptop quirk to select ALC268_TOSHIBA
by Tony Vroon 04 Jun '08

04 Jun '08

Good morning, Could you please consider the attached patch. The COMPAL IFL90/JFL-92 laptop defaults to an incorrect output channel on autoprobe, resulting in no audio. The user was unable to enable analog audio with the mixer controls provided. Testing confirms that the Toshiba model enables both headphone with automuting and the internal speakers. Embedded sign-off and short patch description. Tested-By: Guri <gurashka(a)gmail.com> Signed-Off-By: Tony Vroon <tony(a)linx.net> Regards, Tony V. LINX sysadmin

1 0

[alsa-devel] asus bracket 'bug'
by Jordi Haarman 03 Jun '08

03 Jun '08

Hi, I was asked in the channel to post a bug and send it to the mailinglist in the hope that somebody can fix it. (they said it would be pretty easy). The bug is located here (with the debug info): https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3965 The problem is that the spdif of the bracket does not work and should be added somehow. -- Groet, J http://startu.net - esperanto startpagina

1 1

[alsa-devel] [PATCH] maestro3: Fix hw volume on HP OmniBook
by Ville Syrjälä 03 Jun '08

03 Jun '08

maestro3: Fix hw volume on HP OmniBook Make the hw volume buttons work correctly on some HP OmniBook laptops. The original quirk was apparently applied a bit too early and it was also lacking some critial register writes. This improved sequence was discovered by trial and error (like the original sequence). Tested and found working on OB500 and OB6000 laptops. Signed-off-by: Ville Syrjala <syrjala(a)sci.fi> --- sound/pci/maestro3.c | 42 +++++++++++++++++++++++++++--------------- 1 file changed, 27 insertions(+), 15 deletions(-) Index: linux-2.6.25.4/sound/pci/maestro3.c =================================================================== --- linux-2.6.25.4.orig/sound/pci/maestro3.c +++ linux-2.6.25.4/sound/pci/maestro3.c @@ -2427,6 +2427,29 @@ snd_m3_amp_enable(struct snd_m3 *chip, i outw(0xffff, io + GPIO_MASK); } +static void +snd_m3_hv_init(struct snd_m3 *chip) +{ + unsigned long io = chip->iobase; + u16 val = GPI_VOL_DOWN | GPI_VOL_UP; + + if (!chip->is_omnibook) + return; + + /* + * Volume buttons on some HP OmniBook laptops + * require some GPIO magic to work correctly. + */ + outw(0xffff, io + GPIO_MASK); + outw(0x0000, io + GPIO_DATA); + + outw(~val, io + GPIO_MASK); + outw(inw(io + GPIO_DIRECTION) & ~val, io + GPIO_DIRECTION); + outw(val, io + GPIO_MASK); + + outw(0xffff, io + GPIO_MASK); +} + static int snd_m3_chip_init(struct snd_m3 *chip) { @@ -2442,21 +2465,6 @@ snd_m3_chip_init(struct snd_m3 *chip) DISABLE_LEGACY); pci_write_config_word(pcidev, PCI_LEGACY_AUDIO_CTRL, w); - if (chip->is_omnibook) { - /* - * Volume buttons on some HP OmniBook laptops don't work - * correctly. This makes them work for the most part. - * - * Volume up and down buttons on the laptop side work. - * Fn+cursor_up (volme up) works. - * Fn+cursor_down (volume down) doesn't work. - * Fn+F7 (mute) works acts as volume up. - */ - outw(~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_MASK); - outw(inw(io + GPIO_DIRECTION) & ~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DIRECTION); - outw((GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DATA); - outw(0xffff, io + GPIO_MASK); - } pci_read_config_dword(pcidev, PCI_ALLEGRO_CONFIG, &n); n &= ~(HV_CTRL_ENABLE | REDUCED_DEBOUNCE | HV_BUTTON_FROM_GD); n |= chip->hv_config; @@ -2644,6 +2652,8 @@ static int m3_resume(struct pci_dev *pci snd_m3_enable_ints(chip); snd_m3_amp_enable(chip, 1); + snd_m3_hv_init(chip); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } @@ -2783,6 +2793,8 @@ snd_m3_create(struct snd_card *card, str snd_m3_amp_enable(chip, 1); + snd_m3_hv_init(chip); + tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip); if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED,

1 0

[alsa-devel] testing of 1.0.17rc1 packages
by Jaroslav Kysela 03 Jun '08

03 Jun '08

Hello, finally, I prepared 1.0.17rc1 packages for testing. This is not release announce - official 1.0.17rc1 packages might be little different. I would like to invite people on devel list to test these packages, because we moved from HG to GIT and the gap between releases was too big. ftp://ftp.alsa-project.org/pub/testing Please, if you reach any major compilation or functionality issue, e-mail me directly or on this list to fix this problem. I'm going to make official release tomorrow. Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

1 0

[alsa-devel] [PATCH][ASoC v2] Update Freescale MPC8610 codec driver naming
by Timur Tabi 03 Jun '08

03 Jun '08

Change the Freescale MPC8610 HPCD codec and fabric drivers to use the codec name and id, instead of just the name, to match codec instances. Previously, the codec and fabric drivers would create a name string based on the I2C bus and address. Instead, keep the name as "cs4270" and combine the bus and address into a unique number. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- My patch, "Add CS4270 i2c data to fsl_soc.c" must be applied for these drivers to work. sound/soc/codecs/cs4270.c | 21 ++++-- sound/soc/fsl/mpc8610_hpcd.c | 147 +++++++++++++++++++++--------------------- 2 files changed, 88 insertions(+), 80 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 410fed5..df3294d 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -23,6 +23,8 @@ * 7) The only supported control is volume and hardware mute (if enabled) */ +#define DEBUG + #include <linux/module.h> #include <sound/core.h> #include <sound/pcm.h> @@ -119,7 +121,6 @@ */ struct cs4270_private { struct snd_soc_codec codec; - char name[16]; unsigned int mclk; unsigned int mode; struct snd_soc_dai *dai; @@ -557,9 +558,6 @@ static int cs4270_i2c_probe(struct i2c_client *client, goto error; } - sprintf(cs4270->name, "%s@%s", dev_driver_string(&client->dev), - dev_name(&client->dev)); - /* Initialize the playback and capture stream data. The sample rate * fields are initialized later when the fabric driver calls * cs4270_set_dai_sysclk(). Hopefully, this won't cause any problems. @@ -574,14 +572,19 @@ static int cs4270_i2c_probe(struct i2c_client *client, cs4270->capture.channels_max = 2; cs4270->capture.formats = CS4270_FORMATS; + /* Register the codec */ + codec = &cs4270->codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->list); INIT_LIST_HEAD(&codec->dai_list); - codec->name = cs4270->name; + + codec->name = "cs4270"; + codec->num = client->adapter->nr << 16 | client->addr; snd_soc_card_config_codec(codec, NULL, NULL, client); + codec->init = cs4270_codec_init; codec->reg_cache = cs4270->reg_cache; codec->codec_read = cs4270_read_reg_cache; @@ -593,8 +596,12 @@ static int cs4270_i2c_probe(struct i2c_client *client, dev_err(&client->dev, "failed to register card\n"); goto error; } - memset(&dai_new, 0, sizeof(dai_new)); - dai_new.name = cs4270->name; + + /* Register the codec DAI */ + + memset(&dai_new, 0, sizeof(dai_new)); + dai_new.name = codec->name; + dai_new.id = codec->num; dai_new.playback = &cs4270->playback; dai_new.capture = &cs4270->capture; dai_new.ops = &dai_ops; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 4beb61e..45c3311 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -42,7 +42,7 @@ struct mpc8610_hpcd_data { struct snd_soc_card soc_card; struct ccsr_guts __iomem *guts; unsigned int num_configs; - struct snd_soc_pcm_config pcm_configs[MAX_SSI]; + struct snd_soc_pcm_config configs[MAX_SSI]; struct mpc8610_hpcd_names { char pcm_name[32]; char ssi_name[32]; @@ -265,72 +265,33 @@ static void mpc8610_hpcd_shutdown(struct snd_pcm_substream *substream) } /** - * get_codec_name: get the name of the codec that a given SSI is using - * @ssi_np: pointer to the SSI node - * @name: returned name of the codec driver - * - * This function returns the name of the codec device that matches a given - * SSI. - * - * The purpose behind this function is to make the device tree the sole - * source of information concerning the linkage between devices in the ASoC - * device chain. Unfortunately, it's not really as elegant as it should be. - * - * With this function, we should be able to mix-and-match codecs just by - * loading the right codec driver. + * find_codec: return the node for the codec of a given SSI + * @ssi_np: the device node for the SSI to look up */ -static int get_codec_name(struct device_node *ssi_np, char *name) +static struct device_node *find_codec(struct device_node *ssi_np) { const phandle *codec_ph; - struct device_node *codec_np; - struct device_node *i2c_np; - unsigned int bus; - unsigned int addr; - const unsigned int *iprop; - const char *compat; codec_ph = of_get_property(ssi_np, "codec-handle", NULL); - if (!codec_ph) { - pr_debug("%s: no codec handle\n", ssi_np->full_name); - return 0; - } - - codec_np = of_find_node_by_phandle(*codec_ph); - if (!codec_np) { - pr_debug("%s: codec node does not exist\n", ssi_np->full_name); - return -1; - } + if (!codec_ph) + return NULL; - /* We assume that the first (or only) string in the compatible field is - * the one that counts. - */ - compat = of_get_property(codec_np, "compatible", NULL); + return of_find_node_by_phandle(*codec_ph); +} - /* We only care about the part after the comma */ - compat = strchr(compat, ',') + 1; +static int of_get_integer(struct device_node *np, const char *string) +{ + const u32 *iprop; + int len; - /* Now determine the I2C bus and address it's on */ - i2c_np = of_get_parent(codec_np); - iprop = of_get_property(i2c_np, "cell-index", NULL); - if (!iprop) { - pr_debug("%s: cannot determine I2C bus number\n", - codec_np->full_name); + iprop = of_get_property(np, string, &len); + if (!iprop) return -1; - } - bus = *iprop; - iprop = of_get_property(codec_np, "reg", NULL); - if (!iprop) { - pr_debug("%s: cannot determine I2C address\n", - codec_np->full_name); + if (len != sizeof(u32)) return -1; - } - addr = *iprop; - - /* Now build the string */ - sprintf(name, "%s@%d-%04x", compat, bus, addr); - return 1; + return *iprop; } /** @@ -356,7 +317,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) struct mpc8610_hpcd_data *soc_card_data; struct device_node *guts_np = NULL; struct device_node *ssi_np; - unsigned int i = 0; + unsigned int ssi = 0; int ret = -ENODEV; soc_card_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); @@ -396,12 +357,17 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) * codec is registered. Remember, we only support MAX_SSI of these. */ for_each_compatible_node(ssi_np, NULL, "fsl,mpc8610-ssi") { - const unsigned int *iprop; struct snd_soc_pcm_config *pcm_config; struct mpc8610_hpcd_names *name; + struct device_node *codec_np; + const char *compat; + const char *p; + unsigned int bus; + unsigned int address; + int i; - name = &soc_card_data->names[i]; - pcm_config = &soc_card_data->pcm_configs[i]; + name = &soc_card_data->names[ssi]; + pcm_config = &soc_card_data->configs[ssi]; pcm_config->name = name->pcm_name; pcm_config->codec = name->codec_name; @@ -412,39 +378,74 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) pcm_config->playback = 1; pcm_config->capture = 1; - sprintf(name->pcm_name, "MPC8610HPCD.%u", i); + sprintf(name->pcm_name, "MPC8610HPCD.%u", ssi); /* Get the SSI name */ - iprop = of_get_property(ssi_np, "cell-index", NULL); - if (!iprop) { + i = of_get_integer(ssi_np, "cell-index"); + if (i < 0) { dev_err(soc_card->dev, "no cell-index for %s\n", ssi_np->full_name); continue; } - sprintf(name->ssi_name, "ssi%u", *iprop); + sprintf(name->ssi_name, "ssi%u", i); - /* Get the codec name */ - ret = get_codec_name(ssi_np, name->codec_name); - if (ret < 0) - dev_err(soc_card->dev, "could not get codec for %s\n", + /* Get the codec name and ID */ + codec_np = find_codec(ssi_np); + if (!codec_np) { + dev_err(soc_card->dev, "missing codec node for %s\n", ssi_np->full_name); - if (ret < 1) continue; + } + + /* We assume that the first (or only) string in the compatible + * field is the one that counts. + */ + compat = of_get_property(codec_np, "compatible", NULL); + if (!compat) { + dev_err(soc_card->dev, + "missing compatible property for %s\n", + ssi_np->full_name); + continue; + } + + /* We only care about the part after the comma */ + p = strchr(compat, ','); + strcpy(name->codec_name, p ? p + 1 : compat); + + /* Now determine the I2C bus and address of the codec */ + bus = of_get_integer(of_get_parent(codec_np), "cell-index"); + if (bus < 0) { + dev_err(soc_card->dev, + "cannot determine I2C bus number for %s\n", + codec_np->full_name); + continue; + } + + address = of_get_integer(codec_np, "reg"); + if (address < 0) { + dev_err(soc_card->dev, + "cannot determine I2C address for %s\n", + codec_np->full_name); + continue; + } + + pcm_config->codec_num = bus << 16 | address; - dev_dbg(soc_card->dev, "registering cpu %s with codec %s\n", - pcm_config->cpu_dai, pcm_config->codec); + dev_dbg(soc_card->dev, "registering cpu %s with codec %s-%x\n", + pcm_config->cpu_dai, pcm_config->codec, + pcm_config->codec_num); - if (++i == MAX_SSI) + if (++ssi == MAX_SSI) break; } - ret = snd_soc_card_create_pcms(soc_card, soc_card_data->pcm_configs, i); + ret = snd_soc_card_create_pcms(soc_card, soc_card_data->configs, ssi); if (ret) { dev_err(soc_card->dev, "could not create PCMs\n"); return ret; } - soc_card_data->num_configs = i; + soc_card_data->num_configs = ssi; platform_set_drvdata(pdev, soc_card); -- 1.5.5

2 1

[alsa-devel] [PATCH][ASoC v2] Fix Kconfig and Makefile for Freescale MPC8610 sound drivers
by Timur Tabi 03 Jun '08

03 Jun '08

Fix the Kconfig for the Freescale MPC8610 sound drivers so that only the board is selectable, but the SOC is still a Kconfig option. Fix the Makefile so that all of the modules are named snd-soc-xxx.ko. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- My patch, "Add CS4270 i2c data to fsl_soc.c" must be applied for these drivers to work. sound/soc/fsl/Kconfig | 13 ++++++------- sound/soc/fsl/Makefile | 7 +++++-- 2 files changed, 11 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 257101f..056dc2f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,20 +1,19 @@ -menu "ALSA SoC audio for Freescale SOCs" config SND_SOC_MPC8610 - bool "ALSA SoC support for the MPC8610 SOC" - depends on SND_SOC && MPC8610_HPCD - default y if MPC8610 + tristate + # ALSA SoC support for the MPC8610 SOC + depends on MPC8610 help Say Y if you want to add support for codecs attached to the SSI device on an MPC8610. config SND_SOC_MPC8610_HPCD - bool "ALSA SoC support for the Freescale MPC8610 HPCD board" - depends on SND_SOC_MPC8610 + tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" + depends on SND_SOC && MPC8610_HPCD + select SND_SOC_MPC8610 select SND_SOC_CS4270 select SND_SOC_CS4270_VD33_ERRATA default y if MPC8610_HPCD help Say Y if you want to enable audio on the Freescale MPC8610 HPCD. -endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 62f680a..3a2b64e 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -1,6 +1,9 @@ # MPC8610 HPCD Machine Support -obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o +snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o +obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o # MPC8610 Platform Support -obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o +snd-soc-fsl-ssi-objs := fsl_ssi.o +snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o -- 1.5.5

2 1

[alsa-devel] [PATCH][ASoC V2] Make Freescale MPC8610 drivers more dynamic
by Timur Tabi 02 Jun '08

02 Jun '08

More miscellaneous changes to the Freescale MPC8610 driver set, with the goal of supporting multiple SSIs. Minor documentation improvements. Allow teh CS4270 registers to be read via sysfs. Create the device names from the device tree, to allow multiple devices. Remove separate Kconfig option for the MPC8610 SOC. Various minor code optimizations. Improve programming of the global utilties registers. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- My patch, "Add CS4270 i2c data to fsl_soc.c" must be applied for these drivers to work. sound/soc/codecs/cs4270.c | 85 ++++++---- sound/soc/fsl/Kconfig | 28 +--- sound/soc/fsl/Makefile | 7 +- sound/soc/fsl/fsl_dma.c | 7 +- sound/soc/fsl/fsl_dma.h | 5 - sound/soc/fsl/fsl_ssi.c | 57 +++---- sound/soc/fsl/fsl_ssi.h | 19 ++- sound/soc/fsl/mpc8610_hpcd.c | 350 ++++++++++++++++++++++++++++++------------ 8 files changed, 354 insertions(+), 204 deletions(-) rewrite sound/soc/fsl/Kconfig (77%) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 6bb4bc7..410fed5 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -104,18 +104,28 @@ #define CS4270_MUTE_DAC_A 0x01 #define CS4270_MUTE_DAC_B 0x02 -/* Private data for the CS4270 */ +/** + * struct cs4270_private: per-codec private data for the CS4270 + * @codec: codec structure + * @name: name of this codec + * @mclk: input frequency of the MCLK pin + * @mode: mode (I2S or left-justified) + * @reg_cache: register cache + * + * The register cache is NUMREGS+1 because the hardware registers are + * indexed from 1, not 0. The ASoC function that reads the registers + * assumes it's indexed from 0, so to maintain compatibility we pretend + * there is a register 0. + */ struct cs4270_private { struct snd_soc_codec codec; char name[16]; - unsigned int mclk; /* Input frequency of the MCLK pin */ - unsigned int mode; /* The mode (I2S or left-justified) */ - struct snd_soc_dai_ops dai_ops; + unsigned int mclk; + unsigned int mode; struct snd_soc_dai *dai; - struct snd_soc_dai_new dai_new; struct snd_soc_dai_caps playback; struct snd_soc_dai_caps capture; - u8 reg_cache[CS4270_NUMREGS]; + u8 reg_cache[CS4270_NUMREGS + 1]; }; /* @@ -169,8 +179,8 @@ static struct { /* The number of MCLK/LRCK ratios supported by the CS4270 */ #define NUM_MCLK_RATIOS ARRAY_SIZE(cs4270_mode_ratios) -/* - * Determine the CS4270 samples rates. +/** + * cs4270_set_dai_sysclk: determine the CS4270 samples rates. * * 'freq' is the input frequency to MCLK. The other parameters are ignored. * @@ -284,10 +294,10 @@ static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec, container_of(codec, struct cs4270_private, codec); u8 *cache = cs4270->reg_cache; - if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) + if (reg > CS4270_LASTREG) return -EIO; - return cache[reg - CS4270_FIRSTREG]; + return cache[reg]; } /* @@ -311,7 +321,7 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, return -EIO; /* Only perform an I2C operation if the new value is different */ - if (cache[reg - CS4270_FIRSTREG] != value) { + if (cache[reg] != value) { struct i2c_client *client = codec->control_data; if (i2c_smbus_write_byte_data(client, reg, value)) { @@ -320,7 +330,7 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, } /* We've written to the hardware, so update the cache */ - cache[reg - CS4270_FIRSTREG] = value; + cache[reg] = value; } return 0; @@ -481,6 +491,13 @@ static int cs4270_codec_init(struct snd_soc_codec *codec, return ret; } +static struct snd_soc_dai_ops dai_ops = { + .hw_params = cs4270_hw_params, + .set_sysclk = cs4270_set_dai_sysclk, + .set_fmt = cs4270_set_dai_fmt, + .digital_mute = cs4270_mute, +}; + /* * Initialize the I2C interface of the CS4270 * @@ -495,8 +512,8 @@ static int cs4270_i2c_probe(struct i2c_client *client, { struct cs4270_private *cs4270; struct snd_soc_codec *codec = NULL; + struct snd_soc_dai_new dai_new; int ret = 0; - int registered = 0; /* 1 == the codec has been registered */ /* Verify that we have a CS4270 */ @@ -515,8 +532,9 @@ static int cs4270_i2c_probe(struct i2c_client *client, return -ENODEV; } - dev_info(&client->dev, "found device at address %X\n", client->addr); - dev_info(&client->dev, "hardware revision %X\n", ret & 0xF); + dev_info(&client->dev, + "found device at address %X, hardware revision %u\n", + client->addr, ret & 0xF); /* * Normally, we'd call snd_soc_new_codec, but that function @@ -532,15 +550,20 @@ static int cs4270_i2c_probe(struct i2c_client *client, /* The I2C interface is set up, so pre-fill our register cache */ ret = i2c_smbus_read_i2c_block_data(client, CS4270_FIRSTREG | 0x80, - CS4270_NUMREGS, cs4270->reg_cache); + CS4270_NUMREGS, cs4270->reg_cache + 1); if (ret != CS4270_NUMREGS) { dev_err(&client->dev, "failed to fill register cache\n"); ret = -EIO; goto error; } - strcpy(cs4270->name, "cirrus,cs4270"); + sprintf(cs4270->name, "%s@%s", dev_driver_string(&client->dev), + dev_name(&client->dev)); + /* Initialize the playback and capture stream data. The sample rate + * fields are initialized later when the fabric driver calls + * cs4270_set_dai_sysclk(). Hopefully, this won't cause any problems. + */ cs4270->playback.stream_name = "Playback"; cs4270->playback.channels_min = 1; cs4270->playback.channels_max = 2; @@ -551,11 +574,6 @@ static int cs4270_i2c_probe(struct i2c_client *client, cs4270->capture.channels_max = 2; cs4270->capture.formats = CS4270_FORMATS; - cs4270->dai_ops.hw_params = cs4270_hw_params; - cs4270->dai_ops.set_sysclk = cs4270_set_dai_sysclk; - cs4270->dai_ops.set_fmt = cs4270_set_dai_fmt; - cs4270->dai_ops.digital_mute = cs4270_mute; - codec = &cs4270->codec; mutex_init(&codec->mutex); @@ -563,29 +581,29 @@ static int cs4270_i2c_probe(struct i2c_client *client, INIT_LIST_HEAD(&codec->dai_list); codec->name = cs4270->name; - codec->control_data = client; + snd_soc_card_config_codec(codec, NULL, NULL, client); codec->init = cs4270_codec_init; codec->reg_cache = cs4270->reg_cache; codec->codec_read = cs4270_read_reg_cache; codec->codec_write = cs4270_i2c_write; + codec->reg_cache_size = sizeof(cs4270->reg_cache); ret = snd_soc_register_codec(codec, &client->dev); if (ret < 0) { dev_err(&client->dev, "failed to register card\n"); goto error; } - registered = 1; + memset(&dai_new, 0, sizeof(dai_new)); + dai_new.name = cs4270->name; + dai_new.playback = &cs4270->playback; + dai_new.capture = &cs4270->capture; + dai_new.ops = &dai_ops; - cs4270->dai_new.name = cs4270->name; - cs4270->dai_new.playback = &cs4270->playback; - cs4270->dai_new.capture = &cs4270->capture; - cs4270->dai_new.ops = &cs4270->dai_ops; - - cs4270->dai = - snd_soc_register_codec_dai(&cs4270->dai_new, &client->dev); + cs4270->dai = snd_soc_register_codec_dai(&dai_new, &client->dev); if (!cs4270->dai) { dev_err(&client->dev, "failed to register DAI\n"); ret = -EINVAL; + snd_soc_unregister_codec(codec); goto error; } @@ -597,9 +615,6 @@ error: if (cs4270->dai) snd_soc_unregister_codec_dai(cs4270->dai); - if (registered) - snd_soc_unregister_codec(codec); - kfree(cs4270); return ret; @@ -646,7 +661,7 @@ static int __init cs4270_init(void) { int ret; - printk(KERN_INFO "Cirrus Logic CS4270 ASoC codec driver\n"); + pr_info("Cirrus Logic CS4270 ASoC codec driver\n"); /* i2c_add_driver() will call cs4270_i2c_probe() */ ret = i2c_add_driver(&cs4270_i2c_driver); diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig dissimilarity index 77% index 257101f..5e80774 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,20 +1,8 @@ -menu "ALSA SoC audio for Freescale SOCs" - -config SND_SOC_MPC8610 - bool "ALSA SoC support for the MPC8610 SOC" - depends on SND_SOC && MPC8610_HPCD - default y if MPC8610 - help - Say Y if you want to add support for codecs attached to the SSI - device on an MPC8610. - -config SND_SOC_MPC8610_HPCD - bool "ALSA SoC support for the Freescale MPC8610 HPCD board" - depends on SND_SOC_MPC8610 - select SND_SOC_CS4270 - select SND_SOC_CS4270_VD33_ERRATA - default y if MPC8610_HPCD - help - Say Y if you want to enable audio on the Freescale MPC8610 HPCD. - -endmenu +config SND_SOC_MPC8610_HPCD + tristate "Freescale MPC8610 HPCD support" + depends on SND_SOC && MPC8610_HPCD + select SND_SOC_CS4270 + select SND_SOC_CS4270_VD33_ERRATA + default y if SND_SOC && MPC8610_HPCD + help + Say Y or M if you want to enable audio on the Freescale MPC8610 HPCD. diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 62f680a..0392962 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -1,6 +1,3 @@ -# MPC8610 HPCD Machine Support -obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o - -# MPC8610 Platform Support -obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o +# MPC8610 HPCD Support +obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o fsl_ssi.o fsl_dma.o diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 2925ec0..fed18ac 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -794,8 +794,7 @@ static struct snd_pcm_ops fsl_dma_ops = { .pointer = fsl_dma_pointer, }; -const char fsl_platform_id[] = "fsl_pcm"; -EXPORT_SYMBOL_GPL(fsl_platform_id); +static const char fsl_platform_id[] = "fsl-elo"; static struct snd_soc_platform_new fsl_dma_platform = { .name = fsl_platform_id, @@ -852,7 +851,7 @@ static __init int fsl_dma_init(void) { int ret; - printk(KERN_INFO "Freescale Elo DMA ASoC PCM driver\n"); + pr_info("Freescale Elo DMA ASoC platform driver\n"); ret = platform_driver_register(&fsl_dma_driver); if (ret < 0) { @@ -880,6 +879,6 @@ module_init(fsl_dma_init); module_exit(fsl_dma_exit); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); -MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM driver"); +MODULE_DESCRIPTION("Freescale Elo DMA ASoC platform driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h index ed4592f..8bba2bf 100644 --- a/sound/soc/fsl/fsl_dma.h +++ b/sound/soc/fsl/fsl_dma.h @@ -140,9 +140,4 @@ struct fsl_dma_info { unsigned int channel_id; }; -extern const char fsl_platform_id[]; -extern struct snd_soc_dai fsl_ssi; /* TODO: probably not needed */ - -int fsl_dma_configure(struct fsl_dma_info *dma_info); - #endif diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 5fd0312..1d4c3f4 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -424,7 +424,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { - struct fsl_ssi_info *ssi_info = cpu_dai->private_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -652,8 +651,6 @@ static int fsl_ssi_probe(struct of_device *ofdev, return -ENOMEM; } - ssi_info->dev = &ofdev->dev; - /* * We are only interested in SSIs with a codec phandle in them, so let's * make sure this SSI has one. @@ -689,7 +686,7 @@ static int fsl_ssi_probe(struct of_device *ofdev, } ssi_info->id = *iprop; - strcpy(ssi_info->name, "fsl,mpc8610-ssi"); + sprintf(ssi_info->name, "ssi%u", ssi_info->id); /* Get the serial format and clock direction. */ sprop = of_get_property(np, "fsl,mode", NULL); @@ -703,20 +700,6 @@ static int fsl_ssi_probe(struct of_device *ofdev, ssi_info->dai_format = SND_SOC_DAIFMT_I2S; ssi_info->codec_clk_direction = SND_SOC_CLOCK_OUT; ssi_info->cpu_clk_direction = SND_SOC_CLOCK_IN; - - /* - * In i2s-slave mode, the codec has its own clock source, so we - * need to get the frequency from the device tree and pass it to - * the codec driver. - */ - iprop = of_get_property(codec_np, "clock-frequency", NULL); - if (!iprop || !*iprop) { - dev_err(&ofdev->dev, "codec bus-frequency property " - "is missing or invalid\n"); - ret = -EINVAL; - goto error; - } - ssi_info->clk_frequency = *iprop; } else if (strcasecmp(sprop, "i2s-master") == 0) { ssi_info->dai_format = SND_SOC_DAIFMT_I2S; ssi_info->codec_clk_direction = SND_SOC_CLOCK_IN; @@ -752,6 +735,19 @@ static int fsl_ssi_probe(struct of_device *ofdev, goto error; } + /* If the codec is the clock source, then the codec node should + contain the clock frequency. */ + if (ssi_info->codec_clk_direction == SND_SOC_CLOCK_OUT) { + iprop = of_get_property(codec_np, "clock-frequency", NULL); + if (!iprop || !*iprop) { + dev_err(&ofdev->dev, "codec clock-frequency property " + "is missing or invalid\n"); + ret = -EINVAL; + goto error; + } + ssi_info->clk_frequency = *iprop; + } + if (!ssi_info->clk_frequency) { dev_err(&ofdev->dev, "unknown clock frequency\n"); ret = -EINVAL; @@ -825,6 +821,8 @@ static int fsl_ssi_probe(struct of_device *ofdev, dai_template.capture = &capture; dai_template.ops = &ops; + /* If the other three drivers have loaded, then the call to + snd_soc_register_platform_dai() will initialize the system. */ ssi_info->dai = snd_soc_register_platform_dai(&dai_template, &ofdev->dev); if (!ssi_info->dai) { @@ -835,7 +833,7 @@ static int fsl_ssi_probe(struct of_device *ofdev, ssi_info->dai->private_data = ssi_info; - ssi_info->dev_attr.attr.name = ssi_info->name; + ssi_info->dev_attr.attr.name = "stats"; ssi_info->dev_attr.attr.mode = S_IRUGO; ssi_info->dev_attr.show = fsl_sysfs_ssi_show; @@ -848,12 +846,9 @@ static int fsl_ssi_probe(struct of_device *ofdev, dev_set_drvdata(&ofdev->dev, ssi_info); - - return 0; error: - if (ssi_info->dai) snd_soc_unregister_platform_dai(ssi_info->dai); @@ -877,18 +872,14 @@ static int fsl_ssi_remove(struct of_device *ofdev) { struct fsl_ssi_info *ssi_info = dev_get_drvdata(&ofdev->dev); - if (ssi_info->dai) + if (ssi_info) { + device_remove_file(&ofdev->dev, &ssi_info->dev_attr); snd_soc_unregister_platform_dai(ssi_info->dai); - - if (ssi_info->irq) irq_dispose_mapping(ssi_info->irq); - - if (ssi_info->ssi) iounmap(ssi_info->ssi); - - kfree(ssi_info); - - dev_set_drvdata(&ofdev->dev, NULL); + kfree(ssi_info); + dev_set_drvdata(&ofdev->dev, NULL); + } return 0; } @@ -918,7 +909,7 @@ static int __init fsl_ssi_init(void) { int ret; - printk(KERN_INFO "Freescale SSI ASoC driver\n"); + pr_info("Freescale SSI ASoC CPU driver\n"); ret = of_register_platform_driver(&fsl_ssi_of_driver); @@ -943,5 +934,5 @@ module_init(fsl_ssi_init); module_exit(fsl_ssi_exit); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); -MODULE_DESCRIPTION("Freescale SSI ASoC driver"); +MODULE_DESCRIPTION("Freescale SSI ASoC CPU driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 9a22b04..91af81c 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -200,14 +200,22 @@ struct ccsr_ssi { * fsl_ssi_info: per-SSI private data * * @name: short name for this device ("SSI0", "SSI1", etc) + * @id: 0=SS1, 1=SSI2, etc * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI - * @dev: struct device pointer - * @playback: the number of playback streams opened - * @capture: the number of capture streams opened - * @cpu_dai: the CPU DAI for this device + * @playback: 1=playback stream open, 0 otherwise + * @capture: 1=capture stream open, 0 otherwise + * @lock: spinlock for enabling/disabling the SSI + * @dai: the CPU DAI for this device * @dev_attr: the sysfs device attribute structure + * @pmuxcr: saved value of the PMUXCR register + * @dmcr: saved value of the DMACR register + * @dai_format: the serial protocol format (I2S, left-justified, etc) + * @codec_clk_direction: whether the codec is the clock master or slave + * @cpu_clk_direction: whether the SSI is the clock master or slave + * @clk_frequency: the base frequency of the clock + * @dma_info: information on the two DMA channels used by this SSI * @stats: SSI statistics */ struct fsl_ssi_info { @@ -216,12 +224,13 @@ struct fsl_ssi_info { struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; - struct device *dev; unsigned int playback; unsigned int capture; spinlock_t lock; struct snd_soc_dai *dai; struct device_attribute dev_attr; + u32 pmuxcr; + u32 dmacr; unsigned int dai_format; unsigned int codec_clk_direction; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 10c7fd7..4beb61e 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -1,5 +1,5 @@ /** - * Freescale MPC8610HPCD ALSA SoC Fabric driver + * Freescale MPC8610 HPCD ASoC fabric driver * * Author: Timur Tabi <timur(a)freescale.com> * @@ -9,6 +9,8 @@ * express or implied. */ +#define DEBUG + #include <linux/module.h> #include <linux/interrupt.h> #include <linux/of_device.h> @@ -24,6 +26,11 @@ #include "fsl_dma.h" #include "fsl_ssi.h" +/* To keep memory allocations simple, we define a limit to the number of + * SSIs we support. + */ +#define MAX_SSI 2 + /** * mpc8610_hpcd_data: fabric-specific ASoC device data * @@ -31,75 +38,82 @@ * MPC8610 HPCD. Some of the data is taken from the device tree. */ struct mpc8610_hpcd_data { + char codec_name[32]; struct snd_soc_card soc_card; - unsigned int dai_format; - unsigned int codec_clk_direction; - unsigned int cpu_clk_direction; - unsigned int clk_frequency; struct ccsr_guts __iomem *guts; + unsigned int num_configs; + struct snd_soc_pcm_config pcm_configs[MAX_SSI]; + struct mpc8610_hpcd_names { + char pcm_name[32]; + char ssi_name[32]; + char codec_name[32]; + char platform_name[32]; + char cpu_dai_name[32]; + } names[MAX_SSI]; }; /** + * guts_dmacr_mask: return the DMACR bitmask for a given DMA channel + * @co: The DMA controller (0 or 1) + * @ch: The channel on the DMA controller (0, 1, 2, or 3) + */ +static inline u32 guts_dmacr_mask(unsigned int co, unsigned int ch) +{ + unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch)); + + return 3 << shift; +} + +/** + * guts_pmuxcr_dma_mask: return the mask for the DMA bits in the PMXUCR + * @co: The DMA controller (0 or 1) + * @ch: The channel on the DMA controller (0, 1, 2, or 3) + * + * If ch is not 0 or 3, then a value of 0 is returned. + */ +static inline u32 guts_pmuxcr_dma_mask(unsigned int co, unsigned int ch) +{ + if ((ch == 0) || (ch == 3)) { + unsigned int shift = 2 * (co + 1) - (ch & 1) - 1; + + return 1 << shift; + } else + return 0; +} + +/** * mpc8610_hpcd_audio_init: initalize the board * * This function is called when platform_device_add() is called. It is used * to initialize the board-specific hardware. * - * Here we program the DMACR and PMUXCR registers. + * If the platform (SSI) driver is loaded last, then cpu_dai->private_data + * is not yet initialized, so therefore we cannot get ssi_info. */ static int mpc8610_hpcd_audio_init(struct snd_soc_card *soc_card) { + struct snd_soc_codec *codec; struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; - struct snd_soc_dai *cpu_dai = - snd_soc_card_get_dai(soc_card, "fsl,mpc8610-ssi"); - struct snd_soc_codec *codec = - snd_soc_card_get_codec(soc_card, "cirrus,cs4270"); - struct fsl_ssi_info *ssi_info = cpu_dai->private_data; - int ret; - - /* This is stupid. snd_soc_card_config_codec() initializes the - * codec->control_data structure, which is supposed to be a pointer to - * the i2c_client structure. But I already assigned that variable in - * the codec driver. After all, the codec driver is the one that - * supposed to get probed by the I2C bus. So in order to avoid - * overwriting codec->control_data, I pass it as the 4th parameter. - */ - snd_soc_card_config_codec(codec, NULL, NULL, codec->control_data); - ret = snd_soc_card_init_codec(codec, soc_card); - if (ret < 0) { - dev_err(soc_card->dev, "could not initialize codec\n"); - return ret; - } - - /* Program the signal routing between the SSI and the DMA */ - guts_set_dmacr(soc_card_data->guts, - ssi_info->dma_info[0].controller_id, - ssi_info->dma_info[0].channel_id, CCSR_GUTS_DMACR_DEV_SSI); - guts_set_dmacr(soc_card_data->guts, - ssi_info->dma_info[1].controller_id, - ssi_info->dma_info[1].channel_id, CCSR_GUTS_DMACR_DEV_SSI); - - guts_set_pmuxcr_dma(soc_card_data->guts, - ssi_info->dma_info[0].controller_id, - ssi_info->dma_info[0].channel_id, 0); - guts_set_pmuxcr_dma(soc_card_data->guts, - ssi_info->dma_info[1].controller_id, - ssi_info->dma_info[1].channel_id, 0); - - /* FIXME: Magic numbers? */ - guts_set_pmuxcr_dma(soc_card_data->guts, 1, 0, 0); - guts_set_pmuxcr_dma(soc_card_data->guts, 1, 3, 0); - guts_set_pmuxcr_dma(soc_card_data->guts, 0, 3, 0); + unsigned int i; + int ret = 0; - switch (ssi_info->id) { - case 0: - clrsetbits_be32(&soc_card_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI); - break; - case 1: - clrsetbits_be32(&soc_card_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI); - break; + for (i = 0; i < soc_card_data->num_configs; i++) { + codec = snd_soc_card_get_codec(soc_card, + soc_card_data->names[i].codec_name); + + if (!codec) { + dev_err(soc_card->dev, "could not find codec\n"); + return -ENODEV; + } + + /* The codec driver should have called + * snd_soc_card_config_codec() by now. + */ + ret = snd_soc_card_init_codec(codec, soc_card); + if (ret < 0) { + dev_err(soc_card->dev, "could not initialize codec\n"); + continue; + } } return 0; @@ -138,8 +152,42 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct fsl_ssi_info *ssi_info = cpu_dai->private_data; + struct snd_soc_card *soc_card = rtd->soc_card; + struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; int ret = 0; + /* Save the PMUXCR and DMACR registers so that we can restore them + * properly later. + */ + ssi_info->dmacr = in_be32(&soc_card_data->guts->dmacr); + ssi_info->pmuxcr = in_be32(&soc_card_data->guts->pmuxcr); + + /* Program the signal routing between the SSI and the DMA */ + guts_set_dmacr(soc_card_data->guts, + ssi_info->dma_info[0].controller_id, + ssi_info->dma_info[0].channel_id, CCSR_GUTS_DMACR_DEV_SSI); + guts_set_dmacr(soc_card_data->guts, + ssi_info->dma_info[1].controller_id, + ssi_info->dma_info[1].channel_id, CCSR_GUTS_DMACR_DEV_SSI); + + guts_set_pmuxcr_dma(soc_card_data->guts, + ssi_info->dma_info[0].controller_id, + ssi_info->dma_info[0].channel_id, 0); + guts_set_pmuxcr_dma(soc_card_data->guts, + ssi_info->dma_info[1].controller_id, + ssi_info->dma_info[1].channel_id, 0); + + switch (ssi_info->id) { + case 0: + clrsetbits_be32(&soc_card_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI); + break; + case 1: + clrsetbits_be32(&soc_card_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI); + break; + } + /* Tell the CPU driver what the serial protocol is. */ ret = snd_soc_dai_set_fmt(cpu_dai, ssi_info->dai_format); if (ret < 0) { @@ -183,38 +231,106 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) return 0; } +static void mpc8610_hpcd_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct fsl_ssi_info *ssi_info = cpu_dai->private_data; + struct snd_soc_card *soc_card = rtd->soc_card; + struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; + u32 mask; + + soc_card_data = soc_card->private_data; + ssi_info = cpu_dai->private_data; + + /* Restore the DMACR register */ + mask = guts_dmacr_mask(ssi_info->dma_info[0].controller_id, + ssi_info->dma_info[0].channel_id) | + guts_dmacr_mask(ssi_info->dma_info[1].controller_id, + ssi_info->dma_info[1].channel_id); + + clrsetbits_be32(&soc_card_data->guts->dmacr, mask, + ssi_info->dmacr & mask); + + /* Restore the PMUXCR register */ + mask = guts_pmuxcr_dma_mask(ssi_info->dma_info[0].controller_id, + ssi_info->dma_info[0].channel_id) | + guts_pmuxcr_dma_mask(ssi_info->dma_info[1].controller_id, + ssi_info->dma_info[1].channel_id) | + (ssi_info->id ? CCSR_GUTS_PMUXCR_SSI2_MASK : + CCSR_GUTS_PMUXCR_SSI1_MASK); + + clrsetbits_be32(&soc_card_data->guts->pmuxcr, mask, + ssi_info->pmuxcr & mask); +} + /** - * mpc8610_hpcd_audio_exit: Remove the sound device + * get_codec_name: get the name of the codec that a given SSI is using + * @ssi_np: pointer to the SSI node + * @name: returned name of the codec driver + * + * This function returns the name of the codec device that matches a given + * SSI. + * + * The purpose behind this function is to make the device tree the sole + * source of information concerning the linkage between devices in the ASoC + * device chain. Unfortunately, it's not really as elegant as it should be. * - * This function is called to remove the sound device for one SSI. We - * de-program the DMACR and PMUXCR register. + * With this function, we should be able to mix-and-match codecs just by + * loading the right codec driver. */ -void mpc8610_hpcd_audio_exit(struct snd_soc_card *soc_card) +static int get_codec_name(struct device_node *ssi_np, char *name) { - struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; - struct snd_soc_dai *cpu_dai = - snd_soc_card_get_dai(soc_card, "fsl,mpc8610-ssi"); - struct fsl_ssi_info *ssi_info = cpu_dai->private_data; + const phandle *codec_ph; + struct device_node *codec_np; + struct device_node *i2c_np; + unsigned int bus; + unsigned int addr; + const unsigned int *iprop; + const char *compat; + + codec_ph = of_get_property(ssi_np, "codec-handle", NULL); + if (!codec_ph) { + pr_debug("%s: no codec handle\n", ssi_np->full_name); + return 0; + } - /* Restore the signal routing */ + codec_np = of_find_node_by_phandle(*codec_ph); + if (!codec_np) { + pr_debug("%s: codec node does not exist\n", ssi_np->full_name); + return -1; + } - guts_set_dmacr(soc_card_data->guts, - ssi_info->dma_info[0].controller_id, - ssi_info->dma_info[0].channel_id, 0); - guts_set_dmacr(soc_card_data->guts, - ssi_info->dma_info[1].controller_id, - ssi_info->dma_info[1].channel_id, 0); + /* We assume that the first (or only) string in the compatible field is + * the one that counts. + */ + compat = of_get_property(codec_np, "compatible", NULL); + + /* We only care about the part after the comma */ + compat = strchr(compat, ',') + 1; + + /* Now determine the I2C bus and address it's on */ + i2c_np = of_get_parent(codec_np); + iprop = of_get_property(i2c_np, "cell-index", NULL); + if (!iprop) { + pr_debug("%s: cannot determine I2C bus number\n", + codec_np->full_name); + return -1; + } + bus = *iprop; - switch (ssi_info->id) { - case 0: - clrsetbits_be32(&soc_card_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); - break; - case 1: - clrsetbits_be32(&soc_card_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); - break; + iprop = of_get_property(codec_np, "reg", NULL); + if (!iprop) { + pr_debug("%s: cannot determine I2C address\n", + codec_np->full_name); + return -1; } + addr = *iprop; + + /* Now build the string */ + sprintf(name, "%s@%d-%04x", compat, bus, addr); + + return 1; } /** @@ -222,17 +338,7 @@ void mpc8610_hpcd_audio_exit(struct snd_soc_card *soc_card) */ static struct snd_soc_ops mpc8610_hpcd_ops = { .startup = mpc8610_hpcd_startup, -}; - -static struct snd_soc_pcm_config mpc8610_pcm_config = { - .name = "MPC8610 HPCD", - .codec = "cirrus,cs4270", - .codec_dai = "cirrus,cs4270", - .platform = "fsl_pcm", - .cpu_dai = "fsl,mpc8610-ssi", - .ops = &mpc8610_hpcd_ops, - .playback = 1, - .capture = 1, + .shutdown = mpc8610_hpcd_shutdown, }; /** @@ -249,6 +355,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) struct snd_soc_card *soc_card = NULL; struct mpc8610_hpcd_data *soc_card_data; struct device_node *guts_np = NULL; + struct device_node *ssi_np; + unsigned int i = 0; int ret = -ENODEV; soc_card_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); @@ -281,16 +389,63 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) soc_card->longname = "MPC8610HPCD"; soc_card->init = mpc8610_hpcd_audio_init; - soc_card->exit = mpc8610_hpcd_audio_exit; soc_card->private_data = soc_card_data; soc_card->dev = &pdev->dev; - ret = snd_soc_card_create_pcms(soc_card, &mpc8610_pcm_config, 1); + /* Scan the device tree for SSI nodes. Each one we find that has a + * codec is registered. Remember, we only support MAX_SSI of these. + */ + for_each_compatible_node(ssi_np, NULL, "fsl,mpc8610-ssi") { + const unsigned int *iprop; + struct snd_soc_pcm_config *pcm_config; + struct mpc8610_hpcd_names *name; + + name = &soc_card_data->names[i]; + pcm_config = &soc_card_data->pcm_configs[i]; + + pcm_config->name = name->pcm_name; + pcm_config->codec = name->codec_name; + pcm_config->codec_dai = name->codec_name; + pcm_config->platform = "fsl-elo"; + pcm_config->cpu_dai = name->ssi_name; + pcm_config->ops = &mpc8610_hpcd_ops; + pcm_config->playback = 1; + pcm_config->capture = 1; + + sprintf(name->pcm_name, "MPC8610HPCD.%u", i); + + /* Get the SSI name */ + iprop = of_get_property(ssi_np, "cell-index", NULL); + if (!iprop) { + dev_err(soc_card->dev, "no cell-index for %s\n", + ssi_np->full_name); + continue; + } + sprintf(name->ssi_name, "ssi%u", *iprop); + + /* Get the codec name */ + ret = get_codec_name(ssi_np, name->codec_name); + if (ret < 0) + dev_err(soc_card->dev, "could not get codec for %s\n", + ssi_np->full_name); + if (ret < 1) + continue; + + dev_dbg(soc_card->dev, "registering cpu %s with codec %s\n", + pcm_config->cpu_dai, pcm_config->codec); + + if (++i == MAX_SSI) + break; + } + + ret = snd_soc_card_create_pcms(soc_card, soc_card_data->pcm_configs, i); if (ret) { - dev_err(&pdev->dev, "could not create PCMs\n"); - goto error; + dev_err(soc_card->dev, "could not create PCMs\n"); + return ret; } + soc_card_data->num_configs = i; + platform_set_drvdata(pdev, soc_card); ret = snd_soc_card_register(soc_card); @@ -330,7 +485,7 @@ static int mpc8610_hpcd_remove(struct platform_device *pdev) static struct platform_driver mpc8610_hpcd_driver = { .driver = { - .name = "mpc8610_hpcd", + .name = "MPC8610HPCD", .owner = THIS_MODULE, }, .probe = mpc8610_hpcd_probe, @@ -348,7 +503,7 @@ static int __init mpc8610_hpcd_init(void) { int ret; - printk(KERN_INFO "Freescale MPC8610 HPCD ASoC fabric driver\n"); + pr_info("Freescale MPC8610 HPCD ASoC fabric driver\n"); ret = platform_driver_register(&mpc8610_hpcd_driver); if (ret < 0) { @@ -356,7 +511,8 @@ static int __init mpc8610_hpcd_init(void) return ret; } - pdev = platform_device_register_simple("mpc8610_hpcd", 0, NULL, 0); + pdev = platform_device_register_simple(mpc8610_hpcd_driver.driver.name, + 0, NULL, 0); if (!pdev) { pr_err("mpc8610-hpcd: could not register device\n"); platform_driver_unregister(&mpc8610_hpcd_driver); -- 1.5.5

2 3

[alsa-devel] hda intel - power save - need more help
by Harsha priya gupta 02 Jun '08

02 Jun '08

I checked on these articles on powertop...i manually set it the value to 10 for *POWER_SAVE_DEFAULT but still don't see any power saving... Has anyone tried this before and seen any power saving? please help me with more clues... -- -Harsha

2 5

[alsa-devel] playback write error (DMA or IRQ trouble)
by Harsha priya gupta 02 Jun '08

02 Jun '08

Can anyone give me a clue as to when i would get such an error? -- -Harsha

2 11

[alsa-devel] [PATCH] improved snd-aloop quality when using certain samplerates and kernel HZ
by Ahmet İnan 02 Jun '08

02 Jun '08

after using a HZ:=300 kernel for some time now and needing snd-aloop again i found out, that i either had to correct the hardcoded number, or fix the problem permanently, which couses stuttering and other problems. i also removed the SYNC on start code, not just because its more or less useless and brings problems, but ive got a nice idea how to fix this overrun/underrun problem in a much nicer way. so stay tuned for the next patch, until its implemented. this patch here is more or less experimental, id like to hear some feedback. http://www.mathematik.uni-freiburg.de/IAM/homepages/ainan/alsa-driver-1.0.1… Signed-off-by: Ahmet İnan <ainan <at> mathematik.uni-freiburg.de> please also include my email address when responding - i have no intention to enable recieving emails from the list. ahmet -- admin der abteilung für angewandte mathematik, tel. 0761-203-5626

3 16

Re: [alsa-devel] power save of snd_hda_intel
by Harsha priya gupta 02 Jun '08

02 Jun '08

I checked on these articles...i manually set it the value to 10 but still dont see any power saving... has anyone tried this before and seen any power saving? please help me with more clues... On Wed, May 28, 2008 at 9:44 PM, Chris Pemberton < cjpembo.mailing.lists(a)gmail.com> wrote: > Harsha priya gupta wrote: > >> Hi, >> >> I am trying to get a power_save option of snd_hda_intel working. >> We are using latest ALSA build and ALC262 codec. when we start the player >> and close it, we dont see any power savings. [ We see the power saving in >> windows with the same ALC262 card] >> >> Any idea what i could be doing wrong here? >> >> >> > powertop is your friend > > www.lesswatts.org > > specifically: > > http://www.lesswatts.org/tips/misc.php > > Chris > -- -Harsha

1 0

Re: [alsa-devel] [Alsa-user] Async behavior and -EAGAIN on write?
by Paul Adolph 02 Jun '08

02 Jun '08

>> For playback I am using the ALSA callback to wake up a thread that >> then feeds the ALSA ring buffer with a period-worth of samples using >> snd_pcm_writei(). I've set snd_pcm_sw_params_set_avail_min() to one >> period. The problem is that most of the time when my thread wakes up >> and tries to write the data, snd_pcm_writei() fails with -EAGAIN, and >> I must retry until it works. >> >> Presumably this write failure is because there is not enough space in >> the ring buffer, but then why did the ALSA callback fire in the first >> place? As it stands my retries are eating needless amounts of CPU. >> > > What does snd_pcm_avail_update return if you call it before > snd_pcm_writei in your callback? Taking the sleep out of my thread so I respond to the ALSA callback wakeups as often as possible, I found that snd_pcm_avail_update() returns very small values, usually less than 1/2 the period size and sometimes even 0! It seems as if I'm getting callbacks whenever ALSA feels like giving them. This behavior does not seem to change at all even with different settings of snd_pcm_sw_params_set_avail_min()! I must be missing some key thing in all this. Here are some of the readings. This is with snd_pcm_sw_params_set_avail_min(period_size) and period_size = 470 frames. avail = 68 avail = 202 avail = 336 avail = 0 avail = 134 >What period and buffer sizes do you use? I am using set_buffer/period_time_near and read back the actual values. I get the following with the settings I'm using. buffer_size : 7526 frames period_size : 470 frames I'm set to 22050 sampling rate, 16-bit signed interleaved samples. > FYI, you should seriously consider using poll() rather than the > async/callback interface. It's more efficient than signals and more > compatible with various IO plugins like pulseaudio. Yes - I'm looking into this for sure. This async mode has caused me lots of problems. My main issue is that I'm almost out of time to spend on this part of the project. :-( Thanks for your help.

1 0

Re: [alsa-devel] [Alsa-user] Async behavior and -EAGAIN on write?
by Paul Adolph 02 Jun '08

02 Jun '08

>> For playback I am using the ALSA callback to wake up a thread that >> then feeds the ALSA ring buffer with a period-worth of samples using >> snd_pcm_writei(). I've set snd_pcm_sw_params_set_avail_min() to one >> period. The problem is that most of the time when my thread wakes up >> and tries to write the data, snd_pcm_writei() fails with -EAGAIN, and >> I must retry until it works. >> >> Presumably this write failure is because there is not enough space in >> the ring buffer, but then why did the ALSA callback fire in the first >> place? As it stands my retries are eating needless amounts of CPU. > No definitive help. I don't know the internals, but I have a guess. It > probably leads the > amount so that on busy systems, underruns don't occur. You could try > setting it to 2 buffer > min and still writing only a single buffer. That way, you have a buffer of > slack. It seems to make no difference what snd_pcm_sw_set_avail_min() value I use. I tried setting it from twice to many times the period size. I still get ALSA callbacks very often. When I check available space at each callback invocation using snd_pcm_avail_update(), I almost always get much less than this setting. Indeed most usually much less than a single period size. > You might have better luck trying this on the alsa-devel list. The > developers live there and they > know the internals. It is possible they will answer. Good idea. I am cross-posting this email. (Hopefully to no-one's annoyance...) > I went and looked at the function you are using. > (snip) > Are you using a buffer size compatible with your card. It is possible it is > using the closest one and that is less than the buffer size you have > defined. Just a thought. To set buffer sizes, I am using set_buffer/period_time_near() and taking whatever the card gives me - I read them back with the get functions. One weird thing I noticed is that my card give me a period size of 470 frames *always*, regardless of whatever desired buffer and period time I ask for. I don't think this issue is related to the async behavior weirdness though. Thanks for your help.

1 0

[PATCH] Revert "wm9713: Expose the PLL configuration"
by Robert Jarzmik 01 Jun '08

01 Jun '08

This reverts commit 0d03f161680832984dd03f6bfc9a3423fc67501d. There's a bug in set_pll call from resume function. --- sound/soc/codecs/wm9713.c | 8 ++------ 1 files changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 5defac8..eae042c 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -659,7 +659,7 @@ struct pll_ { unsigned int k; }; -static struct pll_ pll[] = { +struct pll_ pll[] = { {13000000, 0, 1, 0, 0, 7, 0x23f488}, {2048000, 1, 0, 0, 0, 12, 0x0}, {4096000, 1, 0, 0, 0, 6, 0x0}, @@ -667,10 +667,9 @@ static struct pll_ pll[] = { /* liam - add more entries */ }; -static int wm9713_set_pll(struct snd_soc_dai *codec_dai, +static int wm9713_set_pll(struct snd_soc_codec *codec, int pll_id, unsigned int freq_in, unsigned int freq_out) { - struct snd_soc_codec *codec = codec_dai->codec; struct wm9713_data *wm9713 = codec->private_data; int i; u16 reg, reg2; @@ -1056,13 +1055,11 @@ static struct snd_soc_dai_caps wm9713_voice_playback = { static struct snd_soc_dai_ops wm9713_hifi_dai_ops = { /* alsa ops */ .prepare = ac97_hifi_prepare, - .set_pll = wm9713_set_pll, }; static struct snd_soc_dai_ops wm9713_aux_dai_ops = { /* alsa ops */ .prepare = ac97_aux_prepare, - .set_pll = wm9713_set_pll, }; static struct snd_soc_dai_ops wm9713_voice_dai_ops = { @@ -1074,7 +1071,6 @@ static struct snd_soc_dai_ops wm9713_voice_dai_ops = { .set_clkdiv = wm9713_set_dai_clkdiv, .set_fmt = wm9713_set_dai_fmt, .set_tristate = wm9713_set_dai_tristate, - .set_pll = wm9713_set_pll, }; /* for modprobe */ -- 1.5.5.3 --=-=-= Content-Type: application/octet-stream Content-Disposition: attachment; filename=sound_soc_pxa.tar.gz Content-Transfer-Encoding: base64 Content-Description: mioa701 board code H4sIALyrXkgAA+w8/XfaxrL9Ff8VG3riBw7GgO2ktROfSzB2aDBwALfNy8vRkUHYqoWkqw9/tM37 29/M7K60+gKcNr3vnltOG8PuzOzs7OzM7MxKvhPa871vvuqnAZ9Xh4f4t/nqsKH+lZ9vms1G69Vh 6/Cg1fqm0QSAl9+ww6/LFv+EfqB7jH3j/bIabl3/v+nHp/X3ndlX1IGnrf8hrv/LxuHf6/9XfOL1 dx/0r6QDT1r//X1Y/+b+q4O/1/+v+CTXf2k6+qtGU7tffv+quV+f/Tlj4AK/PDgoWP+D1v5+M1p/ /N7A5Yf1b/w5w6/+/Iev/97OFtth6XWHNmweO1eGF7AfdO/XpXmLLYyZtu+anjFnV49sqcOvwLGN CBFb+6a+ZOemN793nDkg7W1tfWvaMyucG+y1Zdrhw97SmYeWUb85KehxdU9f5nXPjTtzlototma5 zd4/c5vtwPC80A3yOu8d7/afoRHmj2O7YS6Wa+nBwvGWWh6PfJvNHC+33Z0t85phVxY07851NxfF tM3gTrfyuvTZ96+0mTM3iKjSrfvLPd2b3aAF2PWMaz+Jjd3XrulkWwlJD+e8L+4s1+t7NJC/J/Ti pqz0wjCth4ddmrTaHpEU2oi9W9/OjYVpG+x81Bu2WtpkMmpNPkw6/ffaxWmp0mqx36lLa/en2tlA a2nDy2k1wmp3YM7UP7rs90ulxsPhd1t7O0x0BzcG82eGrXum46OiSrwLpHh5Cv8Oz85KgJfoOZ9c aJ12vy9A3rUHp5PutNRcCdRtE1BrLaWzcbdb2l8F9rZ/2Z0Oh9N3pYME2GTaHXeH2hS+jbrt991x qXRYAIDsjN4NB91J6WUCpNMeTS/H3WhSpVf53WI6pe9yu2MOS99vbRkPsNlssBxBZGiWug9tGqpr xQ+8cBYw355roNraTPfmbAf/rR5L1DvHnKdxZze6fW1UkKpcwxSC98utRlrvekD3SWMlUedg4irY DlBb4C8CcyZno8mx2Zuk1iQh3Ru0kjC6oS819DirWSnGdNxNEX0jCF3tF312q5m+J7FSVortuPAl RuSTN5bOnfFFqDcuYQXO9bVlSES0p5r4vhPaoW8IQVILCnFpPhi+FrDftkqwrB7bgQZbXxrHWyWc CgOTdrz1ORbpzAG/w3II8L+eAZPXdMv6+AmWBYjClreMRcDejQjCw71e+q3cF20X1DbqsLeG4bKR pT9ewSzY5N4MZjflGmt8rmXBf3RQCBngEssHb4cPmwO/fXR138/hJB981LnYnPaFYzs9e2PaHd0N Qs9g7wx9TrrILsKHiDqJ1jOvbzKyHcvGDYWbgi+QLmMF8EXiLQBfK+AM+/kSLgBfK+MUfIGQmSJk 3zX0WwBVZTwRbRuKOAm+Vn+T4Gv0Nwm8VrxpzlfqbxJ4rWxT4GAummxizg0MVNPUSbRLoJiQKw6x qVAV2LU6q8DmipOxXNi10kzwmyPKArpS63iT2NUFsGulrsLmizxFN3c7MVXlb6LdsAyBrp+w2lHf MAzEbknurQKA5GYKU85A6k4B0fxuIglN+yo9Vkag/QTYb2wAYSj8Yp+f4M2u/SU4ecvSUB7g2BSn BoEh8RI4NF/oBJvSZyM9uJFcbOrjmuxzje3tAQQC7J4opsh8UCjliLUFqKUS4CZwDA+IjPrFfI4T fG5mQVnEKAfAITiGSuSJPI6Jxz4EsX3kEJmFoQVfG+zDUhNoA2lOYfeEmI+ZIgppTogXhbbAihFy VojwAO3SXoaBoWD/cc3S7WLNOvMcO1b9/+/6leSWdqXK7ka+JNYyJtUsgUeUTtsdUJM75JE1k+uq AOPCAlQzYxEUcOxHOGSWpnPR6zRzNBEWtM0mTujNjDItdaRHCI8a1Ou000pbZIxxggK3LbWPfYHS DgfDf6XSLjzDyFfbsQEHCbkSk9H7/go92FxtFbwnaAGXVaQHBa7mgElhSY1AttfNa/zF+l3K12/F kmZ53I94zEwPeflbfdepL+YuDAdOyHBK1T0RiEgNzlHMvHBO+BvsRLl/oWn6Y0r5R3QkK7koo3EL 4gs8x8IUAE/33M4Cz9KuHjEfUJz7qDGeOLCrWyBHBWwWWJphGUsNExvmHEYrLY0lOLvKNvwEHmrM N381nEUFflarx1sl+Fs3FzqE8m/YZHA6/lHrTPtat9+90Hpn7U5Xu+j93B0f0yj2zH1ExDpyV2O2 SozaOEUD4mw7YmdhwhdzXkG+d08498gMgH7e2qI8Ck/e8LB35aRXKtvHTyQNSiM5PkyncZyQTSxs ELJ1nC+3O90KDRYK0GORoVni5D4eHHxCgd7fmJbBKnxMGOlTXeRvqqjUUt5LLiMQ+MEBSiWSn+jI ogPkPkEie8B+rkpwQRANgpWrKzlWl1i20aqUzAWrIGXOJQ2ye3INuPgNliSCRmA0bOMf3rO5cRVe o3W7MkSibI69cG5xPRD0baUs1g5V3j9iUsRH7LnPnsN6P5//j12uCRw+ZqQ/csQ6Cb1u2OHS8PTA AIUMjOXHxieJp8gKQDmDaGdLpbUkSI4q9nE8eTfMn/xn+B/AX7yAH59RSWXSd/BW6w5OR1K+hj13 7eqe+MV3ZDWVlIwyphpCO9C6UsMT+V2+t7mZJeI0ILeerJwI/cAOlVVXyUVeYmX0Shipo8FCoOh3 z45g5CEFvA+CcMLih+grs8/HMTO0DZHV4NGlecHe+yiEI2w7WL1GLfOfDDPV1DHBNqm/WYu/qLD5 JQhlkKaKuwKR5/ElYvNpI2KpIjNmdmJF5Qtl1CKZZEoaBXJs5uLEVY48tNQEU5WPDEYzw1yqGLLB EmfqI7Cd0KJG6kMKjZY61iOp/J9EMtxEi7twPFYx3zSOmcley10IP168qArDxpHNT/i7JDcWlg2B rH5lGZpr2sJ28k0EoLjVDcs3Mihz01+BkwD1H+1ZRVQjIs/XIM+m2IFrxQ5UCjySMN/ETKFf2ons FNoG+WP3hFs/LPNeGx63hdL2xTWb42IGTX+EB80LZ54qMUmH6tgEwJ0qyIBiM1aJ4KiWofurKoZH KyC4Sq2AiFRoFRW5SQGmFDPcJCsezV10qBLgMQhNSaOoAsWwxkjbxr2WkdK97kdiqrF707ISUou6 oUGVuLpIqGEqogqXGJNCu2RFTu1HRc33PiKCSI65RbvoWcTh9nbMPu6pnIqe0HpCVPGerUJ03Hi3 JKI+yVWioEXbTahaguGEumWsJq5/IfHcEwnZgitouD3eRI+LqadzTJsTloq7IWnMBGxGPNpbm5Dm 3j5BGGIfPbSCI6XpM9e9nPp0Sq2SVsb/OmawcKfCllDJA4+mW7mNIz2uu8lN9Iats6qo1bEdlQoa Gw4u2LVUYmvcTMvpSeVzkkDCJXE8486wA8FMGW+GsHfmmSljPjzqaZNhRzttjy5gB427bfzTHk+r K9zEU6rzfypfw1GGLTigXPr8psx3DAME3M5RqUNfupa5MGd64FCFQpkEwmnQr7nOveFlZoA3eWAK +IdbegLj/tb2zWvbAF9x43hI6FrqEIehuBdahd4hORgMTr6cWHw9p3N2zo+EAureg9NKLlgNR2G/ s8YDXhMnpE0GwPs/m4wwosQEH6LSfP26eUhnRTgAYWD01eazzf73q08IxkjMqFirI4XgqplSCNLd e3MOMRzb4X/FsbQwtcD1hqjlGihFxWDinObuCf0WUa83k5qV2A/dH7uDqTYcVDhtskRIIqXSQipN nLgIcFeANdTQFQf+nEruyTsqsZn0ZQavPFwscOvSsfIdd3nxb3F0TPSj34pa3oItDBwnuKEWTPAp x9dyVP/w6Wd0a0AZRr1JkG6LiWPWLS9diYuBGYNEMEQtcn4o+e7gEkxQb3De72rdn6eVZiMZPfnV WmFCNOF+IDyLcjqizc8MhCOUL8ATYcRXruWwxtMiVDtWzxW1hHsVPO1Fl13ZUgf3BxEa6rPQOV+x jBm207ovOVfaYu4TRnv0vpJJTmAGFHjKgUxmLVKbMYNy0etUopxEAdl+b9CtpBMfm0BiSiQfUB1W ZEaK4dT8iYASa2E8oF8yorWg26QQQbm2MQtMWIGNFsRzMDcf4Sr1oA7EcFRDEBc74NtbU/cFH4nU jria0Os0494IXlb5lRnLawMgsX6MoYpEXKcCgHEhQFTduaCLz5LP4WDYG8RIiaXjZEedt93uaCVI dhlxSkC6nL1hAcJFkYMrxxr9CCCxQM+5SWoXzmHUF0PkdI0j6guSrQxAsiMML6f7Yoi83ZEYqAAA SUTjJeKdxHBYsqjxYpm4nZbcY4IagonxCgFoekJ743r0D1ifWYR2Rmlzw0FxS1JE5Eg7tpY+Jk+1 ACOqe40DSgcs713iBUyZCE50aXoQeOYVboYd/CqBuL+6ChfofMUtG5aoCsjGcL/FgqUL7MgW+AUg eAFcQ+NK0XoFI4tmS3s3+qHdea/1BpMuhckSB8m+gABmZlNOfFGBoV9AY42N2uddcB3/3WW7jBrK PB3Onj2DgRQSwvsCDLrf0+6PvU5Xa0+n4wqXCaym1htfng9rqqSEecnc0lUeblh3D3hNeJIsktAV 4avQ1xwXFp1+wTeZqEPbHRwXRcky2lF5IBFThzgPCJ6pSTOj7ILA5bqjHL92u4MhyArJS2byRpB9 YhD3QedNN3BAAItqKiM9k7A8lhc1jvfd8UDrjsesfEnJRNxjzlWgmzZFnsxxseaAWwEWV4S0SQYp /qQ9e23icRkccHDDkUGedHzR57s3zozNPfMOzy+zmeHzK2BAS5kV5njiZxtkDCfZrjG1ytbgwR3N DKm8hp8xc7RaxNXcYTpYEAvvQqOxRnbwYQERpFjIjz3HJpsF3uNWie0Awr3uLSMEy4KQkp6S0W2Q BwpDt9I0hWbtxcu1e0K9fBq7J9iMDMfdOIiWC4OTUsnIUwOHEGcCiJdPh2Otd9qsov40CpaVFsIy 7VtmeJ7jpVaRqzWuYEK1cGclVyHKbIFMsdfULbBuUQTGp++a3GI+MT0HJNtzWAPDJx8zw8OtXCBO jyfJycBRmrw9Hrc/kPWppKPOKs+dkzBQjpVYxzDTos8T1VpxMqdPJAHQw8p2Jpo1P3EpCMtUrUqd y8pSTIhCXUoh8VnkPTKhyhVxpEAjOSjhbGq7UMCEe0ayWJGVZFwxmlkpL7IVXTkyVKGqkqsJDBq6 WcYoSGMu3lhJxHqFrILoeXwnGY3Q1GVQ2Ir6qzk7/dpBSxXKfT7y8FEzzBJ2GD3YJbhYefzeP9U6 w8F0POyrxY/s8TuGi3MWs0aDny4LayalZHCAfwSzp1R59oNwseBsZh9vyam7wGSPthLKlkrzp9yj 8bDWPa7lhZ6X4cnOb83F3FgwEMRZ71wbXaxwzH4IWmLPVz5uItcchLYEd6BfU6gEJI21ic+IHnrA uXc31wO9Ih5hWe3xv7qvLk5DxAvJ9vYSowtpae5sKXcGl0M6mZOSMriOML5Ik/9Mz3+YKLlIYkkK zaVEkbzokK+rxEgRDCfLBAxAm4utgmMBxmkSm4IwHrj9pqZRVqDo4cMGGDA/NPoL85rFX6OMBXo+ uoVSKr0B/48XubCFBIhNqSWJOzFgTPcjQzyOJDipLwCGsWakPpKMGwoiUSSaJUBX4ACkSQ0wW2Rq OyUB6KPjW2IuUTb9CyejRMVfOJscCsnpzHiGrnB6Uif4/D6vPN+ABb76I/tbJlx5MKxUjaL9MoNN Br6uLAaG09s2kt09UQy0uDl32j1rX/anEG/+3Kyl2ibTcTPa2TRM7sa+6F5IRnZPLMe+xoUFlso/ XeB8y8eyDz2gcklMOe5FIOjcsiDYGoFAKHwHs9PQpqGpgzlBXyRHP2XxlBg3GbwoklJttLLzEmGk EsAoIIUhDETlSDQeViQAxIAL0zIqyqKwbfiXsgPioLyCrv/oL/yC+cjTWqLUnUND6FHiXCWaj7B+ SryKx0w34VWydbQlJn6UOnpgEj1iKj/YQaXWcB+QEmRsNfLy9bziKnafLo7cwk2Gd35uTk1UtJLJ r5OpQEOTZ0JAPeucJ4SIJKe5lVzZYea3LtxilqLo4DTRLeaMyjsQhDOJIElLLhB2OQK36c69zUGn 73oT7WJ4etnv5ppJjUxB7DN9GTTTg924uDLnlJRgrPTbubKsZiJqTUuoGI1EYXU0UnqI0N5oEP5y DM51Zh4AIvpprMzoVZ6zFHlm08bxKUeDATyXm9a+nL4bjivl5Is/YIv/wr/941G/cZz6wqtiPkAg nXYnnXFvNO0NB5Vyuz/BS/4dxq0z3r9gbXQSMXy/1+kOJlhmGPXLgqtzkFfbB7S7FrIDLddmcLS3 54DW+PTIQP3esRa+Yy/NmeeAs17u0Ws3dlv1l7s4R4b/7N61dkFPkcS/+sUq/yaf5Pt/oiPcn/Xq H/qsfv9P4+XLVvz+n4MGvf9nv/Xy7/f//BUf/v4f1nHcR/4Yf2VWxaQlbH3Ht/Q79v7RNyydvXYN z3j4B9hxoz779YSQLm00Tb5uMXqdDj0ygTm3NuV4OnRY+6/vX4mXCbEz6Fo6ngGxSqCbls8sx7nF zHG7g2AM9rTr2IYdp/O4efLAP/r4pVVvESFgrwcDWjCE5zo80cwqN0HggsVAN2Vh/pkuGFloKap1 zgJ/gdH0xvQZ6Pi1py8ZfEVfzHxnEdzrnnHMHp0QQjocdQ4GWVRTwDrpsE1oAnCMexSvQgoYbB6D p4ZBAEufOQv6cT64ZOcGOCaQzSi8ssAz9MHH274BZoq52OLfUHaYU0KcM+RjIvgAYQFpmtkxM0zo 9xjKmsQgRxEka8zxOJmKHiD/nsg3V4HpRwauJkauF4ohnu0clpPo34AY4QsQhZniHUZ85gEUYBFa NU4EwNlPPfAZl1PWHnxgP2EYO5h+OKZ0vhPyOx9EzMT7R0AbZufpdvAIk+A0LrrjzjtAar/t9XvT DzAZdtabDrqTCTsbjlmbjdrjaa9z2W+P2ehyPBpOunU4Xhj0+h5OYoW4FymliwXwARbaBx6tObvR 7/BhjpkB+jynzL/7uH4pORkdjyS8ehEoIj1mEBrbTlBjlBhENc8sMicQr3QN1HpWr7HD79nUoLL4 yII9BeemEEns7zdq7K2DL9yqsYs2Y41Ws9ncbe43XrHLSZtPLPGyLW7ceQSR98oq2OBXuS/hAi14 SL0lavUbrHT6Yyye+Bqr6HVU2fdYLTZ5qdQWN1/osWIbJKWgJh7xTsJp9+3leV5yBAfTwO54j2zn 2nKu+HcITNKFO5nkpJTwymty2ZKfGmHmpZYRJH7+KxqKp5WfOFamDZ8RUq8fJpLVhMCfIoLh6fVY JmyGQKnNkQrjVqKnn65N0Pi4xJ3zwiR8i1iWdVXO9KeWWYWrcLEweK16Ie4Yrr4rRnSSp3b5iBro svHxJT6hFldgeWaDz/c4enDtWTQ+Ht0Qr8IZqBGR6PEx/IElHFkl8n3wFLwZK9kP7Dk+/rhNA2yj RNmzN6xF1R6stsDeMuIKd1QsF4+NjX943zoiUaN4UZZUNQCqR/APFcmVlSoliqdLiGLQOJO1pefA +MH8Gg/mD9818BZ6hd8BFHU/bAFK7DV8f1jAh9ekkmqXVRB6+iuVyyxaf9RuDV8RuFp9V6uAeAAs vDLnD9FTBfEKiiIfr9g0jlk8Zf79Bcqf1gsn+ya1hRNbj9+kNOMbCzR6+bl59LzRegDc540Dvgyc m6xcNhHImv0QpYS+zsaIr1esWi4hFimCRp5d2uSVaikmM9OVG5UeJd1vfYqehOL3VVzPWJgP6ysK cgNleWN4hd+Y8+r1Vy89cH4xM4lcYF7SF8rEH+4sP/e/fYGiBhXisJJ4hUvlTSwmkccjDqkz8bAw pxe34DLjMjpOUCWbQ0yS2kcUMXEJXj2gex9C26T+5ysDcFfkREsloV54v5/9/gYv4/x0ORkfxz2z Oo5W567jTaF/OJbWRFhUyW6UDyFyvGrOH9ONQDBwF8LhQDh6SYqS36qiR1fhv9ity61RpNPxu/4o UZOrW//X3rX+NBEE8c/yV5wmNFytQGvwg7UmqJUYeUWRaNQQQtGQUmp4iTH8785rX3d7LyA10ZkP 0HZfs3M7u7O7M78bkcdcMLaQeddIOhfn08vRzzDF/F7P+twijv/LvgYYZX0XBwHl+//uSm/5cQb/ d6W78lj3/7OgZvi/+CPtfg7AluK979r2+mUvjvJ7OQEViRv/d4+EO3NA26WlSkjbKPJuTcDbjVWM 4tvbWP2I8BNJL0jovVx992phksLSIYsjRlkYF6HzpJ1OUl4abLG3UEsPyy6MoVhBqWQcWgxz0Xy/ Mwb1AfTyXFxhbFza+a8f6JuB/8w638Z16+xz2DMT1nwCJlUkqeSa8rrvZLK7td6jZbYLBuPoqstr ZI++gAH4Ze43XvACmzDTwq90pUD35YkUouwYqb2IjFAK14MV4O9f5u4tYmfyoCA7n7aHe282d4Zr w3eY79qxtfFhZ/h3+XqxtbU+XN1kvkJ5mYtYkZ5Ashhf5+U00w2XnXuVzx/ZWgdhQP8VjIt/Qc+K g/lr9Hyf7sBIJ9qT/atOXJwUVlUsJLA76IrHlxN+9wVVVJD62hbIFXFOJ/uGBAGsmbQIDIugsHD+ tgNXkc94Sf9tuiAxYcQRWW3YUQztxlOfMDoUEsJgNS5sjdwCWVMzlQG14sFZJMD2BTHoCTkzX/In YNybXP1gWir/6LmoJ+fmr32baLTfpJrJVJIz0jg6ERSfeDJJcVl2FX57sPEvmR3woZZU2K1foUyD FRX2Vp6UxNSKfBGI5xZPsCIuuhH0Uf9mA6DUW8BbxqmS1EUNwDzRSUj1p8ejDNrHs3CcOK9lliy6 l/MIgMpG3rCi5fXoa1/2Wff9OUkcrE1OWqTRb7lFPLRIczOHWA48KYOY5Pyb/czM1sXBYiz0G1iw fAVDbFC61FIjXDH+beOoSpawiquHXXNCZvZvbkwhDtPn+dFXFPBgfgRfj2H2nl6cDyQSJQPJxIPM PZLyMHYUG3CT9vkUrioz814RjSu8IzbT/6UPEYU4OfzZTCEwxHQ2CsEtVT3xmw50rh3/ts0Qh8FO E2lTZcyrBQytjFqwVkBztbWiULXpobFCkEYUZuT+hXNLAEgW7U5Oc6B7Lw7pDvWpPEJ0prNH5NNv yRk6EKLgOxTFQ7d/9BuvxftndFWHkwrtjakO2uGyv3wA7xFXIWu0RfUD+eGzbDsQiMVWKxmnNHLN T2xWBtqR2aGZI0QqOM65L7qqZFaJAS3WeReHMU8nxj6FAVDWx6L5hWK+aY45sWd0hdOQO8Yj+IDT UxzsVBatTz5dWMhbs35baSdZe729h5FNw3Ub1EZ1eKe4xsoVU5mSC0zloHKskDKDatTZakhe8WCd yCsuTA1ocw58C9alfT90NiKuYi4Fg1YG3hoBKXZGDENIRKJBwJL0lEu46Ek5lT49LY6BSngWdAez Y3JqZMmiEocP0FNTXmBYU9E97L2JDcpFzYYja3I05c+XU/tij2AjKxinGSw93t067ASHp7o7Pb6Y EBZtR4rF8Y1gtuN19QHDSefLo/PjbdoxAHLSTFkdXa8tA6/asEcM2ydtFdfhWooIuBjTrz5vEXC/ BkzVFHUcDbCRoCPdz4EFVmY0j7gqo5NkPpKjQB/QXcMqRHBWU6ARfIAT6717u8XNVMKVJ+ndpqFQ J6LPygEi+63lBk+9TvlaERmBsbaa6UV99goVoyZfNSVephq15V1LOSpzWu2oylmmHjn86EtE/8UP qCGQvfkbyNCK4YpAvRK/jOB1YbSunTJawRLFkGbSur+wh6WtUFuhQlPxjG1b9z1oEqXph3Vvbu28 eTm0PvSZ0vOjlF0bbIQ1xx5noexIomE3LT5p2o8UwK7kuuYXmfmNp5KSkpKSkpKSkpKSkpKSkpKS kpKSkpKSkpKSkpKSkpKSkpKS0r9BfwBAnKAHAKAAAA== --=-=-=--

1 0

[alsa-devel] 2.6.26-rc1 regression: ISA DMA broken (bisected)
by Rene Herman 31 May '08

31 May '08

Good day. commit 8779f2fc3b84ebb6c5181fb13d702e9944c16069 "x86: don't try to allocate from DMA zone at first" breaks all of ISA DMA. Or all of ALSA ISA DMA at least. All ISA soundcards are silent following that commit -- no error messages, everything appears fine, just silence. It won't just revert due to 32/64 merge. Rene.

7 32

[alsa-devel] FW: [Alsa-user] raw spdif frame capture
by Brett Andrews 31 May '08

31 May '08

Hi: Anyone have any ideas about the following? As a side note, when I saw raw I don't mean 61937 raw (just preambles and payload), I mean 60958 raw (block sync, payload, VUCP). Thanks, Brett -----Original Message----- From: stan [mailto:ghjeold_i_mwee@cox.net] Sent: Friday, May 30, 2008 18:31 To: Brett Andrews Cc: alsa-user(a)lists.sourceforge.net Subject: Re: [Alsa-user] raw spdif frame capture Brett Andrews wrote: > Hi: > > I would like to perform raw spdif frame capture (all 64-bits, in 2 > 32-bit subframes) using an EMU 1212M under ALSA. Can this be done? > If so, has anyone else already done it? If not, does anyone have any > idea how to do it? EMU has claimed that it is possible with the > hardware, but that the driver would need to support it (windows driver > does not, they didn't know about the ALSA driver). Does the current > emu10k1 driver support this? How would I enter this mode? > > Thanks, > > Brett Andrews > Software Engineer > BSE Reference Software > I haven't got a lot of help for you as I'm not knowledgeable about this but I can give you some suggestions. You could ask you question on alsa-devel mailing list as they know the internals and might be able to answer your question directly. You could look at the alsa driver code. Go to ftp://ftp.alsa-project.org/pub/driver/ and download File:alsa-driver-1.0.16.tar.bz2. <ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.16.tar.bz2> This is the latest official alsa version. Unpack it and then look in "top level"/alsa-kernel/pci/emu10k1. That has the C source.

1 0

[PATCH] ISA: set 24-bit dma_mask for ISA devices.
by Rene Herman 31 May '08

31 May '08

Set the ISA device dma_mask in preparation for using the actual device with the DMA API. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- drivers/base/isa.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/drivers/base/isa.c b/drivers/base/isa.c index d222239..efd5775 100644 --- a/drivers/base/isa.c +++ b/drivers/base/isa.c @@ -7,6 +7,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <linux/init.h> +#include <linux/dma-mapping.h> #include <linux/isa.h> static struct device isa_bus = { @@ -141,6 +142,9 @@ int isa_register_driver(struct isa_driver *isa_driver, unsigned int ndev) isa_dev->dev.release = isa_dev_release; isa_dev->id = id; + isa_dev->dev.coherent_dma_mask = DMA_24BIT_MASK; + isa_dev->dev.dma_mask = &isa_dev->dev.coherent_dma_mask; + error = device_register(&isa_dev->dev); if (error) { put_device(&isa_dev->dev); -- 1.5.2.2 --------------050209050208090004020909--

1 0

[PATCH] ISA: set 24-bit dma_mask for ISA devices.
by Rene Herman 31 May '08

31 May '08

dma_alloc_coherent() on x86 currently takes a passed in NULL device pointer to mean that it should allocate an ISA compatible (24-bit) buffer which is a bit of a hack. The ALSA ISA drivers are the main consumers of this but have a struct device in fact readily available. For the legacy drivers, this sets the device dma_mask in preparation for using the actual device with the DMA API so as to eventually not need the NULL hack in dma_alloc_coherent(). This does not fix a current bug -- 2.6.26-rc1 stumbled over the NULL hack in dma_alloc_coherent() but this has already been fixed in commit 4a367f3a9dbf2e7ffcee4702203479809236ee6e by Takashi Iwai. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> Cc: Bjorn Helgaas <bjorn.helgaas(a)hp.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Alan Cox <alan(a)lxorguk.ukuu.org.uk> --- drivers/base/isa.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/drivers/base/isa.c b/drivers/base/isa.c index d222239..efd5775 100644 --- a/drivers/base/isa.c +++ b/drivers/base/isa.c @@ -7,6 +7,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <linux/init.h> +#include <linux/dma-mapping.h> #include <linux/isa.h> static struct device isa_bus = { @@ -141,6 +142,9 @@ int isa_register_driver(struct isa_driver *isa_driver, unsigned int ndev) isa_dev->dev.release = isa_dev_release; isa_dev->id = id; + isa_dev->dev.coherent_dma_mask = DMA_24BIT_MASK; + isa_dev->dev.dma_mask = &isa_dev->dev.coherent_dma_mask; + error = device_register(&isa_dev->dev); if (error) { put_device(&isa_dev->dev); -- 1.5.2.2

1 0

[PATCH] PNP: set the pnp_card dma_mask for use by ISAPnP cards.
by Rene Herman 31 May '08

31 May '08

This makes the pnp_card->dev available as a DMAable device same as the pnp_dev->dev. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- drivers/pnp/card.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/drivers/pnp/card.c b/drivers/pnp/card.c index a762a41..b00ef10 100644 --- a/drivers/pnp/card.c +++ b/drivers/pnp/card.c @@ -8,6 +8,7 @@ #include <linux/ctype.h> #include <linux/slab.h> #include <linux/pnp.h> +#include <linux/dma-mapping.h> #include "base.h" LIST_HEAD(pnp_cards); @@ -167,6 +168,9 @@ struct pnp_card *pnp_alloc_card(struct pnp_protocol *protocol, int id, char *pnp sprintf(card->dev.bus_id, "%02x:%02x", card->protocol->number, card->number); + card->dev.coherent_dma_mask = DMA_24BIT_MASK; + card->dev.dma_mask = &card->dev.coherent_dma_mask; + dev_id = pnp_add_card_id(card, pnpid); if (!dev_id) { kfree(card); -- 1.5.2.2 --------------030605020307000902060404--

1 0

[PATCH] PNP: set the pnp_card dma_mask for use by ISAPnP cards.
by Rene Herman 31 May '08

31 May '08

dma_alloc_coherent() on x86 currently takes a passed in NULL device pointer to mean that it should allocate an ISA compatible (24-bit) buffer which is a bit of a hack. The ALSA ISA drivers are the main consumers of this but have a struct device in fact readily available. For the PnP drivers, the specific pnp_dev->dev device pointer is not always available at the right time so for now we want to pass the pnp_card->dev instead which is always available. Set its dma_mask in preparation for doing so. This does not fix a current bug -- 2.6.26-rc1 stumbled over the NULL hack in dma_alloc_coherent() but this has already been fixed in commit 4a367f3a9dbf2e7ffcee4702203479809236ee6e by Takashi Iwai. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> Acked-by: Bjorn Helgaas <bjorn.helgaas(a)hp.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Alan Cox <alan(a)lxorguk.ukuu.org.uk> --- drivers/pnp/card.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/drivers/pnp/card.c b/drivers/pnp/card.c index a762a41..b00ef10 100644 --- a/drivers/pnp/card.c +++ b/drivers/pnp/card.c @@ -8,6 +8,7 @@ #include <linux/ctype.h> #include <linux/slab.h> #include <linux/pnp.h> +#include <linux/dma-mapping.h> #include "base.h" LIST_HEAD(pnp_cards); @@ -167,6 +168,9 @@ struct pnp_card *pnp_alloc_card(struct pnp_protocol *protocol, int id, char *pnp sprintf(card->dev.bus_id, "%02x:%02x", card->protocol->number, card->number); + card->dev.coherent_dma_mask = DMA_24BIT_MASK; + card->dev.dma_mask = &card->dev.coherent_dma_mask; + dev_id = pnp_add_card_id(card, pnpid); if (!dev_id) { kfree(card); -- 1.5.2.2 --------------050004000406080804070800--

1 0

Re: [alsa-devel] hdsp rpm support
by Martin Tufnel 30 May '08

30 May '08

Hi Flo That's great news! Unfortunately, I don't have the unit yet so I'm unable to test it for you. However, I'm sure Roman would be very happy to give it a go. Roman? Just to clarify, given the driver is finished but the user tools are still under development, this means that the unit will basically work but not have its full functionality - is this correct? Also, did you use the new patched version of the firmware that Karl worked on? And, when do you think the driver will be incorporated into the main Alsa code branch? Thanks for your work! Regards, Martin On Tue, May 27, 2008 at 1:30 AM, <faber(a)faberman.de> wrote: > Martin, > > > Nice to hear from you. I saw your other posts and had wondered if you > had > > chucked in the towel. I would really like to see the RPM supported as it > > suits my needs exactly. > > RPM support is finished, I just need more time for the new userspace tools > (new totalmix etc.). > > If you want to test the new driver, drop me a note. > > > Flo >

4 9

[alsa-devel] Digital Audio Multiplexer of the i.MX27 processor
by Juergen Beisert 30 May '08

30 May '08

Hi, does anybody understand the DAM unit in the i.MX27 processor? This routing device drivers me crazy, as I do not understand how to route clocks, syncs and the data signals of my internal i2s unit to the processor's pins. All I can see is, when I enable the internal SSI unit in i2s mode, the DMA unit transfers some data into it, but the SSI unit does not shift it out. So everything stops, the DMA timeouts and I cannot hear anything. I can see the external device generates clock and frame signals, but it seems they do not reach the internal SSI unit. Has anybody a working DAM configuration which uses the internal SSI#1 as an i2s slave, an external i2s device connected to the SSI#1 pin group of this processor which acts as a master (I did not find anything in the manual, how the relation between the internal so called port-1 to port-6 at the DAM unit and the 4 external SSI1...SSI4 pin groups are). Means the external i2s device generates the clock and frame signal and the internal SSI#1 unit in i2s mode should use it to send serial sound data to the external device. Juergen -- Dipl.-Ing. Juergen Beisert | http://www.pengutronix.de Pengutronix - Linux Solutions for Science and Industry Handelsregister: Amtsgericht Hildesheim, HRA 2686 Vertretung Sued/Muenchen, Germany Phone: +49-8766-939 228 | Fax: +49-5121-206917-9

3 4

[alsa-devel] ASOC, SPDIF and AC97
by Jon Smirl 29 May '08

29 May '08

I'm back to working on an ASOC driver for the STAC9766 used in the MPC5200 based Efika. I'm having trouble with SPDIF. I can make two devices like this in the codec file. But how do I get the SPDIF device hooked up to the right AC97 slot, 10/11? Are any of the ASOC codecs implementing SPDIF? I also see that the STAC9766 is missing an ac97_patch. From my reading of the datasheet it needs somethings fixed up. Does the ac97 support belong in the PCI directory? It took me a while to locate it while working on ASOC. With a model more like ASOC codecs, the AC97 patches could be pushed into the codec files. static struct snd_soc_dai stac9766_dai[] = { { .name = "STAC9766-Analog", .id = 0, /* stream cababilities */ .playback = { .stream_name = "STAC9766 Analog", .channels_min = 1, .channels_max = 6, .rates = STAC9766_RATES, .formats = SNDRV_PCM_FMTBIT_S16, }, .capture = { .stream_name = "STAC9766 Analog", .channels_min = 1, .channels_max = 2, .rates = STAC9766_RATES, .formats = SNDRV_PCM_FMTBIT_S16, }, /* alsa ops */ .prepare = ac97_analog_prepare, }, { .name = "stac9766-S/PDIF", .id = 1, /* stream cababilities */ .playback = { .stream_name = "STAC9766 Digital", .channels_min = 1, .channels_max = 2, .rates = STAC9766_RATES, .formats = SNDRV_PCM_FMTBIT_S16, }, /* alsa ops */ .prepare = ac97_digital_prepare, }}; -- Jon Smirl jonsmirl(a)gmail.com

2 5

[alsa-devel] my ALS drv gives DMA underrun
by Norbert van Bolhuis 29 May '08

29 May '08

probably this question is asked many times. Anyway, I hope the experts are willing to help me here. I simply don't know how to avoid the DMA overrun that I'm facing when playing a tone. Here's the thing. I wrote a (new) ALSA driver (for AD1938 CODEC). I test it with the ALSA-LIB pcm.c example program and it always results in a DMA underrun. I really wonder whether this is due to the driver (me doing something stupid) or a CPU shortage or something else beyond my control. (I cannot really imagine it's due to CPU shortage or anything). I put may printk's in the kernel (in my driver and PCM middle layer). Here's the full test output/log. root:/> /mnt/tprogs/pcm -c 2 -r 48000 -o S32_LE -v Playback device is hw:0,0 Stream parameters are 48000Hz, S32_LE, 2 channels Sine wave rate is 440.0000Hz Using transfer method: write setting start_threshold to (buffer_size=8192 / period_size=1024) * period_size=1024 = 8192 setting avail_min to 1024 Hardware PCM card 0 'ADI ad1938' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tick_time : 4000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1024 xfer_align : 1024 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 1073741824 root:/proc/asound/card0> root:/proc/asound/card0> snd_pcm_lib_write1: (still) to write 1024 frames snd_pcm_playback_avail(nr=0x0000): avail = (hwptr=0 + bz=8192) - apptr=0) = 8192 snd_pcm_lib_write1: actually wrote 0 frames, err=1024, offset=0, appl_ptr=0, appl_ofs=0 snd_pcm_playback_avail(nr=0x0100): avail = (hwptr=0 + bz=8192) - apptr=1024) = 7168 snd_pcm_lib_write1: (still) to write 1024 frames snd_pcm_playback_avail(nr=0x0000): avail = (hwptr=0 + bz=8192) - apptr=1024) = 7168 snd_pcm_lib_write1: actually wrote 0 frames, err=1024, offset=0, appl_ptr=1024, appl_ofs=1024 snd_pcm_playback_avail(nr=0x0100): avail = (hwptr=0 + bz=8192) - apptr=2048) = 6144 snd_pcm_lib_write1: (still) to write 1024 frames snd_pcm_playback_avail(nr=0x0000): avail = (hwptr=0 + bz=8192) - apptr=2048) = 6144 snd_pcm_lib_write1: actually wrote 0 frames, err=1024, offset=0, appl_ptr=2048, appl_ofs=2048 snd_pcm_playback_avail(nr=0x0100): avail = (hwptr=0 + bz=8192) - apptr=3072) = 5120 snd_pcm_lib_write1: (still) to write 1024 frames snd_pcm_playback_avail(nr=0x0000): avail = (hwptr=0 + bz=8192) - apptr=3072) = 5120 snd_pcm_lib_write1: actually wrote 0 frames, err=1024, offset=0, appl_ptr=3072, appl_ofs=3072 snd_pcm_playback_avail(nr=0x0100): avail = (hwptr=0 + bz=8192) - apptr=4096) = 4096 snd_pcm_lib_write1: (still) to write 1024 frames snd_pcm_playback_avail(nr=0x0000): avail = (hwptr=0 + bz=8192) - apptr=4096) = 4096 snd_pcm_lib_write1: actually wrote 0 frames, err=1024, offset=0, appl_ptr=4096, appl_ofs=4096 snd_pcm_playback_avail(nr=0x0100): avail = (hwptr=0 + bz=8192) - apptr=5120) = 3072 snd_pcm_lib_write1: (still) to write 1024 frames snd_pcm_playback_avail(nr=0x0000): avail = (hwptr=0 + bz=8192) - apptr=5120) = 3072 snd_pcm_lib_write1: actually wrote 0 frames, err=1024, offset=0, appl_ptr=5120, appl_ofs=5120 snd_pcm_playback_avail(nr=0x0100): avail = (hwptr=0 + bz=8192) - apptr=6144) = 2048 snd_pcm_lib_write1: (still) to write 1024 frames snd_pcm_playback_avail(nr=0x0000): avail = (hwptr=0 + bz=8192) - apptr=6144) = 2048 snd_pcm_lib_write1: actually wrote 0 frames, err=1024, offset=0, appl_ptr=6144, appl_ofs=6144 snd_pcm_playback_avail(nr=0x0100): avail = (hwptr=0 + bz=8192) - apptr=7168) = 1024 snd_pcm_lib_write1: (still) to write 1024 frames snd_pcm_playback_avail(nr=0x0000): avail = (hwptr=0 + bz=8192) - apptr=7168) = 1024 snd_pcm_lib_write1: actually wrote 0 frames, err=1024, offset=0, appl_ptr=7168, appl_ofs=7168 snd_pcm_playback_avail(nr=0x0100): avail = (hwptr=0 + bz=8192) - apptr=8192) = 0 snd_pcm_playback_avail(nr=0x0300): avail = (hwptr=0 + bz=8192) - apptr=8192) = 0 snd_pcm_lib_write1: (still) to write 1024 frames snd_ad1938_playback_pointer: bytes_per_frame=32, diff=24352, frames=761, dma_offset=0x0, dma_last_pos=0x0, dma_pos_base=0x0 snd_pcm_playback_avail(nr=0x0010): avail = (hwptr=761 + bz=8192) - apptr=8192) = 761 snd_pcm_playback_avail(nr=0x0000): avail = (hwptr=761 + bz=8192) - apptr=8192) = 761 snd_pcm_lib_write1: WAIT snd_ad1938_playback_pointer: bytes_per_frame=32, diff=45840, frames=1432, dma_offset=0x0, dma_last_pos=0x2f9, dma_pos_base=0x0 snd_pcm_update_hw_ptr_interrupt: pos=1432, new_hw_ptr=(hwpb=0 + pos)=1432, hwpi=1024, pz=1024, delta=-408, bz=8192 snd_pcm_playback_avail(nr=0x0010): avail = (hwptr=1432 + bz=8192) - apptr=8192) = 1432 snd_pcm_playback_avail(nr=0x0001): avail = (hwptr=1432 + bz=8192) - apptr=8192) = 1432 snd_pcm_lib_write1: WAKE-UP snd_pcm_lib_write1: actually wrote 0 frames, err=1024, offset=0, appl_ptr=8192, appl_ofs=0 snd_ad1938_playback_pointer: bytes_per_frame=32, diff=81920, frames=2560, dma_offset=0x0, dma_last_pos=0x598, dma_pos_base=0x0 snd_pcm_update_hw_ptr_interrupt: pos=2560, new_hw_ptr=(hwpb=0 + pos)=2560, hwpi=2048, pz=1024, delta=-512, bz=8192 snd_pcm_playback_avail(nr=0x0010): avail = (hwptr=2560 + bz=8192) - apptr=9216) = 1536 snd_ad1938_playback_pointer: bytes_per_frame=32, diff=114688, frames=3584, dma_offset=0x0, dma_last_pos=0xa00, dma_pos_base=0x0 snd_pcm_update_hw_ptr_interrupt: pos=3584, new_hw_ptr=(hwpb=0 + pos)=3584, hwpi=3072, pz=1024, delta=-512, bz=8192 snd_pcm_playback_avail(nr=0x0010): avail = (hwptr=3584 + bz=8192) - apptr=9216) = 2560 snd_pcm_lib_write1: (still) to write 1024 frames snd_ad1938_playback_pointer: bytes_per_frame=32, diff=150368, frames=4699, dma_offset=0x0, dma_last_pos=0xe00, dma_pos_base=0x0 snd_pcm_playback_avail(nr=0x0010): avail = (hwptr=4699 + bz=8192) - apptr=9216) = 3675 snd_pcm_playback_avail(nr=0x0000): avail = (hwptr=4699 + bz=8192) - apptr=9216) = 3675 snd_ad1938_playback_pointer: bytes_per_frame=32, diff=170664, frames=5333, dma_offset=0x0, dma_last_pos=0x125b, dma_pos_base=0x0 snd_pcm_update_hw_ptr_interrupt: pos=5333, new_hw_ptr=(hwpb=0 + pos)=5333, hwpi=4096, pz=1024, delta=-1237, bz=8192 snd_pcm_playback_avail(nr=0x0010): avail = (hwptr=5333 + bz=8192) - apptr=9216) = 4309 snd_pcm_lib_write1: actually wrote 0 frames, err=1024, offset=0, appl_ptr=9216, appl_ofs=1024 snd_ad1938_playback_pointer: bytes_per_frame=32, diff=199248, frames=6226, dma_offset=0x0, dma_last_pos=0x14d5, dma_pos_base=0x0 snd_pcm_update_hw_ptr_interrupt: pos=6226, new_hw_ptr=(hwpb=0 + pos)=6226, hwpi=6144, pz=1024, delta=-82, bz=8192 snd_pcm_playback_avail(nr=0x0010): avail = (hwptr=6226 + bz=8192) - apptr=9216) = 5202 snd_ad1938_playback_pointer: bytes_per_frame=32, diff=229376, frames=7168, dma_offset=0x0, dma_last_pos=0x1852, dma_pos_base=0x0 snd_pcm_update_hw_ptr_interrupt: pos=7168, new_hw_ptr=(hwpb=0 + pos)=7168, hwpi=7168, pz=1024, delta=0, bz=8192 snd_pcm_playback_avail(nr=0x0010): avail = (hwptr=7168 + bz=8192) - apptr=10240) = 5120 snd_ad1938_playback_pointer: bytes_per_frame=32, diff=262144, frames=8192, dma_offset=0x0, dma_last_pos=0x1c00, dma_pos_base=0x0 snd_pcm_update_hw_ptr_interrupt: pos=0, new_hw_ptr=(hwpb=0 + pos)=0, hwpi=8192, pz=1024, delta=8192, bz=8192 snd_pcm_playback_avail(nr=0x0010): avail = (hwptr=8192 + bz=8192) - apptr=10240) = 6144 snd_pcm_lib_write1: (still) to write 1024 frames snd_ad1938_playback_pointer: bytes_per_frame=32, diff=27384, frames=855, dma_offset=0x0, dma_last_pos=0x0, dma_pos_base=0x2000 snd_pcm_playback_avail(nr=0x0010): avail = (hwptr=9047 + bz=8192) - apptr=10240) = 6999 snd_pcm_playback_avail(nr=0x0000): avail = (hwptr=9047 + bz=8192) - apptr=10240) = 6999 snd_ad1938_playback_pointer: bytes_per_frame=32, diff=47744, frames=1492, dma_offset=0x0, dma_last_pos=0x357, dma_pos_base=0x2000 snd_pcm_update_hw_ptr_interrupt: pos=1492, new_hw_ptr=(hwpb=8192 + pos)=9684, hwpi=9216, pz=1024, delta=-468, bz=8192 snd_pcm_playback_avail(nr=0x0010): avail = (hwptr=9684 + bz=8192) - apptr=10240) = 7636 snd_pcm_lib_write1: actually wrote 0 frames, err=1024, offset=0, appl_ptr=10240, appl_ofs=2048 snd_ad1938_playback_pointer: bytes_per_frame=32, diff=81920, frames=2560, dma_offset=0x0, dma_last_pos=0x5d4, dma_pos_base=0x2000 snd_pcm_update_hw_ptr_interrupt: pos=2560, new_hw_ptr=(hwpb=8192 + pos)=10752, hwpi=10240, pz=1024, delta=-512, bz=8192 snd_pcm_playback_avail(nr=0x0010): avail = (hwptr=10752 + bz=8192) - apptr=11264) = 7680 snd_ad1938_playback_pointer: bytes_per_frame=32, diff=114688, frames=3584, dma_offset=0x0, dma_last_pos=0xa00, dma_pos_base=0x2000 snd_pcm_update_hw_ptr_interrupt: pos=3584, new_hw_ptr=(hwpb=8192 + pos)=11776, hwpi=11264, pz=1024, delta=-512, bz=8192 snd_pcm_playback_avail(nr=0x0010): avail = (hwptr=11776 + bz=8192) - apptr=11264) = 8704 snd_pcm_stop: stopping stream, state=4 snd_pcm_do_stop: SNDRV_PCM_TRIGGER_STOP snd_pcm_lib_write1: (still) to write 1024 frames snd_pcm_playback_avail(nr=0x0000) = snd_pcm_lib_write1 calling snd_pcm_playback_avail (1st time) snd_pcm_playback_avail(nr=0x0001) = snd_pcm_lib_write1 calling snd_pcm_playback_avail (2nd time) snd_pcm_playback_avail(nr=0x0010) = snd_pcm_update_hw_ptr_post calling snd_pcm_playback_avail snd_pcm_playback_avail(nr=0x0100) = snd_pcm_playback_hw_avail calling snd_pcm_playback_avail snd_pcm_playback_avail(nr=0x0300) = snd_pcm_playback_data calling snd_pcm_playback_avail I hope you're still with me. so the APP fills the DMA TX buf with 8192 frames. Then DMA starts. DMA stops because there's more available in the DMA buffer then stop_threshold. stop_threshold is max = 8192 frames. somehow DMA is running faster than the APP. Probably this is a classic example of a playback underrun. Most of the times the underrun happens sooner, the APP then not even gets a chance of writing more frames (than 8192). so I understand why DMA stops but I don't want it to stop. Is my driver doing something wrong ? What can I do to avoid having this underrun ? Btw. what exactly is the purpose of boundary ? thanks for any advice, N. van Bolhuis -------------- -- This message has been scanned for viruses and is believed to be clean

2 2

[alsa-devel] basic question regarding copy unctions
by Harsha priya gupta 29 May '08

29 May '08

Hi, I have implemented copy function to send the buffer directly to the hardware. But i dont see the player playing after it sends the first buffer down. It does not seem to be sending more buffer. Can anyone tell me where I am going wrong? -- -Harsha

2 1

[alsa-devel] [PATCH 1/2] ASoC: Add SOC_DOUBLE_S8_TLV control type
by Mark Brown 29 May '08

29 May '08

The SOC_DOUBLE_S8_TLV control type was originally implemented in the UDA1380 driver by Philipp Zabel and was moved into the core by me. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 15 ++++++++++ sound/soc/soc-core.c | 72 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 87 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index bca9538..9fa2093 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -73,6 +73,15 @@ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ .private_value = (reg_left) | ((shift) << 8) | \ ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } +#define SOC_DOUBLE_S8_TLV(xname, reg, min, max, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ + .put = snd_soc_put_volsw_s8, \ + .private_value = (reg) | (((signed char)max) << 16) | \ + (((signed char)min) << 24) } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .texts = xtexts } @@ -267,6 +276,12 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); /* SoC PCM stream information */ struct snd_soc_pcm_stream { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a3f091e..f594ab8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1589,6 +1589,78 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); +/** + * snd_soc_info_volsw_s8 - signed mixer info callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int max = (signed char)((kcontrol->private_value >> 16) & 0xff); + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = max-min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); + +/** + * snd_soc_get_volsw_s8 - signed mixer get callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to get the value of a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + int val = snd_soc_read(codec, reg); + + ucontrol->value.integer.value[0] = + ((signed char)(val & 0xff))-min; + ucontrol->value.integer.value[1] = + ((signed char)((val >> 8) & 0xff))-min; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); + +/** + * snd_soc_put_volsw_sgn - signed mixer put callback + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to set the value of a signed mixer control. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + unsigned short val; + + val = (ucontrol->value.integer.value[0]+min) & 0xff; + val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; + + return snd_soc_update_bits(codec, reg, 0xffff, val); +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); + static int __devinit snd_soc_init(void) { printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION); -- 1.5.5.1

2 2

Re: [alsa-devel] snd_hda_intel reports -ENOMEM on probe
by Takashi Iwai 29 May '08

29 May '08

At Wed, 28 May 2008 23:33:43 +0200, Pierre Ossman wrote: > > I missed one line that was part of the module's output: > > cannot find the slot for index 1 (range 0-1), error: -16 > > This comes before the other two. Then I guess it was just a coincidence that the kernel with CONFIG_SND_DEBUG=y worked. Take a look at /etc/modprobe.d/sound or whatever module config file containing the sound setup. You must have a line with options snd-XXX index=1 and this conflicts with the actual hardware. Take a look at /proc/asound/cards. The secondary slot must be already occupied by other drivers that are auto-loaded beforehand. Either add a proper index to that driver as well, or use slots option for snd module. The latter is a new option introduced on 2.6.25. See ALSA-Configuration.txt for details. Takashi

1 0

[alsa-devel] [PATCH] ASoC Platform Driver for AT32AP7000 (AVR32)
by Geoffrey Wossum 28 May '08

28 May '08

Add ASoC platform driver for AT32AP7000 (AVR32). This is basically a port of the at91 ASoC platform code to the AVR32. This is currently against the Linux 2.6.24.3.atmel.3 kernel, the latest kernel Atmel has released for the AVR32. playpaq_wm8510.c isn't necessarily intended for inclusion into the mainline kernel or linux-2.6-asoc repository. It's provided more as a reference machine driver for an AT32AP7000 system. Signed-off-by: Geoffrey Wossum <gwossum(a)acm.org> --- /dev/null 2008-05-27 03:29:50.500019981 -0500 +++ linux-2.6.24.3.atmel.3/sound/soc/at32/Makefile 2008-05-27 17:11:38.000000000 -0500 @@ -0,0 +1,15 @@ +# AT32 Platform Support +snd-soc-at32-objs := at32-pcm.o +snd-soc-at32-ssc-objs := at32-ssc.o + + +obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o +obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o + + + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + + +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o --- /dev/null 2008-05-27 03:29:50.500019981 -0500 +++ linux-2.6.24.3.atmel.3/sound/soc/at32/at32-pcm.h 2008-05-27 17:04:54.000000000 -0500 @@ -0,0 +1,79 @@ +/* sound/soc/at32/at32-pcm.h + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_PCM_H +#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__ + +#include <linux/atmel-ssc.h> + + +/* + * Registers and status bits that are required by the PCM driver + * TODO: Is ptcr really used? + */ +struct at32_pdc_regs { + u32 xpr; /* PDC RX/TX pointer */ + u32 xcr; /* PDC RX/TX counter */ + u32 xnpr; /* PDC next RX/TX pointer */ + u32 xncr; /* PDC next RX/TX counter */ + u32 ptcr; /* PDC transfer control */ +}; + + + +/* + * SSC mask info + */ +struct at32_ssc_mask { + u32 ssc_enable; /* SSC RX/TX enable */ + u32 ssc_disable; /* SSC RX/TX disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */ + u32 pdc_enable; /* PDC RX/TX enable */ + u32 pdc_disable; /* PDC RX/TX disable */ +}; + + + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct at32_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct at32_pdc_regs *pdc; /* PDC register info */ + struct at32_ssc_mask *mask; /* SSC mask info */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler) (u32, struct snd_pcm_substream *); +}; + + + +/* + * The AT32 ASoC platform driver + */ +extern struct snd_soc_platform at32_soc_platform; + + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* __SOUND_SOC_AT32_AT32_PCM_H */ --- /dev/null 2008-05-27 03:29:50.500019981 -0500 +++ linux-2.6.24.3.atmel.3/sound/soc/at32/at32-pcm.c 2008-05-27 16:58:44.000000000 -0500 @@ -0,0 +1,505 @@ +/* sound/soc/at32/at32-pcm.c + * ASoC PCM interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-pcm.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "at32-pcm.h" + + + +/*--------------------------------------------------------------------------*\ + * Hardware definition +\*--------------------------------------------------------------------------*/ +/* TODO: These values were taken from the AT91 platform driver, check + * them against real values for AT32 + */ +static const struct snd_pcm_hardware at32_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE), + + .formats = SNDRV_PCM_FMTBIT_S16, + .period_bytes_min = 32, + .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */ + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, +}; + + + +/*--------------------------------------------------------------------------*\ + * Data types +\*--------------------------------------------------------------------------*/ +struct at32_runtime_data { + struct at32_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of DMA buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ + size_t period_size; + + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ + + /* Save PDC registers (for power management) */ + u32 pdc_xpr_save; + u32 pdc_xcr_save; + u32 pdc_xnpr_save; + u32 pdc_xncr_save; +}; + + + +/*--------------------------------------------------------------------------*\ + * Helper functions +\*--------------------------------------------------------------------------*/ +static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *dmabuf = &substream->dma_buffer; + size_t size = at32_pcm_hardware.buffer_bytes_max; + + dmabuf->dev.type = SNDRV_DMA_TYPE_DEV; + dmabuf->dev.dev = pcm->card->dev; + dmabuf->private_data = NULL; + dmabuf->area = dma_alloc_coherent(pcm->card->dev, size, + &dmabuf->addr, GFP_KERNEL); + pr_debug("at32_pcm: preallocate_dma_buffer: " + "area=%p, addr=%p, size=%ld\n", + (void *)dmabuf->area, (void *)dmabuf->addr, size); + + if (!dmabuf->area) { + return -ENOMEM; + } + + dmabuf->bytes = size; + return 0; +} + + + +/*--------------------------------------------------------------------------*\ + * ISR +\*--------------------------------------------------------------------------*/ +static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + static int count = 0; + + count++; + if (ssc_sr & params->mask->ssc_endbuf) { + pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "underrun" : "overrun", params->name, ssc_sr, count); + + /* re-start the PDC */ + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) { + prtd->period_ptr = prtd->dma_buffer; + } + + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + } + + + if (ssc_sr & params->mask->ssc_endx) { + /* Load the PDC next pointer and counter registers */ + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) { + prtd->period_ptr = prtd->dma_buffer; + } + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + } + + + snd_pcm_period_elapsed(substream); +} + + + +/*--------------------------------------------------------------------------*\ + * PCM operations +\*--------------------------------------------------------------------------*/ +static int at32_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* this may get called several times by oss emulation + * with different params + */ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params->dma_intr_handler = at32_pcm_dma_irq; + + prtd->dma_buffer = runtime->dma_addr; + prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; + prtd->period_size = params_period_bytes(params); + + pr_debug("hw_params: DMA for %s initialized " + "(dma_bytes=%ld, period_size=%ld)\n", + prtd->params->name, runtime->dma_bytes, prtd->period_size); + + return 0; +} + + + +static int at32_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + if (params != NULL) { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + prtd->params->dma_intr_handler = NULL; + } + + return 0; +} + + + +static int at32_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + + return 0; +} + + +static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct at32_runtime_data *prtd = rtd->private_data; + struct at32_pcm_dma_params *params = prtd->params; + int ret = 0; + + pr_debug("at32_pcm_trigger: buffer_size = %ld, " + "dma_area = %p, dma_bytes = %ld\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->period_ptr = prtd->dma_buffer; + + ssc_writex(params->ssc->regs, params->pdc->xpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xcr, + prtd->period_size / params->pdc_xfer_size); + + prtd->period_ptr += prtd->period_size; + ssc_writex(params->ssc->regs, params->pdc->xnpr, + prtd->period_ptr); + ssc_writex(params->ssc->regs, params->pdc->xncr, + prtd->period_size / params->pdc_xfer_size); + + pr_debug("trigger: period_ptr=%lx, xpr=%x, " + "xcr=%d, xnpr=%x, xncr=%d\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); + + ssc_writex(params->ssc->regs, SSC_IER, + params->mask->ssc_endx | params->mask->ssc_endbuf); + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_enable); + + pr_debug("sr=%x, imr=%x\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); + break; /* SNDRV_PCM_TRIGGER_START */ + + + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_disable); + break; + + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, + params->mask->pdc_enable); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + + + +static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd = runtime->private_data; + struct at32_pcm_dma_params *params = prtd->params; + dma_addr_t ptr; + snd_pcm_uframes_t x; + + ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr); + x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); + + if (x == runtime->buffer_size) { + x = 0; + } + + return x; +} + + + +static int at32_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct at32_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + goto out; + } + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + runtime->private_data = prtd; + + +out: + return ret; +} + + + +static int at32_pcm_close(struct snd_pcm_substream *substream) +{ + struct at32_runtime_data *prtd = substream->runtime->private_data; + + kfree(prtd); + return 0; +} + + +static int at32_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + + + +static struct snd_pcm_ops at32_pcm_ops = { + .open = at32_pcm_open, + .close = at32_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = at32_pcm_hw_params, + .hw_free = at32_pcm_hw_free, + .prepare = at32_pcm_prepare, + .trigger = at32_pcm_trigger, + .pointer = at32_pcm_pointer, + .mmap = at32_pcm_mmap, +}; + + + +/*--------------------------------------------------------------------------*\ + * ASoC platform driver +\*--------------------------------------------------------------------------*/ +static u64 at32_pcm_dmamask = 0xffffffff; + +static int at32_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) { + card->dev->dma_mask = &at32_pcm_dmamask; + } + if (!card->dev->coherent_dma_mask) { + card->dev->coherent_dma_mask = 0xffffffff; + } + + if (dai->playback.channels_min) { + ret = at32_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) { + goto out; + } + } + + if (dai->capture.channels_min) { + pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n"); + ret = at32_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) { + goto out; + } + } + + +out: + return ret; +} + + + +static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (substream == NULL) { + continue; + } + + buf = &substream->dma_buffer; + if (!buf->area) { + continue; + } + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + + + +#ifdef CONFIG_PM +static int at32_pcm_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) { + return 0; + } + prtd = runtime->private_data; + params = prtd->params; + + /* Disable the PDC and save the PDC registers */ + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + + return 0; +} + + + +static int at32_pcm_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct snd_pcm_runtime *runtime = dai->runtime; + struct at32_runtime_data *prtd; + struct at32_pcm_dma_params *params; + + if (runtime == NULL) { + return 0; + } + prtd = runtime->private_data; + params = prtd->params; + + /* Restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + + ssc_writex(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + return 0; +} +#else /* CONFIG_PM */ +# define at32_pcm_suspend NULL +# define at32_pcm_resume NULL +#endif /* CONFIG_PM */ + + + +struct snd_soc_platform at32_soc_platform = { + .name = "at32-audio", + .pcm_ops = &at32_pcm_ops, + .pcm_new = at32_pcm_new, + .pcm_free = at32_pcm_free_dma_buffers, + .suspend = at32_pcm_suspend, + .resume = at32_pcm_resume, +}; +EXPORT_SYMBOL_GPL(at32_soc_platform); + + + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("Atmel AT32 PCM module"); +MODULE_LICENSE("GPL"); --- /dev/null 2008-05-27 03:29:50.500019981 -0500 +++ linux-2.6.24.3.atmel.3/sound/soc/at32/at32-ssc.c 2008-05-27 17:03:30.000000000 -0500 @@ -0,0 +1,855 @@ +/* sound/soc/at32/at32-ssc.c + * ASoC platform driver for AT32 using SSC as DAI + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Note that this is basically a port of the sound/soc/at91-ssc.c to + * the AVR32 kernel. Thanks to Frank Mandarino for that code. + */ + +/* #define DEBUG */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <asm/io.h> + +#include "at32-pcm.h" +#include "at32-ssc.h" + + + +/*-------------------------------------------------------------------------*\ + * Constants +\*-------------------------------------------------------------------------*/ +#define NUM_SSC_DEVICES 3 + +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + + + +/*-------------------------------------------------------------------------*\ + * Module data +\*-------------------------------------------------------------------------*/ +/* + * SSC PDC registered required by the PCM DMA engine + */ +static struct at32_pdc_regs pdc_tx_reg = { + .xpr = SSC_PDC_TPR, + .xcr = SSC_PDC_TCR, + .xnpr = SSC_PDC_TNPR, + .xncr = SSC_PDC_TNCR, +}; + + + +static struct at32_pdc_regs pdc_rx_reg = { + .xpr = SSC_PDC_RPR, + .xcr = SSC_PDC_RCR, + .xnpr = SSC_PDC_RNPR, + .xncr = SSC_PDC_RNCR, +}; + + + +/* + * SSC and PDC status bits for transmit and receive + */ +static struct at32_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS), +}; + + + +static struct at32_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN), + .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS), +}; + + + +/* + * DMA parameters for each SSC + */ +static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + { + { + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, + { + { + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + }, + }, +}; + + + +static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +}; + + + + +/*-------------------------------------------------------------------------*\ + * ISR +\*-------------------------------------------------------------------------*/ +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt + * handler in the PCM driver. + */ +static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id) +{ + struct at32_ssc_info *ssc_p = dev_id; + struct at32_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) & + ssc_readl(ssc_p->ssc->regs, IMR)); + + /* + * Loop through substreams attached to this SSC. If a DMA-related + * interrupt occured on that substream, call the DMA interrupt + * handler function, if one has been registered in the dma_param + * structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + + return IRQ_HANDLED; +} + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int at32_ssc_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + + + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void at32_ssc_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + int dir_mask; + + dma_params = ssc_p->dma_params[substream->stream]; + + if (dma_params != NULL) { + ssc_writel(dma_params->ssc->regs, CR, + dma_params->mask->ssc_disable); + pr_debug("%s disabled SSC_SR=0x%08x\n", + (substream->stream ? "receiver" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[substream->stream] = NULL; + } + + + dir_mask = 1 << substream->stream; + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + /* Shutdown the SSC clock */ + pr_debug("at32-ssc: Stopping user %d clock\n", + ssc_p->ssc->user); + clk_disable(ssc_p->ssc->clk); + + if (ssc_p->initialized) { + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + /* clear the SSC dividers */ + ssc_p->cmr_div = 0; + ssc_p->tcmr_period = 0; + ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} + + + +/* + * Set the SSC system clock rate + */ +static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + /* TODO: What the heck do I do here? */ + return 0; +} + + + +/* + * Record DAI format for use by hw_params() + */ +static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, + unsigned int fmt) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + + + +/* + * Record SSC clock dividers for use in hw_params() + */ +static int at32_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, + int div_id, int div) +{ + struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case AT32_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value + */ + if (ssc_p->cmr_div == 0) { + ssc_p->cmr_div = div; + } else if (div != ssc_p->cmr_div) { + return -EBUSY; + } + break; + + case AT32_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case AT32_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + + + +/* + * Configure the SSC + */ +static int at32_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int id = rtd->dai->cpu_dai->id; + struct at32_ssc_info *ssc_p = &ssc_info[id]; + struct at32_pcm_dma_params *dma_params; + int channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + + /* + * Currently, there is only one set of dma_params for each direction. + * If more are added, this code will have to be changed to select + * the proper set + */ + dma_params = &ssc_dma_params[id][substream->stream]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[substream->stream] = dma_params; + + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the PCM driver's hw_params() + * function. It should not be used for other purposes as it + * is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + + /* + * Determine sample size in bits and the PDC increment + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + + case SNDRV_PCM_FORMAT_S16: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + + case SNDRV_PCM_FORMAT_S24: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + + case SNDRV_PCM_FORMAT_S32: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + + default: + pr_warning("at32-ssc: Unsupported PCM format %d", + params_format(params)); + return -EINVAL; + } + pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n", + bits, dma_params->pdc_xfer_size, channels); + + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) { + if (bits > 16) { + pr_warning("at32-ssc: " + "sample size %d is too large for I2S\n", + bits); + return -EINVAL; + } + } + + + /* + * Compute the SSC register settings + */ + switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_MASTER_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRS clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, SSC_START_FALLING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(RFMR_FSLEN, bits - 1) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, SSC_START_FALLING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) | + SSC_BF(TFMR_FSLEN, bits - 1) | + SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) | + SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clock. + * + * The SSC transmit clock is obtained from the BCLK signal + * on the TK line, and the SSC receive clock is generated from + * the transmit clock. + * + * For single channel data, one sample is transferred on the + * falling edge of the LRC clock. For two channel data, one + * sample is transferred on both edges of the LRC clock. + */ + pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n"); + start_event = ((channels == 1) ? + SSC_START_FALLING_RF : SSC_START_EDGE_RF); + + rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) | + SSC_BF(RCMR_START, start_event) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_CLOCK)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) | + SSC_BF(TCMR_START, start_event) | + SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | + SSC_BF(TCMR_CKO, SSC_CKO_NONE) | + SSC_BF(TCMR_CKS, SSC_CKS_PIN)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output on the SSC TK line + */ + pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n"); + rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | + SSC_BF(RCMR_STTDLY, 1) | + SSC_BF(RCMR_START, SSC_START_RISING_RF) | + SSC_BF(RCMR_CKI, SSC_CKI_RISING) | + SSC_BF(RCMR_CKO, SSC_CKO_NONE) | + SSC_BF(RCMR_CKS, SSC_CKS_DIV)); + + rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(RFMR_DATNB, channels - 1) | + SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); + + tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | + SSC_BF(TCMR_STTDLY, 1) | + SSC_BF(TCMR_START, SSC_START_RISING_RF) | + SSC_BF(TCMR_CKI, SSC_CKI_RISING) | + SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | + SSC_BF(TCMR_CKS, SSC_CKS_DIV)); + + tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | + SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) | + SSC_BF(TFMR_DATNB, channels - 1) | + SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); + break; + + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + pr_warning("at32-ssc: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + + if (!ssc_p->initialized) { + /* enable peripheral clock */ + pr_debug("at32-ssc: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + pr_warning("at32-ssc: request irq failed (%d)\n", ret); + pr_debug("at32-ssc: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* Set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("at32-ssc: SSC initialized\n"); + return 0; +} + + + +static int at32_ssc_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct at32_pcm_dma_params *dma_params; + + dma_params = ssc_p->dma_params[substream->stream]; + + ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable); + + return 0; +} + + + +#ifdef CONFIG_PM +static int at32_ssc_suspend(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + + if (!cpu_dai->active) { + return 0; + } + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int at32_ssc_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) { + return 0; + } + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define at32_ssc_suspend NULL +# define at32_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define AT32_SSC_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + + +#define AT32_SSC_FORMATS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \ + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) + + +struct snd_soc_cpu_dai at32_ssc_dai[NUM_SSC_DEVICES] = { + { + .name = "at32-ssc0", + .id = 0, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[0], + }, + { + .name = "at32-ssc1", + .id = 1, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[1], + }, + { + .name = "at32-ssc2", + .id = 2, + .type = SND_SOC_DAI_PCM, + .suspend = at32_ssc_suspend, + .resume = at32_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = AT32_SSC_RATES, + .formats = AT32_SSC_FORMATS, + }, + .ops = { + .startup = at32_ssc_startup, + .shutdown = at32_ssc_shutdown, + .prepare = at32_ssc_prepare, + .hw_params = at32_ssc_hw_params, + }, + .dai_ops = { + .set_sysclk = at32_ssc_set_dai_sysclk, + .set_fmt = at32_ssc_set_dai_fmt, + .set_clkdiv = at32_ssc_set_dai_clkdiv, + }, + .private_data = &ssc_info[2], + }, +}; +EXPORT_SYMBOL_GPL(at32_ssc_dai); + + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("AT32 SSC ASoC Interface"); +MODULE_LICENSE("GPL"); --- /dev/null 2008-05-27 03:29:50.500019981 -0500 +++ linux-2.6.24.3.atmel.3/sound/soc/at32/at32-ssc.h 2008-05-27 17:05:14.000000000 -0500 @@ -0,0 +1,59 @@ +/* sound/soc/at32/at32-ssc.h + * ASoC SSC interface for Atmel AT32 SoC + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SOC_AT32_AT32_SSC_H +#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__ + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "at32-pcm.h" + + + +struct at32_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + + +struct at32_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct at32_pcm_dma_params *dma_params[2]; + struct at32_ssc_state ssc_state; +}; + + +/* SSC divider ids */ +#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ + + +extern struct snd_soc_cpu_dai at32_ssc_dai[]; + + + +#endif /* __SOUND_SOC_AT32_AT32_SSC_H */ --- /dev/null 2008-05-27 03:29:50.500019981 -0500 +++ linux-2.6.24.3.atmel.3/sound/soc/at32/Kconfig 2008-05-27 15:44:04.000000000 -0500 @@ -0,0 +1,32 @@ +config SND_AT32_SOC + tristate "SoC Audio for the Atmel AT32 System-on-a-Chip" + depends on AVR32 && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the AT32 SSC interface. You will also need to + to select the audio interfaces to support below. + + +config SND_AT32_SOC_SSC + tristate + + + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_AT32_SOC + select SND_AT32_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + + + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq --- /dev/null 2008-05-27 03:29:50.500019981 -0500 +++ linux-2.6.24.3.atmel.3/sound/soc/at32/playpaq_wm8510.c 2008-05-27 17:06:51.000000000 -0500 @@ -0,0 +1,550 @@ +/* sound/soc/at32/playpaq_wm8510.c + * ASoC machine driver for PlayPaq using WM8510 codec + * + * Copyright (C) 2008 Long Range Systems + * Geoffrey Wossum <gwossum(a)acm.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c + */ + +//#define DEBUG + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/arch/at32ap700x.h> +#include <asm/arch/portmux.h> + +#include "../codecs/wm8510.h" +#include "at32-pcm.h" +#include "at32-ssc.h" + + + +/*-------------------------------------------------------------------------*\ + * constants +\*-------------------------------------------------------------------------*/ +#define MCLK_PIN GPIO_PIN_PA(30) +#define MCLK_PERIPH GPIO_PERIPH_A + + +/*-------------------------------------------------------------------------*\ + * data types +\*-------------------------------------------------------------------------*/ +/* SSC clocking data */ +struct ssc_clock_data { + /* CMR div */ + unsigned int cmr_div; + + /* Frame period (as needed by xCMR.PERIOD) */ + unsigned int period; + + /* The SSC clock rate these settings where calculated for */ + unsigned long ssc_rate; +}; + + +/*-------------------------------------------------------------------------*\ + * module data +\*-------------------------------------------------------------------------*/ +static struct clk *_gclk0; +static struct clk *_pll0; + +#define CODEC_CLK (_gclk0) + + + +/*-------------------------------------------------------------------------*\ + * Sound SOC operations +\*-------------------------------------------------------------------------*/ +static int playpaq_wm8510_startup( + struct snd_pcm_substream *substream) +{ + return 0; +} + + + +static void playpaq_wm8510_shutdown( + struct snd_pcm_substream *substream) +{ +} + + + +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE +static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( + struct snd_pcm_hw_params *params, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + struct ssc_clock_data cd; + unsigned int rate, width_bits, channels; + unsigned int bitrate, ssc_div; + unsigned actual_rate; + + + /* + * Figure out required bitrate + */ + rate = params_rate(params); + channels = params_channels(params); + width_bits = snd_pcm_format_physical_width(params_format(params)); + bitrate = rate * width_bits * channels; + + + /* + * Figure out required SSC divider and period for required bitrate + */ + cd.ssc_rate = clk_get_rate(ssc->clk); + ssc_div = cd.ssc_rate / bitrate; + cd.cmr_div = ssc_div / 2; + if (ssc_div & 1) { + /* round cmr_div up */ + cd.cmr_div++; + } + cd.period = width_bits - 1; + + + + /* + * Find actual rate, compare to requested rate + */ + actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); + pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n", + rate, actual_rate); + + + return cd; +} +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + +static int playpaq_wm8510_hw_params( + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct at32_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + unsigned int pll_out = 0, bclk = 0, mclk_div = 0; + int ret; + + /* Due to difficulties with getting the correct clocks from the AT32's + * PLL0, we're going to let the CODEC be in charge of all the clocks + */ +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +#else + struct ssc_clock_data cd; + const unsigned int fmt = (SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +#endif + + + + if (ssc == NULL) { + pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + + + /* + * Figure out PLL and BCLK dividers for WM8510 + */ + switch (params_rate(params)) { + case 48000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + + case 44100: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_1; + bclk = WM8510_BCLKDIV_8; + break; + + + case 22050: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_2; + bclk = WM8510_BCLKDIV_8; + break; + + case 16000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_3; + bclk = WM8510_BCLKDIV_8; + break; + + + case 11025: + pll_out = 11289600; + mclk_div = WM8510_MCLKDIV_4; + bclk = WM8510_BCLKDIV_8; + break; + + + case 8000: + pll_out = 12288000; + mclk_div = WM8510_MCLKDIV_6; + bclk = WM8510_BCLKDIV_8; + break; + + + + default: + pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + + + + /* + * set CPU and CODEC DAI configuration + */ + ret = codec_dai->dai_ops.set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI format (%d)\n", ret); + return ret; + } + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU DAI format (%d)\n", ret); + return ret; + } + + + + /* + * Set CPU clock configuration + */ +#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); + pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", + cd.cmr_div, cd.period); + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", ret); + return ret; + } + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + cd.period); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CPU transmit period (%d)\n", + ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + + /* + * Set CODEC clock configuration + */ + pr_debug("playpaq_wm8510: pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", + clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); + +#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", ret); + return ret; + } +#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ + + + ret = codec_dai->dai_ops.set_pll(codec_dai, 0, + clk_get_rate(CODEC_CLK), + pll_out); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", ret); + return ret; + } + + + ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); + if (ret < 0) { + pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", ret); + return ret; + } + + return 0; +} + + + +static struct snd_soc_ops playpaq_wm8510_ops = { + .startup = playpaq_wm8510_startup, + .hw_params = playpaq_wm8510_hw_params, + .shutdown = playpaq_wm8510_shutdown, +}; + + + +static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + + + +static const char *intercon[][3] = { + /* speaker connected to SPKOUT */ + { "Ext Spk", NULL, "SPKOUTP" }, + { "Ext Spk", NULL, "SPKOUTN" }, + + { "Mic Bias", NULL, "Int Mic" }, + { "MICN", NULL, "Mic Bias" }, + { "MICP", NULL, "Mic Bias" }, + + /* Terminator */ + { NULL, NULL, NULL }, +}; + + + +static int playpaq_wm8510_init( + struct snd_soc_codec *codec) +{ + int i; + + + /* + * Add DAPM widgets + */ + for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) { + snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]); + } + + + /* + * Setup audio path interconnects + */ + for (i = 0; intercon[i][0] != NULL; i++) { + snd_soc_dapm_connect_input(codec, + intercon[i][0], + intercon[i][1], + intercon[i][2]); + } + + + /* always connected endpoints */ + snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); + snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + + + snd_soc_dapm_sync_endpoints(codec); + + + /* Make CSB show PLL rate */ + codec->dai->dai_ops.set_clkdiv(codec->dai, WM8510_OPCLKDIV, + WM8510_OPCLKDIV_1 | 4); + + return 0; +} + + + +static struct snd_soc_dai_link playpaq_wm8510_dai = { + .name = "WM8510", + .stream_name = "WM8510 PCM", + .cpu_dai = &at32_ssc_dai[0], + .codec_dai = &wm8510_dai, + .init = playpaq_wm8510_init, + .ops = &playpaq_wm8510_ops, +}; + + + +static struct snd_soc_machine snd_soc_machine_playpaq = { + .name = "LRS_PlayPaq_WM8510", + .dai_link = &playpaq_wm8510_dai, + .num_links = 1, +}; + + + +static struct wm8510_setup_data playpaq_wm8510_setup = { + .i2c_address = 0x1a, +}; + + + +static struct snd_soc_device playpaq_wm8510_snd_devdata = { + .machine = &snd_soc_machine_playpaq, + .platform = &at32_soc_platform, + .codec_dev = &soc_codec_dev_wm8510, + .codec_data = &playpaq_wm8510_setup, +}; + + +static struct platform_device *playpaq_snd_device; + + +static int __init playpaq_asoc_init(void) +{ + int ret = 0; + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + + + /* + * Configure MCLK for WM8510 + */ + _gclk0 = clk_get(NULL, "gclk0"); + if (IS_ERR(_gclk0)) { + _gclk0 = NULL; + goto err_gclk0; + } + _pll0 = clk_get(NULL, "pll0"); + if (IS_ERR(_pll0)) { + _pll0 = NULL; + goto err_pll0; + } + if (clk_set_parent(_gclk0, _pll0)) { + pr_warning("snd-soc-playpaq: " + "Failed to set PLL0 as parent for DAC clock\n"); + goto err_set_clk; + } + clk_set_rate(CODEC_CLK, 12000000); + clk_enable(CODEC_CLK); + + at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); + + + + + + /* + * Create and register platform device + */ + playpaq_snd_device = platform_device_alloc("soc-audio", 0); + if (playpaq_snd_device == NULL) { + ret = -ENOMEM; + goto err_device_alloc; + } + + + + platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata); + playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev; + + + + ret = platform_device_add(playpaq_snd_device); + if (ret) { + pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", + ret); + goto err_device_add; + } + + return 0; + + +err_device_add: + if (playpaq_snd_device != NULL) { + platform_device_put(playpaq_snd_device); + playpaq_snd_device = NULL; + } +err_device_alloc: +err_set_clk: + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } +err_pll0: + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } +err_gclk0: + if (ssc != NULL) { + ssc_free(ssc); + ssc = NULL; + } +err_ssc: + return ret; +} + + + +static void __exit playpaq_asoc_exit(void) +{ + struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) { + ssc_free(ssc); + } + ssc_p->ssc = NULL; + } + + + if (_gclk0 != NULL) { + clk_put(_gclk0); + _gclk0 = NULL; + } + if (_pll0 != NULL) { + clk_put(_pll0); + _pll0 = NULL; + } + + at32_free_pin(MCLK_PIN); + + platform_device_unregister(playpaq_snd_device); + playpaq_snd_device = NULL; +} + + +module_init(playpaq_asoc_init); +module_exit(playpaq_asoc_exit); + + +MODULE_AUTHOR("Geoffrey Wossum <gwossum(a)acm.org>"); +MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); +MODULE_LICENSE("GPL");

2 3

[alsa-devel] power save of snd_hda_intel
by Harsha priya gupta 28 May '08

28 May '08

Hi, I am trying to get a power_save option of snd_hda_intel working. We are using latest ALSA build and ALC262 codec. when we start the player and close it, we dont see any power savings. [ We see the power saving in windows with the same ALC262 card] Any idea what i could be doing wrong here? -- -Harsha

3 3

Re: [alsa-devel] snd_hda_intel reports -ENOMEM on probe
by Takashi Iwai 28 May '08

28 May '08

At Tue, 27 May 2008 19:01:34 +0200, Pierre Ossman wrote: > > Since I upgraded my bios (in which the changelog mentioned nothing > about sound changes, damn MSI), my onboard audio no longer work. I get > the following in dmesg: > > hda-intel: Error creating card! > HDA Intel: probe of 0000:00:14.2 failed with error -12 > > -12 is -ENOMEM so I'm guessing some BIOS table is involved and tricks > the driver into allocating too much memory. Try to turn on CONFIG_SND_DEBUG=y. If you still don't get any relevant error messages, it's really an error from kmalloc(). Usually, it's a conflict of sound slots to be assigned, etc. Takashi

1 0

Re: [alsa-devel] problem with usb-audio and oss emulation
by kunal＠singhkunal.com 28 May '08

28 May '08

Hi, The second issue is resolved too. I was using /dev/dsp to access the Default sound card -> card0. (1) The /dev/dsp is mapped (through a symbolic link) to /dev/sound/dsp by default, when the system boots up. So, all works fine for me. (2) When I connect a usb-audio-device to my system, (a) the /dev/dsp1 gets mapped to /sound/dev/dsp1 (b) And /dev/dsp too gets mapped to /sound/dev/dsp1 So, I could not access /sound/dev/dsp. Is it not a bug? (3) When I dis-connect the usb-audio-device from my system both the "/dev/dsp" and "/dev/dsp1" entries dis-appear from my system. Is this not a bug too? Every thing will work fine for me as long as I use entries /dev/sound/dspN, insteading of /dev/dspN. I think this issue must be fixed. Shall I report a bug or something? In any case, I will spend some time in fixing this issue, when I get free from current assignment. Thanks and Regards, Kunal ***************** Hi All, The first issue was resolved by disabling the "CONFIG_SND_PCM_OSS_PLUGINS <http://lxr.linux.no/linux/+code=CONFIG_SND_PCM_OSS_PLUGINS>".i.e. disable the OSS plug-in support under KernelConfigMenu->sound->alsa CAN SOME ONE PLEASE HELP ON THIS? (2) Once I connect the iPod to my target, I can not any more use the default sound card. I read on the oss-emulation tutorial that this could be a problem with "dsp_map". But I am not able to check the current settings. Thanks and Regards, Kunal Date: Mon, 26 May 2008 20:10:11 +0530 From: Kunal Singh <kunal at singhkunal.com> Subject: [alsa-devel] problem with usb-audio and oss emulation To: alsa-devel at alsa-project.org Message-ID: <483ACBCB.2020800 at singhkunal.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, I am working on a ARM based target with 2.6.23 Kernel. My project involves streaming data from a USB-Audio Device (an iPod) and playing it back on the target. I am using OSS-emulation API to communicate with the audio devices. What I am able to do: (1) I have a sound card in my target. This is mapped to card0. I can open the "dsp" device and can playback any audio data on this. (2) I have enabled the usb-audio module support in kernel configuration. When I connect iPod to the target, I can see a new card "card1" being added in the /proc/asound/ directory. (3) I can open the "dsp1" device and can read from it. When I connect a "USB-Analyzer" I can see some valid data being streamed to the target. However, I am stuck with following issues: (1) The data being received on the target (in user mode) is junk. I put some printfs in the function "retire_capture_urb" in file "/linux/sound/usb/usbaudio.c", and I can see that the "data" which I see through usb-analzyer is reaching till this point. However, there seem to be some problem with one of the "snd_pmc_oss_ReadN" functions in file "http://lxr.linux.no/linux/sound/core/oss/pcm_oss.c". I am not able to exactly trace the problem, but I think that the "byte count" and "buffer address" being passed to snd-usb-audio is getting messed up some where. Has any one faced similar problems before? (2) Once I connect the iPod to my target, I can not any more use the default sound card. I read on the oss-emulation tutorial that this could be a problem with "dsp_map". But I am not able to check the current settings. Can some one give me some pointers for further debugging? Thanks and Regards, Kunal Singh www.singhkunal.com -------------------------------------------------------------------- myhosting.com - Premium Microsoft® Windows® and Linux web and application hosting - http://link.myhosting.com/myhosting

1 0

Re: [alsa-devel] bandwidth issue with usb-audio in alsa
by kunal＠singhkunal.com 28 May '08

28 May '08

Hi Clemens/Jon, I tried to understand the "lsusb -v" and started scanning through it. After a few scan I realized that Bus-001 where I have connected my device is a FS bus. The Bus-002 is HS. So, I moved my device to Bus-002. It all works without any Bandwidth issues. Thanks a lot for guiding me in to the right direction. Can some one also suggest, why is there a problem with oss-emulation on my system. I am using OSS-emulation API. As I mentioned earlier, I have a sound card "card0" on my system which is mapped to /dev/dsp and I am able to playback any data on this. As soon as I connect a USB-Audio device (which reflects as Card1 in /proc/asound and new nodes /dev/dsp1 /dev/mixer1 etc created), I can no longer use the Card0. The /dev/dsp tries to open a USB-Audio type devices (which should actually happen when I try to open the /dev/dsp1). I think there is some dynamic remapping which happens, and I am unable to trace it. Regards, Kunal .... Hi Clemens, I have copied the "lsusb -v" output below (end of this e-mail). The "HUB" and the "iPod" are connected to Bus-001. Bus-002 is not being used. As I can see from the Log, BUS-001 and BUS-002 both have usb (root ?) hubs with single TT. Each of these buses have only one port available. So, I guess that single "TT" should not be a problem. Am I right? The Belking hub which I have connected to "Bus-001" has "TT" per port. It also says "bInterval" equal to 255. The iPod device which I am using has two configurations. I am using the second configuration. Second configuration again has two interfaces. Second (HID Class) interface is being used and works fine. Problem is with the the first interface (with altsetting==1), which is an audio-streaming-class. The parameters for this interface are: (copied from the log below) bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Here "bInterval" is "1" and the transactions are not getting scheduled at this desired speed. Regards, Kunal Bus 002 Device 001: ID 0000:0000 Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 1 Single TT bMaxPacketSize0 64 idVendor 0x0000 idProduct 0x0000 bcdDevice 2.06 iManufacturer 3 Linux 2.6.22.6 ehci_hcd iProduct 2 Freescale On-Chip EHCI Host Controller iSerial 1 fsl-ehci.1 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 Hub Descriptor: bLength 9 bDescriptorType 41 nNbrPorts 1 wHubCharacteristic 0x0009 Per-port power switching Per-port overcurrent protection TT think time 8 FS bits bPwrOn2PwrGood 10 * 2 milli seconds bHubContrCurrent 0 milli Ampere DeviceRemovable 0x00 PortPwrCtrlMask 0x00 Hub Port Status: Port 1: 0000.0100 power Bus 001 Device 008: ID 05ac:1262 Apple Computer, Inc. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x05ac Apple Computer, Inc. idProduct 0x1262 bcdDevice 0.01 iManufacturer 1 Apple Inc. iProduct 2 iPod iSerial 3 000A27001A2E8B0C bNumConfigurations 2 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 32 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 8 Mass Storage bInterfaceSubClass 6 SCSI bInterfaceProtocol 80 Bulk (Zip) iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 149 bNumInterfaces 3 bConfigurationValue 2 iConfiguration 4 iPod USB Interface bmAttributes 0xc0 Self Powered MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 30 bInCollection 1 baInterfaceNr( 0) 1 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 2 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 1 bSourceID 1 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 35 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 9 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 11025 tSamFreq[ 2] 12000 tSamFreq[ 3] 16000 tSamFreq[ 4] 22050 tSamFreq[ 5] 24000 tSamFreq[ 6] 32000 tSamFreq[ 7] 44100 tSamFreq[ 8] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Devices bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.01 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 96 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 2 Bus 001 Device 006: ID 050d:0234 Belkin Components F5U234 USB 2.0 4-Port Hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x050d Belkin Components idProduct 0x0234 F5U234 USB 2.0 4-Port Hub bcdDevice 0.00 iManufacturer 0 iProduct 0 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 2mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0001 1x 1 bytes bInterval 255 Hub Descriptor: bLength 9 bDescriptorType 41 nNbrPorts 4 wHubCharacteristic 0x0000 Ganged power switching Ganged overcurrent protection bPwrOn2PwrGood 50 * 2 milli seconds bHubContrCurrent 1 milli Ampere DeviceRemovable 0x00 PortPwrCtrlMask 0x00 Hub Port Status: Port 1: 0000.0103 power enable connect Port 2: 0000.0100 power Port 3: 0000.0100 power Port 4: 0000.0100 power Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 2 TT per port bMaxPacketSize0 64 bNumConfigurations 1 Bus 001 Device 001: ID 0000:0000 Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 1 Single TT bMaxPacketSize0 64 idVendor 0x0000 idProduct 0x0000 bcdDevice 2.06 iManufacturer 3 Linux 2.6.22.6 ehci_hcd iProduct 2 Freescale On-Chip EHCI Host Controller iSerial 1 fsl-ehci.0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 Hub Descriptor: bLength 9 bDescriptorType 41 nNbrPorts 1 wHubCharacteristic 0x0009 Per-port power switching Per-port overcurrent protection TT think time 8 FS bits bPwrOn2PwrGood 10 * 2 milli seconds bHubContrCurrent 0 milli Ampere DeviceRemovable 0x00 PortPwrCtrlMask 0x00 Hub Port Status: Port 1: 0000.0103 power enable connect Clemens Ladisch wrote: > Kunal Singh wrote: > >> I have some update on the issue which I had reported earlier. In my last e-mail, I had not mentioned that iPod is connected to the target through a USB (HS) Hub. When I connect the >> same hub to my Desktop and read its configuration using "usbview" - I see that "interval" for this hub is mentioned >> as "256 msec". (I have two hubs and I get all the results same with both the hubs). >> > > Please show the output of "lsusb -v". > > > Regards, > Clemens -------------------------------------------------------------------- mail2web.com What can On Demand Business Solutions do for you? http://link.mail2web.com/Business/SharePoint

1 0

Re: [alsa-devel] bandwidth issue with usb-audio in alsa
by kunal＠singhkunal.com 28 May '08

28 May '08

Hi Clemens, I have copied the "lsusb -v" output below (end of this e-mail). The "HUB" and the "iPod" are connected to Bus-001. Bus-002 is not being used. As I can see from the Log, BUS-001 and BUS-002 both have usb (root ?) hubs with single TT. Each of these buses have only one port available. So, I guess that single "TT" should not be a problem. Am I right? The Belking hub which I have connected to "Bus-001" has "TT" per port. It also says "bInterval" equal to 255. The iPod device which I am using has two configurations. I am using the second configuration. Second configuration again has two interfaces. Second (HID Class) interface is being used and works fine. Problem is with the the first interface (with altsetting==1), which is an audio-streaming-class. The parameters for this interface are: (copied from the log below) bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Here "bInterval" is "1" and the transactions are not getting scheduled at this desired speed. Regards, Kunal Bus 002 Device 001: ID 0000:0000 Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 1 Single TT bMaxPacketSize0 64 idVendor 0x0000 idProduct 0x0000 bcdDevice 2.06 iManufacturer 3 Linux 2.6.22.6 ehci_hcd iProduct 2 Freescale On-Chip EHCI Host Controller iSerial 1 fsl-ehci.1 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 Hub Descriptor: bLength 9 bDescriptorType 41 nNbrPorts 1 wHubCharacteristic 0x0009 Per-port power switching Per-port overcurrent protection TT think time 8 FS bits bPwrOn2PwrGood 10 * 2 milli seconds bHubContrCurrent 0 milli Ampere DeviceRemovable 0x00 PortPwrCtrlMask 0x00 Hub Port Status: Port 1: 0000.0100 power Bus 001 Device 008: ID 05ac:1262 Apple Computer, Inc. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x05ac Apple Computer, Inc. idProduct 0x1262 bcdDevice 0.01 iManufacturer 1 Apple Inc. iProduct 2 iPod iSerial 3 000A27001A2E8B0C bNumConfigurations 2 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 32 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 8 Mass Storage bInterfaceSubClass 6 SCSI bInterfaceProtocol 80 Bulk (Zip) iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 149 bNumInterfaces 3 bConfigurationValue 2 iConfiguration 4 iPod USB Interface bmAttributes 0xc0 Self Powered MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 30 bInCollection 1 baInterfaceNr( 0) 1 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0201 Microphone bAssocTerminal 2 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 1 bSourceID 1 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 35 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 9 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 11025 tSamFreq[ 2] 12000 tSamFreq[ 3] 16000 tSamFreq[ 4] 22050 tSamFreq[ 5] 24000 tSamFreq[ 6] 32000 tSamFreq[ 7] 44100 tSamFreq[ 8] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Devices bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.01 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 96 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 2 Bus 001 Device 006: ID 050d:0234 Belkin Components F5U234 USB 2.0 4-Port Hub Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x050d Belkin Components idProduct 0x0234 F5U234 USB 2.0 4-Port Hub bcdDevice 0.00 iManufacturer 0 iProduct 0 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 2mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0001 1x 1 bytes bInterval 255 Hub Descriptor: bLength 9 bDescriptorType 41 nNbrPorts 4 wHubCharacteristic 0x0000 Ganged power switching Ganged overcurrent protection bPwrOn2PwrGood 50 * 2 milli seconds bHubContrCurrent 1 milli Ampere DeviceRemovable 0x00 PortPwrCtrlMask 0x00 Hub Port Status: Port 1: 0000.0103 power enable connect Port 2: 0000.0100 power Port 3: 0000.0100 power Port 4: 0000.0100 power Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 2 TT per port bMaxPacketSize0 64 bNumConfigurations 1 Bus 001 Device 001: ID 0000:0000 Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 9 Hub bDeviceSubClass 0 Unused bDeviceProtocol 1 Single TT bMaxPacketSize0 64 idVendor 0x0000 idProduct 0x0000 bcdDevice 2.06 iManufacturer 3 Linux 2.6.22.6 ehci_hcd iProduct 2 Freescale On-Chip EHCI Host Controller iSerial 1 fsl-ehci.0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 25 bNumInterfaces 1 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 9 Hub bInterfaceSubClass 0 Unused bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0004 1x 4 bytes bInterval 12 Hub Descriptor: bLength 9 bDescriptorType 41 nNbrPorts 1 wHubCharacteristic 0x0009 Per-port power switching Per-port overcurrent protection TT think time 8 FS bits bPwrOn2PwrGood 10 * 2 milli seconds bHubContrCurrent 0 milli Ampere DeviceRemovable 0x00 PortPwrCtrlMask 0x00 Hub Port Status: Port 1: 0000.0103 power enable connect Clemens Ladisch wrote: > Kunal Singh wrote: > >> I have some update on the issue which I had reported earlier. In my last e-mail, I had not mentioned that iPod is connected to the target through a USB (HS) Hub. When I connect the >> same hub to my Desktop and read its configuration using "usbview" - I see that "interval" for this hub is mentioned >> as "256 msec". (I have two hubs and I get all the results same with both the hubs). >> > > Please show the output of "lsusb -v". > > > Regards, > Clemens -------------------------------------------------------------------- mail2web.com - Microsoft® Exchange solutions from a leading provider - http://link.mail2web.com/Business/Exchange

1 0

[alsa-devel] AT32AP7000 (AVR32) + WM8510
by Geoffrey Wossum 28 May '08

28 May '08

Hi all, I'm attempting to bring the sound up on a board using an Atmel AT32AP7000 (AVR32) microcontroller with a Wolfson WM8510 mono CODEC. I'm using the Linux 2.6.24.3.atmel.3 kernel, the latest available for the AVR32. I've patched this kernel with the ASoC subsystem and WM8510 driver from git://opensource.wolfsonmicro.com/linux-2.6-asoc, as it appeared when v2.6.24 was tagged. I've also written the platform driver for the AT32, which is very close to the AT91 driver, and a machine driver for my specific hardware. I'm letting the WM8510 provide the I2S clocking from its internal PLL. I'm having issues getting the sound to work. I found what I believe are several bugs in the wm8510.c. I also had to lie about the WM8510 supporting 2 channels. Otherwise, the upper ALSA seem to include only 1 channel in the data stream, and the WM8510 seems to still expect 2 channels in the I2S stream. A patch is found a the bottom of this email. With this patch, the registers on the WM8510 seem to be getting configured properly (all the clock rates and such look correct), but I get no signal at the speaker outputs or the mono output, no matter what the mixer settings are. They just sit VMID after the initial power up of the chip. Is anyone else using the WM8510? Any hints or suggestions? TIA, --- Geoffrey Index: linux-2.6.24.3.atmel.3/sound/soc/codecs/wm8510.c =================================================================== --- linux-2.6.24.3.atmel.3.orig/sound/soc/codecs/wm8510.c +++ linux-2.6.24.3.atmel.3/sound/soc/codecs/wm8510.c @@ -186,7 +186,7 @@ SOC_SINGLE("Speaker Playback Volume", WM SOC_SINGLE("Speaker Boost", WM8510_OUTPUT, 2, 1, 0), SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0), -SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 0), +SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1), }; /* add non dapm controls */ @@ -209,14 +209,14 @@ static int wm8510_add_controls(struct sn static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0), SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0), -SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 1), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 0), }; /* Mono Output Mixer */ static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0), SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0), -SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 1), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0), }; /* AUX Input boost vol */ @@ -251,7 +251,7 @@ SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_ &wm8510_mono_mixer_controls[0], ARRAY_SIZE(wm8510_mono_mixer_controls)), SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0), -SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER3, 0, 0), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER2, 0, 0), SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), @@ -393,8 +393,8 @@ static int wm8510_set_dai_pll(struct snd wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18); - wm8510_write(codec, WM8510_PLLK1, (pll_div.k >> 9) && 0x1ff); - wm8510_write(codec, WM8510_PLLK1, pll_div.k && 0x1ff); + wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff); + wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff); reg = wm8510_read_reg_cache(codec, WM8510_POWER1); wm8510_write(codec, WM8510_POWER1, reg | 0x020); return 0; @@ -412,23 +412,23 @@ static int wm8510_set_dai_clkdiv(struct switch (div_id) { case WM8510_OPCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_GPIO & 0x1cf); + reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf; wm8510_write(codec, WM8510_GPIO, reg | div); break; case WM8510_MCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK & 0x1f); + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f; wm8510_write(codec, WM8510_CLOCK, reg | div); break; case WM8510_ADCCLK: - reg = wm8510_read_reg_cache(codec, WM8510_ADC & 0x1f7); + reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7; wm8510_write(codec, WM8510_ADC, reg | div); break; case WM8510_DACCLK: - reg = wm8510_read_reg_cache(codec, WM8510_DAC & 0x1f7); + reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7; wm8510_write(codec, WM8510_DAC, reg | div); break; case WM8510_BCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK & 0x1e3); + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3; wm8510_write(codec, WM8510_CLOCK, reg | div); break; default: @@ -597,14 +597,14 @@ struct snd_soc_codec_dai wm8510_dai = { .name = "WM8510 HiFi", .playback = { .stream_name = "Playback", - .channels_min = 1, - .channels_max = 1, + .channels_min = 2, + .channels_max = 2, .rates = WM8510_RATES, .formats = WM8510_FORMATS,}, .capture = { .stream_name = "Capture", - .channels_min = 1, - .channels_max = 1, + .channels_min = 2, + .channels_max = 2, .rates = WM8510_RATES, .formats = WM8510_FORMATS,}, .ops = {

2 9

Re: [alsa-devel] bandwidth issue with usb-audio in alsa
by Kunal Singh 27 May '08

27 May '08

Hi, I have some update on the issue which I had reported earlier. In my last e-mail, I had not mentioned that iPod is connected to the target through a USB (HS) Hub. When I connect the same hub to my Desktop and read its configuration using "usbview" - I see that "interval" for this hub is mentioned as "256 msec". (I have two hubs and I get all the results same with both the hubs). With 36 URBs, the gap between two set of transactions is appx 220 msec (220+36 => 256). So, I am wondering if the problem which I am seeing is because the external hub. I can not connect iPod directly to my target board, because this board does not support enough current (as needed by iPod). Can some one suggest, where is the problem - is it with the host controller driver? Regards, Kunal Singh ----------------------------- Kunal Singh wrote: Hi, My Platform ------------- I am trying to stream audio data from a USB-device (an iPod) to my arm-based embedded device (running kernel 2.6.23). The USB-device confirms to usb digital audio (subclass - streaming) class. The endpoint buffer size 196 bytes and transfer period is 1 msec. So, I should be reading data every msec. Problem ----------- I am facing some bandwidth issues. The data is not being captured fast enough. I notice that once one set of URBs are captured (one set equal to MAX_URBS), there is a significant delay in capturing the next set of URBs. I have set MAX_URBS equal to 32. When I connect a USB-Analyzer and analyze the timings, I can see that 32 URBs are captured at a interval of 1 msec each. After that there is a break of around 250 msec. And then again 32 URBs captured in sequence. If I change the MAX_URBs to any other value N, N URBs get captured at interval of 1 msec each, and then there is a break. Can some one suggest a way to improve this problem. Is there some configuration or parameter which I can change, so that URBs get submitted fast enough? May be as soon as the old one has retired. Thanks and Regards, Kunal www.singhkunal.com

3 2

Re: [alsa-devel] [Alsa-user] ICH8 HDA not detected after upgrade from Ubuntu 7.10 to 8.04
by Takashi Iwai 27 May '08

27 May '08

At Tue, 27 May 2008 15:17:40 +0200, Janto Schraiber wrote: > > Takashi Iwai wrote: > > You should have more messages than that. It's just the end result, > > not the cause... > > I'm not sure what's caused this, but I guess this is the whole part: > > [ 41.554112] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ > 21 > [ 41.554119] PCI: Unable to reserve mem region #1:4000@febf8000 for device > 0000:00:1b.0 So, this is no sound-driver but rather a deeper kernel problem. Please report this on kernel bugzilla, or let distro bug master to point to there. Takashi

1 0

Re: [alsa-devel] problem with usb-audio and oss emulation
by Kunal Singh 27 May '08

27 May '08

Hi All, The first issue was resolved by disabling the "CONFIG_SND_PCM_OSS_PLUGINS <http://lxr.linux.no/linux/+code=CONFIG_SND_PCM_OSS_PLUGINS>".i.e. disable the OSS plug-in support under KernelConfigMenu->sound->alsa CAN SOME ONE PLEASE HELP ON THIS? (2) Once I connect the iPod to my target, I can not any more use the default sound card. I read on the oss-emulation tutorial that this could be a problem with "dsp_map". But I am not able to check the current settings. Thanks and Regards, Kunal Date: Mon, 26 May 2008 20:10:11 +0530 From: Kunal Singh <kunal(a)singhkunal.com> Subject: [alsa-devel] problem with usb-audio and oss emulation To: alsa-devel(a)alsa-project.org Message-ID: <483ACBCB.2020800(a)singhkunal.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, I am working on a ARM based target with 2.6.23 Kernel. My project involves streaming data from a USB-Audio Device (an iPod) and playing it back on the target. I am using OSS-emulation API to communicate with the audio devices. What I am able to do: (1) I have a sound card in my target. This is mapped to card0. I can open the "dsp" device and can playback any audio data on this. (2) I have enabled the usb-audio module support in kernel configuration. When I connect iPod to the target, I can see a new card "card1" being added in the /proc/asound/ directory. (3) I can open the "dsp1" device and can read from it. When I connect a "USB-Analyzer" I can see some valid data being streamed to the target. However, I am stuck with following issues: (1) The data being received on the target (in user mode) is junk. I put some printfs in the function "retire_capture_urb" in file "/linux/sound/usb/usbaudio.c", and I can see that the "data" which I see through usb-analzyer is reaching till this point. However, there seem to be some problem with one of the "snd_pmc_oss_ReadN" functions in file "http://lxr.linux.no/linux/sound/core/oss/pcm_oss.c". I am not able to exactly trace the problem, but I think that the "byte count" and "buffer address" being passed to snd-usb-audio is getting messed up some where. Has any one faced similar problems before? (2) Once I connect the iPod to my target, I can not any more use the default sound card. I read on the oss-emulation tutorial that this could be a problem with "dsp_map". But I am not able to check the current settings. Can some one give me some pointers for further debugging? Thanks and Regards, Kunal Singh www.singhkunal.com

1 0

[alsa-devel] snd_kcontrol questions
by Harsha priya gupta 27 May '08

27 May '08

Hi, Can anyone help me to with the following questions on controls? 1. I want to define a snd_kcontrol that is of type mono. I am basically trying to define a mono output control that is exposed in KMIX and whose 'split channel' does not split into two channels. 2. My input controls give out 9 channels when i say 'split channels' in kmix. can anyone help me how to fix that to 2 channels 3. I would want to get the individual setting of each channel from KMIX into my ALSA driver. How do I get them? will .get and .set functions help in this? -- -Harsha

2 3

[alsa-devel] bandwidth issue with usb-audio in alsa
by kunal＠singhkunal.com 27 May '08

27 May '08

Hi, My Platform ------------- I am trying to stream audio data from a USB-device (an iPod) to my arm-based embedded device (running kernel 2.6.23). The USB-device confirms to usb digital audio (subclass - streaming) class. The endpoint buffer size 196 bytes and transfer period is 1 msec. So, I should be reading data every msec. Problem ----------- I am facing some bandwidth issues. The data is not being captured fast enough. I notice that once one set of URBs are captured (one set equal to MAX_URBS), there is a significant delay in capturing the next set of URBs. I have set MAX_URBS equal to 32. When I connect a USB-Analyzer and analyze the timings, I can see that 32 URBs are captured at a interval of 1 msec each. After that there is a break of around 250 msec. And then again 32 URBs captured in sequence. If I change the MAX_URBs to any other value N, N URBs get captured at interval of 1 msec each, and then there is a break. Can some one suggest a way to improve this problem. Is there some configuration or parameter which I can change, so that URBs get submitted fast enough? May be as soon as the old one has retired. Thanks and Regards, Kunal www.singhkunal.com -------------------------------------------------------------------- mail2web.com What can On Demand Business Solutions do for you? http://link.mail2web.com/Business/SharePoint

1 0

[alsa-devel] Recording problem with ALSA: trying to set up some constraints
by Sedji Gaouaou 26 May '08

26 May '08

Hi, I have a problem with my recording driver. I have set up a pcm_hardware structure in which I define the format supported by my hardware: static struct snd_pcm_hardware snd_at91_ac97_capture_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BLOCK_TRANSFER), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | SNDRV_PCM_FMTBIT_U20_3BE | SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_U18_3BE), .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000, .channels_min = 1, .channels_max = 2, .buffer_bytes_max = 256*1024, .period_bytes_min = 1024, .period_bytes_max = 4*1024, .periods_min = 1, .periods_max = 64, }; Then in my open function I pass it to the runtime struct. Nevertheless if I try to run arecord with this option: arecord -d 1 test.wav, I will have a kernel oops. The default format is U8, but it is not supported by my Hardware, abd it is not defined in my structure, so I though that it should exit arecord but not crash. Do I miss something when defining my pcm_hardware strucutre, or do I forgot to use a specific function which should return me an error? I have try to set up some constraints as well(for the rate and the format), but still I have the same crash. Regards, Sedji

3 4

[alsa-devel] Surround and SPDIF on STAC9277
by Jon Smirl 26 May '08

26 May '08

How are you supposed to get 5.1 audio out of your PC? When I hook the SPDIF device from my STAC9227/HDA audio up to pulse, pulse says that the SPDIF channel only supports stereo. So I go digging around the in STAC9227 datasheet. http://www.idt.com/?genID=STAC9227 STAC9227 codec only supports 192Khz SPDIF with 4/1*48Khz channels. It doesn't support 384Khz with 8/1*48Khz channels. So there are two problems, first the existing Intel HDA/STAC9227 driver is not exposing the 4 channel SPDIF support. Or I can't figure out how to get to it. Second, the hardware doesn't seem to support any way to get digital 5.1 audio out of the codec and into my Sony amp. STAC9227 does have six channel analog out, but my Sony amp doesn't have 6 channel analog in. Sony amp supports 384Khz 8-channel SPDIF in. -- Jon Smirl jonsmirl(a)gmail.com

2 4

[alsa-devel] Building alsa drivers as part of alsa vs kernel external module
by Ben Stanley 26 May '08

26 May '08

Hi, I'm trying to compile the AudioScience driver as an external kernel module (i.e. using the alsa in the kernel, rather than as a module within the alsa tree). What source code transformations can I expect to have to make? Is there some documentation anywhere on what transformations are performed as standard? Or is there a proper way that the module should be written so that it compiles both ways? Of course, I understand that patches will most likely be required for the module to compile against older kernels in particular. This kind of problem is usually dealt with by autoconf. Is there a standard way of doing this for just compiling a module directly for the kernel? I see that alsa has its own set of autoconf macros, and when I compile directly for the kernel (and its embedded alsa) I will lose the benefit of that. Ben Stanley.

2 1

[alsa-devel] problem with usb-audio and oss emulation
by Kunal Singh 26 May '08

26 May '08

Hi, I am working on a ARM based target with 2.6.23 Kernel. My project involves streaming data from a USB-Audio Device (an iPod) and playing it back on the target. I am using OSS-emulation API to communicate with the audio devices. What I am able to do: (1) I have a sound card in my target. This is mapped to card0. I can open the "dsp" device and can playback any audio data on this. (2) I have enabled the usb-audio module support in kernel configuration. When I connect iPod to the target, I can see a new card "card1" being added in the /proc/asound/ directory. (3) I can open the "dsp1" device and can read from it. When I connect a "USB-Analyzer" I can see some valid data being streamed to the target. However, I am stuck with following issues: (1) The data being received on the target (in user mode) is junk. I put some printfs in the function "retire_capture_urb" in file "/linux/sound/usb/usbaudio.c", and I can see that the "data" which I see through usb-analzyer is reaching till this point. However, there seem to be some problem with one of the "snd_pmc_oss_ReadN" functions in file "http://lxr.linux.no/linux/sound/core/oss/pcm_oss.c". I am not able to exactly trace the problem, but I think that the "byte count" and "buffer address" being passed to snd-usb-audio is getting messed up some where. Has any one faced similar problems before? (2) Once I connect the iPod to my target, I can not any more use the default sound card. I read on the oss-emulation tutorial that this could be a problem with "dsp_map". But I am not able to check the current settings. Can some one give me some pointers for further debugging? Thanks and Regards, Kunal Singh www.singhkunal.com

1 0

[alsa-devel] [PATCH] Added support to HDAIntel driver for the Foxconn P35AX-S mainboard
by Travis Place 26 May '08

26 May '08

Without forcing model type to 6stack-dig, the user gets no sound. Patch is attached, and alsa-info.sh output is at http://pastebin.ca/1023781. Changelog: Added IDs for the Foxconn P35AX-S mainboard to patch_realtek.c, so that ALC883_6ST_DIG is used by default. Signed-off-by: Travis Place <wishie at wishie dot net> Also, on a side note, are we to use the GIT repo from now on ? I just tried to pull from it, and got an error. HG seems to be working for me still though. Thanks, Travis Place. -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

2 4

[alsa-devel] hdsp rpm support
by Martin Tufnel 26 May '08

26 May '08

Hi All I was wondering if there is any more news in regard to the development effort toward supporting the RME HDSP RPM? The last couple of emails I could find on the development were from Karl Grill (please see below). Does anyone know if Karl (or anyone else) is still working on this? Thanks in advance for any help anyone can provide. Regards, Martin On Mon, 2008-02-25 at 08:45 +0100, Karl Grill wrote: >* Dear Roman and all, *>* I'm sorry i promised a little too much - in fact I was too busy with *>* other things to get back to the rpm stuff in time; last night, I started *>* to check the current state of affairs, and it looks quite promising; in *>* the next three weeks, I won't have too much time to devote to this *>* project, but then there's the University's easter break which I hope *>* will give me the time to get things done, so, if all goes well, at least *>* an experimental version should be ready by the end of March. * * *On Tue, 2008-02-26 at 21:58 +0100, Karl Grill wrote: > *Hi all, > as a first step towards implementing support for the hdsp rpm, I've > uploaded a patch to the alsa-firmware branch as bug report #3810. > Besides the patch, I've put up the firmware binaries in .tar.gz format. > This contains replacements for the rev.11 firmware for the > digiface/multiface and also for the rpm. The latter is of little use > right now, but I'd be very happy if anybody could try those for the > digiface/multiface and report if it works, so I know if my assumptions > on the drivers are correct. > > best regards > kg > > PS. sorry I didn't file this report in the firmware section, but the web > form there doesn't give me any usable choice for Category yet insists > that I supply one.*

2 2

[alsa-devel] [PATCH] Added another unknown card to hda-intel driver. ASUS P5K-E/WIFI-AP
by Travis Place 26 May '08

26 May '08

Changelog: Added a config table entry for the ASUS P5K-E/WIFI-AP mainboard (ID 1043:8227) to use AD1988_6STACK_DIG Signed-off-by: Travis Place <wishie at wishie dot net> -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

2 1

Re: [alsa-devel] [Alsa-user] ICH8 HDA not detected after upgrade from Ubuntu 7.10 to 8.04
by stan 26 May '08

26 May '08

Janto Schraiber wrote: > I got the ASUS P5B mainboard with an Intel HDA compatible Soundchip > (Analog Devices... AD 1988B afaik but I'm not realy sure about that). > Under Ubunt 7.10 it works out of the box but I just recently updated > to the new Ubuntu 8.04 where the card is no longer detected. > > There are some post on the web about the problem, but most deal with > issues, after you get details per 'head -n 1 > /proc/asound/card0/codec*' or the like, but sine the card is not > detected at all, i get no info from commands like that... see: > > $ cat /proc/asound/cards > --- no soundcards --- > > I installed the latest drivers from source, but it doesn't help. I > also tried to update the kernel, etc. from the 'hardy-proposed' > repository, still no luck. > > Here's the output from the 'alsa-info.sh' script: > http://pastebin.ca/1028944 > > And here's my dmesg: > http://pastebin.ca/1028967 > > What should I do to get my card working again? From your dmesg here's the problem. HDA Intel: probe of 0000:00:1b.0 failed with error -16 I'm not a developer so I don't know what that means, just that it is not good. I think this is an issue for some hda-intel cards in kernel version 2.6.25. In the cases I have seen the card was working on the 24 version kernel, but when they moved to 25 it stopped working. That is, it's a regression. I don't think there is anything you can do until the problem is fixed there. Using the earlier kernel is probably not possible with the rest of the system. I don't know this, but suspect it. You could try compiling an earlier kernel on your system. I'll cross post this to the alsa-devel mailing list so the developers become aware of the problem.

2 1

[alsa-devel] [PATCH] Added correct model selection for Intel DG35EC mainboard - HDAIntel driver.
by Travis Place 26 May '08

26 May '08

Changelog: Added the correct model for the Intel DG35EC mainboard (http://www.intel.com/products/motherboard/DG35EC/index.htm) The card was previously 'unknown'. The following patch, adds the card to the quirk table, to automatically select ALC883_3STACK_6CH. Confirmed working for all analogue outputs, and inputs. Patch attached. Signed-off-by: Travis Place <wishie at wishie dot net> -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

2 1

[alsa-devel] [patch] snd-pcsp: fix pcsp_treble_info() to honour the item number
by Stas Sergeev 26 May '08

26 May '08

Hello. The attached patch fixes the problem with mixers not properly displaying the PWM freq. The bug was that the pcsp_treble_info() didn't take the item number into an account. Takashi, could you please apply?

2 1

[alsa-devel] Kernel crash in snd-aloop (2)
by Benjamin van den Hout 26 May '08

26 May '08

Hello all, I did some more experimenting, this time with a stock Debian 2.6.18-6-686 SMP kernel and alsa-driver 10.0.16 source. This is on a 32 bit machine and the problem happens here as well. What I've noticed is that prior to the whole machine crashing (about 2 or 3 seconds), I get a number of "ALSA /home/benjamin/alsa-driver-1.0.16/acore/../alsa-kernel/core/pcm_lib.c:1471: BUG? (runtime != ((void *)0))?" lines on my console. After that, the kernel panics completely. This line is causes by the assertion 'snd_assert(runtime != NULL, return);' failing in the snd_pcm_period_elapsed function of pcm_lib.c Any ideas? The crash dump is a bit more informative so that's why I'm posting it here (the tainted indication comes from the vmware driver): ----Start 32bit dump--- ALSA /home/benjamin/alsa-driver-1.0.16/acore/../alsa-kernel/core/pcm_lib.c:1471: BUG? (runtime != ((void *)0)) [<c0125607>] run_timer_softirq+0xfb/0x151 [<c0121838>] __do_softirq+0x5a/0xbb [<c01218cf>] do_softirq+0x36/0x3a [<c0103747>] apic_timer_interrupt+0x1f/0x24 ALSA /home/benjamin/alsa-driver-1.0.16/acore/../alsa-kernel/core/pcm_lib.c:1471: BUG? (runtime != ((void *)0)) [<c0125607>] run_timer_softirq+0xfb/0x151 [<c0121838>] __do_softirq+0x5a/0xbb [<c01218cf>] do_softirq+0x36/0x3a [<c0103747>] apic_timer_interrupt+0x1f/0x24 [<c014c26c>] __handle_mm_fault+0x3d2/0x740 [<c01154b6>] do_page_fault+0x18a/0x481 [<c011532c>] do_page_fault+0x0/0x481 [<c01037f9>] error_code+0x39/0x40 ALSA /home/benjamin/alsa-driver-1.0.16/acore/../alsa-kernel/core/pcm_lib.c:1471: BUG? (runtime != ((void *)0)) [<c0125607>] run_timer_softirq+0xfb/0x151 [<c0121838>] __do_softirq+0x5a/0xbb [<c01218cf>] do_softirq+0x36/0x3a [<c0103747>] apic_timer_interrupt+0x1f/0x24 [<c0102bbe>] sysenter_past_esp+0x3/0x79 ALSA /home/benjamin/alsa-driver-1.0.16/acore/../alsa-kernel/core/pcm_lib.c:1471: BUG? (runtime != ((void *)0)) [<c0125607>] run_timer_softirq+0xfb/0x151 [<c0121838>] __do_softirq+0x5a/0xbb [<c01218cf>] do_softirq+0x36/0x3a [<c0103747>] apic_timer_interrupt+0x1f/0x24 [<c0101a5a>] default_idle+0x0/0x59 [<c0101a8b>] default_idle+0x31/0x59 [<c0101b52>] cpu_idle+0x9f/0xb9 ALSA /home/benjamin/alsa-driver-1.0.16/acore/../alsa-kernel/core/pcm_lib.c:1471: BUG? (runtime != ((void *)0)) [<c0125607>] run_timer_softirq+0xfb/0x151 [<c0121838>] __do_softirq+0x5a/0xbb [<c01218cf>] do_softirq+0x36/0x3a [<c0103747>] apic_timer_interrupt+0x1f/0x24 ALSA /home/benjamin/alsa-driver-1.0.16/acore/../alsa-kernel/core/pcm_lib.c:1471: BUG? (runtime != ((void *)0)) [<c0125607>] run_timer_softirq+0xfb/0x151 [<c0121838>] __do_softirq+0x5a/0xbb [<c01218cf>] do_softirq+0x36/0x3a [<c0103747>] apic_timer_interrupt+0x1f/0x24 [<c0102bbe>] sysenter_past_esp+0x3/0x79 ALSA /home/benjamin/alsa-driver-1.0.16/acore/../alsa-kernel/core/pcm_lib.c:1471: BUG? (runtime != ((void *)0)) [<c0125607>] run_timer_softirq+0xfb/0x151 [<c0121838>] __do_softirq+0x5a/0xbb [<c01218cf>] do_softirq+0x36/0x3a [<c0103747>] apic_timer_interrupt+0x1f/0x24 ALSA /home/benjamin/alsa-driver-1.0.16/acore/../alsa-kernel/core/pcm_lib.c:1471: BUG? (runtime != ((void *)0)) [<c0125607>] run_timer_softirq+0xfb/0x151 [<c0121838>] __do_softirq+0x5a/0xbb [<c01218cf>] do_softirq+0x36/0x3a [<c0103747>] apic_timer_interrupt+0x1f/0x24 ALSA /home/benjamin/alsa-driver-1.0.16/acore/../alsa-kernel/core/pcm_lib.c:1471: BUG? (runtime != ((void *)0)) [<c0125607>] run_timer_softirq+0xfb/0x151 [<c0121838>] __do_softirq+0x5a/0xbb [<c01218cf>] do_softirq+0x36/0x3a [<c0103747>] apic_timer_interrupt+0x1f/0x24 ALSA /home/benjamin/alsa-driver-1.0.16/acore/../alsa-kernel/core/pcm_lib.c:1471: BUG? (runtime != ((void *)0)) [<c0125607>] run_timer_softirq+0xfb/0x151 [<c0121838>] __do_softirq+0x5a/0xbb [<c01218cf>] do_softirq+0x36/0x3a [<c0103747>] apic_timer_interrupt+0x1f/0x24 ALSA /home/benjamin/alsa-driver-1.0.16/acore/../alsa-kernel/core/pcm_lib.c:1471: BUG? (runtime != ((void *)0)) [<c0125607>] run_timer_softirq+0xfb/0x151 [<c0121838>] __do_softirq+0x5a/0xbb [<c01218cf>] do_softirq+0x36/0x3a [<c0103747>] apic_timer_interrupt+0x1f/0x24 divide error: 0000 [#1] SMP Modules linked in: snd_aloop kfusd ppdev lp button ac battery vmblock vmxnet vmmemctl ipv6 vmhgfs dm_snapshot dm_mirror dm_mod loop tsdev snd_ens1371 gameport snd_seq_dummy snd_seq_oss snd_seq_midi snd_seq_midi_event snd_seq snd_rawmidi snd_seq_device snd_ac97_codec ac97_bus snd_pcm_oss snd_mixer_oss snd_pcm snd_timer snd i2c_piix4 serio_raw shpchp rtc floppy soundcore i2c_core intel_agp agpgart pci_hotplug parport_pc parport snd_page_alloc pcspkr psmouse evdev ext3 jbd mbcache sd_mod ide_cd cdrom generic mptspi mptscsih mptbase scsi_transport_spi scsi_mod pcnet32 mii piix ide_core thermal processor fan CPU: 1 EIP: 0060:[<d8ab2331>] Tainted: P VLI EFLAGS: 00210046 (2.6.18-6-686 #1) EIP is at snd_card_loopback_timer_function+0x41/0x82 [snd_aloop] eax: 00000000 ebx: c942e654 ecx: 00000000 edx: 00000000 esi: c13a8000 edi: d8ab22f0 ebp: 00000001 esp: cabebdf8 ds: 007b es: 007b ss: 0068 Process jackd (pid: 3746, ti=cabea000 task=d7041550 task.ti=cabea000) Stack: c13a8000 00000100 c13a8000 c0125607 cabebe08 cabebe08 00000011 c030df48 0000000a c0121838 00200046 00000000 ca447000 d776b394 c01218cf 00000400 c0103747 00000400 00000000 caaa1814 00000000 ca447000 d776b394 ca445000 Call Trace: [<c0125607>] run_timer_softirq+0xfb/0x151 [<c0121838>] __do_softirq+0x5a/0xbb [<c01218cf>] do_softirq+0x36/0x3a [<c0103747>] apic_timer_interrupt+0x1f/0x24 [<d89c2184>] snd_pcm_attach_substream+0x2bf/0x311 [snd_pcm] [<d89c3ea7>] snd_pcm_open_substream+0x11/0x91 [snd_pcm] [<d89c3fc8>] snd_pcm_open+0xa1/0x1d5 [snd_pcm] [<c0117778>] default_wake_function+0x0/0xc [<d89d4718>] snd_open+0xdb/0x17e [snd] [<c0161879>] chrdev_open+0x126/0x141 [<c0161753>] chrdev_open+0x0/0x141 [<c0158b45>] __dentry_open+0xc8/0x1ac [<c0158c8d>] nameidata_to_filp+0x19/0x28 [<c0158cc7>] do_filp_open+0x2b/0x31 [<d89d75c0>] snd_ctl_ioctl+0x0/0x960 [snd] [<c0158d0b>] do_sys_open+0x3e/0xb3 [<c0158dad>] sys_open+0x16/0x18 [<c0102c11>] sysenter_past_esp+0x56/0x79 Code: 08 74 08 0f 0b 52 00 65 26 ab d8 8b 50 08 e8 df 30 67 e7 8d 43 04 e8 9e e6 7c e7 8b 43 38 31 d2 8b 4b 3c 01 c1 03 43 40 89 4b 3c <f7> 73 24 89 53 40 8b 53 28 39 d1 72 2a 89 d6 89 c8 31 d2 f7 f6 EIP: [<d8ab2331>] snd_card_loopback_timer_function+0x41/0x82 [snd_aloop] SS:ESP 0068:cabebdf8 <0>Kernel panic - not syncing: Fatal exception in interrupt BUG: warning at arch/i386/kernel/smp.c:547/smp_call_function() [<c010f5a3>] smp_call_function+0x53/0xfe [<c011d97e>] printk+0x14/0x18 [<c010f661>] smp_send_stop+0x13/0x1c [<c011cfc6>] panic+0x4c/0xe2 [<c0104013>] die+0x256/0x28a [<c0104681>] do_divide_error+0x0/0x9a [<c010470f>] do_divide_error+0x8e/0x9a [<d8ab2331>] snd_card_loopback_timer_function+0x41/0x82 [snd_aloop] [<c015be05>] __getblk+0x27/0x266 [<c0146aec>] __do_page_cache_readahead+0x69/0x1e8 [<d88dbabb>] do_get_write_access+0x435/0x462 [jbd] [<d89104cf>] __ext3_get_inode_loc+0x109/0x2b9 [ext3] [<c0116412>] __activate_task+0x1c/0x29 [<c01037f9>] error_code+0x39/0x40 [<d8ab22f0>] snd_card_loopback_timer_function+0x0/0x82 [snd_aloop] [<d8ab2331>] snd_card_loopback_timer_function+0x41/0x82 [snd_aloop] [<c0125607>] run_timer_softirq+0xfb/0x151 [<c0121838>] __do_softirq+0x5a/0xbb [<c01218cf>] do_softirq+0x36/0x3a [<c0103747>] apic_timer_interrupt+0x1f/0x24 [<d89c2184>] snd_pcm_attach_substream+0x2bf/0x311 [snd_pcm] [<d89c3ea7>] snd_pcm_open_substream+0x11/0x91 [snd_pcm] [<d89c3fc8>] snd_pcm_open+0xa1/0x1d5 [snd_pcm] [<c0117778>] default_wake_function+0x0/0xc [<d89d4718>] snd_open+0xdb/0x17e [snd] [<c0161879>] chrdev_open+0x126/0x141 [<c0161753>] chrdev_open+0x0/0x141 [<c0158b45>] __dentry_open+0xc8/0x1ac [<c0158c8d>] nameidata_to_filp+0x19/0x28 [<c0158cc7>] do_filp_open+0x2b/0x31 [<d89d75c0>] snd_ctl_ioctl+0x0/0x960 [snd] [<c0158d0b>] do_sys_open+0x3e/0xb3 [<c0158dad>] sys_open+0x16/0x18 [<c0102c11>] sysenter_past_esp+0x56/0x79 ---Stop 32bit dump--- Kind regards, Benjamin

1 2

[alsa-devel] [PATCH] Add EM-X270 ASoC driver
by Mike Rapoport 26 May '08

26 May '08

This patch adds ASoC support for EM-X270 machine. Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> --- sound/soc/pxa/Kconfig | 9 ++++ sound/soc/pxa/Makefile | 3 +- sound/soc/pxa/em-x270.c | 102 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 113 insertions(+), 1 deletions(-) create mode 100644 sound/soc/pxa/em-x270.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 484f883..e31ab49 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -62,3 +62,12 @@ config SND_PXA2XX_SOC_E800 help Say Y if you want to add support for SoC audio on the Toshiba e800 PDA + +config SND_PXA2XX_SOC_EM_X270 + tristate "SoC Audio support for CompuLab EM-x270" + depends on SND_PXA2XX_SOC && MACH_EM_X270 + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9712 + help + Say Y if you want to add support for SoC audio on + CompuLab EM-x270. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 04e5646..5bc8edf 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -13,10 +13,11 @@ snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o +snd-soc-em-x270-objs := em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o - +obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c new file mode 100644 index 0000000..02dcac3 --- /dev/null +++ b/sound/soc/pxa/em-x270.c @@ -0,0 +1,102 @@ +/* + * em-x270.c -- SoC audio for EM-X270 + * + * Copyright 2007 CompuLab, Ltd. + * + * Author: Mike Rapoport <mike(a)compulab.co.il> + * + * Copied from tosa.c: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Richard Purdie <richard(a)openedhand.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <asm/arch/pxa-regs.h> +#include <asm/arch/hardware.h> +#include <asm/arch/audio.h> + +#include "../codecs/wm9712.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + +static struct snd_soc_dai_link em_x270_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], + }, +}; + +static struct snd_soc_machine em_x270 = { + .name = "EM-X270", + .dai_link = em_x270_dai, + .num_links = ARRAY_SIZE(em_x270_dai), +}; + +static struct snd_soc_device em_x270_snd_devdata = { + .machine = &em_x270, + .platform = &pxa2xx_soc_platform, + .codec_dev = &soc_codec_dev_wm9712, +}; + +static struct platform_device *em_x270_snd_device; + +static int __init em_x270_init(void) +{ + int ret; + + if (!machine_is_em_x270()) + return -ENODEV; + + em_x270_snd_device = platform_device_alloc("soc-audio", -1); + if (!em_x270_snd_device) + return -ENOMEM; + + platform_set_drvdata(em_x270_snd_device, &em_x270_snd_devdata); + em_x270_snd_devdata.dev = &em_x270_snd_device->dev; + ret = platform_device_add(em_x270_snd_device); + + if (ret) + platform_device_put(em_x270_snd_device); + + return ret; +} + +static void __exit em_x270_exit(void) +{ + platform_device_unregister(em_x270_snd_device); +} + +module_init(em_x270_init); +module_exit(em_x270_exit); + +/* Module information */ +MODULE_AUTHOR("Mike Rapoport"); +MODULE_DESCRIPTION("ALSA SoC EM-X270"); +MODULE_LICENSE("GPL"); -- 1.5.3.7

2 1

[alsa-devel] [PATCH] ASoC: Remove in-code changelogs
by Mark Brown 26 May '08

26 May '08

The overwhelming majority just say "initial version" anyway. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ac97.c | 3 --- sound/soc/codecs/wm9712.c | 3 --- sound/soc/codecs/wm9713.c | 3 --- sound/soc/pxa/corgi.c | 4 ---- sound/soc/pxa/pxa2xx-i2s.c | 3 --- sound/soc/pxa/spitz.c | 3 --- sound/soc/pxa/tosa.c | 3 --- sound/soc/s3c24xx/neo1973_wm8753.c | 4 ---- sound/soc/s3c24xx/s3c2443-ac97.c | 3 --- sound/soc/s3c24xx/s3c24xx-i2s.c | 5 ----- sound/soc/s3c24xx/s3c24xx-pcm.c | 4 ---- sound/soc/s3c24xx/smdk2443_wm9710.c | 3 --- sound/soc/soc-core.c | 4 ---- sound/soc/soc-dapm.c | 5 ----- 14 files changed, 0 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 2a1ffe3..7bf2081 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -10,9 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 17th Oct 2005 Initial version. - * * Generic AC97 support. */ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index aabfa0d..5ca0eb1 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -9,9 +9,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * Revision history - * 4th Feb 2006 Initial version. */ #include <linux/init.h> diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 1d292e7..86c1445 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -10,9 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 4th Feb 2006 Initial version. - * * Features:- * * o Support for AC97 Codec, Voice DAC and Aux DAC diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 852f16d..edeea63 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -11,10 +11,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * Revision history - * 30th Nov 2005 Initial version. - * */ #include <linux/module.h> diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 4250710..35090c2 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -9,9 +9,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * Revision history - * 12th Aug 2005 Initial version. */ #include <linux/init.h> diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 73a6df8..092b5c7 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -12,9 +12,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 30th Nov 2005 Initial version. - * */ #include <linux/module.h> diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index fda2aa0..465ff0f 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -12,9 +12,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 30th Nov 2005 Initial version. - * * GPIO's * 1 - Jack Insertion * 5 - Hookswitch (headset answer/hang up switch) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 79c5027..c1a0161 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -10,10 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 20th Jan 2007 Initial version. - * 05th Feb 2007 Rename all to Neo1973 - * */ #include <linux/module.h> diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index e81d9a6..0eed140 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -10,9 +10,6 @@ * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. - * - * Revision history - * 21st Mar 2007 Initial Version */ #include <linux/init.h> diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 1ed6afd..ddf8724 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -12,11 +12,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * - * Revision history - * 11th Dec 2006 Merged with Simtec driver - * 10th Nov 2006 Initial version. */ #include <linux/init.h> diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 7806ae6..ef59974 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -12,10 +12,6 @@ * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * - * Revision history - * 11th Dec 2006 Merged with Simtec driver - * 10th Nov 2006 Initial version. */ #include <linux/module.h> diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index b4a5630..8515d6f 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -10,9 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 8th Mar 2007 Initial version. - * */ #include <linux/module.h> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e148db9..0318d8a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -14,10 +14,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 12th Aug 2005 Initial version. - * 25th Oct 2005 Working Codec, Interface and Platform registration. - * * TODO: * o Add hw rules to enforce rates, etc. * o More testing with other codecs/machines. diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1ef6d94..8a3192b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -10,11 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 12th Aug 2005 Initial version. - * 25th Oct 2005 Implemented path power domain. - * 18th Dec 2005 Implemented machine and stream level power domain. - * * Features: * o Changes power status of internal codec blocks depending on the * dynamic configuration of codec internal audio paths and active -- 1.5.5.1

4 11

Re: [alsa-devel] Question about "control names"
by Juergen Beisert 23 May '08

23 May '08

On Friday 23 May 2008 11:36, Clemens Ladisch wrote: > Juergen Beisert wrote: > > I'm playing with the control names of my "snd_kcontrol_new" controls. > > When I'm using "PCM Playback Switch" and "PCM Playback Volume" alsamixer > > shows me a slider with a switch at its bottom. That's what I expected. > > But when I'm using a second set ("line in" to "line out" bypass), with > > "Line Bypass Capture Switch" and "Line Bypass Capture Volume" control > > names, its confusing (BTW: "Line Bypass Capture Volume" is of type > > SNDRV_CTL_ELEM_TYPE_ENUMERATED). > > This may be caused by the control type. Why is the volume "enumerated"? Its not linear. Its defined in dB and has some "holes" in it. When I change it to SNDRV_CTL_ELEM_TYPE_INTEGER I see the slider in the [Capture] screen, and also something at its bottom, but I can't switch anything: │▒▒│ │▒▒│ └──┘ ------ 7 <Line Byp> This I see in the [Playback] screen now with identical code but "PCM Playback Switch" and "PCM Playback Volume" as control names. │ │ ├──┤ │MM│ └──┘ 0<>0 PCM If I change the "Line Bypass Capture ..." control names to "CD Playback ..." the control looks like the PCM one above. Seems "Line Bypass Capture" is something alsamixer can't handle. Is there a better control name for such a feature? I can route "Line In" through a amplifier to "Line Out". Juergen -- Dipl.-Ing. Juergen Beisert | http://www.pengutronix.de Pengutronix - Linux Solutions for Science and Industry Handelsregister: Amtsgericht Hildesheim, HRA 2686 Vertretung Sued/Muenchen, Germany Phone: +49-8766-939 228 | Fax: +49-5121-206917-9

3 2

[alsa-devel] [PATCH][ASoC V2] Additional fixes for the Freescale MPC8610 drivers
by Timur Tabi 23 May '08

23 May '08

More miscellaneous fixes for the Freescale MPC8610 ASoC V2 drivers. CS4270 driver: Replace printk() with dev_xxx(). Use snd_soc_add_new_controls(). Report an error if a CS4270 is not found at the given I2C address. Additional initialization of the 'codec' structure. DMA driver: when an OSS application requests a sync, ASoC calls fsl_dma_hw_params() after calling fsl_dma_prepare(). This caused the DMA controller to be reprogrammed, which made it stop working. So the bulk of DMA programming was moved to fsl_dma_open(). Initialized some more fields in dma_private, so now the ISR won't panic. SSI driver: The SSI is now enabled only when it starts playback or capture, to prevent premature communication with the DMA controller. Added spinlocks to prevent the SSI from being enabled inadvertently by another stream. Fabric driver: Turned the driver into a platform driver. Added calls to snd_soc_card_config_codec() and snd_soc_card_init_codec(). Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 54 +++++++++-------- sound/soc/fsl/fsl_dma.c | 139 ++++++++++++++++++++++------------------- sound/soc/fsl/fsl_ssi.c | 79 +++++++++++++++++------- sound/soc/fsl/fsl_ssi.h | 1 + sound/soc/fsl/mpc8610_hpcd.c | 130 ++++++++++++++++++++++++--------------- 5 files changed, 240 insertions(+), 163 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 13b9d56..6bb4bc7 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -223,7 +223,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, rates &= ~SNDRV_PCM_RATE_KNOT; if (!rates) { - printk(KERN_ERR "cs4270: could not find a valid sample rate\n"); + dev_err(codec_dai->dev, "could not find a valid sample rate\n"); return -EINVAL; } @@ -263,7 +263,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, cs4270->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; break; default: - printk(KERN_ERR "cs4270: invalid DAI format\n"); + dev_err(codec_dai->dev, "invalid DAI format\n"); ret = -EINVAL; } @@ -315,7 +315,7 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, struct i2c_client *client = codec->control_data; if (i2c_smbus_write_byte_data(client, reg, value)) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "I2C write failed\n"); return -EIO; } @@ -359,7 +359,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, if (i == NUM_MCLK_RATIOS) { /* We did not find a matching ratio */ - printk(KERN_ERR "cs4270: could not find matching ratio\n"); + dev_err(codec->dev, "could not find matching ratio\n"); return -EINVAL; } @@ -369,7 +369,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | CS4270_PWRCTL_PDN); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "I2C write failed\n"); return ret; } @@ -381,7 +381,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, ret = cs4270_i2c_write(codec, CS4270_MODE, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "I2C write failed\n"); return ret; } @@ -398,13 +398,13 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg |= CS4270_FORMAT_DAC_LJ | CS4270_FORMAT_ADC_LJ; break; default: - printk(KERN_ERR "cs4270: unknown format\n"); + dev_err(codec->dev, "unknown format\n"); return -EINVAL; } ret = cs4270_i2c_write(codec, CS4270_FORMAT, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "I2C write failed\n"); return ret; } @@ -415,7 +415,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg &= ~CS4270_MUTE_AUTO; ret = cs4270_i2c_write(codec, CS4270_MUTE, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "I2C write failed\n"); return ret; } @@ -423,7 +423,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, ret = cs4270_i2c_write(codec, CS4270_PWRCTL, 0); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "I2C write failed\n"); return ret; } @@ -469,23 +469,16 @@ static int cs4270_codec_init(struct snd_soc_codec *codec, struct snd_soc_card *soc_card) { int ret; - unsigned int i; /* Add the non-DAPM controls */ - for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { - struct snd_kcontrol *kctrl; - - kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); + ret = snd_soc_add_new_controls(soc_card, cs4270_snd_controls, codec, + ARRAY_SIZE(cs4270_snd_controls)); - ret = snd_ctl_add(soc_card->card, kctrl); - if (ret < 0) { - dev_err(soc_card->card->dev, "could not add control\n"); - return ret; - } - } + if (ret < 0) + dev_err(soc_card->card->dev, "could not add control\n"); - return 0; + return ret; } /* @@ -513,9 +506,14 @@ static int cs4270_i2c_probe(struct i2c_client *client, return ret; } /* The top four bits of the chip ID should be 1100. */ - if ((ret & 0xF0) != 0xC0) - /* The device at this address is not a CS4270 codec */ + if ((ret & 0xF0) != 0xC0) { + /* This is a new-style I2C driver, which means we're supposed + * to be given the actual I2C address that the chip is on. If + * this device is not a CS4270, then that's a bug. + */ + dev_err(&client->dev, "device is not a CS4270\n"); return -ENODEV; + } dev_info(&client->dev, "found device at address %X\n", client->addr); dev_info(&client->dev, "hardware revision %X\n", ret & 0xF); @@ -567,10 +565,13 @@ static int cs4270_i2c_probe(struct i2c_client *client, codec->control_data = client; codec->init = cs4270_codec_init; + codec->reg_cache = cs4270->reg_cache; + codec->codec_read = cs4270_read_reg_cache; + codec->codec_write = cs4270_i2c_write; ret = snd_soc_register_codec(codec, &client->dev); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to register card\n"); + dev_err(&client->dev, "failed to register card\n"); goto error; } registered = 1; @@ -583,6 +584,7 @@ static int cs4270_i2c_probe(struct i2c_client *client, cs4270->dai = snd_soc_register_codec_dai(&cs4270->dai_new, &client->dev); if (!cs4270->dai) { + dev_err(&client->dev, "failed to register DAI\n"); ret = -EINVAL; goto error; } @@ -650,7 +652,7 @@ static int __init cs4270_init(void) ret = i2c_add_driver(&cs4270_i2c_driver); if (ret) - printk(KERN_ERR "cs4270: failed to register I2C driver\n"); + pr_err("cs4270: failed to register I2C driver\n"); return ret; } diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 4b48f8a..5a266fc 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -232,10 +232,8 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) ret = IRQ_HANDLED; if (sr & CCSR_DMA_SR_EOSI) { - struct snd_pcm_substream *substream = dma_private->substream; - /* Tell ALSA we completed a period. */ - snd_pcm_period_elapsed(substream); + snd_pcm_period_elapsed(dma_private->substream); /* * Update our link descriptors to point to the next period. We @@ -332,9 +330,13 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) struct fsl_dma_info *dma_info = cpu_dai->dma_data; struct fsl_dma_private *dma_private; + struct ccsr_dma_channel __iomem *dma_channel; dma_addr_t ld_buf_phys; + u64 temp_link; /* Pointer to next link descriptor */ + u32 mr; unsigned int channel; int ret = 0; + unsigned int i; /* * Reject any DMA buffer whose size is not a multiple of the period @@ -363,6 +365,10 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) else dma_private->dma_info = &dma_info[1]; + dma_private->substream = substream; + dma_private->ld_buf_phys = ld_buf_phys; + dma_private->dma_buf_phys = substream->dma_buffer.addr; + ret = request_irq(dma_private->dma_info->irq, fsl_dma_isr, 0, "DMA", dma_private); if (ret) { @@ -379,6 +385,66 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); runtime->private_data = dma_private; + /* Program the fixed DMA controller parameters */ + + dma_channel = dma_private->dma_info->channel; + + temp_link = dma_private->ld_buf_phys + + sizeof(struct fsl_dma_link_descriptor); + + for (i = 0; i < NUM_DMA_LINKS; i++) { + struct fsl_dma_link_descriptor *link = &dma_private->link[i]; + + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->next = cpu_to_be64(temp_link); + + temp_link += sizeof(struct fsl_dma_link_descriptor); + } + /* The last link descriptor points to the first */ + dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); + + /* Tell the DMA controller where the first link descriptor is */ + out_be32(&dma_channel->clndar, + CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys)); + out_be32(&dma_channel->eclndar, + CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys)); + + /* The manual says the BCR must be clear before enabling EMP */ + out_be32(&dma_channel->bcr, 0); + + /* + * Program the mode register for interrupts, external master control, + * and source/destination hold. Also clear the Channel Abort bit. + */ + mr = in_be32(&dma_channel->mr) & + ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); + + /* + * We want External Master Start and External Master Pause enabled, + * because the SSI is controlling the DMA controller. We want the DMA + * controller to be set up in advance, and then we signal only the SSI + * to start transfering. + * + * We want End-Of-Segment Interrupts enabled, because this will generate + * an interrupt at the end of each segment (each link descriptor + * represents one segment). Each DMA segment is the same thing as an + * ALSA period, so this is how we get an interrupt at the end of every + * period. + * + * We want Error Interrupt enabled, so that we can get an error if + * the DMA controller is mis-programmed somehow. + */ + mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | + CCSR_DMA_MR_EMS_EN; + + /* For playback, we want the destination address to be held. For + capture, set the source address to be held. */ + mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE; + + out_be32(&dma_channel->mr, mr); + return 0; } @@ -452,12 +518,8 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; - struct ccsr_dma_channel __iomem *dma_channel = - dma_private->dma_info->channel; dma_addr_t temp_addr; /* Pointer to next period */ - u64 temp_link; /* Pointer to next link descriptor */ - uint32_t mr; /* Temporary variable for MR register */ unsigned int i; @@ -475,8 +537,6 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, dma_private->dma_buf_next = dma_private->dma_buf_phys; /* - * Initialize each link descriptor. - * * The actual address in STX0 (destination for playback, source for * capture) is based on the sample size, but we don't know the sample * size in this function, so we'll have to adjust that later. See @@ -492,16 +552,11 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, * buffer itself. */ temp_addr = substream->dma_buffer.addr; - temp_link = dma_private->ld_buf_phys + - sizeof(struct fsl_dma_link_descriptor); for (i = 0; i < NUM_DMA_LINKS; i++) { struct fsl_dma_link_descriptor *link = &dma_private->link[i]; link->count = cpu_to_be32(period_size); - link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->next = cpu_to_be64(temp_link); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) link->source_addr = cpu_to_be32(temp_addr); @@ -509,51 +564,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, link->dest_addr = cpu_to_be32(temp_addr); temp_addr += period_size; - temp_link += sizeof(struct fsl_dma_link_descriptor); } - /* The last link descriptor points to the first */ - dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); - - /* Tell the DMA controller where the first link descriptor is */ - out_be32(&dma_channel->clndar, - CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys)); - out_be32(&dma_channel->eclndar, - CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys)); - - /* The manual says the BCR must be clear before enabling EMP */ - out_be32(&dma_channel->bcr, 0); - - /* - * Program the mode register for interrupts, external master control, - * and source/destination hold. Also clear the Channel Abort bit. - */ - mr = in_be32(&dma_channel->mr) & - ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); - - /* - * We want External Master Start and External Master Pause enabled, - * because the SSI is controlling the DMA controller. We want the DMA - * controller to be set up in advance, and then we signal only the SSI - * to start transfering. - * - * We want End-Of-Segment Interrupts enabled, because this will generate - * an interrupt at the end of each segment (each link descriptor - * represents one segment). Each DMA segment is the same thing as an - * ALSA period, so this is how we get an interrupt at the end of every - * period. - * - * We want Error Interrupt enabled, so that we can get an error if - * the DMA controller is mis-programmed somehow. - */ - mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | - CCSR_DMA_MR_EMS_EN; - - /* For playback, we want the destination address to be held. For - capture, set the source address to be held. */ - mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE; - - out_be32(&dma_channel->mr, mr); return 0; } @@ -793,8 +804,6 @@ static struct snd_soc_platform_new fsl_dma_platform = { .pcm_free = fsl_dma_free_dma_buffers, }; -/* - */ static int fsl_dma_probe(struct platform_device *pdev) { struct snd_soc_platform *platform; @@ -806,15 +815,16 @@ static int fsl_dma_probe(struct platform_device *pdev) return -ENOMEM; } - platform_set_drvdata(pdev, platform); - ret = snd_soc_register_platform(platform, &pdev->dev); if (ret < 0) { dev_err(&pdev->dev, "could not register platform\n"); snd_soc_free_platform(platform); + return ret; } - return ret; + platform_set_drvdata(pdev, platform); + + return 0; } static int fsl_dma_remove(struct platform_device *pdev) @@ -852,7 +862,8 @@ static __init int fsl_dma_init(void) pdev = platform_device_register_simple(fsl_platform_id, 0, NULL, 0); if (!pdev) { - pr_err("fsl-dma: could not register platform\n"); + pr_err("fsl-dma: could not register device\n"); + platform_driver_unregister(&fsl_dma_driver); return ret; } diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9263fd5..0d531e9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -245,11 +245,13 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, /* * Program the SSI into I2S Slave Non-Network Synchronous mode. - * Also enable the transmit and receive FIFO. + * Also enable the transmit and receive FIFO. Make sure TE and + * RE are disabled. * * FIXME: Little-endian samples require a different shift dir */ - clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK, + clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | + CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE, CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE | CCSR_SSI_SCR_SYN); @@ -284,6 +286,8 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, out_be32(&ssi->sfcsr, CCSR_SSI_SFCSR_TFWM0(6) | CCSR_SSI_SFCSR_RFWM0(2)); + spin_unlock(&ssi_info->lock); + /* * We keep the SSI disabled because if we enable it, then the * DMA controller will start. It's not supposed to start until @@ -296,10 +300,9 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ssi_info->playback++; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ssi_info->capture++; + ssi_info->playback = 1; + else + ssi_info->capture = 1; cpu_dai->dma_data = ssi_info->dma_info; @@ -326,20 +329,33 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream, struct fsl_ssi_info *ssi_info = cpu_dai->private_data; struct ccsr_ssi __iomem *ssi = ssi_info->ssi; uint32_t wl; + u32 scr; + u32 sxccr; wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); - /* FIXME: We should read and write the registers manually so as to - minimize the amount of time the SSI is disabled. */ + spin_lock(&ssi_info->lock); - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + scr = in_be32(&ssi->scr); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); - else - clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + /* If the SSI is currently enabled, then we want to keep it disabled for + * as short as time as possible, so we pre-calculate all the values for + * the appropriate registers, and then program them rapidly. + */ - setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sxccr = (in_be32(&ssi->stccr) & ~CCSR_SSI_SxCCR_WL_MASK) | wl; + out_be32(&ssi->scr, scr & ~CCSR_SSI_SCR_SSIEN); + out_be32(&ssi->stccr, sxccr); + out_be32(&ssi->scr, scr); + } else { + sxccr = (in_be32(&ssi->srccr) & ~CCSR_SSI_SxCCR_WL_MASK) | wl; + out_be32(&ssi->scr, scr & ~CCSR_SSI_SCR_SSIEN); + out_be32(&ssi->srccr, sxccr); + out_be32(&ssi->scr, scr); + } + + spin_unlock(&ssi_info->lock); return 0; } @@ -364,10 +380,14 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - setbits32(&ssi->scr, CCSR_SSI_SCR_TE); - } else { - setbits32(&ssi->scr, CCSR_SSI_SCR_RE); + spin_lock(&ssi_info->lock); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); + else { + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); /* * I think we need this delay to allow time for the SSI @@ -376,6 +396,8 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, */ mdelay(1); } + + spin_unlock(&ssi_info->lock); break; case SNDRV_PCM_TRIGGER_STOP: @@ -406,10 +428,9 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, struct fsl_ssi_info *ssi_info = cpu_dai->private_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ssi_info->playback--; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ssi_info->capture--; + ssi_info->playback = 0; + else + ssi_info->capture = 0; /* * If this is the last active substream, disable the SSI and release @@ -427,7 +448,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, /** * fsl_ssi_set_sysclk: set the clock frequency and direction * - * This function is called by the soc_card driver to tell us what the clock + * This function is called by the fabric driver to tell us what the clock * frequency and direction are. * * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN), @@ -523,6 +544,16 @@ static struct snd_soc_dai_ops ops = { .set_fmt = fsl_ssi_set_fmt, }; +/** + * find_dma_node: find the DMA node to use in a legacy device tree + * + * This function returns a device node matching a given DMA channel on a + * given DMA controller. + * + * The original device tree for the MPC8610 HPCD did not include phandles + * from the SSI nodes to the DMA nodes. To support these device trees, we + * declare that SSIx will use DMA channels 0 and 1 of DMA controller DMAx. + */ static struct device_node *find_dma_node(unsigned int controller, unsigned int channel) { @@ -698,7 +729,7 @@ static int fsl_ssi_probe(struct of_device *ofdev, ssi_info->dai_format = SND_SOC_DAIFMT_LEFT_J; ssi_info->codec_clk_direction = SND_SOC_CLOCK_IN; ssi_info->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "rj-master") == 0) { + } else if (strcasecmp(sprop, "rj-slave") == 0) { ssi_info->dai_format = SND_SOC_DAIFMT_RIGHT_J; ssi_info->codec_clk_direction = SND_SOC_CLOCK_OUT; ssi_info->cpu_clk_direction = SND_SOC_CLOCK_IN; diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 05fd84d..9a22b04 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -219,6 +219,7 @@ struct fsl_ssi_info { struct device *dev; unsigned int playback; unsigned int capture; + spinlock_t lock; struct snd_soc_dai *dai; struct device_attribute dev_attr; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index a857e57..10c7fd7 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -52,7 +52,24 @@ static int mpc8610_hpcd_audio_init(struct snd_soc_card *soc_card) struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; struct snd_soc_dai *cpu_dai = snd_soc_card_get_dai(soc_card, "fsl,mpc8610-ssi"); + struct snd_soc_codec *codec = + snd_soc_card_get_codec(soc_card, "cirrus,cs4270"); struct fsl_ssi_info *ssi_info = cpu_dai->private_data; + int ret; + + /* This is stupid. snd_soc_card_config_codec() initializes the + * codec->control_data structure, which is supposed to be a pointer to + * the i2c_client structure. But I already assigned that variable in + * the codec driver. After all, the codec driver is the one that + * supposed to get probed by the I2C bus. So in order to avoid + * overwriting codec->control_data, I pass it as the 4th parameter. + */ + snd_soc_card_config_codec(codec, NULL, NULL, codec->control_data); + ret = snd_soc_card_init_codec(codec, soc_card); + if (ret < 0) { + dev_err(soc_card->dev, "could not initialize codec\n"); + return ret; + } /* Program the signal routing between the SSI and the DMA */ guts_set_dmacr(soc_card_data->guts, @@ -94,17 +111,37 @@ static int mpc8610_hpcd_audio_init(struct snd_soc_card *soc_card) * This function takes board-specific information, like clock frequencies * and serial data formats, and passes that information to the codec and * transport drivers. + * + * The clock information is stored in the ssi_info data structure because + * it's gathered by the SSI driver from the device tree. This isn't really + * the best way to handle it. In fact, this problem is a core dillema with + * ASoC V2: whether the fabric driver should hand the information to the + * platform/codec drivers, or whether these drivers should get the + * information themselves. With device trees, the necessary data may be + * scattered in various device nodes, so it might be impossible to obtain + * that information in a generic manner. + * + * What makes this problem even more thorny is that some of these drivers + * are OF drivers and the rest aren't. The CS4270 can't be an OF driver, + * yet the information it needs is in an OF device node. So perhaps the + * best approach is to have generic codec OF->platform code in the fabric + * driver (normally this code is in fsl_soc.c). + * + * Even more frustrating is the fact that the CS4270 driver is an I2C + * driver, so it's probed via the I2C bus. We need to update the powerpc + * platform to initialize client->dev->archdata.of_node to point to the + * device node. Then the CS4270 driver can get its own clock rate. */ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct mpc8610_hpcd_data *soc_card_data = rtd->soc_card->private_data; + struct fsl_ssi_info *ssi_info = cpu_dai->private_data; int ret = 0; /* Tell the CPU driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(cpu_dai, soc_card_data->dai_format); + ret = snd_soc_dai_set_fmt(cpu_dai, ssi_info->dai_format); if (ret < 0) { dev_err(substream->pcm->card->dev, "could not set CPU driver audio format\n"); @@ -112,7 +149,7 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) } /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(codec_dai, soc_card_data->dai_format); + ret = snd_soc_dai_set_fmt(codec_dai, ssi_info->dai_format); if (ret < 0) { dev_err(substream->pcm->card->dev, "could not set codec driver audio format\n"); @@ -123,8 +160,8 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) * Tell the CPU driver what the clock frequency is, and whether it's a * slave or master. */ - ret = snd_soc_dai_set_sysclk(cpu_dai, 0, soc_card_data->clk_frequency, - soc_card_data->cpu_clk_direction); + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, ssi_info->clk_frequency, + ssi_info->cpu_clk_direction); if (ret < 0) { dev_err(substream->pcm->card->dev, "could not set CPU driver clock parameters\n"); @@ -135,8 +172,8 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) * Tell the codec driver what the MCLK frequency is, and whether it's * a slave or master. */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, soc_card_data->clk_frequency, - soc_card_data->codec_clk_direction); + ret = snd_soc_dai_set_sysclk(codec_dai, 0, ssi_info->clk_frequency, + ssi_info->codec_clk_direction); if (ret < 0) { dev_err(substream->pcm->card->dev, "could not set codec driver clock params\n"); @@ -207,39 +244,38 @@ static struct snd_soc_pcm_config mpc8610_pcm_config = { * respect to audio hardware connections. Therefore, we create a new ASoC * device for each new SSI node that has a codec attached. */ -static int mpc8610_hpcd_probe(struct of_device *ofdev, - const struct of_device_id *match) +static int mpc8610_hpcd_probe(struct platform_device *pdev) { struct snd_soc_card *soc_card = NULL; - struct device_node *guts_np = NULL; struct mpc8610_hpcd_data *soc_card_data; + struct device_node *guts_np = NULL; int ret = -ENODEV; soc_card_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); if (!soc_card_data) { - dev_err(&ofdev->dev, "could not allocate card structure\n"); + dev_err(&pdev->dev, "could not allocate card structure\n"); return -ENOMEM; } /* Map the global utilities registers. */ guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); if (!guts_np) { - dev_err(&ofdev->dev, "could not obtain address of GUTS\n"); + dev_err(&pdev->dev, "could not obtain address of GUTS\n"); ret = -EINVAL; goto error; } soc_card_data->guts = of_iomap(guts_np, 0); of_node_put(guts_np); if (!soc_card_data->guts) { - dev_err(&ofdev->dev, "could not map GUTS\n"); + dev_err(&pdev->dev, "could not map GUTS\n"); ret = -EINVAL; goto error; } - soc_card = snd_soc_card_create("MPC8610", &ofdev->dev, + soc_card = snd_soc_card_create("MPC8610", &pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (!soc_card) { - dev_err(&ofdev->dev, "could not create card\n"); + dev_err(&pdev->dev, "could not create card\n"); goto error; } @@ -247,19 +283,19 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, soc_card->init = mpc8610_hpcd_audio_init; soc_card->exit = mpc8610_hpcd_audio_exit; soc_card->private_data = soc_card_data; - soc_card->dev = &ofdev->dev; + soc_card->dev = &pdev->dev; ret = snd_soc_card_create_pcms(soc_card, &mpc8610_pcm_config, 1); if (ret) { - dev_err(&ofdev->dev, "could not create PCMs\n"); + dev_err(&pdev->dev, "could not create PCMs\n"); goto error; } - /* every has been added at this point */ - dev_set_drvdata(&ofdev->dev, soc_card); + platform_set_drvdata(pdev, soc_card); + ret = snd_soc_card_register(soc_card); if (ret) { - dev_err(&ofdev->dev, "could not register card\n"); + dev_err(&pdev->dev, "could not register card\n"); goto error; } @@ -277,14 +313,9 @@ error: return ret; } -/** - * mpc8610_hpcd_remove: remove the OF device - * - * This function is called when the OF device is removed. - */ -static int mpc8610_hpcd_remove(struct of_device *ofdev) +static int mpc8610_hpcd_remove(struct platform_device *pdev) { - struct snd_soc_card *soc_card = dev_get_drvdata(&ofdev->dev); + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; if (soc_card_data->guts) @@ -294,27 +325,20 @@ static int mpc8610_hpcd_remove(struct of_device *ofdev) snd_soc_card_free(soc_card); - dev_set_drvdata(&ofdev->dev, NULL); - return 0; } -static struct of_device_id mpc8610_hpcd_match[] = { - { - .compatible = "fsl,mpc8610-ssi", +static struct platform_driver mpc8610_hpcd_driver = { + .driver = { + .name = "mpc8610_hpcd", + .owner = THIS_MODULE, }, - {} -}; -MODULE_DEVICE_TABLE(of, mpc8610_hpcd_match); - -static struct of_platform_driver mpc8610_hpcd_of_driver = { - .owner = THIS_MODULE, - .name = "mpc8610_hpcd", - .match_table = mpc8610_hpcd_match, .probe = mpc8610_hpcd_probe, - .remove = mpc8610_hpcd_remove, + .remove = __devexit_p(mpc8610_hpcd_remove), }; +static struct platform_device *pdev; + /** * mpc8610_hpcd_init: fabric driver initialization. * @@ -324,15 +348,22 @@ static int __init mpc8610_hpcd_init(void) { int ret; - printk(KERN_INFO "Freescale MPC8610 HPCD ALSA SoC fabric driver\n"); + printk(KERN_INFO "Freescale MPC8610 HPCD ASoC fabric driver\n"); - ret = of_register_platform_driver(&mpc8610_hpcd_of_driver); + ret = platform_driver_register(&mpc8610_hpcd_driver); + if (ret < 0) { + pr_err("mpc8610-hpcd: could not register platform\n"); + return ret; + } - if (ret) - printk(KERN_ERR - "mpc8610-hpcd: failed to register platform driver\n"); + pdev = platform_device_register_simple("mpc8610_hpcd", 0, NULL, 0); + if (!pdev) { + pr_err("mpc8610-hpcd: could not register device\n"); + platform_driver_unregister(&mpc8610_hpcd_driver); + return ret; + } - return ret; + return 0; } /** @@ -342,12 +373,13 @@ static int __init mpc8610_hpcd_init(void) */ static void __exit mpc8610_hpcd_exit(void) { - of_unregister_platform_driver(&mpc8610_hpcd_of_driver); + platform_device_unregister(pdev); + platform_driver_unregister(&mpc8610_hpcd_driver); } module_init(mpc8610_hpcd_init); module_exit(mpc8610_hpcd_exit); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); -MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC fabric driver"); +MODULE_DESCRIPTION("Freescale MPC8610 HPCD ASoC fabric driver"); MODULE_LICENSE("GPL"); -- 1.5.5

2 1

[alsa-devel] Problem with intel-hda integrated audio
by Matteo Sasso 23 May '08

23 May '08

Hello, I'm having a problem with a soundcard under ubuntu. I tried nearly every workaround I could find on the Internet but I have not been successful so far. The computer in question is a Toshiba Satellite Pro U200 series. The soundcard seems to be correctly identified by ALSA but it seems muted, nothing I tried seems to work. I followed the instructions given at http://www.mjmwired.net/kernel/Documentation/sound/alsa/ALSA-Configuration.… I determined that my card uses a AD1981 codec, so I added the following to /etc/modprobe.d/alsa-base: options snd-hda-intel model=toshiba I tried many other combinations (3stack, basic, laptop), with and without "position_fix=1". I also tried using underscores instead of hypens ("snd_hda_intel"). I run update-initramfs -u after every try. lspci -vn has to say the following: 00:1b.0 0403: 8086:27d8 (rev 02) Sybsystem: 1179:0001 Flags: bus master, fast devsel, latency 0, IRQ 23 Memory at 50080000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 I'll keep tabs on this mailing list, so feel free to ask for more info. Any help is appreciated. Thanks.

2 2

[alsa-devel] Question about "control names"
by Juergen Beisert 23 May '08

23 May '08

Hi, I'm playing with the control names of my "snd_kcontrol_new" controls. When I'm using "PCM Playback Switch" and "PCM Playback Volume" alsamixer shows me a slider with a switch at its bottom. That's what I expected. But when I'm using a second set ("line in" to "line out" bypass), with "Line Bypass Capture Switch" and "Line Bypass Capture Volume" control names, its confusing (BTW: "Line Bypass Capture Volume" is of type SNDRV_CTL_ELEM_TYPE_ENUMERATED). The "Line Bypass Capture Volume" control is shown on the [Playback] page with the name "Line Byp" and the default value. I can walk through the enumerated values. But the "Line Bypass Capture Switch" is shown on the [Capture] page in alsamixer as: L R CAPTUR <Line Byp> What does it mean? Problem of alsamixer or the name of this set? Juergen -- Dipl.-Ing. Juergen Beisert | http://www.pengutronix.de Pengutronix - Linux Solutions for Science and Industry Handelsregister: Amtsgericht Hildesheim, HRA 2686 Vertretung Sued/Muenchen, Germany Phone: +49-8766-939 228 | Fax: +49-5121-206917-9

1 0

[alsa-devel] softvol plug-in dependancies?
by Simon Dunn 23 May '08

23 May '08

Hi Guys The softvol plug-in, are there any pre-requisites or dependancies to using this? I've tried various entries in the /etc/asound.conf configuration file all with the same result. I can see the control in alsamixer, once I've run the speaker-test through it, but cannot adjust any settings. Trying to run 'alsactl store 0' results in this error : . alsactl: control.c:2437: snd_ctl_elem_value_get_integer: Assertion `idx < sizeof (obj->value.integer.value) / sizeof(obj->value.integer.value[0])' failed. Aborted . Curiously using amixer the device is listed but its properties are displayed like this . Simple mixer control 'PCM Vol',0 Capabilities: volume Playback channels: Front Left - Front Right - Rear Left - Rear Right - Front C enter - Woofer - Side Left - Side Right - Rear Center - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? Capture channels: Front Left - Front Right - Rear Left - Rear Right - Front Ce nter - Woofer - Side Left - Side Right - Rear Center - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? Limits: 0 - 0 Front Left: 0 [0%] Front Right: 0 [0%] Rear Left: 0 [0%] Rear Right: 0 [0%] Front Center: 0 [0%] Woofer: 0 [0%] Side Left: 0 [0%] Side Right: 0 [0%] Rear Center: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] Any clues as to what's going on here would be much appreciated before I have to start dissecting the library. TIA for any hints Simon -- Simon Dunn, Software Engineer MPC Data Limited e-mail: sdunn(a)mpc-data.co.uk web: www.mpc-data.co.uk tel: +44 (0) 1225 710600 fax: +44 (0) 1225 710601 ddi: +44 (0) 1225 710631 MPC Data Limited is a company registered in England and Wales with company number 05507446 Registered Address: County Gate, County Way, Trowbridge, Wiltshire, BA14 7FJ VAT no: 850625238 The information in this email and in the attached documents is confidential and may be legally privileged. Any unauthorized review, copying, disclosure or distribution is prohibited and may be unlawful. It is intended solely for the addressee. Access to this email by anyone else is unauthorized. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. When addressed to our clients any opinions or advice contained in this email is subject to the terms and conditions expressed in the governing contract.

2 3

[alsa-devel] HG -> GIT migration
by Jaroslav Kysela 23 May '08

23 May '08

Hi all, I finished migration from HG to GIT. All repositories are converted and we have one "new" repository - alsa-kernel.git which is full Linux 2.6 GIT tree. The old HG alsa-kernel repository was renamed to alsa-kmirror - just to track our changes in continuous timeline for releases and packaging. HG access to all repositories is closed now. More information can be obtained here: http://www.alsa-project.org/main/index.php/GIT_Server Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

10 35

Re: [alsa-devel] [Alsa-user] Pc Speaker
by Rene Herman 23 May '08

23 May '08

On 13-05-08 12:44, Armin ranjbar wrote: >> Device Drivers -> >> Input device support -> >> Miscelaneous devices >> >> With current mainline, it's: >> >> General setup -> >> Configure standard kernel features (for small systems) >> Enable PC-Speaker support > > Thank you very much for your reply , > > but i have to say that i have this options activated , still no pc > speaker . Rechecked. Yes, you need this option, you need the ALSA one disabled (not modular, disabled) and then the PC speaker driver option appears in its input menu where you want to enable it. Paths as given in last reply. Yes, that would seem to be amazingly clumsily done. Perhaps there was a reason (adding alsa-devel). Rene.

3 6

[alsa-devel] [PATCH] Patch for alsa-info.sh to check if pastebin.ca is down, and handle with grace.
by Travis Place 22 May '08

22 May '08

Changelog: This patch will allow alsa-info.sh to check if the pastebin site (www.pastebin.ca) is down, before trying to upload to it. It will fail gracefully, with a message letting the user know what happened. The script will also keep the ALSA information it collected (same way it does with --no-upload). Patch is attached. Signed-off-by: Travis Place <wishie at wishie dot net> -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

1 0

[alsa-devel] Moving sound/* to drivers/ ?
by Takashi Iwai 22 May '08

22 May '08

While we are at the issue ALSA HG <-> GIT transition... One thing that annoys me sometimes is that the sound driver codes are on a different subdirectory than other normal drivers, namely under /sound. I don't remember exactly why it came there, but this often makes people forget to change the necessary code for the sound subsystem, or slip from the statistics (I'm not sure whether Linus counted sound/* with recent his posts). If I understand correctly, with git, we can move the files in relatively little costs. So, what about moving sound/* back to drivers/sound/* or drivers/media/sound/*? Of course, the primary question is whether it's really worth. The obvious drawback is that patches won't be applicable after the move. So, if we do it, doing at the last seems practical. But, this can be a bit problem with keeping such a change on linux-next until the next merge... And, another question is to where. drivers/sound or drivers/media/sound, or whatever. Any comments appreciated. thanks, Takashi

9 20

Re: [alsa-devel] [pulseaudio-discuss] Asynchronous notification in ALSA - I/O Plugin - Pulseaudio
by Lennart Poettering 22 May '08

22 May '08

On Thu, 22.05.08 14:11, Colin Guthrie (gmane(a)colin.guthr.ie) wrote: > > That freeze is an unfixable race condition that Flash has when > > shutting down audio. > > > > See the explanation down on http://pulseaudio.org/ticket/267 > > I thought the whole point was that by adding async code to ioplug, the > libflashsupport stuff wasn't needed and flash could just use it's native > alsa support + alsa's pulse plugin.... or am I missing the point here? The destruction dead lock issue is present both in libflashsupport and the native ALSA backend. It's just that it is almost never triggered if a hw device is used. Adobe released the sources of libflashsupport where you can actually check this. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH] Add async support to ioplug
by Takashi Iwai 22 May '08

22 May '08

Hi, the patch below adds a framework to support async on ioplug. Now an external ioplug plugin can implement appropriate async methods if possible. The async callback is just a copy of snd_pcm_async(). When sig >= 0, activate the async, and with a negative value, deactivate async. The test patch to alsa-plugins will follow. Takashi --- diff -r 8b22022a861d include/local.h --- a/include/local.h Fri May 09 16:02:02 2008 +0200 +++ b/include/local.h Tue May 20 09:02:35 2008 +0200 @@ -149,7 +149,13 @@ void *private_data; struct list_head glist; struct list_head hlist; + int no_signal; }; + +int _snd_async_add_handler(snd_async_handler_t **handler, int fd, + snd_async_callback_t callback, + void *private_data, int no_signal); +void _snd_async_call_handlers(int fd); typedef enum _snd_set_mode { SND_CHANGE, diff -r 8b22022a861d include/pcm_ioplug.h --- a/include/pcm_ioplug.h Fri May 09 16:02:02 2008 +0200 +++ b/include/pcm_ioplug.h Tue May 20 09:02:35 2008 +0200 @@ -66,7 +66,7 @@ */ #define SND_PCM_IOPLUG_VERSION_MAJOR 1 /**< Protocol major version */ #define SND_PCM_IOPLUG_VERSION_MINOR 0 /**< Protocol minor version */ -#define SND_PCM_IOPLUG_VERSION_TINY 1 /**< Protocol tiny version */ +#define SND_PCM_IOPLUG_VERSION_TINY 2 /**< Protocol tiny version */ /** * IO-plugin protocol version */ @@ -114,6 +114,10 @@ unsigned int rate; /**< rate; filled after hw_params is called */ snd_pcm_uframes_t period_size; /**< period size; filled after hw_params is called */ snd_pcm_uframes_t buffer_size; /**< buffer size; filled after hw_params is called */ + /** + * fields added by version 1.0.2 + */ + unsigned int no_signal_async; /**< async support without signal */ }; /** Callback table of ioplug */ @@ -189,6 +193,10 @@ * get the delay for the running PCM; optional */ int (*delay)(snd_pcm_ioplug_t *io, snd_pcm_sframes_t *delayp); + /** + * set up async handler; optional; added in version 1.0.2 + */ + int (*async)(snd_pcm_ioplug_t *io, int sig, pid_t pid); }; @@ -212,6 +220,9 @@ /* change PCM status */ int snd_pcm_ioplug_set_state(snd_pcm_ioplug_t *ioplug, snd_pcm_state_t state); +/* call async handlers */ +void snd_pcm_ioplug_call_async_handlers(snd_pcm_ioplug_t *ioplug); + /** \} */ #endif /* __ALSA_PCM_IOPLUG_H */ diff -r 8b22022a861d src/async.c --- a/src/async.c Fri May 09 16:02:02 2008 +0200 +++ b/src/async.c Tue May 20 09:02:35 2008 +0200 @@ -48,19 +48,48 @@ #endif static LIST_HEAD(snd_async_handlers); +static int sig_count; -static void snd_async_handler(int signo ATTRIBUTE_UNUSED, siginfo_t *siginfo, void *context ATTRIBUTE_UNUSED) +#ifndef DOC_HIDDEN +void _snd_async_call_handlers(int fd) { - int fd; struct list_head *i; - //assert(siginfo->si_code == SI_SIGIO); - fd = siginfo->si_fd; list_for_each(i, &snd_async_handlers) { snd_async_handler_t *h = list_entry(i, snd_async_handler_t, glist); if (h->fd == fd && h->callback) h->callback(h); } } +#endif + +static void snd_async_handler(int signo ATTRIBUTE_UNUSED, siginfo_t *siginfo, void *context ATTRIBUTE_UNUSED) +{ + int fd; + //assert(siginfo->si_code == SI_SIGIO); + fd = siginfo->si_fd; + _snd_async_call_handlers(fd); +} + +#ifndef DOC_HIDDEN +int _snd_async_add_handler(snd_async_handler_t **handler, int fd, + snd_async_callback_t callback, + void *private_data, int no_signal) +{ + snd_async_handler_t *h; + assert(handler); + h = malloc(sizeof(*h)); + if (!h) + return -ENOMEM; + h->fd = fd; + h->callback = callback; + h->private_data = private_data; + list_add_tail(&h->glist, &snd_async_handlers); + INIT_LIST_HEAD(&h->hlist); + h->no_signal = no_signal; + *handler = h; + return 0; +} +#endif /** * \brief Registers an async handler. @@ -94,20 +123,13 @@ int snd_async_add_handler(snd_async_handler_t **handler, int fd, snd_async_callback_t callback, void *private_data) { - snd_async_handler_t *h; - int was_empty; - assert(handler); - h = malloc(sizeof(*h)); - if (!h) - return -ENOMEM; - h->fd = fd; - h->callback = callback; - h->private_data = private_data; - was_empty = list_empty(&snd_async_handlers); - list_add_tail(&h->glist, &snd_async_handlers); - INIT_LIST_HEAD(&h->hlist); - *handler = h; - if (was_empty) { + int err; + + err = _snd_async_add_handler(handler, fd, callback, private_data, 0); + if (err < 0) + return err; + + if (!sig_count) { int err; struct sigaction act; memset(&act, 0, sizeof(act)); @@ -116,10 +138,15 @@ sigemptyset(&act.sa_mask); err = sigaction(snd_async_signo, &act, NULL); if (err < 0) { + err = -errno; SYSERR("sigaction"); - return -errno; + list_del(&(*handler)->glist); + free(*handler); + *handler = NULL; + return err; } } + sig_count++; return 0; } @@ -133,15 +160,18 @@ int err = 0; assert(handler); list_del(&handler->glist); - if (list_empty(&snd_async_handlers)) { - struct sigaction act; - memset(&act, 0, sizeof(act)); - act.sa_flags = 0; - act.sa_handler = SIG_DFL; - err = sigaction(snd_async_signo, &act, NULL); - if (err < 0) { - SYSERR("sigaction"); - return -errno; + if (!handler->no_signal) { + sig_count--; + if (!sig_count) { + struct sigaction act; + memset(&act, 0, sizeof(act)); + act.sa_flags = 0; + act.sa_handler = SIG_DFL; + err = sigaction(snd_async_signo, &act, NULL); + if (err < 0) { + SYSERR("sigaction"); + /* return -errno; */ + } } } if (handler->type == SND_ASYNC_HANDLER_GENERIC) diff -r 8b22022a861d src/pcm/pcm.c --- a/src/pcm/pcm.c Fri May 09 16:02:02 2008 +0200 +++ b/src/pcm/pcm.c Tue May 20 09:02:36 2008 +0200 @@ -1984,8 +1984,10 @@ int err; int was_empty; snd_async_handler_t *h; - err = snd_async_add_handler(&h, _snd_pcm_async_descriptor(pcm), - callback, private_data); + + err = _snd_async_add_handler(&h, _snd_pcm_async_descriptor(pcm), + callback, private_data, + pcm->no_signal_async); if (err < 0) return err; h->type = SND_ASYNC_HANDLER_PCM; diff -r 8b22022a861d src/pcm/pcm_ioplug.c --- a/src/pcm/pcm_ioplug.c Fri May 09 16:02:02 2008 +0200 +++ b/src/pcm/pcm_ioplug.c Tue May 20 09:02:36 2008 +0200 @@ -698,11 +698,15 @@ return 0; } -static int snd_pcm_ioplug_async(snd_pcm_t *pcm ATTRIBUTE_UNUSED, - int sig ATTRIBUTE_UNUSED, - pid_t pid ATTRIBUTE_UNUSED) +static int snd_pcm_ioplug_async(snd_pcm_t *pcm, int sig, pid_t pid) { - return -ENOSYS; + ioplug_priv_t *io = pcm->private_data; + + if (io->data->version >= 0x010002 && + io->data->callback->async) + return io->data->callback->async(io->data, sig, pid); + else + return -ENOSYS; } static int snd_pcm_ioplug_munmap(snd_pcm_t *pcm ATTRIBUTE_UNUSED) @@ -1037,6 +1041,9 @@ ioplug->pcm->poll_events = ioplug->poll_events; ioplug->pcm->monotonic = (ioplug->flags & SND_PCM_IOPLUG_FLAG_MONOTONIC) != 0; ioplug->pcm->mmap_rw = ioplug->mmap_rw; + if (ioplug->version >= 0x010002) + ioplug->pcm->no_signal_async = ioplug->no_signal_async; + return 0; } @@ -1070,3 +1077,16 @@ ioplug->state = state; return 0; } + +/** + * \brief Call the associated async handlers + * \param ioplug the ioplug handle + * + * Call the async handlers associated with the given ioplug. + * This function is usually called from the plugin's async handling + * routine appropriately. + */ +void snd_pcm_ioplug_call_async_handlers(snd_pcm_ioplug_t *ioplug) +{ + _snd_async_call_handlers(ioplug->poll_fd); +} diff -r 8b22022a861d src/pcm/pcm_local.h --- a/src/pcm/pcm_local.h Fri May 09 16:02:02 2008 +0200 +++ b/src/pcm/pcm_local.h Tue May 20 09:02:36 2008 +0200 @@ -220,6 +220,7 @@ * use the mmaped buffer of the slave */ unsigned int donot_close: 1; /* don't close this PCM */ + unsigned int no_signal_async :1; /* async without signal */ snd_pcm_channel_info_t *mmap_channels; snd_pcm_channel_area_t *running_areas; snd_pcm_channel_area_t *stopped_areas;

2 3

Re: [alsa-devel] [Bug #10679] 2.6.26-rc1 regression: e5e1d3cb20034a3cbcfff1f0bae12201aa2ce17e breaks artsd
by Stas Sergeev 22 May '08

22 May '08

Hello. Dave Jones wrote: > More seriously, it doesn't seem that there's any way we can build > this driver modular without it screwing up already working systems Actually, I have just upgraded f8->f9, and guess what? Lost the sound... And not because of snd-pcsp, but rather because snd-hda-intel started to take the first slot and snd-intel8x0 takes the second. On f8 snd-hda-intel have never been loaded for me at all. I don't need it, it finds a HDA chip on an ATI video board, which it never did before. So... it doesn't look like this problem is specific to snd-pcsp. Anyone can get into that trouble it seems.

2 3

[alsa-devel] No audio from laptop speakers, and very faint audio through headphones
by Adam Wiseman 22 May '08

22 May '08

Note: I originally posted my problem on the alsa-user list, but due to the fact that I've got an intel "integrated-everything" chipset, which alsa apparently seems to be having all sorts of trouble with, I was referred to the devel list. So here's my problem: On my Panasonic Toughbook, I get no sound from the laptop speakers, and just barely audible sound from the headphones. I've googled, searched the ubuntu forums, and chatted with a very helpful individual (wishie) on IRC, to no avail. Here is my pastebin of my alsa-info.sh output: http://pastebin.ca/1021657 Here is my pastebin of my amixer output: http://pastebin.ca/1021660 I would be ridiculously grateful if anyone can help me get the sound working on this thing. Thanks, Adam

1 0

[alsa-devel] Asynchronous notification in ALSA - I/O Plugin - Pulseaudio
by Juha Erkkilä 22 May '08

22 May '08

I'm trying to get Abode Flash Player working properly on a thin client, with sound, using Pulseaudio as a sound server. I'm using ALSA version 1.10.15 and Pulseaudio version 0.9.10 on Ubuntu Hardy, with Pulseaudio server running as a system-wide daemon on the thin client. I already outlined the issues on Pulseaudio mailing list on this post: https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008-May/001793.html, see also Lennart Poettering's answer: https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008-May/001796.html. I feel the proper solution would involve writing support for asynchronous notification into ALSA library and I/O-plugin layer. Currently, Pulse I/O-plugin does not work in the Flash case, because Flash requires the asynchronous API. On the other hand, Pulse requires the I/O-plugin layer to work. See the posts by Alexander Indenbaum, http://mailman.alsa-project.org/pipermail/alsa-devel/2008-April/007033.html, and Takashi Iwai, http://mailman.alsa-project.org/pipermail/alsa-devel/2008-April/007335.html, on this subject. I tried to tackle this issue, and did get the signal handling code to call the Flash audio callback, but I'm new to both ALSA and Pulseaudio, and it appears there are some issues with it I don't fully understand. Thus I'm asking: are there any other people who have been looking into this, or anyone who can tell whether this is a sane thing to do or not, will it require some nasty hack, or can help with some pointers to relevant information? I'm hoping that implementing the asynchronous API on I/O-plugin layer would not only fix Flash audio, but might help some other programs too to achieve network transparent audio. Juha

3 5

[alsa-devel] Will ALSA ever call _prepare or _hw_params while it's playing?
by Timur Tabi 21 May '08

21 May '08

I have an ASoC driver with a _hw_params, _prepare, and _trigger functions. I've recently discovered that I need to keep one of my devices disabled until playback starts. So I would like to know if ALSA will ever call _hw_params, _prepare, or _startup if playback or capture is currently in process, or will ALSA first call the _trigger to stop/pause the driver. If ALSA does stop/pause the driver before changing parameters, then I can safely assume the device is always disabled during those functions. -- Timur Tabi Linux kernel developer at Freescale

2 2

[alsa-devel] Possible problem with snd_pcm_oss_sync()
by Timur Tabi 21 May '08

21 May '08

-------- Original Message -------- Subject: RE: Regarding MPC8610 BSP question and issue Date: Tue, 20 May 2008 08:55:46 -0700 From: Hayasaka Manabu <manabu.hayasaka(a)freescale.com> To: Hayasaka Manabu <manabu.hayasaka(a)freescale.com>, Johnson Kelly <Kelly.Johnson(a)freescale.com> CC: sun york <yorksun(a)freescale.com>, Namiki Ayumi <namiki(a)freescale.com>, Beck Glenn <Glenn.Beck(a)freescale.com>, Cox Kevin <K.Cox(a)freescale.com>, Hamano Masahiro <M.Hamano(a)freescale.com>, Tabi Timur <timur(a)freescale.com> References: <E03EE6E38549E949AE8138F60493E1E801C49A0C(a)zja67exm20.fsl.freescale.net> <4807899D.3000901(a)freescale.com> <E03EE6E38549E949AE8138F60493E1E801DA1B11(a)zja67exm20.fsl.freescale.net> <48110FCF.7070302(a)freescale.com> <BD243C8D8F65694A966948E9D51F9815C29409(a)zja67exm21.fsl.freescale.net> <48189540.9060309(a)freescale.com> <BD243C8D8F65694A966948E9D51F9815C29422(a)zja67exm21.fsl.freescale.net> <E03EE6E38549E949AE8138F60493E1E801DA25D9(a)zja67exm20.fsl.freescale.net> <4821CA02.2020504(a)freescale.com> <E03EE6E38549E949AE8138F60493E1E801FA11C8(a)zja67exm20.fsl.freescale.net> <4829C444.7010206(a)freescale.com> <E03EE6E38549E949AE8138F60493E1E801FA13D2(a)zja67exm20.fsl.freescale.net> <482B1D10.7090402(a)freescale.com> <482B2699.8020409(a)freescale.com> I'm working with a customer that has discovered a possible problem with OSS emulation. Unfortunately, I don't have a lot of experience with OSS, so I need help understanding whether the customer's observations point to a real problem or not. The customer is using an old application called "bplay" to test OSS audio support on our hardware. The application works fine on x86, but fails on our PowerPC board. It could be an endian issue, but I doubt it. Here is the forwarded email (slightly edited). Can someone familiar with the OSS "sync" functionality comment? --------------- They reported that it seemed some ioctl commands in the OSS emulation did not work correctly. The tested procedure of the ioctl commands are as following, #1) Stopped audio stream output after doing followings: 1. Opening /dev/mixer 2. Getting a channel value (0x00000001) that is used for "ioctl(fd, SOUND_MIXER_READ_DEVMASK, &Sndsts);". 3. Write audio volume information by "ioctl(fd, SOUND_MIXER_WRITE_VOLUME, &Value);", #2) Stopped audio stream output after doing following: 1. Opening /dev/dsp 2. Executing command "ioctl(fd, SNDCTL_DSP_SYNC, NULL);", XXXX found that by commenting out the "sync_audio" function on the source code of "bplay" made "bplay" work. So XXXX suspects the implementation of the ioctl on the OSS emulation may have a problem. -- Timur Tabi Linux kernel developer at Freescale

3 4

[alsa-devel] AC'97 recording on AT91sam9263
by Sedji Gaouaou 21 May '08

21 May '08

Hi, I am trying to write the AC'97 driver(recording) and I encounter some problems. I would like to set up some constraints but if I do so I will have a seg fault when running arecord. here is my open function: /* define the constraints */ static unsigned int rates[] = { 8000, 44100, }; static struct snd_pcm_hw_constraint_list hw_constraints_rates = { .count = ARRAY_SIZE(rates), .list = rates, .mask = 0, }; ... static int snd_at91_ac97_capture_open(struct snd_pcm_substream *substream) { at91_ac97_t *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; int err; runtime->hw = snd_at91_ac97_capture_hw; chip->capture_substream = substream; chip->period = 0; if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0) return err; snd_printd(KERN_DEBUG "%s : snd_at91_ac97_capture_open\n\r", driver_name); return 0; } I have another problem if I try to set up some constraint on the format. Indeed I do exactly the same as for the rate, but then it will fail in aplay.c in the function set_params, returning the error message:"Broken configuration for this PCM: no configuration available." Do I miss something when trying to set up the constraints? Is there any particular way for setting the format constraints? Regards, Sedji

1 0

[alsa-devel] External Plugin Non-Interleaved
by Charles Eidsness 21 May '08

21 May '08

I'm working on a small ALSA pcm/ctl plugin to add support for LADSPA modules that include real-time controls (unless I'm missing something it looks like the native ALSA LADSPA module doesn't include real-time adjustable controls). I want to use it as a system wide equalizer. I'm using snd_pcm_extplug, I want to force access to be non-interleaved only but I can't figure out how. Is there some way to force non-interleaved access for an extplug? If you're interested you can find the first pass of my plugin here: http://www.thedigitalmachine.net/alsaequal.html Thanks, Charles

2 2

[alsa-devel] Kernel crash in snd-aloop
by Benjamin van den Hout 21 May '08

21 May '08

Hello all, Thanks to the excellent advice here on the list, I'm one step further with getting my project running with the snd-aloop driver! Now I'm looking to scale my solution to more concurrent users. If I check aloop-kernel.c (in the alsa 10.0.16 release tarball), there are two lines that interest me: #define MAX_PCM_SUBSTREAMS 8 and static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8}; The standard aloop driver supplies 8 loopback devices; I need to have more, so I changed both occurences of '8' to 16 and there are no problems. I'm bundling all these subdevices with a 'type multi' in .asoundrc to one device that is used as a JACK backend. So far so good! If I change the 16 to 32 (and modifying .asoundrc to bundle all 32 devices to the 'multi' device) however, restarting JACK will sometimes panic my kernel. I'm running a stock 2.6.22-4 kernel (64 bit) and this is the oops output (redirected to a serial console luckily, otherwise it would be a lot of typing): ----Start oops output---- ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> [<ffffffff80208e47>] mwait_idle+0x42/0x45 [<ffffffff80208d89>] cpu_idle+0x88/0xbf ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> [<ffffffff80208e47>] mwait_idle+0x42/0x45 [<ffffffff80208d89>] cpu_idle+0x88/0xbf ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> [<ffffffff80208e47>] mwait_idle+0x42/0x45 [<ffffffff80208d89>] cpu_idle+0x88/0xbf ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> [<ffffffff80208e47>] mwait_idle+0x42/0x45 [<ffffffff80208d89>] cpu_idle+0x88/0xbf ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> [<ffffffff80208e47>] mwait_idle+0x42/0x45 [<ffffffff80208d89>] cpu_idle+0x88/0xbf ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> [<ffffffff8026669d>] get_page_from_freelist+0x2b/0x534 [<ffffffff80266d0b>] __alloc_pages+0x5a/0x2bc [<ffffffff8026664f>] get_zeroed_page+0x11/0x34 [<ffffffff8026eba6>] __pte_alloc+0x1a/0xb8 [<ffffffff8026ed7f>] __handle_mm_fault+0x13b/0xa35 [<ffffffff80220e9a>] do_page_fault+0x42b/0x7ad [<ffffffff80273f66>] do_mmap_pgoff+0x60a/0x776 [<ffffffff803fe7dd>] error_exit+0x0/0x84 ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> [<ffffffff80208e47>] mwait_idle+0x42/0x45 [<ffffffff80208d89>] cpu_idle+0x88/0xbf ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> [<ffffffff8026e33b>] unmap_vmas+0x3a0/0x741 [<ffffffff8026e26e>] unmap_vmas+0x2d3/0x741 [<ffffffff80272394>] exit_mmap+0x80/0x117 [<ffffffff8022fdfe>] mmput+0x2c/0x9e [<ffffffff8028aee7>] flush_old_exec+0x69d/0x9b6 [<ffffffff8028702b>] vfs_read+0x13f/0x153 [<ffffffff802b60e2>] load_elf_binary+0x462/0x1979 [<ffffffff80244a85>] autoremove_wake_function+0x0/0x2e [<ffffffff80266d0b>] __alloc_pages+0x5a/0x2bc [<ffffffff802b5c80>] load_elf_binary+0x0/0x1979 [<ffffffff8028a135>] search_binary_handler+0xf1/0x27e [<ffffffff8028bbae>] do_execve+0x188/0x231 [<ffffffff80208b6f>] sys_execve+0x36/0x90 [<ffffffff8020a207>] stub_execve+0x67/0xb0 ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> [<ffffffff8026655a>] free_hot_cold_page+0x135/0x167 [<ffffffff802665ad>] __pagevec_free+0x21/0x2e [<ffffffff80268e15>] release_pages+0x123/0x169 [<ffffffff8026945d>] __pagevec_release+0x19/0x22 [<ffffffff80269988>] truncate_inode_pages_range+0xc5/0x28d [<ffffffff8027ffd2>] shmem_delete_inode+0x3f/0xe7 [<ffffffff8027ff93>] shmem_delete_inode+0x0/0xe7 [<ffffffff802991a7>] generic_delete_inode+0xc5/0x13d [<ffffffff80296307>] d_kill+0x38/0x59 [<ffffffff80296435>] dput+0x10d/0x119 [<ffffffff802877dd>] __fput+0x145/0x16e [<ffffffff802d1141>] shm_destroy+0x41/0x5a [<ffffffff802d1e5f>] sys_shmctl+0x49d/0x632 [<ffffffff8023d659>] sys_kill+0x133/0x15c [<ffffffff80286ed7>] vfs_write+0x142/0x157 [<ffffffff802873c1>] sys_write+0x45/0x6e [<ffffffff80209e3e>] system_call+0x7e/0x83 ALSA */home/benjamin/alsa/alsa-driver-1.0.16/acore/*..*/alsa-kernel/core/*pcm_lib.c) Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> [<ffffffff80208e47>] mwait_idle+0x42/0x45 [<ffffffff80208d89>] cpu_idle+0x88/0xbf divide error: 0000 [1] PREEMPT SMP CPU 2 Modules linked in: snd_aloop snd_pcm_oss snd_mixer_oss snd_pcm snd_timer snd snn Pid: 0, comm: swapper Not tainted 2.6.22-4-amd64-preempt-blktrace #1 RIP: 0010:[<ffffffff882e73dd>] [<ffffffff882e73dd>] :snd_aloop:snd_card_loopba0 RSP: 0018:ffff81042faefed0 EFLAGS: 00010046 RAX: 0000000000000000 RBX: ffff810404cd78a8 RCX: 00000001003d172d RDX: 0000000000000000 RSI: 0000000000000217 RDI: 0000000000000001 RBP: ffff810404cd78b0 R08: 0000000000000091 R09: 00000001003d172d R10: 0000000000000000 R11: 0000000000000217 R12: ffff81042fadc000 R13: ffffffff882e7391 R14: 0000000000000000 R15: 0000000000000000 FS: 0000000000000000(0000) GS:ffff81042fa7d440(0000) knlGS:0000000000000000 CS: 0010 DS: 0018 ES: 0018 CR0: 000000008005003b CR2: 00000000005b2b50 CR3: 0000000000201000 CR4: 00000000000006e0 Process swapper (pid: 0, threadinfo ffff81042fae8000, task ffff81042fae75a0) Stack: ffff810404cd78a8 ffff810404cd78a8 0000000000000102 ffffffff8023a720 ffff81042faefef0 ffff81042faefef0 0000012c0000000a 0000000000000021 ffffffff804f1110 000000000000000a 0000000000000002 ffffffff8023723b Call Trace: <IRQ> [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> [<ffffffff80208e47>] mwait_idle+0x42/0x45 [<ffffffff80208d89>] cpu_idle+0x88/0xbf Code: f7 73 64 8b 4b 68 89 93 80 00 00 00 8b 53 7c 39 ca 72 35 89 RIP [<ffffffff882e73dd>] :snd_aloop:snd_card_loopback_timer_function+0x4c/0xa0 RSP <ffff81042faefed0> Kernel panic - not syncing: Aiee, killing interrupt handler! Unable to handle kernel paging request at ffff81042af0c800 RIP: [<ffff81042af0c800>] PGD 8063 PUD 19063 PMD 800000042ae001e3 PTE 0 Oops: 0011 [2] PREEMPT SMP CPU 2 Modules linked in: snd_aloop snd_pcm_oss snd_mixer_oss snd_pcm snd_timer snd snn Pid: 0, comm: swapper Not tainted 2.6.22-4-amd64-preempt-blktrace #1 RIP: 0010:[<ffff81042af0c800>] [<ffff81042af0c800>] RSP: 0018:ffff81042faefb60 EFLAGS: 00010003 RAX: ffff81042fae9fd8 RBX: 0000000000000000 RCX: 0000000000000000 RDX: ffffffff8059e900 RSI: 0000000000000000 RDI: 0000000000000000 RBP: ffff81042faefb90 R08: 0000000000000000 R09: ffff81042af0c800 R10: ffffffff8046d656 R11: 00000000000000fc R12: ffff81042af0c800 R13: 0000000000000000 R14: 000000000000000b R15: 0000000000000000 FS: 0000000000000000(0000) GS:ffff81042fa7d440(0000) knlGS:0000000000000000 CS: 0010 DS: 0018 ES: 0018 CR0: 000000008005003b CR2: ffff81042af0c800 CR3: 0000000000201000 CR4: 00000000000006e0 Process swapper (pid: 0, threadinfo ffff81042fae8000, task ffff81042fae75a0) Stack: ffffffff802189f1 ffffffff8046d636 ffff81042fae75a0 ffffffff80468144 ffffffff8020a9c6 ffff81042faefb90 0000000000000000 00000000000000fc ffffffff8046d656 ffff81042af0c800 0000000000000000 ffffffff8037f3ca Call Trace: <IRQ> [<ffffffff802189f1>] smp_call_function_interrupt+0x40/0x5e [<ffffffff8020a9c6>] call_function_interrupt+0x66/0x70 [<ffffffff8037f3ca>] i8042_panic_blink+0x1/0x147 [<ffffffff80232206>] panic+0x126/0x148 [<ffffffff80232193>] panic+0xb3/0x148 [<ffffffff802298b0>] __wake_up_common+0x3e/0x68 [<ffffffff802352ec>] do_exit+0x8a/0x849 [<ffffffff8020b964>] kernel_math_error+0x0/0x71 [<ffffffff882e7391>] :snd_aloop:snd_card_loopback_timer_function+0x0/0xa0 [<ffffffff8020be38>] do_divide_error+0x7f/0x89 [<ffffffff882e73dd>] :snd_aloop:snd_card_loopback_timer_function+0x4c/0xa0 [<ffffffff882e7391>] :snd_aloop:snd_card_loopback_timer_function+0x0/0xa0 [<ffffffff80232b1f>] printk+0x4e/0x56 [<ffffffff8024d569>] __module_text_address+0x5e/0x68 [<ffffffff803fe7dd>] error_exit+0x0/0x84 [<ffffffff882e7391>] :snd_aloop:snd_card_loopback_timer_function+0x0/0xa0 [<ffffffff882e73dd>] :snd_aloop:snd_card_loopback_timer_function+0x4c/0xa0 [<ffffffff8023a720>] run_timer_softirq+0x15b/0x1cf [<ffffffff8023723b>] __do_softirq+0x49/0xb8 [<ffffffff8020affc>] call_softirq+0x1c/0x28 [<ffffffff8020ca75>] do_softirq+0x2c/0x7d [<ffffffff802371e6>] irq_exit+0x36/0x42 [<ffffffff8021a549>] smp_apic_timer_interrupt+0x49/0x5c [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff80208e05>] mwait_idle+0x0/0x45 [<ffffffff8020aaa6>] apic_timer_interrupt+0x66/0x70 <EOI> [<ffffffff80208e47>] mwait_idle+0x42/0x45 [<ffffffff80208d89>] cpu_idle+0x88/0xbf Code: 00 00 00 00 00 00 00 00 00 a0 f2 2b 04 81 ff ff 00 20 18 29 RIP [<ffff81042af0c800>] RSP <ffff81042faefb60> CR2: ffff81042af0c800 Kernel panic - not syncing: Aiee, killing interrupt handler! ----End oops output---- This crash can be reproduced by automatically restarting jackd in a 'while true' loop and simultaneously killing it every 0.5 seconds from a different terminal. That seems like a nasty thing to do, but I have had it happen some times when jackd was running for a few hours and I needed to restart it for testing something else. Sometimes it takes down the machine, sometimes it doesn't. This constant starting/killing will reproduce it within a minute or so. I'm convinced that jackd should never be able to crash the kernel right? Is there anything I'm missing here by blindly changing the value from 16 to 32, anything else I need to consider? Any input is greatly appreciated. Thanks in advance! Kind regards, Benjamin

1 0

[alsa-devel] [PATCH][ASoC V2] Fix null-pointer dereference in soc_create_pcm()
by Timur Tabi 21 May '08

21 May '08

When soc_create_pcm() finds all four components it needs, it displays a status message. This message references the local variables 'codec_dai' and 'cpu_dai', but these variables are initialized only when the codec DAI and the cpu DAI are the last to be registered. This patch changes the code to use the stored values instead of the local variables. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/soc-core.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7dc6d68..520469b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -921,8 +921,8 @@ check: } pcm->private_free = platform->pcm_free; - printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, - cpu_dai->name); + printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", + pcm_runtime->codec_dai->name, pcm_runtime->cpu_dai->name); list_add(&pcm_runtime->list, &soc_card->pcm_list); return 1; err: -- 1.5.5

3 4

[alsa-devel] MIDI on ice1724 - preliminary findings and questions
by Pavel Hofman 20 May '08

20 May '08

Hi, I am trying to fix the ancient MIDI problem with ice1724 cards. I applied Takashi's patch from http://mailman.alsa-project.org/pipermail/alsa-devel/2007-April/000641.html and made some minor changes (MPU401_INFO_INPUT | MPU401_INFO_OUTPUT to mpu401_uart info_flags, added a few debugs for testing). I can test only MIDI input using my MIDI keyboard, I have no MIDI output device available. The input works fine now, the output most probably too since the method snd_vt1724_mpu401_write gets called when transmitting some data with amidi. But here is the problem I do not understand: When no MIDI input/output is used, the interrupt handler snd_vt1724_interrupt gets called with interrupt status 0x10 which correctly refers to the audio data interrupt (VT1724_IRQ_MTPCM). However, when MIDI input or output is used (opened) even for a short time after the module is loaded, any subsequent audio playback generates interrupt status of 0x30 which refers to VT1724_IRQ_MTPCM AND VT1724_IRQ_MPU_TX. I have no idea why VT1724_IRQ_MPU_TX gets generated. The first run of the snd_vt1724_interrupt loop tries to handle the MPU TX status, calling uart_interrupt_tx in mpu401_uart.c: if (test_bit(MPU401_MODE_BIT_OUTPUT, &mpu->mode) && test_bit(MPU401_MODE_BIT_OUTPUT_TRIGGER, &mpu->mode)) { spin_lock_irqsave(&mpu->output_lock, flags); snd_mpu401_uart_output_write(mpu); spin_unlock_irqrestore(&mpu->output_lock, flags); } However, since the MPU output is not open at this time (no MIDI is being transmitted), the call to snd_mpu401_uart_output_write is skipped. Even though I think the interrupt status should get cleared by outb(status, ICEREG1724(ice, IRQSTAT)) the loop repeats with the MPU TX interrupt status enabled until the timeout check breaks the while loop, returning to the interrupt handler the next moment again. Apr 19 22:20:47 nahore kernel: [ 2402.938245] uart interrupt: status 0x30 Apr 19 22:20:47 nahore kernel: [ 2402.938260] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938267] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938274] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938281] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938288] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938295] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938302] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938309] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938316] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938323] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.938326] ice1724: Too long irq loop, status = 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961247] uart interrupt: status 0x30 Apr 19 22:20:47 nahore kernel: [ 2402.961278] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961285] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961292] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961299] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961306] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961313] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961320] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961327] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961334] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961341] uart interrupt: status 0x20 Apr 19 22:20:47 nahore kernel: [ 2402.961344] ice1724: Too long irq loop, status = 0x20 After the playback stops, the interrupts are gone too. It seems as if the playback interrupt initiates the MPU TX interrupt. If we could avoid generating the MPU TX interrupt during regular playback, I believe the major problem would be resolved. Even if I mask the interrupts (CCS01) and do not explicitly unmask them (according to proc ice1724 the CCS01 register stays at 0xa0), the interrupt gets generated. I am enclosing the testing patch producing the above given results. Thanks a lot for any suggestion or help. Regards, Pavel.

3 19

[alsa-devel] Addition to soundcard vendor matrix
by Joshua Boyd 20 May '08

20 May '08

http://www.alsa-project.org/main/index.php/Matrix:Main Could someone with proper permissions add Marian (their web site is www.marian.de) to the vendor section? While there is no support in ALSA for any of their stuff yet, there is some information that seems like it would be worth adding, plus there is a guy working on a driver for one of their cards.

2 1

[alsa-devel] ca0106 Digital Output at 44.1kHz
by Ben Stanley 20 May '08

20 May '08

Dear List, Hello. I'm new here. This is my first attempt to hack an alsa driver. I recently purchased a "Sound Blaster Audigy Value!" card for use in a MythTV box. As I have a digital receiver, I was mainly concerned with getting a working S/PDIF output (coax or optical) and not too much more. Summary: I can make 44.1kHz digital output work, but I think that the driver model of the card having 3 or 4 digital output channels is incorrect. I believe the card only has 1 digital output channel. Details: The card I have is characterised by the following: http://www.soundblaster.com/products/product.asp?category=1&subcategory=205… Model: SB0570 serial: 100a1102 Also known as: Sound Blaster Audigy SE During initial testing, I noticed that 44.1kHz playback was not implemented. Subsequent testing shows that speaker-test works fine with 48kHz, 96kHz and 192kHz. (I only tested 16bit output so far.) I noticed in the source code that 44.1k was explicitly disabled. I added code in snd_ca0106_pcm_prepare_playback to set up this rate for S/PDIF output as per the comments in ca0106.h . Initial tests using hw:0,0 at 44.1kHz produced recognisable signals with some noisy corruption. Later I accidentally discovered that serially opening hw:0,2 , hw:0,1 and hw:0,0 at 44.1kHz then produces perfectly good 44.1kHz sampled digital audio output. Removing hw:0,1 or hw:0,2 from this sequence causes noisy corruption. It seems that channels 0-2 in reg40 must all be set to the same sampling frequency for S/PDIF to work where 44.1kHz is concerned. Conversely, to sucessfully output 48kHz again, I have to open hw:0,2 , hw:0,1 and hw:0,0 at 48kHz to restore proper output. I do not have such troubles with 96kHz and 192kHz, for which it suffices to just open hw:0,0 at the relevant sampling rate. Using these tricks, I can successfully output 44.1kHz encoded Dolby Digital and DTS signals (found at http://www.sr.se/cgi-bin/mall/index.asp?programid=2445 , as mentioned at http://alsa.opensrc.org/DigitalOut . ) Later, I examined the product manual (from a Windows compressed helpfile on the CD), and noticed that Creative instruct you to connect to the blue socket in only two ways: 1) with a mono 3.5mm jack plug to RCA plug -> S/PDIF input to external device, and 2) a proprietary digital I/O module (see http://www.soundblaster.com/products/product.asp?category=1&subcategory=16&… ) The digital I/O module only has one input and one output (the optical/coaxial pairs are duplicates of each other). The manual does not say how to connect a S/PDIF input signal other than through the digital I/O module. This suggests to me that the following comment near the top of ca0106_main.c is incorrect for the SB0570: * ( In theory one could output 3 different AC3 streams at once, to 3 different SPDIF outputs. ) I think you can only output one AC3 stream, and that there is only one S/PDIF output. Given this knowledge of the digital I/O module, I suggest that the pinout *may* possibly be: 1 (tip): S/PDIF output 2 : S/PDIF input 3 : power to digital I/O module 4 : gnd I have not been able to get any S/PDIF signal out of pin 2 (I only have a stereo cable to test with at the moment). I will attempt to test pin 2 as an input at some point. I think that this card can only output one S/PDIF channel, given the discovery of the required clock settings above, and that Creative's own hardware only gives you access to one output. Anyway, I don't have access to documentation. From reading around, I believe James has the docs under NDA from Creative Labs. Perhaps James could confirm these suspicions? Anyway, this is speculation. I'd love to have the docs. I haven't tried to get them. Is it likely/unlikely that I would get them? I'm currently working on kernel 2.6.22 from Ubuntu. They don't seem to have changed the ca0106 driver, and not much has happened there (for output) in the newer kernels. Please find attached my current patch, which is only suitable for the purposes of discussion. I need to clarify some things about channel arrangements with those who have the documentation before I begin to try to fix things up. This patch was written with a faulty understanding of the interaction between the channel model and the spdif_enable. I don't know how any of this affects other models in the ca0106 range. So, I think that before I go any further here, I need some feedback, and access to some documentation would be useful. Ben Stanley.

2 4

[alsa-devel] [PATCH] alsa-info.sh - Add another distro to search for
by Travis Place 20 May '08

20 May '08

Just allows finding of "Pardus" distro. diff -r e167861901c7 alsa-info.sh --- a/alsa-info.sh Tue Feb 26 10:55:47 2008 +0100 +++ b/alsa-info.sh Tue Apr 29 00:09:20 2008 +1000 @@ -1,6 +1,6 @@ #!/bin/bash -SCRIPT_VERSION=0.4.41 +SCRIPT_VERSION=0.4.42 CHANGELOG="http://hg.alsa-project.org/alsa/log/tip/alsa-info.sh" ################################################################################# @@ -270,7 +270,7 @@ mkdir $TEMPDIR 2>/dev/null mkdir $TEMPDIR 2>/dev/null #Fetch the info and store in temp files/variables -DISTRO=`grep -ihs "buntu\|SUSE\|Fedora\|PCLinuxOS\|MEPIS\|Mandriva\|Debian\| Damn\|Sabayon\|Slackware\|KNOPPIX\|Gentoo\|Zenwalk\|Mint\|Kubuntu\|FreeBSD\| Puppy\|Freespire\|Vector\|Dreamlinux\|CentOS\|Arch\|Xandros\|Elive\|SLAX\| Red\|BSD\|KANOTIX\|Nexenta\|Foresight\|GeeXboX\|Frugalware\|64\| SystemRescue\|Novell\|Solaris\|BackTrack\| KateOS" /etc/{issue,*release,*version}` +DISTRO=`grep -ihs "buntu\|SUSE\|Fedora\|PCLinuxOS\|MEPIS\|Mandriva\|Debian\| Damn\|Sabayon\|Slackware\|KNOPPIX\|Gentoo\|Zenwalk\|Mint\|Kubuntu\|FreeBSD\| Puppy\|Freespire\|Vector\|Dreamlinux\|CentOS\|Arch\|Xandros\|Elive\|SLAX\| Red\|BSD\|KANOTIX\|Nexenta\|Foresight\|GeeXboX\|Frugalware\|64\| SystemRescue\|Novell\|Solaris\|BackTrack\|KateOS\| Pardus" /etc/{issue,*release,*version}` KERNEL_VERSION=`uname -r` KERNEL_PROCESSOR=`uname -p` KERNEL_MACHINE=`uname -m` -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

3 6

[alsa-devel] ALSA fixes for non-coherent archs (Re: [PATCH] Sam440ep support)
by Takashi Iwai 20 May '08

20 May '08

At Tue, 06 May 2008 11:16:22 +0200, Gerhard Pircher wrote: > > -------- Original-Nachricht -------- > > Datum: Tue, 06 May 2008 18:48:39 +1000 > > Von: Benjamin Herrenschmidt <benh(a)kernel.crashing.org> > > An: Gerhard Pircher <gerhard_pircher(a)gmx.net> > > CC: linuxppc-dev(a)ozlabs.org, Takashi Iwai <tiwai(a)suse.de>, cjg(a)cruxppc.org, galak(a)kernel.crashing.org > > Betreff: Re: [PATCH] Sam440ep support > > > > > On Tue, 2008-05-06 at 09:51 +0200, Gerhard Pircher wrote: > > > Takashi Iwai posted a preliminary patch a long time ago. I tested it > > > on my machine and it failed with non coherent scatter-gather DMA > > > allocations (I guess almost all ALSA PCI drivers use SG DMA?). > > > > How does Alsa allocate such SG ? > I can't answer this question. *ducked* :-) Takashi? > > FYI: I posted the results of the test with Takashi's dma_mmap_coherent > patch here: > http://ozlabs.org/pipermail/linuxppc-dev/2006-June/024078.html > > On the other side it looks like this problem does not only affect ALSA. > As far as I can tell also some V4L(2) drivers have a problem with mmaping > non coherent DMA allocations, but I'm not sure (it's a long time since I > did some tests with video cards on my AmigaOne). > > Naturally I can do some tests, if you or Takashi come up with a new > patch. OK, here is another patch for testing. Since I lost my old patch somewhere (and it's not worth to dig the archive), I wrote it up quickly from scratch. This version should cover both SG and non-SG buffers. It's against the latest git tree. The patch adds a hackish verison of dma_mmap_coherent() for some architectures in sound/core/pcm_native.c. I'm not sure whether this works. I just tested it on X86. It'd be appreciated if someone can test this. Also, this disables HDSPM driver for non-X86/IA64 since the driver has own copy and silence methods that are incompatible with the new SG-buffer data. And, yes, I know we need to clean up huge messes in ALSA memory handling routines. But, let's fix obvious bugs before starting a big rewrite... thanks, Takashi --- diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index ae2921d..ccf3dfa 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -62,6 +62,18 @@ struct snd_dma_buffer { void *private_data; /* private for allocator; don't touch */ }; +/* needs to use dma_mmap_coherent() for pages allocated via + * dma_alloc_coherent() + */ +#ifdef CONFIG_HAS_DMA +#if (defined(CONFIG_PPC32) && !defined(CONFIG_CONFIG_NOT_COHERENT_CACHE)) || \ + defined(CONFIG_ARM) || \ + defined(CONFIG_MIPS) || \ + defined(CONFIG_PARISC) +#define SND_NEEDS_DMA_MMAP_COHERENT +#endif /* archs */ +#endif /* CONFIG_HAS_DMA */ + /* * Scatter-Gather generic device pages */ @@ -75,7 +87,9 @@ struct snd_sg_buf { int pages; /* allocated pages */ int tblsize; /* allocated table size */ struct snd_sg_page *table; /* address table */ +#ifndef SND_NEEDS_DMA_MMAP_COHERENT struct page **page_table; /* page table (for vmap/vunmap) */ +#endif struct device *dev; }; diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 51d58cc..2e68420 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -969,10 +969,25 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size); int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream); +/* + * SG-buffer + */ #define snd_pcm_substream_sgbuf(substream) ((substream)->runtime->dma_buffer_p->private_data) #define snd_pcm_sgbuf_pages(size) snd_sgbuf_aligned_pages(size) #define snd_pcm_sgbuf_get_addr(sgbuf,ofs) snd_sgbuf_get_addr(sgbuf,ofs) + +#ifdef SND_NEEDS_DMA_MMAP_COHERENT +int snd_pcm_sgbuf_ops_copy(struct snd_pcm_substream *substream, + int channel, snd_pcm_uframes_t pos, + void __user *buf, snd_pcm_uframes_t count); +int snd_pcm_sgbuf_ops_silence(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, snd_pcm_uframes_t count); +#define snd_pcm_sgbuf_ops_page NULL +#else +#define snd_pcm_sgbuf_ops_copy NULL +#define snd_pcm_sgbuf_ops_silence NULL struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, unsigned long offset); +#endif /* handle mmap counter - PCM mmap callback should handle this counter properly */ static inline void snd_pcm_mmap_data_open(struct vm_area_struct *area) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index ff07b4a..9e57032 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -306,6 +306,126 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages_for_all); +#ifdef SND_NEEDS_DMA_MMAP_COHERENT +/* + * snd_pcm_sgbuf_ops_copy - copy callback for DMA SG-buffer + */ +int snd_pcm_sgbuf_ops_copy(struct snd_pcm_substream *substream, + int channel, snd_pcm_uframes_t pos, + void __user *buf, snd_pcm_uframes_t count) +{ + struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int ofs, idx; + char *ptr; + + if (channel < 0) { + pos = frames_to_bytes(runtime, pos); + count = frames_to_bytes(runtime, count); + } else { + pos = channel * (runtime->dma_bytes / runtime->channels) + + samples_to_bytes(runtime, pos); + count = samples_to_bytes(runtime, count); + } + + idx = pos >> PAGE_SHIFT; + ofs = pos & (PAGE_SIZE - 1); + ptr = sgbuf->table[idx].buf + ofs; + + for (;;) { + unsigned int size, ret; + size = count; + if (ofs + size > PAGE_SIZE) + size = PAGE_SIZE - ofs; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = copy_from_user(ptr, buf, size); + else + ret = copy_to_user(buf, ptr, size); + if (ret) + return -EFAULT; + count -= size; + if (!count) + return 0; + ofs = 0; + idx++; + ptr = sgbuf->table[idx].buf; + buf += size; + } +} +EXPORT_SYMBOL(snd_pcm_sgbuf_ops_copy); + +/* + * snd_pcm_sgbuf_ops_silence - fill with silence data + */ +int snd_pcm_sgbuf_ops_silence(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, snd_pcm_uframes_t count) +{ + struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int idx, ofs, width; + const char *silence; + char *ptr; + + if (channel < 0) { + pos = frames_to_bytes(runtime, pos); + count = frames_to_bytes(runtime, count); + } else { + pos = channel * (runtime->dma_bytes / runtime->channels) + + samples_to_bytes(runtime, pos); + count = samples_to_bytes(runtime, count); + } + + idx = pos >> PAGE_SHIFT; + if (idx >= (unsigned int)sgbuf->pages) + return -EFAULT; + ofs = pos & (PAGE_SIZE - 1); + ptr = sgbuf->table[idx].buf + ofs; + + width = snd_pcm_format_physical_width(runtime->format); + if (width < 8) { + count /= 2; + width = 8; + } + width /= 8; + silence = snd_pcm_format_silence_64(runtime->format); + + if (PAGE_SIZE % width) { + unsigned int pat = 0; + for (;;) { + *ptr = silence[pat]; + pat = (pat + 1) % width; + count--; + if (!count) + return 0; + ofs++; + if (ofs == PAGE_SIZE) { + ofs = 0; + idx++; + ptr = sgbuf->table[idx].buf; + } else + ptr++; + } + } else { + for (;;) { + unsigned int size, samples; + size = count; + if (ofs + size > PAGE_SIZE) + size = PAGE_SIZE - ofs; + samples = bytes_to_samples(runtime, size); + snd_pcm_format_set_silence(runtime->format, ptr, + samples); + count -= size; + if (!count) + return 0; + ofs = 0; + idx++; + ptr = sgbuf->table[idx].buf; + } + } +} +EXPORT_SYMBOL(snd_pcm_sgbuf_ops_silence); + +#else /* !SND_NEEDS_DMA_MMAP_COHERENT */ /** * snd_pcm_sgbuf_ops_page - get the page struct at the given offset * @substream: the pcm substream instance @@ -323,9 +443,10 @@ struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, unsigne return NULL; return sgbuf->page_table[idx]; } - EXPORT_SYMBOL(snd_pcm_sgbuf_ops_page); +#endif /* SND_NEEDS_DMA_MMAP_COHERENT */ + /** * snd_pcm_lib_malloc_pages - allocate the DMA buffer * @substream: the substream to allocate the DMA buffer to diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 61f5d42..e3f68d2 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3124,19 +3124,96 @@ static struct vm_operations_struct snd_pcm_vm_ops_data = { .open = snd_pcm_mmap_data_open, .close = snd_pcm_mmap_data_close, +}; + +static struct vm_operations_struct snd_pcm_vm_ops_data_fault = +{ + .open = snd_pcm_mmap_data_open, + .close = snd_pcm_mmap_data_close, .fault = snd_pcm_mmap_data_fault, }; +#ifdef SND_NEEDS_DMA_MMAP_COHERENT + +/* + * FIXME!! + * dma_mmap_coherent is missing on most architectures... + */ +#ifndef CONFIG_ARM +static int dma_mmap_coherent(struct device *dev, struct vm_area_struct *vma, + void *cpu_addr, dma_addr_t handle, size_t size) +{ +#if defined(CONFIG_PPC32) && !defined(CONFIG_CONFIG_NOT_COHERENT_CACHE) + cpu_addr = bus_to_virt(handle); +#elif defined(CONFIG_MIPS) + cpu_addr = phys_to_virt(plat_dma_addr_to_phys(handle)); +#elif defined(CONFIG_PARISC) + cpu_addr = __va(handle); +#endif + return remap_pfn_range(vma, vma->vm_start, + page_to_pfn(virt_to_page(cpu_addr)), + size, vma->vm_page_prot); +} +#endif /* !ARM */ + +/* + * snd_pcm_sgbuf_ops_mmap - mmap SG DMA pages + */ +static int snd_pcm_mmap_sgbuf(struct snd_pcm_substream *substream, + struct vm_area_struct *area) +{ + struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream); + unsigned long start, offset, offset_saved, size; + int err = 0; + + start = area->vm_start; + offset_saved = offset = area->vm_pgoff; + size = area->vm_end - area->vm_start; + size = PAGE_ALIGN(size); + while (size > 0) { + if (offset >= sgbuf->pages) { + err = -EFAULT; + break; + } + err = dma_mmap_coherent(sgbuf->dev, area, + sgbuf->table[offset].buf, + sgbuf->table[offset].addr, + PAGE_SIZE); + if (err < 0) + break; + offset++; + area->vm_start += PAGE_SIZE; + size -= PAGE_SIZE; + } + area->vm_start = start; + area->vm_pgoff = offset_saved; + return err; +} +#endif /* SND_NEEDS_DMA_MMAP_COHERENT */ + /* * mmap the DMA buffer on RAM */ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { - area->vm_ops = &snd_pcm_vm_ops_data; - area->vm_private_data = substream; area->vm_flags |= VM_RESERVED; - atomic_inc(&substream->mmap_count); +#ifdef SND_NEEDS_DMA_MMAP_COHERENT + if (!substream->ops->page) { + switch (substream->dma_buffer.dev.type) { + case SNDRV_DMA_TYPE_DEV: + return dma_mmap_coherent(substream->dma_buffer.dev.dev, + area, + substream->runtime->dma_area, + substream->runtime->dma_addr, + area->vm_end - area->vm_start); + case SNDRV_DMA_TYPE_DEV_SG: + return snd_pcm_mmap_sgbuf(substream, area); + } + } +#endif /* SND_NEEDS_DMA_MMAP_COHERENT */ + /* mmap with fault handler */ + area->vm_ops = &snd_pcm_vm_ops_data_fault; return 0; } @@ -3144,12 +3221,6 @@ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, * mmap the DMA buffer on I/O memory area */ #if SNDRV_PCM_INFO_MMAP_IOMEM -static struct vm_operations_struct snd_pcm_vm_ops_data_mmio = -{ - .open = snd_pcm_mmap_data_open, - .close = snd_pcm_mmap_data_close, -}; - int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_struct *area) { @@ -3159,8 +3230,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, #ifdef pgprot_noncached area->vm_page_prot = pgprot_noncached(area->vm_page_prot); #endif - area->vm_ops = &snd_pcm_vm_ops_data_mmio; - area->vm_private_data = substream; area->vm_flags |= VM_IO; size = area->vm_end - area->vm_start; offset = area->vm_pgoff << PAGE_SHIFT; @@ -3168,7 +3237,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, (substream->runtime->dma_addr + offset) >> PAGE_SHIFT, size, area->vm_page_prot)) return -EAGAIN; - atomic_inc(&substream->mmap_count); return 0; } @@ -3185,6 +3253,7 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, long size; unsigned long offset; size_t dma_bytes; + int err; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (!(area->vm_flags & (VM_WRITE|VM_READ))) @@ -3210,10 +3279,15 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, if (offset > dma_bytes - size) return -EINVAL; + area->vm_ops = &snd_pcm_vm_ops_data; + area->vm_private_data = substream; if (substream->ops->mmap) - return substream->ops->mmap(substream, area); + err = substream->ops->mmap(substream, area); else - return snd_pcm_default_mmap(substream, area); + err = snd_pcm_default_mmap(substream, area); + if (!err) + atomic_inc(&substream->mmap_count); + return err; } EXPORT_SYMBOL(snd_pcm_mmap_data); diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index cefd228..67c6631 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -46,12 +46,14 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab) tmpb.bytes = PAGE_SIZE; snd_dma_free_pages(&tmpb); } +#ifndef SND_NEEDS_DMA_MMAP_COHERENT if (dmab->area) vunmap(dmab->area); + kfree(sgbuf->page_table); +#endif dmab->area = NULL; kfree(sgbuf->table); - kfree(sgbuf->page_table); kfree(sgbuf); dmab->private_data = NULL; @@ -77,9 +79,11 @@ void *snd_malloc_sgbuf_pages(struct device *device, sgbuf->table = kcalloc(sgbuf->tblsize, sizeof(*sgbuf->table), GFP_KERNEL); if (! sgbuf->table) goto _failed; +#ifndef SND_NEEDS_DMA_MMAP_COHERENT sgbuf->page_table = kcalloc(sgbuf->tblsize, sizeof(*sgbuf->page_table), GFP_KERNEL); if (! sgbuf->page_table) goto _failed; +#endif /* allocate each page */ for (i = 0; i < pages; i++) { @@ -91,14 +95,20 @@ void *snd_malloc_sgbuf_pages(struct device *device, } sgbuf->table[i].buf = tmpb.area; sgbuf->table[i].addr = tmpb.addr; +#ifndef SND_NEEDS_DMA_MMAP_COHERENT sgbuf->page_table[i] = virt_to_page(tmpb.area); +#endif sgbuf->pages++; } sgbuf->size = size; +#ifndef SND_NEEDS_DMA_MMAP_COHERENT dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, PAGE_KERNEL); if (! dmab->area) goto _failed; +#else + dmab->area = sgbuf->table[0].buf; +#endif return dmab->area; _failed: diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7e47421..09af3f5 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -651,6 +651,7 @@ config SND_HDSP config SND_HDSPM tristate "RME Hammerfall DSP MADI" depends on SND + depends on X86 || ALPHA || IA64 select SND_HWDEP select SND_RAWMIDI select SND_PCM diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index f9a58b4..222e599 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -409,6 +409,8 @@ static struct snd_pcm_ops snd_vortex_playback_ops = { .prepare = snd_vortex_pcm_prepare, .trigger = snd_vortex_pcm_trigger, .pointer = snd_vortex_pcm_pointer, + .copy = snd_pcm_sgbuf_ops_copy, + .silence = snd_pcm_sgbuf_ops_silence, .page = snd_pcm_sgbuf_ops_page, }; diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 4ecdd63..49b8ebc 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -548,6 +548,7 @@ static struct snd_pcm_ops snd_bt87x_pcm_ops = { .prepare = snd_bt87x_prepare, .trigger = snd_bt87x_trigger, .pointer = snd_bt87x_pointer, + .copy = snd_pcm_sgbuf_ops_copy, .page = snd_pcm_sgbuf_ops_page, }; diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index e16dc92..e2c3f3d 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -801,6 +801,8 @@ static struct snd_pcm_ops analog_playback_ops = { .prepare = pcm_prepare, .trigger = pcm_trigger, .pointer = pcm_pointer, + .copy = snd_pcm_sgbuf_ops_copy, + .silence = snd_pcm_sgbuf_ops_silence, .page = snd_pcm_sgbuf_ops_page, }; static struct snd_pcm_ops analog_capture_ops = { @@ -812,6 +814,7 @@ static struct snd_pcm_ops analog_capture_ops = { .prepare = pcm_prepare, .trigger = pcm_trigger, .pointer = pcm_pointer, + .copy = snd_pcm_sgbuf_ops_copy, .page = snd_pcm_sgbuf_ops_page, }; #ifdef ECHOCARD_HAS_DIGITAL_IO @@ -825,6 +828,8 @@ static struct snd_pcm_ops digital_playback_ops = { .prepare = pcm_prepare, .trigger = pcm_trigger, .pointer = pcm_pointer, + .copy = snd_pcm_sgbuf_ops_copy, + .silence = snd_pcm_sgbuf_ops_silence, .page = snd_pcm_sgbuf_ops_page, }; #endif /* !ECHOCARD_HAS_VMIXER */ @@ -837,6 +842,7 @@ static struct snd_pcm_ops digital_capture_ops = { .prepare = pcm_prepare, .trigger = pcm_trigger, .pointer = pcm_pointer, + .copy = snd_pcm_sgbuf_ops_copy, .page = snd_pcm_sgbuf_ops_page, }; #endif /* ECHOCARD_HAS_DIGITAL_IO */ diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index cf9276d..2d72959 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1323,6 +1323,8 @@ static struct snd_pcm_ops snd_emu10k1_playback_ops = { .prepare = snd_emu10k1_playback_prepare, .trigger = snd_emu10k1_playback_trigger, .pointer = snd_emu10k1_playback_pointer, + .copy = snd_pcm_sgbuf_ops_copy, + .silence = snd_pcm_sgbuf_ops_silence, .page = snd_pcm_sgbuf_ops_page, }; @@ -1347,6 +1349,8 @@ static struct snd_pcm_ops snd_emu10k1_efx_playback_ops = { .prepare = snd_emu10k1_efx_playback_prepare, .trigger = snd_emu10k1_efx_playback_trigger, .pointer = snd_emu10k1_efx_playback_pointer, + .copy = snd_pcm_sgbuf_ops_copy, + .silence = snd_pcm_sgbuf_ops_silence, .page = snd_pcm_sgbuf_ops_page, }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b3a618e..e5ac9e1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1461,6 +1461,8 @@ static struct snd_pcm_ops azx_pcm_ops = { .prepare = azx_pcm_prepare, .trigger = azx_pcm_trigger, .pointer = azx_pcm_pointer, + .copy = snd_pcm_sgbuf_ops_copy, + .silence = snd_pcm_sgbuf_ops_silence, .page = snd_pcm_sgbuf_ops_page, }; diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 979f7da..7fcb3d7 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1696,6 +1696,8 @@ static struct snd_pcm_ops snd_riptide_playback_ops = { .hw_params = snd_riptide_hw_params, .hw_free = snd_riptide_hw_free, .prepare = snd_riptide_prepare, + .copy = snd_pcm_sgbuf_ops_copy, + .silence = snd_pcm_sgbuf_ops_silence, .page = snd_pcm_sgbuf_ops_page, .trigger = snd_riptide_trigger, .pointer = snd_riptide_pointer, @@ -1707,6 +1709,7 @@ static struct snd_pcm_ops snd_riptide_capture_ops = { .hw_params = snd_riptide_hw_params, .hw_free = snd_riptide_hw_free, .prepare = snd_riptide_prepare, + .copy = snd_pcm_sgbuf_ops_copy, .page = snd_pcm_sgbuf_ops_page, .trigger = snd_riptide_trigger, .pointer = snd_riptide_pointer, diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index bbcee2c..156e457 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -2081,6 +2081,8 @@ static struct snd_pcm_ops snd_trident_nx_playback_ops = { .prepare = snd_trident_playback_prepare, .trigger = snd_trident_trigger, .pointer = snd_trident_playback_pointer, + .copy = snd_pcm_sgbuf_ops_copy, + .silence = snd_pcm_sgbuf_ops_silence, .page = snd_pcm_sgbuf_ops_page, }; @@ -2126,6 +2128,8 @@ static struct snd_pcm_ops snd_trident_nx_foldback_ops = { .prepare = snd_trident_foldback_prepare, .trigger = snd_trident_trigger, .pointer = snd_trident_playback_pointer, + .copy = snd_pcm_sgbuf_ops_copy, + .silence = snd_pcm_sgbuf_ops_silence, .page = snd_pcm_sgbuf_ops_page, }; diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index b585cc3..4a9f132 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1304,6 +1304,8 @@ static struct snd_pcm_ops snd_via686_playback_ops = { .prepare = snd_via686_playback_prepare, .trigger = snd_via82xx_pcm_trigger, .pointer = snd_via686_pcm_pointer, + .copy = snd_pcm_sgbuf_ops_copy, + .silence = snd_pcm_sgbuf_ops_silence, .page = snd_pcm_sgbuf_ops_page, }; @@ -1317,6 +1319,7 @@ static struct snd_pcm_ops snd_via686_capture_ops = { .prepare = snd_via686_capture_prepare, .trigger = snd_via82xx_pcm_trigger, .pointer = snd_via686_pcm_pointer, + .copy = snd_pcm_sgbuf_ops_copy, .page = snd_pcm_sgbuf_ops_page, }; @@ -1330,6 +1333,8 @@ static struct snd_pcm_ops snd_via8233_playback_ops = { .prepare = snd_via8233_playback_prepare, .trigger = snd_via82xx_pcm_trigger, .pointer = snd_via8233_pcm_pointer, + .copy = snd_pcm_sgbuf_ops_copy, + .silence = snd_pcm_sgbuf_ops_silence, .page = snd_pcm_sgbuf_ops_page, }; @@ -1343,6 +1348,8 @@ static struct snd_pcm_ops snd_via8233_multi_ops = { .prepare = snd_via8233_multi_prepare, .trigger = snd_via82xx_pcm_trigger, .pointer = snd_via8233_pcm_pointer, + .copy = snd_pcm_sgbuf_ops_copy, + .silence = snd_pcm_sgbuf_ops_silence, .page = snd_pcm_sgbuf_ops_page, }; @@ -1356,6 +1363,7 @@ static struct snd_pcm_ops snd_via8233_capture_ops = { .prepare = snd_via8233_capture_prepare, .trigger = snd_via82xx_pcm_trigger, .pointer = snd_via8233_pcm_pointer, + .copy = snd_pcm_sgbuf_ops_copy, .page = snd_pcm_sgbuf_ops_page, }; diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 31f64ee..1431b08 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -803,6 +803,8 @@ static struct snd_pcm_ops snd_via686_playback_ops = { .prepare = snd_via82xx_pcm_prepare, .trigger = snd_via82xx_pcm_trigger, .pointer = snd_via686_pcm_pointer, + .copy = snd_pcm_sgbuf_ops_copy, + .silence = snd_pcm_sgbuf_ops_silence, .page = snd_pcm_sgbuf_ops_page, }; @@ -816,6 +818,7 @@ static struct snd_pcm_ops snd_via686_capture_ops = { .prepare = snd_via82xx_pcm_prepare, .trigger = snd_via82xx_pcm_trigger, .pointer = snd_via686_pcm_pointer, + .copy = snd_pcm_sgbuf_ops_copy, .page = snd_pcm_sgbuf_ops_page, };

3 5

[alsa-devel] [PATCH] snd-usb-us122l v0.2 for US-122L
by Karsten Wiese 20 May '08

20 May '08

Hi, attached is a fresh US-122L linux(test-)driver. If everything is installed on top a kernel 2.6.25-rc5 and alsa-lib 1.0.16, everything works, erm at least here. Without the ehci patch, it still works when connected to the uhci-hcd. (Maybe the ehci-patch makes it into .26) alsa-lib < 1.0.16 lets the plugin misbehave. PCM access needs an ~/.asoundrc containing whats in .asoundrc. Its established by patching/compiling/installing alsa-plugins. You need to adjust the path to the kernel's sound/usb/usx2y/usb_stream.h Only S24_3LE, 2 channels interleaved, 2 periods is supported. The userspace pcm devicename then is forexample "usb_stream:1" (To let alsa convert to other formats, use forexample "plug:usb_stream:1") Tested with aplay, arecord, amidi & jackd using alsa driver. Propably doesn't do 32bit PCM access on x86_64, tested on x86_64 only. Karsten

2 19

Re: [alsa-devel] problems compiling alsa-drivers-1.0.16 into 2.6.25.4 kernel
by Takashi Iwai 20 May '08

20 May '08

At Tue, 20 May 2008 03:03:44 +0200, Dragan Noveski wrote: > > hallo list, > steven rostedt asked me to pass this problem to this list, so here it goes. No, it should be alsa-devel ML. > when running make in the alsa-driver source i get this error: > > ..... > make[1]: Entering directory `/usr/src/linux-2.6.25' ALSA 1.0.16 doesn't work for 2.6.25 kernel. Get the latest HG version. Takashi

2 1

[alsa-devel] Fwd: Re: Incorrect model selection for HDA Intel based card.
by Travis Place 19 May '08

19 May '08

Below (as well as attached) is my patch, for the 0x814e variant of the ASUS P5GD1 w/SPDIF. diff -r e2b60934d1c8 pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Mon May 19 12:32:25 2008 +0200 +++ b/pci/hda/patch_realtek.c Tue May 20 01:50:04 2008 +1000 @@ -2981,7 +2981,7 @@ static struct snd_pci_quirk alc880_cfg_t /* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */ SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x814e, "ASUS", ALC880_ASUS), + SND_PCI_QUIRK(0x1043, 0x814e, "ASUS P5GD1 w/SPDIF", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST), SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST), Changelog: Corrected the model assignment for the ASUS P5GD1 w/SPDIF after reports of surround sound not being possible. Signed-off-by: Travis Place <wishie at wishie dot net>

1 0

[alsa-devel] Incorrect model selection for HDA Intel based card.
by Travis Place 19 May '08

19 May '08

Just had a user in #alsa on freenode, with an Asus P5GD1 mainboard. They couldnt seem to get surround sound working at all. It has an ALC880 based HDA-Intel sound device on board. Now, the problem lies in the Susbsystem ID's (1043:814e). These match in patch_realtek.c as follows: SND_PCI_QUIRK(0x1043, 0x814e, "ASUS", ALC880_ASUS) The users mainboard (The ASUS P5GD1) actually has 6 output jacks, aswell as SPDIF, not the standard 3-jacks that the "ASUS" model quirk implies. I talked this user through forcing the driver to use '6stack-digout' for his sound device, and things all work fine now. Below is a link to his alsa-info output, with the problem fixed. (so you can check IDs etc) http://pastebin.ca/1022042 Up until this point, i hadnt come across a wrong detection like that. Is there anyway to differentiate this users card from the normal "3stack" model that actually requires the "ASUS" model quirk ? Thanks, Travis Place -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

2 8

[alsa-devel] [PATCH] Support for Roland SonicCell sound module
by Chris Mennie 19 May '08

19 May '08

I'm not sure if this is the right place for it, but I've added support for the Roland SonicCell sound module. Patches were made against alsa-driver-1.0.16 source. Please let me know if there's a better place to post this: --- usbquirks.h 2008-02-05 04:23:24.000000000 -0500 +++ usbquirks.h 2008-05-18 13:02:32.000000000 -0400 @@ -1367,6 +1367,39 @@ } }, +{ + /* Roland SonicCell */ + USB_DEVICE(0x0582, 0x00c2), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "SonicCell", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, + + /* Guillemot devices */ { /*

2 4

[alsa-devel] [ALC262_FUJITSU patch_realtek.c] Use jack sense for both headphone outputs
by Tony Vroon 19 May '08

19 May '08

Good morning, Could you please consider the following patch for 2.6.27. The driver starts up with the jacks in an inconsistent state (speakers on, laptop headphone output enabled, dock headphone output enabled) and does not consider their state until the first unsolicited event arrives. In some cases it takes *headphone insert* *headphone removal* *headphone insert* to get the driver to mute the speakers successfully. Also, both HP outputs are enabled together, rather then individually enabled based on jack sense results. This patch addresses both problems. Open to comments as usual & sign-off in diff. Regards, Tony V. LINX sysadmin

2 5

[alsa-devel] pyalsa with threads allowed for python (Py_BEGIN_ALLOW_THREADS & Py_END_ALLOW_THREADS)
by ki.ber.kom.uni.st 19 May '08

19 May '08

i develop my app iterating ipython interactive sessions with pasting the good code into files (which eventually being loaded back to ipython sessions)... for that kind of development i need pyalsa to run in python threads... when i did (as it is now) alsaseq.Sequencer.receive_events(100) in infinite loop of pyqt4 thread it always steal interaction from ipython and it makes interaction unusable... it gets back every now and then almost randomly and definitively not frequent enough to be usable.. hope this description explained enough of what's happening... the solution was to make time.sleep(.001) at the end of while True: loop... it took me several hours to find out the solution... i was quite happy with the solution but my guru friends told me that's dirty and i'm lame... the other solution is in particular case of receive_events to put Py_*_ALLOW_THREADS Py_BEGIN_ALLOW_THREADS; ret = poll(self->receive_fds, self->receive_max, timeout); Py_END_ALLOW_THREADS; around poll... this url should explain this strange phenomena (to me) to the real programmers: http://www.python.org/doc/1.5.2/api/threads.html in my case my programming guru akapav sort it out after couple of minuts checking the code... hope this is of some interest to pyalsa devs... i definitively would need most of the functions being ready to run in threads.... thanx.. keep up the good work...

2 1

[alsa-devel] Application doing forward/rewind
by Aggarwal, Anuj 18 May '08

18 May '08

Hi, I am new to ALSA framework and currently working on an embedded platform. My ALSA driver works fine with APLAY and ARECORD utilities, which are coming along with the alsa-utils package. But they don't support forward and rewind. I wrote my own app for the same but the APIs are returning me 0, although some frames have been forwarded. I searched a lot for some simple command line based audio player which can provide me this sort of functionality but I could not find one. Can someone help? If I have missed something, please let me know. Thanks & Regards, Anuj Aggarwal

2 6

[alsa-devel] [patch] snd-pcsp: silent misleading warning
by Stas Sergeev 18 May '08

18 May '08

Hello. Roberto Oppedisano wrote: > I noticed that when I run > aplay -D plughw:2,0 /usr/share/sounds/KDE_Logout.wav > I get a flood of: > [ 417.338143] PCSP: playback_ptr inconsistent (4642 4661 18645) It turns out that the buffer size you get, is not evenly devided by period size. 18645 % 4661 = 1. That (wrongly) triggers the warning. This may very well be an alsa bug, or may not, but the code in the driver is handling that properly, so there is no need for such a verbose warning. The attached patch shuts down the warning. Takashi, could you please apply?

2 4

[alsa-devel] v2.6.25.3 BUG (due to usb audio device?)
by Avuton Olrich 18 May '08

18 May '08

This locks up KDE due to kmixer somehow trying to stay attached to the audio device after resume from suspend-to-disk. This is the second time of dealing with this (or a similar bug) with the usb audio device: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-March/006867.html I got it working by using the head of the git tree recommended in that thread (unfortunately I don't have the hash it was related to). http://avuton.googlepages.com/config.gz http://avuton.googlepages.com/proc-version http://avuton.googlepages.com/lspci-vvv [ 1432.710102] BUG: unable to handle kernel paging request at 00100100 [ 1432.710102] IP: [<c039be3d>] snd_ctl_dev_disconnect+0x5d/0x90 [ 1432.710102] *pde = 00000000 [ 1432.710102] Oops: 0000 [#1] PREEMPT [ 1432.710102] Modules linked in: netconsole configfs snd_pcm_oss snd_mixer_oss [ 1432.710102] [ 1432.710102] Pid: 176, comm: khubd Not tainted (2.6.25.3 #1) [ 1432.710102] EIP: 0060:[<c039be3d>] EFLAGS: 00010246 CPU: 0 [ 1432.710102] EIP is at snd_ctl_dev_disconnect+0x5d/0x90 [ 1432.710102] EAX: 00000000 EBX: 00100100 ECX: 00020004 EDX: 0000001d [ 1432.710102] ESI: f7c7dd68 EDI: f7c7dc00 EBP: f7c7dd4c ESP: f7d33e68 [ 1432.710102] DS: 007b ES: 007b FS: 0000 GS: 0000 SS: 0068 [ 1432.710102] Process khubd (pid: 176, ti=f7d32000 task=f7d30000 task.ti=f7d32000) [ 1432.710102] Stack: 00000000 f7d654e0 f7c7dc00 f7c7dd40 00000000 c039e0b6 c01a8630 f7d654e0 [ 1432.710102] c039e35a 00000000 f7c7dc00 c05502c0 f7c7e080 c039a561 f7d33ec4 f7d6d9c0 [ 1432.710102] f7d6c400 f7d6c41c c03be425 f7c7dc00 c04f0755 f7c3dc5c f7d6c400 f7d6c41c [ 1432.710102] Call Trace: [ 1432.710102] [<c039e0b6>] snd_device_disconnect+0x56/0x80 [ 1432.710102] [<c01a8630>] sysfs_ilookup_test+0x0/0x10 [ 1432.710102] [<c039e35a>] snd_device_disconnect_all+0x2a/0x50 [ 1432.710102] [<c039a561>] snd_card_disconnect+0x141/0x1b0 [ 1432.710102] [<c03be425>] usb_audio_disconnect+0x55/0x130 [ 1432.710102] [<c035ba18>] usb_unbind_interface+0x38/0x90 [ 1432.710102] [<c030df81>] __device_release_driver+0x51/0x90 [ 1432.710102] [<c030e3ad>] device_release_driver+0x1d/0x40 [ 1432.710102] [<c030d881>] bus_remove_device+0x51/0x80 [ 1432.710102] [<c030bf3c>] device_del+0xdc/0x150 [ 1432.710102] [<c035902a>] usb_disable_device+0x7a/0xe0 [ 1432.710102] [<c0354be9>] usb_disconnect+0x89/0xe0 [ 1432.710102] [<c0355996>] hub_thread+0x2f6/0xc90 [ 1432.710102] [<c0116e0b>] pick_next_task_fair+0x7b/0xb0 [ 1432.710102] [<c012dc40>] autoremove_wake_function+0x0/0x40 [ 1432.710102] [<c03556a0>] hub_thread+0x0/0xc90 [ 1432.710102] [<c012d912>] kthread+0x42/0x70 [ 1432.710102] [<c012d8d0>] kthread+0x0/0x70 [ 1432.710102] [<c0104b1f>] kernel_thread_helper+0x7/0x18 [ 1432.710102] ======================= [ 1432.710102] Code: 18 b9 01 00 00 00 ba 03 00 00 00 c7 04 24 00 00 00 00 e8 d7 c7 d7 ff 8d 43 20 b9 04 00 02 00 ba 1d 00 00 00 e8 65 9a dd ff 8b 1b <8b> 03 0f 18 00 90 39 f3 75 c9 89 e8 e8 12 5a d9 ff 89 fa 31 c0 [ 1432.710102] EIP: [<c039be3d>] snd_ctl_dev_disconnect+0x5d/0x90 SS:ESP 0068:f7d33e68 [ 1432.710127] ---[ end trace 1c91addd529faaa2 ]--- -- avuton -- "I've got a fever. And the only prescription is more cowbell." -- Christopher Walken

2 5

[alsa-devel] [patch] snd-pcsp: depend on CONFIG_EXPERIMENTAL
by Stas Sergeev 18 May '08

18 May '08

Hello. Considering all the feedbacks I got, depending snd-pcsp on CONFIG_EXPERIMENTAL looks like the only safe way to get out of all the troubles at one go. :) Can the attached patch be applied? Btw, it seems to me that the alsa-kernel hg tree somehow lacks a few snd-pcsp patches that the alsa git tree has... how have that happened and how can be fixed?

5 9

Re: [alsa-devel] [Bug #10679] 2.6.26-rc1 regression: e5e1d3cb20034a3cbcfff1f0bae12201aa2ce17e breaks artsd
by Stas Sergeev 18 May '08

18 May '08

Hello. Dave Jones wrote: > Unless someone has 10 sound cards :-) (Sounds bizarre, but I never tire > of seeing some of the creative situations our users get themselves into). Yes, the hardcoded values sucks... > Something like this is probably the only real supportable way to do it though. > It would be great if alsa took something like -1 to mean 'never make this default'. -1 seems to be already reserved for "use first free slot", but maybe it would be nice to reserve some value for the "movable" index. For example, the index -2 can mean "use the first free slot, but move to the next one when some other driver is registering with -1 or explicitly to that slot". Of course I have no idea whether this is possible and acceptable, but maybe the alsa guys can tell. > But until that lands in a module-init-tools update, we're kind of stuck. Well, they won't take 10 or any other value I guess...

1 1

[alsa-devel] update: usb headset problem (solved)
by Anselm Reisner 17 May '08

17 May '08

Hello I found the problem. It was completely my fault. Every time the computer started I pressed the mute button but the button didn't mute the micro but froze my headset. Sorry for the trouble. Anselm

1 0

Re: [alsa-devel] Auzen X-Fi Prelude
by Kamil Kaminski 17 May '08

17 May '08

---------- Forwarded message ---------- From: Kamil Kaminski <kamilkss(a)gmail.com> Date: Sat, May 17, 2008 at 8:27 AM Subject: Re: [alsa-devel] Auzen X-Fi Prelude To: Sebastian Schneider <Sebastian.Brian(a)gmx.de> Hi Sebastian, I understand you, but I would like to kindly ask for a rough date. I believe I read somewhere in the past here that in order to make basic functionality to work you have to change subsystem id and use existing driver. What patch do I need in order to apply such a hack? I'm willing to test and even contribute, my C is very good and I can read assembly. Regards, Kamil (Kyle). On Sat, May 17, 2008 at 6:28 AM, Sebastian Schneider <Sebastian.Brian(a)gmx.de> wrote: > It's done when it's done > -- > 249 Spiele für nur 1 Preis. Die GMX Spieleflatrate schon ab 9,90 Euro. > Neu: Asterix bei den Olympischen Spielen: http://flat.games.gmx.de > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

1 0

Re: [alsa-devel] pcsp weirdness/report (yes, even with libasound 1.0.16 revert)
by Stas Sergeev 17 May '08

17 May '08

Hi. Andreas Mohr wrote: > Immediately tried playback, and nothing worked properly. > Sound is __EXTREMELY__ silent, about 15 times more silent than the > normal terminal beep. Sorry for the silly question but... do you have the Master volume at maximum? > insmod with nforce_wa=1, no success either. nforce_wa is only needed if your speaker is completely silent otherwise. So you don't need it. > Debug output is (val has astonishingly uniform values): > input: PC Speaker as /class/input/input12 > PCSP: lpj=4012338, min_div=32, res=1 > PCSP: open called > PCSP: prepare called, size=32768 psize=2048 f=16 f1=16 > PCSP: trigger called > PCSP: start_playing called > timer_cnt 32 val 128 CUR_DIV 64 chip->treble 0 chip->max_treble 1 > timer_cnt 27 val 111 CUR_DIV 64 chip->treble 0 chip->max_treble 1 > timer_cnt 35 val 141 CUR_DIV 64 chip->treble 0 chip->max_treble 1 > timer_cnt 28 val 115 CUR_DIV 64 chip->treble 0 chip->max_treble 1 > timer_cnt 27 val 111 CUR_DIV 64 chip->treble 0 chip->max_treble 1 > timer_cnt 32 val 128 CUR_DIV 64 chip->treble 0 chip->max_treble 1 > timer_cnt 36 val 145 CUR_DIV 64 chip->treble 0 chip->max_treble 1 > timer_cnt 32 val 130 CUR_DIV 64 chip->treble 0 chip->max_treble 1 > timer_cnt 30 val 123 CUR_DIV 64 chip->treble 0 chip->max_treble 1 Only timer_cnt and val may change. Others are from the mixer controls. Well, whatever should change, seems changing. :) > BTW, since it has been frequently mentioned that sound device reordering is > an issue: > why not make that hard-coded in the driver itself?? I was thinking about that too - I don't know an answer. Added alsa-devel to CC because of that question. > If this is safe to do (setting it to a "max" value or to e.g. 10, > and all/most ALSA users still evaluate things properly), then one should > probably do it. Hopefully someone else can comment on that. > And mixer controls in gamix are very confusing, too. > E.g. when clicking on the BaseFRQ enum box, I get _two_ 18643 entries listed. I've seen this too, but I think this is a bug in that mixer gui. It doesn't seem to handle the ENUMERATED values right, IIRC. alsamixer works right. > And I'm still not sure (haven't investigated fully) > why that other setting is called "treble". Where? In the GUI or in the code? It shouldn't be called as such in the GUI, and for the code... well, I don't know, that was too long ago. :) > All in all very confusing. > All I'd want is enable playback, enable standard beep, master playback > volume and having the base frequency displayed(/changeable). Could you please see if there are any problems with alsamixer first? If not, then the problems are likely in the mixer you use. > Oh, and maybe oversampling configuration (how many samples to average, > for reduced timer load, see e.g. > http://kerneltrap.org/mailarchive/linux-activists/1993/4/13/29718 ). They seem to be talking about a simple downsampling to reduce the data transfer. I beleive they talk about downsampling in the player, not in the driver. You can't reduce the PWM freq. Making it below 18KHz would just whistle. > Oh, and why does the code order I/O sequence as > outb_p(chip->val61, 0x61); > outb_p(timer_cnt, 0x42); > outb(chip->val61 ^ 1, 0x61); > instead of > outb_p(chip->val61, 0x61); > outb(chip->val61 ^ 1, 0x61); > outb_p(timer_cnt, 0x42); > as mentioned at some other sites? Is there any reason or maybe it just > doesn't matter? I think it doesn't matter, and I did it that way to increase the delays between the port 0x61 writes. I have no proof that it helps or hurts. > Kconfig description should perhaps also be updated to prominently but > shortly display the fact that this driver provides full playback, > not just simple beeps (e.g. mention "digital sound driver" or so). It says something about a "primitive sound card", but I see no problems improving the text. > Thanks a lot for providing a rather important puzzle piece for > Linux usability! :) Ok, the problems should be solved, the texts should be improved, the users should upgrade, etc. And in a mean time, the warning I put there, is hopefully strong enough. :) If not, we can always depend that driver on CONFIG_BROKEN... but I'd like to hear more opinions about that option. Hmm, actually, why not? We can always un-depend it later. Thoughts?

2 2

Re: [alsa-devel] Auzen X-Fi Prelude
by Sebastian Schneider 17 May '08

17 May '08

It's done when it's done -- 249 Spiele für nur 1 Preis. Die GMX Spieleflatrate schon ab 9,90 Euro. Neu: Asterix bei den Olympischen Spielen: http://flat.games.gmx.de

1 0

[alsa-devel] usb headset problem
by Anselm Reisner 17 May '08

17 May '08

Hello Headset: Plantronics DSP 400 (047f:0ca1) Distro: Ubuntu 8.04 output from tsalsa script (from #alsa) : http://nopaste.com/p/a8yymxldp Problemdescription: After some time(~0.5-1.5 hours) listening to music my headset suddenly stops playing. If I try: When the headset crashes dmesg says ALSA /usr/src/modules/alsa-driver/usb/usbaudio.c:943: timeout: still 3 active urbs... at the beginning the number of urbs were 4 but I blacklisted the usbhid module (my headset has a volumecontrol) since then it hangs by 3. /etc/init.d/alsa-utils stop cardid I get the message "amixer: Mixer hw:1 load error: Invalid argument" I also downloaded alsa-source (from the reposity)and built the alsa-driver. Because when I try sudo alsa force-unload or modprobe -r modprobe hangs and I can't Terminate it. When the headset crashes dmesg says ALSA /usr/src/modules/alsa-driver/usb/usbaudio.c:943: timeout: still 3 active urbs... at the beginning the number of urbs were 4 but I blacklisted the usbhid module (my headset has a volumecontrol) since then it hangs by 3. So the only thing what I can do is reboot to get it working again. It's really annoying. I first asked at #also for help but they couldn't help (except that it's maybe somehow related to the usb subsystem) me but gave me the advice to post my problem here. Thanks Anselm

1 0

[alsa-devel] [patch] snd-pcsp: put back the compatibility code
by Stas Sergeev 16 May '08

16 May '08

Hello. The attached patch adds back the compatibility code, allowing the driver to work with older alsa-libs. The removal was premature, it breaks the real-life configs, I am sorry about that. Takashi, could you please apply? It is a straight-forward revert of: http://hg-mirror.alsa-project.org/alsa-kernel/diff/26b0243ab4fa/drivers/pcs…

2 1

[alsa-devel] Planned changes
by Takashi Iwai 16 May '08

16 May '08

Hi, there are some patches floating around on my experimental tree, which I'd like to merge to the ALSA upstream. At least, the following two are ready for merge: - Use menuconfig in ALSA kconfig - Convert CONFIG_SND_DEBUG_DETECT to CONFIG_SND_DEBUG_VERBOSE The first one is to change from menu to menuconfig in ALSA Kconfig files. There have been some patches by others, but they haven't been applied because of the problem with alsa-driver build. Now I hacked alsa-driver build again so that it works with menuconfig somehow. The drawback is that it changes the configure option. Instead of --with-cards and --with-card-options, now we have a single --with-kconfigs. I think, however, it's rather an improvement. The next one, renaming to CONFIG_SND_DEBUG_VERBOSE, is easy. The reason is that CONFIG_SND_DEBUG_DETECT and snd_printdd() are used almostly exclusively for verbose debug messages. This may reduce some misunderstanding about this option. The patches can be found on my experimental tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/alsa-driver-build-unsta… If there is no objection, I'll apply the above two changes in the next week. thanks, Takashi

2 2

[alsa-devel] [PATCH] ASoC V2: Clean up Freescale MPC8610 HPCD drivers
by Timur Tabi 16 May '08

16 May '08

This patch cleans up the drivers for the Freescale MPC8610 HPCD reference board so that they compile and load. Functionality has not been tested, so the drivers themselves probably don't work, but at least the kernel boots. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- A few other patches are needed for before this driver can even pretend to work: Add CS4270 i2c data to fsl_soc.c CS4270 node is misplaced in the MPC8610 device tree Add null pointer check to of_find_property These patches are waiting to be applied to the PowerPC repository. sound/soc/Makefile | 2 +- sound/soc/codecs/cs4270.c | 59 ++- sound/soc/fsl/fsl_dma.c | 296 ++++++------- sound/soc/fsl/fsl_dma.h | 20 +- sound/soc/fsl/fsl_ssi.c | 682 ++++++++++++++++++++--------- sound/soc/fsl/fsl_ssi.h | 64 ++- sound/soc/fsl/mpc8610_hpcd.c | 995 +++++++++++++++--------------------------- 7 files changed, 1066 insertions(+), 1052 deletions(-) rewrite sound/soc/fsl/mpc8610_hpcd.c (60%) diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 478664c..22393cd 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ imx/ au1x/ davinci/ omap/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ imx/ au1x/ davinci/ omap/ fsl/ diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index b47e1d8..13b9d56 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -465,6 +465,29 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) }; +static int cs4270_codec_init(struct snd_soc_codec *codec, + struct snd_soc_card *soc_card) +{ + int ret; + unsigned int i; + + /* Add the non-DAPM controls */ + + for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { + struct snd_kcontrol *kctrl; + + kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); + + ret = snd_ctl_add(soc_card->card, kctrl); + if (ret < 0) { + dev_err(soc_card->card->dev, "could not add control\n"); + return ret; + } + } + + return 0; +} + /* * Initialize the I2C interface of the CS4270 * @@ -474,11 +497,11 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { * Note: snd_soc_new_pcms() must be called before this function can be called, * because of snd_ctl_add(). */ -static int cs4270_i2c_probe(struct i2c_client *client) +static int cs4270_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) { struct cs4270_private *cs4270; struct snd_soc_codec *codec = NULL; - int i; int ret = 0; int registered = 0; /* 1 == the codec has been registered */ @@ -499,7 +522,9 @@ static int cs4270_i2c_probe(struct i2c_client *client) /* * Normally, we'd call snd_soc_new_codec, but that function - * allocates a snd_soc_codec struct, and we don't want that. + * allocates a snd_soc_codec struct, and we don't want that. So we have + * to initialize all the fields ourselves. This might break in the + * future. */ cs4270 = kzalloc(sizeof(struct cs4270_private), GFP_KERNEL); if (!cs4270) { @@ -516,7 +541,7 @@ static int cs4270_i2c_probe(struct i2c_client *client) goto error; } - strcpy(cs4270->name, "CS4270"); + strcpy(cs4270->name, "cirrus,cs4270"); cs4270->playback.stream_name = "Playback"; cs4270->playback.channels_min = 1; @@ -541,6 +566,7 @@ static int cs4270_i2c_probe(struct i2c_client *client) codec->name = cs4270->name; codec->control_data = client; + codec->init = cs4270_codec_init; ret = snd_soc_register_codec(codec, &client->dev); if (ret < 0) { @@ -549,6 +575,7 @@ static int cs4270_i2c_probe(struct i2c_client *client) } registered = 1; + cs4270->dai_new.name = cs4270->name; cs4270->dai_new.playback = &cs4270->playback; cs4270->dai_new.capture = &cs4270->capture; cs4270->dai_new.ops = &cs4270->dai_ops; @@ -560,18 +587,6 @@ static int cs4270_i2c_probe(struct i2c_client *client) goto error; } - /* Add the non-DAPM controls */ - - for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { - struct snd_kcontrol *kctrl; - - kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); - - ret = snd_ctl_add(codec->soc_card->card, kctrl); - if (ret < 0) - goto error; - } - i2c_set_clientdata(client, codec); return 0; @@ -604,12 +619,18 @@ static int cs4270_i2c_remove(struct i2c_client *client) return 0; } +static const struct i2c_device_id cs4270_id[] = { + {"cs4270", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4270_id); + static struct i2c_driver cs4270_i2c_driver = { .driver = { .name = "cs4270", .owner = THIS_MODULE, }, - .id = I2C_DRIVERID_CS4270, + .id_table = cs4270_id, .probe = cs4270_i2c_probe, .remove = cs4270_i2c_remove, }; @@ -623,7 +644,7 @@ static int __init cs4270_init(void) { int ret; - printk(KERN_INFO "Cirrus Logic CS4270 ALSA SoC codec driver\n"); + printk(KERN_INFO "Cirrus Logic CS4270 ASoC codec driver\n"); /* i2c_add_driver() will call cs4270_i2c_probe() */ ret = i2c_add_driver(&cs4270_i2c_driver); @@ -648,5 +669,5 @@ module_init(cs4270_init); module_exit(cs4270_exit); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); -MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver"); +MODULE_DESCRIPTION("Cirrus Logic CS4270 ASoC codec driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 2511eee..94ed933 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -11,6 +11,15 @@ * This driver implements ASoC support for the Elo DMA controller, which is * the DMA controller on Freescale 83xx, 85xx, and 86xx SOCs. In ALSA terms, * the PCM driver is what handles the DMA buffer. + * + * The DMA driver is not really a stand-alone driver as much as it's a library + * for the SSI driver. This is because the DMA channels cannot really stand + * alone. A particular SSI device needs to know which two DMA channels to use + * for playback and capture. If the DMA driver were a regular OF driver, it + * would get probed for each DMA channel, and then it would have to figure out + * which SSI to use for those channels. That's just too complicated. So we + * load like a normal platform driver and wait for the SSI driver to give us the + * information we need via cpu_dai->dma_data. */ #include <linux/module.h> @@ -19,8 +28,9 @@ #include <linux/dma-mapping.h> #include <linux/interrupt.h> #include <linux/delay.h> +#include <linux/of_device.h> +#include <linux/of_platform.h> -#include <sound/driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -52,34 +62,13 @@ #define FSLDMA_PCM_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ SNDRV_PCM_RATE_CONTINUOUS) -/* DMA global data. This structure is used by fsl_dma_open() to determine - * which DMA channels to assign to a substream. Unfortunately, ASoC V1 does - * not allow the soc_card driver to provide this information to the PCM - * driver in advance, and there's no way to differentiate between the two - * DMA controllers. So for now, this driver only supports one SSI device - * using two DMA channels. We cannot support multiple DMA devices. - * - * ssi_stx_phys: bus address of SSI STX register - * ssi_srx_phys: bus address of SSI SRX register - * dma_channel: pointer to the DMA channel's registers - * irq: IRQ for this DMA channel - * assigned: set to 1 if that DMA channel is assigned to a substream - */ -static struct { - dma_addr_t ssi_stx_phys; - dma_addr_t ssi_srx_phys; - struct ccsr_dma_channel __iomem *dma_channel[2]; - unsigned int irq[2]; - unsigned int assigned[2]; -} dma_global_data; - /* * The number of DMA links to use. Two is the bare minimum, but if you * have really small links you might need more. */ #define NUM_DMA_LINKS 2 -/** fsl_dma_private: p-substream DMA data +/** fsl_dma_private: per-substream DMA data * * Each substream has a 1-to-1 association with a DMA channel. * @@ -104,12 +93,8 @@ static struct { */ struct fsl_dma_private { struct fsl_dma_link_descriptor link[NUM_DMA_LINKS]; - unsigned int controller_id; - unsigned int channel_id; - struct ccsr_dma_channel __iomem *dma_channel; - unsigned int irq; + struct fsl_dma_info *dma_info; struct snd_pcm_substream *substream; - dma_addr_t ssi_sxx_phys; dma_addr_t ld_buf_phys; unsigned int current_link; dma_addr_t dma_buf_phys; @@ -143,7 +128,7 @@ static const struct snd_pcm_hardware fsl_dma_hardware = { .rate_min = 5512, .rate_max = 192000, .period_bytes_min = 512, /* A reasonable limit */ - .period_bytes_max = (u32) -1, + .period_bytes_max = (uint32_t) -1, .periods_min = NUM_DMA_LINKS, .periods_max = (unsigned int) -1, .buffer_bytes_max = 128 * 1024, /* A reasonable limit */ @@ -205,9 +190,10 @@ static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private) static irqreturn_t fsl_dma_isr(int irq, void *dev_id) { struct fsl_dma_private *dma_private = dev_id; - struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + struct ccsr_dma_channel __iomem *dma_channel = + dma_private->dma_info->channel; irqreturn_t ret = IRQ_NONE; - u32 sr, sr2 = 0; + uint32_t sr, sr2 = 0; /* We got an interrupt, so read the status register to see what we were interrupted for. @@ -216,9 +202,9 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) if (sr & CCSR_DMA_SR_TE) { dev_err(dma_private->substream->pcm->card->dev, - "DMA transmit error (controller=%u channel=%u irq=%u\n", - dma_private->controller_id, - dma_private->channel_id, irq); + "DMA%u transmit error (channel=%u irq=%u)\n", + dma_private->dma_info->controller_id, + dma_private->dma_info->channel_id, irq); fsl_dma_abort_stream(dma_private->substream); sr2 |= CCSR_DMA_SR_TE; ret = IRQ_HANDLED; @@ -230,8 +216,8 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) if (sr & CCSR_DMA_SR_PE) { dev_err(dma_private->substream->pcm->card->dev, "DMA%u programming error (channel=%u irq=%u)\n", - dma_private->controller_id, - dma_private->channel_id, irq); + dma_private->dma_info->controller_id, + dma_private->dma_info->channel_id, irq); fsl_dma_abort_stream(dma_private->substream); sr2 |= CCSR_DMA_SR_PE; ret = IRQ_HANDLED; @@ -281,9 +267,12 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) * This function is called when the codec driver calls snd_soc_new_pcms(), * once for each .dai_link in the soc_card driver's snd_soc_card * structure. + * + * The DMA controller does support 36-bit addresses for the DMA buffers, but + * supporting that is clunky so we restrict ourselves to 32-bit addresses. */ -static int fsl_dma_new(struct snd_card *card, struct snd_soc_codec_dai *dai, - struct snd_pcm *pcm) +static int fsl_dma_new(struct snd_soc_platform *platform, + struct snd_card *card, int playback, int capture, struct snd_pcm *pcm) { static u64 fsl_dma_dmamask = DMA_BIT_MASK(32); int ret; @@ -294,28 +283,40 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_codec_dai *dai, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = fsl_dma_dmamask; - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, - fsl_dma_hardware.buffer_bytes_max, - &pcm->streams[0].substream->dma_buffer); - if (ret) { - dev_err(card->dev, - "Can't allocate playback DMA buffer (size=%u)\n", - fsl_dma_hardware.buffer_bytes_max); - return -ENOMEM; + if (playback) { + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, + fsl_dma_hardware.buffer_bytes_max, + &pcm->streams[0].substream->dma_buffer); + if (ret) { + dev_err(card->dev, + "cannot allocate playback buffer (size=%u)\n", + fsl_dma_hardware.buffer_bytes_max); + goto error; + } } - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, - fsl_dma_hardware.buffer_bytes_max, - &pcm->streams[1].substream->dma_buffer); - if (ret) { - snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); - dev_err(card->dev, - "Can't allocate capture DMA buffer (size=%u)\n", - fsl_dma_hardware.buffer_bytes_max); - return -ENOMEM; + if (capture) { + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, + fsl_dma_hardware.buffer_bytes_max, + &pcm->streams[1].substream->dma_buffer); + if (ret) { + dev_err(card->dev, + "cannot allocate capture buffer (size=%u)\n", + fsl_dma_hardware.buffer_bytes_max); + goto error; + } } return 0; + +error: + if (pcm->streams[0].substream->dma_buffer.area) + snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); + + if (pcm->streams[1].substream->dma_buffer.area) + snd_dma_free_pages(&pcm->streams[1].substream->dma_buffer); + + return ret; } /** @@ -326,6 +327,10 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_codec_dai *dai, static int fsl_dma_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *pcm_runtime = substream->private_data; + struct snd_soc_dai *cpu_dai = pcm_runtime->cpu_dai; + struct fsl_dma_info *dma_info = cpu_dai->dma_data; + struct fsl_dma_private *dma_private; dma_addr_t ld_buf_phys; unsigned int channel; @@ -345,12 +350,6 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) channel = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - if (dma_global_data.assigned[channel]) { - dev_err(substream->pcm->card->dev, - "DMA channel already assigned\n"); - return -EBUSY; - } - dma_private = dma_alloc_coherent(substream->pcm->dev, sizeof(struct fsl_dma_private), &ld_buf_phys, GFP_KERNEL); if (!dma_private) { @@ -358,34 +357,24 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) "can't allocate DMA private data\n"); return -ENOMEM; } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_private->ssi_sxx_phys = dma_global_data.ssi_stx_phys; + dma_private->dma_info = &dma_info[0]; else - dma_private->ssi_sxx_phys = dma_global_data.ssi_srx_phys; - - dma_private->dma_channel = dma_global_data.dma_channel[channel]; - dma_private->irq = dma_global_data.irq[channel]; - dma_private->substream = substream; - dma_private->ld_buf_phys = ld_buf_phys; - dma_private->dma_buf_phys = substream->dma_buffer.addr; - - /* We only support one DMA controller for now */ - dma_private->controller_id = 0; - dma_private->channel_id = channel; + dma_private->dma_info = &dma_info[1]; - ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private); + ret = request_irq(dma_private->dma_info->irq, + fsl_dma_isr, 0, "DMA", dma_private); if (ret) { dev_err(substream->pcm->card->dev, "can't register ISR for IRQ %u (ret=%i)\n", - dma_private->irq, ret); + dma_private->dma_info->irq, ret); dma_free_coherent(substream->pcm->dev, sizeof(struct fsl_dma_private), dma_private, dma_private->ld_buf_phys); return ret; } - dma_global_data.assigned[channel] = 1; - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); runtime->private_data = dma_private; @@ -463,11 +452,12 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; - struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + struct ccsr_dma_channel __iomem *dma_channel = + dma_private->dma_info->channel; dma_addr_t temp_addr; /* Pointer to next period */ u64 temp_link; /* Pointer to next link descriptor */ - u32 mr; /* Temporary variable for MR register */ + uint32_t mr; /* Temporary variable for MR register */ unsigned int i; @@ -599,13 +589,14 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; - struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; - u32 mr; + struct ccsr_dma_channel __iomem *dma_channel = + dma_private->dma_info->channel; + uint32_t mr; unsigned int i; dma_addr_t ssi_sxx_phys; /* Bus address of SSI STX register */ unsigned int frame_size; /* Number of bytes per frame */ - ssi_sxx_phys = dma_private->ssi_sxx_phys; + ssi_sxx_phys = dma_private->dma_info->ssi_sxx_phys; mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK | CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK); @@ -676,7 +667,8 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; - struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; + struct ccsr_dma_channel __iomem *dma_channel = + dma_private->dma_info->channel; dma_addr_t position; snd_pcm_uframes_t frames; @@ -713,7 +705,7 @@ static int fsl_dma_hw_free(struct snd_pcm_substream *substream) if (dma_private) { struct ccsr_dma_channel __iomem *dma_channel; - dma_channel = dma_private->dma_channel; + dma_channel = dma_private->dma_info->channel; /* Stop the DMA */ out_be32(&dma_channel->mr, CCSR_DMA_MR_CA); @@ -742,11 +734,10 @@ static int fsl_dma_close(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; - int dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; if (dma_private) { - if (dma_private->irq) - free_irq(dma_private->irq, dma_private); + if (dma_private->dma_info->irq) + free_irq(dma_private->dma_info->irq, dma_private); if (dma_private->ld_buf_phys) { dma_unmap_single(substream->pcm->dev, @@ -761,8 +752,6 @@ static int fsl_dma_close(struct snd_pcm_substream *substream) substream->runtime->private_data = NULL; } - dma_global_data.assigned[dir] = 0; - return 0; } @@ -794,95 +783,92 @@ static struct snd_pcm_ops fsl_dma_ops = { .pointer = fsl_dma_pointer, }; -/** - * fsl_dma_configure: store the DMA parameters from the fabric driver. - * - * This function is called by the ASoC fabric driver to give us the DMA and - * SSI channel information. - * - * Unfortunately, ASoC V1 does make it possible to determine the DMA/SSI - * data when a substream is created, so for now we need to store this data - * into a global variable. This means that we can only support one DMA - * controller, and hence only one SSI. - */ -int fsl_dma_configure(struct fsl_dma_info *dma_info) -{ - static int initialized; - - /* We only support one DMA controller for now */ - if (initialized) - return 0; - - dma_global_data.ssi_stx_phys = dma_info->ssi_stx_phys; - dma_global_data.ssi_srx_phys = dma_info->ssi_srx_phys; - dma_global_data.dma_channel[0] = dma_info->dma_channel[0]; - dma_global_data.dma_channel[1] = dma_info->dma_channel[1]; - dma_global_data.irq[0] = dma_info->dma_irq[0]; - dma_global_data.irq[1] = dma_info->dma_irq[1]; - dma_global_data.assigned[0] = 0; - dma_global_data.assigned[1] = 0; - - initialized = 1; - return 1; -} -EXPORT_SYMBOL_GPL(fsl_dma_configure); +const char fsl_platform_id[] = "fsl_pcm"; +EXPORT_SYMBOL_GPL(fsl_platform_id); +static struct snd_soc_platform_new fsl_dma_platform = { + .name = fsl_platform_id, + .pcm_ops = &fsl_dma_ops, + .pcm_new = fsl_dma_new, + .pcm_free = fsl_dma_free_dma_buffers, +}; /* - * TODO: We may want to query the device tree here for our DMA and SSI - * properties. It may also beneficial to merge in the SSI driver. */ -static int fsl_pcm_probe(struct device *dev) +static int fsl_dma_probe(struct platform_device *pdev) { - struct snd_soc_platform *platform = to_snd_soc_platform(dev); + struct snd_soc_platform *platform; int ret; - platform->pcm_ops = &fsl_dma_ops; - platform->pcm_new = fsl_pcm_new, - platform->pcm_free = fsl_dma_free_dma_buffers, -/* TODO: DAI can be config here after dev tree parse */ - ret = snd_soc_platform_add_dai(platform, imx_ssi, ARRAY_SIZE(imx_ssi)); - if (ret < 0) - return ret; -*/ - ret = snd_soc_register_platform(platform); + platform = snd_soc_new_platform(&fsl_dma_platform); + if (!platform) { + dev_err(&pdev->dev, "unable to allocate platform\n"); + return -ENOMEM; + } + + platform_set_drvdata(pdev, platform); + + ret = snd_soc_register_platform(platform, &pdev->dev); + if (ret < 0) { + dev_err(&pdev->dev, "could not register platform\n"); + snd_soc_free_platform(platform); + } + return ret; } -static int fsl_pcm_remove(struct device *dev) +static int fsl_dma_remove(struct platform_device *pdev) { - struct snd_soc_platform *platform = to_snd_soc_platform(dev); + struct snd_soc_platform *platform = platform_get_drvdata(pdev); + + if (platform) + snd_soc_free_platform(platform); - snd_soc_unregister_platform(platform); return 0; } -const char fsl_platform_id[] = "fsl_pcm"; -EXPORT_SYMBOL_GPL(fsl_platform_id); - -static struct device_driver fsl_pcm_driver = { - .name = fsl_platform_id, - .owner = THIS_MODULE, - .bus = &asoc_bus_type, - .probe = fsl_pcm_probe, - .remove = __devexit_p(fsl_pcm_remove), -// .suspend = fsl_pcm_suspend, -// .resume = fsl_pcm_resume, +static struct platform_driver fsl_dma_driver = { + .driver = { + .name = fsl_platform_id, + .owner = THIS_MODULE, + }, + .probe = fsl_dma_probe, + .remove = __devexit_p(fsl_dma_remove), }; -static __init int fsl_pcm_init(void) +static struct platform_device *pdev; + +static __init int fsl_dma_init(void) { - return driver_register(&fsl_pcm_driver); + int ret; + + printk(KERN_INFO "Freescale Elo DMA ASoC PCM driver\n"); + + ret = platform_driver_register(&fsl_dma_driver); + if (ret < 0) { + pr_err("fsl-dma: could not register platform\n"); + return ret; + } + + pdev = platform_device_register_simple(fsl_platform_id, 0, NULL, 0); + if (!pdev) { + pr_err("fsl-dma: could not register platform\n"); + return ret; + } + + return 0; } -static __exit void fsl_pcm_exit(void) +static __exit void fsl_dma_exit(void) { - driver_unregister(&fsl_pcm_driver); + platform_device_unregister(pdev); + platform_driver_unregister(&fsl_dma_driver); } -module_init(fsl_pcm_init); -module_exit(fsl_pcm_exit); +module_init(fsl_dma_init); +module_exit(fsl_dma_exit); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); -MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM module"); +MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM driver"); MODULE_LICENSE("GPL"); + diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h index 05febed..ed4592f 100644 --- a/sound/soc/fsl/fsl_dma.h +++ b/sound/soc/fsl/fsl_dma.h @@ -126,20 +126,18 @@ struct fsl_dma_link_descriptor { u8 res[4]; /* Reserved */ } __attribute__ ((aligned(32), packed)); -/* DMA information needed to create a snd_soc_cpu_dai object +/* Per-stream DMA information that the SSI driver passes to the DMA driver * - * ssi_stx_phys: bus address of SSI STX register to use - * ssi_srx_phys: bus address of SSI SRX register to use - * dma[0]: points to the DMA channel to use for playback - * dma[1]: points to the DMA channel to use for capture - * dma_irq[0]: IRQ of the DMA channel to use for playback - * dma_irq[1]: IRQ of the DMA channel to use for capture + * ssi_sxx_phys: bus address of SSI STX or SRX register to use + * channel: points to the DMA channel device to use + * irq: IRQ of the DMA channel */ struct fsl_dma_info { - dma_addr_t ssi_stx_phys; - dma_addr_t ssi_srx_phys; - struct ccsr_dma_channel __iomem *dma_channel[2]; - unsigned int dma_irq[2]; + dma_addr_t ssi_sxx_phys; + struct ccsr_dma_channel __iomem *channel; + unsigned int irq; + unsigned int controller_id; + unsigned int channel_id; }; extern const char fsl_platform_id[]; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9f194bc..75576b2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -14,8 +14,9 @@ #include <linux/interrupt.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/of_device.h> +#include <linux/of_platform.h> -#include <sound/driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -24,6 +25,7 @@ #include <asm/immap_86xx.h> +#include "fsl_dma.h" #include "fsl_ssi.h" /** @@ -62,56 +64,6 @@ #endif /** - * fsl_ssi_private: per-SSI private data - * - * @name: short name for this device ("SSI0", "SSI1", etc) - * @ssi: pointer to the SSI's registers - * @ssi_phys: physical address of the SSI registers - * @irq: IRQ of this SSI - * @dev: struct device pointer - * @playback: the number of playback streams opened - * @capture: the number of capture streams opened - * @cpu_dai: the CPU DAI for this device - * @dev_attr: the sysfs device attribute structure - * @stats: SSI statistics - */ -struct fsl_ssi_private { - char name[8]; - struct ccsr_ssi __iomem *ssi; - dma_addr_t ssi_phys; - unsigned int irq; - struct device *dev; - unsigned int playback; - unsigned int capture; - struct snd_soc_cpu_dai cpu_dai; - struct device_attribute dev_attr; - - struct { - unsigned int rfrc; - unsigned int tfrc; - unsigned int cmdau; - unsigned int cmddu; - unsigned int rxt; - unsigned int rdr1; - unsigned int rdr0; - unsigned int tde1; - unsigned int tde0; - unsigned int roe1; - unsigned int roe0; - unsigned int tue1; - unsigned int tue0; - unsigned int tfs; - unsigned int rfs; - unsigned int tls; - unsigned int rls; - unsigned int rff1; - unsigned int rff0; - unsigned int tfe1; - unsigned int tfe0; - } stats; -}; - -/** * fsl_ssi_isr: SSI interrupt handler * * Although it's possible to use the interrupt handler to send and receive @@ -121,12 +73,12 @@ struct fsl_ssi_private { * This interrupt handler is used only to gather statistics. * * @irq: IRQ of the SSI device - * @dev_id: pointer to the ssi_private structure for this SSI device + * @dev_id: pointer to the ssi_info structure for this SSI device */ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) { - struct fsl_ssi_private *ssi_private = dev_id; - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + struct fsl_ssi_info *ssi_info = dev_id; + struct ccsr_ssi __iomem *ssi = ssi_info->ssi; irqreturn_t ret = IRQ_NONE; __be32 sisr; __be32 sisr2 = 0; @@ -138,113 +90,113 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) sisr = in_be32(&ssi->sisr) & in_be32(&ssi->sier); if (sisr & CCSR_SSI_SISR_RFRC) { - ssi_private->stats.rfrc++; + ssi_info->stats.rfrc++; sisr2 |= CCSR_SSI_SISR_RFRC; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_TFRC) { - ssi_private->stats.tfrc++; + ssi_info->stats.tfrc++; sisr2 |= CCSR_SSI_SISR_TFRC; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_CMDAU) { - ssi_private->stats.cmdau++; + ssi_info->stats.cmdau++; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_CMDDU) { - ssi_private->stats.cmddu++; + ssi_info->stats.cmddu++; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_RXT) { - ssi_private->stats.rxt++; + ssi_info->stats.rxt++; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_RDR1) { - ssi_private->stats.rdr1++; + ssi_info->stats.rdr1++; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_RDR0) { - ssi_private->stats.rdr0++; + ssi_info->stats.rdr0++; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_TDE1) { - ssi_private->stats.tde1++; + ssi_info->stats.tde1++; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_TDE0) { - ssi_private->stats.tde0++; + ssi_info->stats.tde0++; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_ROE1) { - ssi_private->stats.roe1++; + ssi_info->stats.roe1++; sisr2 |= CCSR_SSI_SISR_ROE1; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_ROE0) { - ssi_private->stats.roe0++; + ssi_info->stats.roe0++; sisr2 |= CCSR_SSI_SISR_ROE0; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_TUE1) { - ssi_private->stats.tue1++; + ssi_info->stats.tue1++; sisr2 |= CCSR_SSI_SISR_TUE1; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_TUE0) { - ssi_private->stats.tue0++; + ssi_info->stats.tue0++; sisr2 |= CCSR_SSI_SISR_TUE0; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_TFS) { - ssi_private->stats.tfs++; + ssi_info->stats.tfs++; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_RFS) { - ssi_private->stats.rfs++; + ssi_info->stats.rfs++; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_TLS) { - ssi_private->stats.tls++; + ssi_info->stats.tls++; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_RLS) { - ssi_private->stats.rls++; + ssi_info->stats.rls++; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_RFF1) { - ssi_private->stats.rff1++; + ssi_info->stats.rff1++; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_RFF0) { - ssi_private->stats.rff0++; + ssi_info->stats.rff0++; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_TFE1) { - ssi_private->stats.tfe1++; + ssi_info->stats.tfe1++; ret = IRQ_HANDLED; } if (sisr & CCSR_SSI_SISR_TFE0) { - ssi_private->stats.tfe0++; + ssi_info->stats.tfe0++; ret = IRQ_HANDLED; } @@ -263,24 +215,24 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) * If this is the first stream open, then grab the IRQ and program most of * the SSI registers. */ -static int fsl_ssi_startup(struct snd_pcm_substream *substream) +static int fsl_ssi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; + struct fsl_ssi_info *ssi_info = cpu_dai->private_data; /* * If this is the first stream opened, then request the IRQ * and initialize the SSI registers. */ - if (!ssi_private->playback && !ssi_private->capture) { - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + if (!ssi_info->playback && !ssi_info->capture) { + struct ccsr_ssi __iomem *ssi = ssi_info->ssi; int ret; - ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, - ssi_private->name, ssi_private); + ret = request_irq(ssi_info->irq, fsl_ssi_isr, 0, + ssi_info->name, ssi_info); if (ret < 0) { dev_err(substream->pcm->card->dev, - "could not claim irq %u\n", ssi_private->irq); + "could not claim irq %u\n", ssi_info->irq); return ret; } @@ -344,10 +296,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream) } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ssi_private->playback++; + ssi_info->playback++; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ssi_private->capture++; + ssi_info->capture++; + + cpu_dai->dma_data = ssi_info->dma_info; return 0; } @@ -365,17 +319,19 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream) * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the * clock master. */ -static int fsl_ssi_prepare(struct snd_pcm_substream *substream) +static int fsl_ssi_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; - - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u32 wl; + struct fsl_ssi_info *ssi_info = cpu_dai->private_data; + struct ccsr_ssi __iomem *ssi = ssi_info->ssi; + uint32_t wl; wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + /* FIXME: We should read and write the registers manually so as to + minimize the amount of time the SSI is disabled. */ + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -397,11 +353,12 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream) * The DMA channel is in external master start and pause mode, which * means the SSI completely controls the flow of data. */ -static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) +static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *cpu_dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + + struct fsl_ssi_info *ssi_info = cpu_dai->private_data; + struct ccsr_ssi __iomem *ssi = ssi_info->ssi; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -417,7 +374,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) * to put data into its FIFO. Without it, ALSA starts * to complain about overruns. */ - msleep(1); + mdelay(1); } break; @@ -442,27 +399,28 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) * * Shutdown the SSI if there are no other substreams open. */ -static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) +static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; + + struct fsl_ssi_info *ssi_info = cpu_dai->private_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ssi_private->playback--; + ssi_info->playback--; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ssi_private->capture--; + ssi_info->capture--; /* * If this is the last active substream, disable the SSI and release * the IRQ. */ - if (!ssi_private->playback && !ssi_private->capture) { - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + if (!ssi_info->playback && !ssi_info->capture) { + struct ccsr_ssi __iomem *ssi = ssi_info->ssi; clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); - free_irq(ssi_private->irq, ssi_private); + free_irq(ssi_info->irq, ssi_info); } } @@ -480,8 +438,8 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) * @freq: the frequency of the given clock ID, currently ignored * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master) */ -static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) +static int fsl_ssi_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) { return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL; @@ -498,7 +456,7 @@ static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, * * @format: one of SND_SOC_DAIFMT_xxx */ -static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) +static int fsl_ssi_set_fmt(struct snd_soc_dai *dai, unsigned int format) { return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; } @@ -511,142 +469,448 @@ static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) static ssize_t fsl_sysfs_ssi_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct fsl_ssi_private *ssi_private = - container_of(attr, struct fsl_ssi_private, dev_attr); + struct fsl_ssi_info *ssi_info = + container_of(attr, struct fsl_ssi_info, dev_attr); ssize_t length; - length = sprintf(buf, "rfrc=%u", ssi_private->stats.rfrc); - length += sprintf(buf + length, "\ttfrc=%u", ssi_private->stats.tfrc); - length += sprintf(buf + length, "\tcmdau=%u", ssi_private->stats.cmdau); - length += sprintf(buf + length, "\tcmddu=%u", ssi_private->stats.cmddu); - length += sprintf(buf + length, "\trxt=%u", ssi_private->stats.rxt); - length += sprintf(buf + length, "\trdr1=%u", ssi_private->stats.rdr1); - length += sprintf(buf + length, "\trdr0=%u", ssi_private->stats.rdr0); - length += sprintf(buf + length, "\ttde1=%u", ssi_private->stats.tde1); - length += sprintf(buf + length, "\ttde0=%u", ssi_private->stats.tde0); - length += sprintf(buf + length, "\troe1=%u", ssi_private->stats.roe1); - length += sprintf(buf + length, "\troe0=%u", ssi_private->stats.roe0); - length += sprintf(buf + length, "\ttue1=%u", ssi_private->stats.tue1); - length += sprintf(buf + length, "\ttue0=%u", ssi_private->stats.tue0); - length += sprintf(buf + length, "\ttfs=%u", ssi_private->stats.tfs); - length += sprintf(buf + length, "\trfs=%u", ssi_private->stats.rfs); - length += sprintf(buf + length, "\ttls=%u", ssi_private->stats.tls); - length += sprintf(buf + length, "\trls=%u", ssi_private->stats.rls); - length += sprintf(buf + length, "\trff1=%u", ssi_private->stats.rff1); - length += sprintf(buf + length, "\trff0=%u", ssi_private->stats.rff0); - length += sprintf(buf + length, "\ttfe1=%u", ssi_private->stats.tfe1); - length += sprintf(buf + length, "\ttfe0=%u\n", ssi_private->stats.tfe0); + length = sprintf(buf, "rfrc=%u", ssi_info->stats.rfrc); + length += sprintf(buf + length, "\ttfrc=%u", ssi_info->stats.tfrc); + length += sprintf(buf + length, "\tcmdau=%u", ssi_info->stats.cmdau); + length += sprintf(buf + length, "\tcmddu=%u", ssi_info->stats.cmddu); + length += sprintf(buf + length, "\trxt=%u", ssi_info->stats.rxt); + length += sprintf(buf + length, "\trdr1=%u", ssi_info->stats.rdr1); + length += sprintf(buf + length, "\trdr0=%u", ssi_info->stats.rdr0); + length += sprintf(buf + length, "\ttde1=%u", ssi_info->stats.tde1); + length += sprintf(buf + length, "\ttde0=%u", ssi_info->stats.tde0); + length += sprintf(buf + length, "\troe1=%u", ssi_info->stats.roe1); + length += sprintf(buf + length, "\troe0=%u", ssi_info->stats.roe0); + length += sprintf(buf + length, "\ttue1=%u", ssi_info->stats.tue1); + length += sprintf(buf + length, "\ttue0=%u", ssi_info->stats.tue0); + length += sprintf(buf + length, "\ttfs=%u", ssi_info->stats.tfs); + length += sprintf(buf + length, "\trfs=%u", ssi_info->stats.rfs); + length += sprintf(buf + length, "\ttls=%u", ssi_info->stats.tls); + length += sprintf(buf + length, "\trls=%u", ssi_info->stats.rls); + length += sprintf(buf + length, "\trff1=%u", ssi_info->stats.rff1); + length += sprintf(buf + length, "\trff0=%u", ssi_info->stats.rff0); + length += sprintf(buf + length, "\ttfe1=%u", ssi_info->stats.tfe1); + length += sprintf(buf + length, "\ttfe0=%u\n", ssi_info->stats.tfe0); return length; } +static struct snd_soc_dai_caps playback = { + /* The SSI does not support monaural audio. */ + .channels_min = 2, + .channels_max = 2, + .rates = FSLSSI_I2S_RATES, + .formats = FSLSSI_I2S_FORMATS, +}; + +static struct snd_soc_dai_caps capture = { + .channels_min = 2, + .channels_max = 2, + .rates = FSLSSI_I2S_RATES, + .formats = FSLSSI_I2S_FORMATS, +}; + +static struct snd_soc_dai_ops ops = { + .startup = fsl_ssi_startup, + .prepare = fsl_ssi_prepare, + .shutdown = fsl_ssi_shutdown, + .trigger = fsl_ssi_trigger, + + .set_sysclk = fsl_ssi_set_sysclk, + .set_fmt = fsl_ssi_set_fmt, +}; + +static struct device_node *find_dma_node(unsigned int controller, + unsigned int channel) +{ + struct device_node *np = NULL; + struct device_node *np2 = NULL; + const uint32_t *iprop; + + for_each_compatible_node(np, NULL, "fsl,eloplus-dma") { + iprop = of_get_property(np, "cell-index", NULL); + if (!iprop) { + pr_err("fsl-ssi: cell-index property not found\n"); + return NULL; + } + if (*iprop == controller) + break; + } + + if (!np) { + pr_err("fsl-ssi: cannot find node for DMA controller %u\n", + controller); + return NULL; + } + + for_each_child_of_node(np, np2) { + if (!of_device_is_compatible(np2, "fsl,eloplus-dma-channel")) + continue; + + iprop = of_get_property(np2, "cell-index", NULL); + if (!iprop) { + pr_err("fsl-ssi: cell-index property not found\n"); + return NULL; + } + if (*iprop == channel) + break; + } + + if (!np2) { + pr_err("fsl-ssi: cannot find node for DMA channel %u\n", + channel); + return NULL; + } + + of_node_put(np); + + return np2; +} + /** - * fsl_ssi_create_dai: create a snd_soc_cpu_dai structure - * - * This function is called by the soc_card driver to create a snd_soc_cpu_dai - * structure. The function creates an ssi_private object, which contains - * the snd_soc_cpu_dai. It also creates the sysfs statistics device. + * Initialize a dma_info structure from a pointer to a DMA node */ -struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) +static int get_dma_info(struct device_node *np, struct fsl_dma_info *dma_info) +{ + const uint32_t *iprop; + + if (!np) + return 0; + + iprop = of_get_property(of_get_parent(np), "cell-index", NULL); + if (!iprop) + return 0; + + dma_info->controller_id = *iprop; + + iprop = of_get_property(np, "cell-index", NULL); + if (!iprop) + return 0; + + dma_info->channel_id = *iprop; + + dma_info->channel = of_iomap(np, 0); + dma_info->irq = irq_of_parse_and_map(np, 0); + + return 1; +} +static int fsl_ssi_probe(struct of_device *ofdev, + const struct of_device_id *match) { - struct snd_soc_cpu_dai *fsl_ssi_dai; - struct fsl_ssi_private *ssi_private; - int ret = 0; - struct device_attribute *dev_attr; - - ssi_private = kzalloc(sizeof(struct fsl_ssi_private), GFP_KERNEL); - if (!ssi_private) { - dev_err(ssi_info->dev, "could not allocate DAI object\n"); - return NULL; + struct device_node *np = ofdev->node; + struct device_node *codec_np = NULL; + struct device_node *dma_np[2] = {NULL, NULL}; + const phandle *codec_ph; + const phandle *dma_ph; + const char *sprop; + const uint32_t *iprop; + struct resource res; + + struct fsl_ssi_info *ssi_info; + int ret = -ENODEV; + + struct snd_soc_dai_new dai_template; + + ssi_info = kzalloc(sizeof(struct fsl_ssi_info), GFP_KERNEL); + if (!ssi_info) { + dev_err(&ofdev->dev, "cannot allocate ssi_info\n"); + return -ENOMEM; + } + + ssi_info->dev = &ofdev->dev; + + /* + * We are only interested in SSIs with a codec phandle in them, so let's + * make sure this SSI has one. + */ + codec_ph = of_get_property(np, "codec-handle", NULL); + if (!codec_ph) { + dev_dbg(&ofdev->dev, "no codec handle for this SSI\n"); + goto error; + } + + codec_np = of_find_node_by_phandle(*codec_ph); + if (!codec_np) { + dev_err(&ofdev->dev, "codec handle does not exist\n"); + goto error; + } + + /* The MPC8610 HPCD only knows about the CS4270 codec, so reject + * anything else. + * FIXME: This should be unnecessary + */ + if (!of_device_is_compatible(codec_np, "cirrus,cs4270")) { + dev_dbg(&ofdev->dev, "unknown codec %s\n", + (char *) of_get_property(codec_np, "compatible", NULL)); + goto error; + } + + /* Get the device ID */ + iprop = of_get_property(np, "cell-index", NULL); + if (!iprop) { + dev_err(&ofdev->dev, "cell-index property not found\n"); + ret = -EINVAL; + goto error; + } + ssi_info->id = *iprop; + + strcpy(ssi_info->name, "fsl,mpc8610-ssi"); + + /* Get the serial format and clock direction. */ + sprop = of_get_property(np, "fsl,mode", NULL); + if (!sprop) { + dev_err(&ofdev->dev, "fsl,mode property not found\n"); + ret = -EINVAL; + goto error; } - memcpy(&ssi_private->cpu_dai, &fsl_ssi_dai_template, - sizeof(struct snd_soc_cpu_dai)); - fsl_ssi_dai = &ssi_private->cpu_dai; - dev_attr = &ssi_private->dev_attr; + if (strcasecmp(sprop, "i2s-slave") == 0) { + ssi_info->dai_format = SND_SOC_DAIFMT_I2S; + ssi_info->codec_clk_direction = SND_SOC_CLOCK_OUT; + ssi_info->cpu_clk_direction = SND_SOC_CLOCK_IN; - sprintf(ssi_private->name, "ssi%u", (u8) ssi_info->id); - ssi_private->ssi = ssi_info->ssi; - ssi_private->ssi_phys = ssi_info->ssi_phys; - ssi_private->irq = ssi_info->irq; - ssi_private->dev = ssi_info->dev; + /* + * In i2s-slave mode, the codec has its own clock source, so we + * need to get the frequency from the device tree and pass it to + * the codec driver. + */ + iprop = of_get_property(codec_np, "clock-frequency", NULL); + if (!iprop || !*iprop) { + dev_err(&ofdev->dev, "codec bus-frequency property " + "is missing or invalid\n"); + ret = -EINVAL; + goto error; + } + ssi_info->clk_frequency = *iprop; + } else if (strcasecmp(sprop, "i2s-master") == 0) { + ssi_info->dai_format = SND_SOC_DAIFMT_I2S; + ssi_info->codec_clk_direction = SND_SOC_CLOCK_IN; + ssi_info->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "lj-slave") == 0) { + ssi_info->dai_format = SND_SOC_DAIFMT_LEFT_J; + ssi_info->codec_clk_direction = SND_SOC_CLOCK_OUT; + ssi_info->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "lj-master") == 0) { + ssi_info->dai_format = SND_SOC_DAIFMT_LEFT_J; + ssi_info->codec_clk_direction = SND_SOC_CLOCK_IN; + ssi_info->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "rj-master") == 0) { + ssi_info->dai_format = SND_SOC_DAIFMT_RIGHT_J; + ssi_info->codec_clk_direction = SND_SOC_CLOCK_OUT; + ssi_info->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "rj-master") == 0) { + ssi_info->dai_format = SND_SOC_DAIFMT_RIGHT_J; + ssi_info->codec_clk_direction = SND_SOC_CLOCK_IN; + ssi_info->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "ac97-slave") == 0) { + ssi_info->dai_format = SND_SOC_DAIFMT_AC97; + ssi_info->codec_clk_direction = SND_SOC_CLOCK_OUT; + ssi_info->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "ac97-master") == 0) { + ssi_info->dai_format = SND_SOC_DAIFMT_AC97; + ssi_info->codec_clk_direction = SND_SOC_CLOCK_IN; + ssi_info->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else { + dev_err(&ofdev->dev, + "unrecognized fsl,mode property \"%s\"\n", sprop); + ret = -EINVAL; + goto error; + } - ssi_private->dev->driver_data = fsl_ssi_dai; + if (!ssi_info->clk_frequency) { + dev_err(&ofdev->dev, "unknown clock frequency\n"); + ret = -EINVAL; + goto error; + } - /* Initialize the the device_attribute structure */ - dev_attr->attr.name = "ssi-stats"; - dev_attr->attr.mode = S_IRUGO; - dev_attr->show = fsl_sysfs_ssi_show; + /* Read the SSI hardware information from the device tree */ + ret = of_address_to_resource(np, 0, &res); + if (ret) { + dev_err(&ofdev->dev, "could not obtain SSI address\n"); + goto error; + } + if (!res.start) { + dev_err(&ofdev->dev, "invalid SSI address\n"); + goto error; + } + ssi_info->ssi_phys = res.start; + + ssi_info->ssi = ioremap(ssi_info->ssi_phys, sizeof(struct ccsr_ssi)); + if (!ssi_info->ssi) { + dev_err(&ofdev->dev, "could not map SSI address %x\n", + ssi_info->ssi_phys); + ret = -EINVAL; + goto error; + } - ret = device_create_file(ssi_private->dev, dev_attr); + /* Get the IRQ of the SSI */ + ssi_info->irq = irq_of_parse_and_map(np, 0); + if (!ssi_info->irq) { + dev_err(&ofdev->dev, "could not get SSI IRQ\n"); + ret = -EINVAL; + goto error; + } + + ssi_info->dma_info[0].ssi_sxx_phys = ssi_info->ssi_phys + + offsetof(struct ccsr_ssi, stx0); + ssi_info->dma_info[1].ssi_sxx_phys = ssi_info->ssi_phys + + offsetof(struct ccsr_ssi, srx0); + + /* + * Get the DMA information. If it's an older device tree (i.e. without + * an "fsl,playback-dma" property), then we assume that SSI1 uses DMA1 + * Channels 0 and 1, and SSI2 uses DMA2 Channels 0 and 1. + */ + dma_ph = of_get_property(np, "fsl,playback-dma", NULL); + if (!dma_ph) { + dev_warn(&ofdev->dev, "please update your device tree\n"); + dma_np[0] = find_dma_node(ssi_info->id, 0); + dma_np[1] = find_dma_node(ssi_info->id, 1); + } else { + dma_np[0] = of_find_node_by_phandle(*dma_ph); + dma_ph = of_get_property(np, "fsl,capture-dma", NULL); + if (dma_ph) + dma_np[1] = of_find_node_by_phandle(*dma_ph); + } + + if (!get_dma_info(dma_np[0], &ssi_info->dma_info[0])) { + dev_err(&ofdev->dev, "could not obtain playback DMA info\n"); + goto error; + } + + if (!get_dma_info(dma_np[1], &ssi_info->dma_info[1])) { + dev_err(&ofdev->dev, "could not obtain capture DMA info\n"); + goto error; + } + + memset(&dai_template, 0, sizeof(dai_template)); + dai_template.name = ssi_info->name; + dai_template.id = ssi_info->id; + dai_template.playback = &playback; + dai_template.capture = &capture; + dai_template.ops = &ops; + + ssi_info->dai = + snd_soc_register_platform_dai(&dai_template, &ofdev->dev); + if (!ssi_info->dai) { + dev_err(&ofdev->dev, "could not register platform DAI\n"); + ret = -ENOMEM; + goto error; + } + + ssi_info->dai->private_data = ssi_info; + + ssi_info->dev_attr.attr.name = ssi_info->name; + ssi_info->dev_attr.attr.mode = S_IRUGO; + ssi_info->dev_attr.show = fsl_sysfs_ssi_show; + + ret = device_create_file(&ofdev->dev, &ssi_info->dev_attr); if (ret) { - dev_err(ssi_info->dev, "could not create sysfs %s file\n", - ssi_private->dev_attr.attr.name); - kfree(fsl_ssi_dai); - return NULL; + dev_err(&ofdev->dev, "could not create sysfs %s file\n", + ssi_info->dev_attr.attr.name); + goto error; } - fsl_ssi_dai->private_data = ssi_private; - fsl_ssi_dai->name = ssi_private->name; - fsl_ssi_dai->id = ssi_info->id; + dev_set_drvdata(&ofdev->dev, ssi_info); + + + + return 0; + +error: + + if (ssi_info->dai) + snd_soc_unregister_platform_dai(ssi_info->dai); + + if (ssi_info->irq) + irq_dispose_mapping(ssi_info->irq); + + if (ssi_info->ssi) + iounmap(ssi_info->ssi); - return fsl_ssi_dai; + kfree(ssi_info); + + return ret; } -EXPORT_SYMBOL_GPL(fsl_ssi_create_dai); /** - * fsl_ssi_destroy_dai: destroy the snd_soc_cpu_dai object + * fsl_ssi_remove: remove the OF device * - * This function undoes the operations of fsl_ssi_create_dai() + * This function is called when the OF device is removed. */ -void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai) +static int fsl_ssi_remove(struct of_device *ofdev) { - struct fsl_ssi_private *ssi_private = - container_of(fsl_ssi_dai, struct fsl_ssi_private, cpu_dai); + struct fsl_ssi_info *ssi_info = dev_get_drvdata(&ofdev->dev); + + if (ssi_info->dai) + snd_soc_unregister_platform_dai(ssi_info->dai); + + if (ssi_info->irq) + irq_dispose_mapping(ssi_info->irq); + + if (ssi_info->ssi) + iounmap(ssi_info->ssi); - device_remove_file(ssi_private->dev, &ssi_private->dev_attr); + kfree(ssi_info); - kfree(ssi_private); + dev_set_drvdata(&ofdev->dev, NULL); + + return 0; } -/* - * TODO: we probably want to remove this dai template and dynamically - * create our DAI in platform probe() based on device tree parse data. +static struct of_device_id fsl_ssi_match[] = { + { + .compatible = "fsl,mpc8610-ssi", + }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_ssi_match); + +static struct of_platform_driver fsl_ssi_of_driver = { + .owner = THIS_MODULE, + .name = "fsl_ssi", + .match_table = fsl_ssi_match, + .probe = fsl_ssi_probe, + .remove = fsl_ssi_remove, +}; + +/** + * fsl_ssi_init: fabric driver initialization. + * + * This function is called when this module is loaded. */ -struct snd_soc_dai fsl_ssi_template = { +static int __init fsl_ssi_init(void) { - .name = "SSI0-0", - .id = IMX_DAI_SSI0, - .new = fsl_ssi_create_dai, - .free = fsl_ssi_destroy_dai, - - .playback = { - /* The SSI does not support monaural audio. */ - .channels_min = 2, - .channels_max = 2, - .rates = FSLSSI_I2S_RATES, - .formats = FSLSSI_I2S_FORMATS, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = FSLSSI_I2S_RATES, - .formats = FSLSSI_I2S_FORMATS, - }, + int ret; - /* alsa ops */ - .startup = fsl_ssi_startup, - .prepare = fsl_ssi_prepare, - .shutdown = fsl_ssi_shutdown, - .trigger = fsl_ssi_trigger, + printk(KERN_INFO "Freescale SSI ASoC driver\n"); - /* dai ops */ - .set_sysclk = fsl_ssi_set_sysclk, - .set_fmt = fsl_ssi_set_fmt, -}, -EXPORT_SYMBOL_GPL(fsl_ssi_template); + ret = of_register_platform_driver(&fsl_ssi_of_driver); + + if (ret) + printk(KERN_ERR + "fsl-ssi: failed to register SSI driver\n"); + + return ret; +} + +/** + * fsl_ssi_exit: fabric driver exit + * + * This function is called when this driver is unloaded. + */ +static void __exit fsl_ssi_exit(void) +{ + of_unregister_platform_driver(&fsl_ssi_of_driver); +} + +module_init(fsl_ssi_init); +module_exit(fsl_ssi_exit); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); -MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); +MODULE_DESCRIPTION("Freescale SSI ASoC driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 0aaafa4..05fd84d 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -196,29 +196,63 @@ struct ccsr_ssi { #define CCSR_SSI_SOR_WAIT(x) (((x) & 3) << CCSR_SSI_SOR_WAIT_SHIFT) #define CCSR_SSI_SOR_SYNRST 0x00000001 -/* Instantiation data for an SSI interface +/** + * fsl_ssi_info: per-SSI private data * - * This structure contains all the information that the the SSI driver needs - * to instantiate an SSI interface with ALSA. The soc_card driver should - * create this structure, fill it in, call fsl_ssi_create_dai(), and then - * delete the structure. - * - * id: which SSI this is (0, 1, etc. ) - * ssi: pointer to the SSI's registers - * ssi_phys: physical address of the SSI registers - * irq: IRQ of this SSI - * dev: struct device, used to create the sysfs statistics file -*/ + * @name: short name for this device ("SSI0", "SSI1", etc) + * @ssi: pointer to the SSI's registers + * @ssi_phys: physical address of the SSI registers + * @irq: IRQ of this SSI + * @dev: struct device pointer + * @playback: the number of playback streams opened + * @capture: the number of capture streams opened + * @cpu_dai: the CPU DAI for this device + * @dev_attr: the sysfs device attribute structure + * @stats: SSI statistics + */ struct fsl_ssi_info { + char name[16]; unsigned int id; struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; struct device *dev; -}; + unsigned int playback; + unsigned int capture; + struct snd_soc_dai *dai; + struct device_attribute dev_attr; -struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); -void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai); + unsigned int dai_format; + unsigned int codec_clk_direction; + unsigned int cpu_clk_direction; + unsigned long clk_frequency; + + struct fsl_dma_info dma_info[2]; + + struct { + unsigned int rfrc; + unsigned int tfrc; + unsigned int cmdau; + unsigned int cmddu; + unsigned int rxt; + unsigned int rdr1; + unsigned int rdr0; + unsigned int tde1; + unsigned int tde0; + unsigned int roe1; + unsigned int roe0; + unsigned int tue1; + unsigned int tue0; + unsigned int tfs; + unsigned int rfs; + unsigned int tls; + unsigned int rls; + unsigned int rff1; + unsigned int rff0; + unsigned int tfe1; + unsigned int tfe0; + } stats; +}; #endif diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c dissimilarity index 60% index e154e3f..a857e57 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -1,642 +1,353 @@ -/** - * Freescale MPC8610HPCD ALSA SoC Fabric driver - * - * Author: Timur Tabi <timur(a)freescale.com> - * - * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed - * under the terms of the GNU General Public License version 2. This - * program is licensed "as is" without any warranty of any kind, whether - * express or implied. - */ - -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/of_device.h> -#include <linux/of_platform.h> -#include <sound/soc.h> -#include <asm/immap_86xx.h> - -#include "../codecs/cs4270.h" -#include "fsl_dma.h" -#include "fsl_ssi.h" - -/** - * mpc8610_hpcd_data: fabric-specific ASoC device data - * - * This structure contains data for a single sound platform device on an - * MPC8610 HPCD. Some of the data is taken from the device tree. - */ -struct mpc8610_hpcd_data { - struct snd_soc_device sound_devdata; - struct snd_soc_dai_link dai; - struct snd_soc_card soc_card; - unsigned int dai_format; - unsigned int codec_clk_direction; - unsigned int cpu_clk_direction; - unsigned int clk_frequency; - struct ccsr_guts __iomem *guts; - struct ccsr_ssi __iomem *ssi; - unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ - unsigned int ssi_irq; - unsigned int dma_id; /* 0 = DMA1, 1 = DMA2, etc */ - unsigned int dma_irq[2]; - struct ccsr_dma_channel __iomem *dma[2]; - unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ -}; - -/** - * mpc8610_hpcd_audio_init: initalize the board - * - * This function is called when platform_device_add() is called. It is used - * to initialize the board-specific hardware. - * - * Here we program the DMACR and PMUXCR registers. - */ -static int mpc8610_hpcd_audio_init(struct snd_soc_card *soc_card) -{ - struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; - - /* Program the signal routing between the SSI and the DMA */ - guts_set_dmacr(soc_card_data->guts, soc_card_data->dma_id + 1, - soc_card_data->dma_channel_id[0], CCSR_GUTS_DMACR_DEV_SSI); - guts_set_dmacr(soc_card_data->guts, soc_card_data->dma_id + 1, - soc_card_data->dma_channel_id[1], CCSR_GUTS_DMACR_DEV_SSI); - - guts_set_pmuxcr_dma(soc_card_data->guts, soc_card_data->dma_id, - soc_card_data->dma_channel_id[0], 0); - guts_set_pmuxcr_dma(soc_card_data->guts, soc_card_data->dma_id, - soc_card_data->dma_channel_id[1], 0); - - guts_set_pmuxcr_dma(soc_card_data->guts, 1, 0, 0); - guts_set_pmuxcr_dma(soc_card_data->guts, 1, 3, 0); - guts_set_pmuxcr_dma(soc_card_data->guts, 0, 3, 0); - - switch (soc_card_data->ssi_id) { - case 0: - clrsetbits_be32(&soc_card_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI); - break; - case 1: - clrsetbits_be32(&soc_card_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI); - break; - } - - return 0; -} - -/** - * mpc8610_hpcd_startup: program the board with various hardware parameters - * - * This function takes board-specific information, like clock frequencies - * and serial data formats, and passes that information to the codec and - * transport drivers. - */ -static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct mpc8610_hpcd_data *soc_card_data = - rtd->socdev->dev->platform_data; - int ret = 0; - - /* Tell the CPU driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(cpu_dai, soc_card_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver audio format\n"); - return ret; - } - - /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(codec_dai, soc_card_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver audio format\n"); - return ret; - } - - /* - * Tell the CPU driver what the clock frequency is, and whether it's a - * slave or master. - */ - ret = snd_soc_dai_set_sysclk(cpu_dai, 0, soc_card_data->clk_frequency, - soc_card_data->cpu_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver clock parameters\n"); - return ret; - } - - /* - * Tell the codec driver what the MCLK frequency is, and whether it's - * a slave or master. - */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, soc_card_data->clk_frequency, - soc_card_data->codec_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver clock params\n"); - return ret; - } - - return 0; -} - -/** - * mpc8610_hpcd_audio_exit: Remove the sound device - * - * This function is called to remove the sound device for one SSI. We - * de-program the DMACR and PMUXCR register. - */ -int mpc8610_hpcd_audio_exit(struct snd_soc_card *soc_card) -{ - struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; - - /* Restore the signal routing */ - - guts_set_dmacr(soc_card_data->guts, soc_card_data->dma_id + 1, - soc_card_data->dma_channel_id[0], 0); - guts_set_dmacr(soc_card_data->guts, soc_card_data->dma_id + 1, - soc_card_data->dma_channel_id[1], 0); - - switch (soc_card_data->ssi_id) { - case 0: - clrsetbits_be32(&soc_card_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); - break; - case 1: - clrsetbits_be32(&soc_card_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); - break; - } - - return 0; -} - -/** - * mpc8610_hpcd_ops: ASoC fabric driver operations - */ -static struct snd_soc_ops mpc8610_hpcd_ops = { - .startup = mpc8610_hpcd_startup, -}; - -/** - * mpc8610_hpcd_probe: OF probe function for the fabric driver - * - * This function gets called when an SSI node is found in the device tree. - * - * Although this is a fabric driver, the SSI node is the "master" node with - * respect to audio hardware connections. Therefore, we create a new ASoC - * device for each new SSI node that has a codec attached. - * - * FIXME: Currently, we only support one DMA controller, so if there are - * multiple SSI nodes with codecs, only the first will be supported. - * - * FIXME: Even if we did support multiple DMA controllers, we have no - * mechanism for assigning DMA controllers and channels to the individual - * SSI devices. We also probably aren't compatible with the generic Elo DMA - * device driver. - */ -static int mpc8610_hpcd_probe(struct of_device *ofdev, - const struct of_device_id *match) -{ - struct snd_soc_card *soc_card; - struct device_node *np = ofdev->node; - struct device_node *codec_np = NULL; - struct device_node *guts_np = NULL; - struct device_node *dma_np = NULL; - struct device_node *dma_channel_np = NULL; - const phandle *codec_ph; - const char *sprop; - const u32 *iprop; - struct resource res; - struct mpc8610_hpcd_data *soc_card_data; - -/* TODO: ssi_info and dma_info could now be created in the platform driver - * during it's probe. A lot of this code could be moved to platform probe() */ - - struct fsl_ssi_info ssi_info; - struct fsl_dma_info dma_info; - int ret = -ENODEV; - - soc_card_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); - if (!soc_card_data) - return -ENOMEM; - - memset(&ssi_info, 0, sizeof(ssi_info)); - memset(&dma_info, 0, sizeof(dma_info)); - - ssi_info.dev = &ofdev->dev; - - /* - * We are only interested in SSIs with a codec phandle in them, so let's - * make sure this SSI has one. - */ - codec_ph = of_get_property(np, "codec-handle", NULL); - if (!codec_ph) - goto error; - - codec_np = of_find_node_by_phandle(*codec_ph); - if (!codec_np) - goto error; - - /* The MPC8610 HPCD only knows about the CS4270 codec, so reject - anything else. */ - if (!of_device_is_compatible(codec_np, "cirrus,cs4270")) - goto error; - - /* Get the device ID */ - iprop = of_get_property(np, "cell-index", NULL); - if (!iprop) { - dev_err(&ofdev->dev, "cell-index property not found\n"); - ret = -EINVAL; - goto error; - } - soc_card_data->ssi_id = *iprop; - ssi_info.id = *iprop; - - /* Get the serial format and clock direction. */ - sprop = of_get_property(np, "fsl,mode", NULL); - if (!sprop) { - dev_err(&ofdev->dev, "fsl,mode property not found\n"); - ret = -EINVAL; - goto error; - } - - if (strcasecmp(sprop, "i2s-slave") == 0) { - soc_card_data->dai_format = SND_SOC_DAIFMT_I2S; - soc_card_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - soc_card_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - - /* - * In i2s-slave mode, the codec has its own clock source, so we - * need to get the frequency from the device tree and pass it to - * the codec driver. - */ - iprop = of_get_property(codec_np, "clock-frequency", NULL); - if (!iprop || !*iprop) { - dev_err(&ofdev->dev, "codec bus-frequency property " - "is missing or invalid\n"); - ret = -EINVAL; - goto error; - } - soc_card_data->clk_frequency = *iprop; - } else if (strcasecmp(sprop, "i2s-master") == 0) { - soc_card_data->dai_format = SND_SOC_DAIFMT_I2S; - soc_card_data->codec_clk_direction = SND_SOC_CLOCK_IN; - soc_card_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "lj-slave") == 0) { - soc_card_data->dai_format = SND_SOC_DAIFMT_LEFT_J; - soc_card_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - soc_card_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "lj-master") == 0) { - soc_card_data->dai_format = SND_SOC_DAIFMT_LEFT_J; - soc_card_data->codec_clk_direction = SND_SOC_CLOCK_IN; - soc_card_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "rj-master") == 0) { - soc_card_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; - soc_card_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - soc_card_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "rj-master") == 0) { - soc_card_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; - soc_card_data->codec_clk_direction = SND_SOC_CLOCK_IN; - soc_card_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "ac97-slave") == 0) { - soc_card_data->dai_format = SND_SOC_DAIFMT_AC97; - soc_card_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - soc_card_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "ac97-master") == 0) { - soc_card_data->dai_format = SND_SOC_DAIFMT_AC97; - soc_card_data->codec_clk_direction = SND_SOC_CLOCK_IN; - soc_card_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else { - dev_err(&ofdev->dev, - "unrecognized fsl,mode property \"%s\"\n", sprop); - ret = -EINVAL; - goto error; - } - - if (!soc_card_data->clk_frequency) { - dev_err(&ofdev->dev, "unknown clock frequency\n"); - ret = -EINVAL; - goto error; - } - - /* Read the SSI information from the device tree */ - ret = of_address_to_resource(np, 0, &res); - if (ret) { - dev_err(&ofdev->dev, "could not obtain SSI address\n"); - goto error; - } - if (!res.start) { - dev_err(&ofdev->dev, "invalid SSI address\n"); - goto error; - } - ssi_info.ssi_phys = res.start; - - soc_card_data->ssi = ioremap(ssi_info.ssi_phys, sizeof(struct ccsr_ssi)); - if (!soc_card_data->ssi) { - dev_err(&ofdev->dev, "could not map SSI address %x\n", - ssi_info.ssi_phys); - ret = -EINVAL; - goto error; - } - ssi_info.ssi = soc_card_data->ssi; - - - /* Get the IRQ of the SSI */ - soc_card_data->ssi_irq = irq_of_parse_and_map(np, 0); - if (!soc_card_data->ssi_irq) { - dev_err(&ofdev->dev, "could not get SSI IRQ\n"); - ret = -EINVAL; - goto error; - } - ssi_info.irq = soc_card_data->ssi_irq; - - - /* Map the global utilities registers. */ - guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); - if (!guts_np) { - dev_err(&ofdev->dev, "could not obtain address of GUTS\n"); - ret = -EINVAL; - goto error; - } - soc_card_data->guts = of_iomap(guts_np, 0); - of_node_put(guts_np); - if (!soc_card_data->guts) { - dev_err(&ofdev->dev, "could not map GUTS\n"); - ret = -EINVAL; - goto error; - } - - /* Find the DMA channels to use. For now, we always use the first DMA - controller. */ - for_each_compatible_node(dma_np, NULL, "fsl,mpc8610-dma") { - iprop = of_get_property(dma_np, "cell-index", NULL); - if (iprop && (*iprop == 0)) { - of_node_put(dma_np); - break; - } - } - if (!dma_np) { - dev_err(&ofdev->dev, "could not find DMA node\n"); - ret = -EINVAL; - goto error; - } - soc_card_data->dma_id = *iprop; - - /* - * Find the DMA channels to use. For now, we always use DMA channel 0 - * for playback, and DMA channel 1 for capture. - */ - while ((dma_channel_np = of_get_next_child(dma_np, dma_channel_np))) { - iprop = of_get_property(dma_channel_np, "cell-index", NULL); - /* Is it DMA channel 0? */ - if (iprop && (*iprop == 0)) { - /* dma_channel[0] and dma_irq[0] are for playback */ - dma_info.dma_channel[0] = of_iomap(dma_channel_np, 0); - dma_info.dma_irq[0] = - irq_of_parse_and_map(dma_channel_np, 0); - soc_card_data->dma_channel_id[0] = *iprop; - continue; - } - if (iprop && (*iprop == 1)) { - /* dma_channel[1] and dma_irq[1] are for capture */ - dma_info.dma_channel[1] = of_iomap(dma_channel_np, 0); - dma_info.dma_irq[1] = - irq_of_parse_and_map(dma_channel_np, 0); - soc_card_data->dma_channel_id[1] = *iprop; - continue; - } - } - if (!dma_info.dma_channel[0] || !dma_info.dma_channel[1] || - !dma_info.dma_irq[0] || !dma_info.dma_irq[1]) { - dev_err(&ofdev->dev, "could not find DMA channels\n"); - ret = -EINVAL; - goto error; - } - - dma_info.ssi_stx_phys = ssi_info.ssi_phys + - offsetof(struct ccsr_ssi, stx0); - dma_info.ssi_srx_phys = ssi_info.ssi_phys + - offsetof(struct ccsr_ssi, srx0); - - /* We have the DMA information, so tell the DMA driver what it is */ - if (!fsl_dma_configure(&dma_info)) { - dev_err(&ofdev->dev, "could not instantiate DMA device\n"); - ret = -EBUSY; - goto error; - } - - -#if 0 /* V1 TODO: remove */ - /* - * Initialize our DAI data structure. We should probably get this - * information from the device tree. - */ - soc_card_data->dai.name = "CS4270"; - soc_card_data->dai.stream_name = "CS4270"; - - soc_card_data->dai.cpu_dai = fsl_ssi_create_dai(&ssi_info); - soc_card_data->dai.codec_dai = &cs4270_dai; /* The codec_dai we want */ - soc_card_data->dai.ops = &mpc8610_hpcd_ops; - - mpc8610_hpcd_soc_card.dai_link = &soc_card_data->dai; - - /* Allocate a new audio platform device structure */ - sound_device = platform_device_alloc("soc-audio", -1); - if (!sound_device) { - dev_err(&ofdev->dev, "platform device allocation failed\n"); - ret = -ENOMEM; - goto error; - } - - soc_card_data->sound_devdata.soc_card = &mpc8610_hpcd_soc_card; - soc_card_data->sound_devdata.codec_dev = &soc_codec_device_cs4270; - soc_card_data->sound_devdata.platform = &fsl_soc_platform; - - sound_device->dev.platform_data = soc_card_data; - - - /* Set the platform device and ASoC device to point to each other */ - platform_set_drvdata(sound_device, &soc_card_data->sound_devdata); - - soc_card_data->sound_devdata.dev = &sound_device->dev; - - - /* Tell ASoC to probe us. This will call mpc8610_hpcd_soc_card.probe(), - if it exists. */ - ret = platform_device_add(sound_device); - - if (ret) { - dev_err(&ofdev->dev, "platform device add failed\n"); - goto error; - } -#else /* V2 */ - soc_card = snd_soc_card_create("MPC8610", &ofdev->dev, - SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (soc_card == NULL) - return -ENOMEM; - - soc_card->longname = "CS4270"; - soc_card->init = mpc8610_hpcd_audio_init; - soc_card->exit = mpc8610_hpcd_audio_exit; - soc_card->private_data = soc_card_data; - - /* TODO: the number, ID and config of codecs, platforms, - * pcm's could be found in dev tree */ - ret = snd_soc_codec_create(soc_card, cs4270_codec_id); - if (ret < 0) - goto err; - - ret = snd_soc_platform_create(soc_card, fsl_platform_id); - if (ret < 0) - goto err; - - ret = snd_soc_pcm_create(soc_card, "HiFi", &mpc8610_hpcd_ops, - CS4270_HIFI_DAI, FSL_DAI_SSI0, 1, 1); - if (ret < 0) - goto err; - - /* every has been added at this point */ - dev_set_drvdata(&ofdev->dev, soc_card); - ret = snd_soc_card_register(soc_card); - if (ret < 0) - goto err; - -#endif - - dev_set_drvdata(&ofdev->dev, sound_device); - - return 0; - -error: - - of_node_put(codec_np); - of_node_put(guts_np); - of_node_put(dma_np); - of_node_put(dma_channel_np); - - if (ssi_info.ssi) - iounmap(ssi_info.ssi); - - if (ssi_info.irq) - irq_dispose_mapping(ssi_info.irq); - - if (dma_info.dma_channel[0]) - iounmap(dma_info.dma_channel[0]); - - if (dma_info.dma_channel[1]) - iounmap(dma_info.dma_channel[1]); - - if (dma_info.dma_irq[0]) - irq_dispose_mapping(dma_info.dma_irq[0]); - - if (dma_info.dma_irq[1]) - irq_dispose_mapping(dma_info.dma_irq[1]); - - if (soc_card_data->guts) - iounmap(soc_card_data->guts); - - kfree(soc_card_data); - - snd_soc_card_free(soc_card); - return ret; -} - -/** - * mpc8610_hpcd_remove: remove the OF device - * - * This function is called when the OF device is removed. - */ -static int mpc8610_hpcd_remove(struct of_device *ofdev) -{ - struct snd_soc_card *soc_card = dev_get_drvdata(&ofdev->dev); - struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; - -/* TODO: some of this will move to platform remove() */ - - if (soc_card_data->dai.cpu_dai) - fsl_ssi_destroy_dai(soc_card_data->dai.cpu_dai); - - if (soc_card_data->ssi) - iounmap(soc_card_data->ssi); - - if (soc_card_data->dma[0]) - iounmap(soc_card_data->dma[0]); - - if (soc_card_data->dma[1]) - iounmap(soc_card_data->dma[1]); - - if (soc_card_data->dma_irq[0]) - irq_dispose_mapping(soc_card_data->dma_irq[0]); - - if (soc_card_data->dma_irq[1]) - irq_dispose_mapping(soc_card_data->dma_irq[1]); - - if (soc_card_data->guts) - iounmap(soc_card_data->guts); - - kfree(soc_card_data); - - snd_soc_card_free(soc_card); - - dev_set_drvdata(&ofdev->dev, NULL); - - return 0; -} - -static struct of_device_id mpc8610_hpcd_match[] = { - { - .compatible = "fsl,mpc8610-ssi", - }, - {} -}; -MODULE_DEVICE_TABLE(of, mpc8610_hpcd_match); - -static struct of_platform_driver mpc8610_hpcd_of_driver = { - .owner = THIS_MODULE, - .name = "mpc8610_hpcd", - .match_table = mpc8610_hpcd_match, - .probe = mpc8610_hpcd_probe, - .remove = mpc8610_hpcd_remove, -}; - -/** - * mpc8610_hpcd_init: fabric driver initialization. - * - * This function is called when this module is loaded. - */ -static int __init mpc8610_hpcd_init(void) -{ - int ret; - - printk(KERN_INFO "Freescale MPC8610 HPCD ALSA SoC fabric driver\n"); - - ret = of_register_platform_driver(&mpc8610_hpcd_of_driver); - - if (ret) - printk(KERN_ERR - "mpc8610-hpcd: failed to register platform driver\n"); - - return ret; -} - -/** - * mpc8610_hpcd_exit: fabric driver exit - * - * This function is called when this driver is unloaded. - */ -static void __exit mpc8610_hpcd_exit(void) -{ - of_unregister_platform_driver(&mpc8610_hpcd_of_driver); -} - -module_init(mpc8610_hpcd_init); -module_exit(mpc8610_hpcd_exit); - -MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); -MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC fabric driver"); -MODULE_LICENSE("GPL"); +/** + * Freescale MPC8610HPCD ALSA SoC Fabric driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed + * under the terms of the GNU General Public License version 2. This + * program is licensed "as is" without any warranty of any kind, whether + * express or implied. + */ + +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/of_device.h> +#include <linux/of_platform.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <asm/immap_86xx.h> + +#include "../codecs/cs4270.h" +#include "fsl_dma.h" +#include "fsl_ssi.h" + +/** + * mpc8610_hpcd_data: fabric-specific ASoC device data + * + * This structure contains data for a single sound platform device on an + * MPC8610 HPCD. Some of the data is taken from the device tree. + */ +struct mpc8610_hpcd_data { + struct snd_soc_card soc_card; + unsigned int dai_format; + unsigned int codec_clk_direction; + unsigned int cpu_clk_direction; + unsigned int clk_frequency; + struct ccsr_guts __iomem *guts; +}; + +/** + * mpc8610_hpcd_audio_init: initalize the board + * + * This function is called when platform_device_add() is called. It is used + * to initialize the board-specific hardware. + * + * Here we program the DMACR and PMUXCR registers. + */ +static int mpc8610_hpcd_audio_init(struct snd_soc_card *soc_card) +{ + struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; + struct snd_soc_dai *cpu_dai = + snd_soc_card_get_dai(soc_card, "fsl,mpc8610-ssi"); + struct fsl_ssi_info *ssi_info = cpu_dai->private_data; + + /* Program the signal routing between the SSI and the DMA */ + guts_set_dmacr(soc_card_data->guts, + ssi_info->dma_info[0].controller_id, + ssi_info->dma_info[0].channel_id, CCSR_GUTS_DMACR_DEV_SSI); + guts_set_dmacr(soc_card_data->guts, + ssi_info->dma_info[1].controller_id, + ssi_info->dma_info[1].channel_id, CCSR_GUTS_DMACR_DEV_SSI); + + guts_set_pmuxcr_dma(soc_card_data->guts, + ssi_info->dma_info[0].controller_id, + ssi_info->dma_info[0].channel_id, 0); + guts_set_pmuxcr_dma(soc_card_data->guts, + ssi_info->dma_info[1].controller_id, + ssi_info->dma_info[1].channel_id, 0); + + /* FIXME: Magic numbers? */ + guts_set_pmuxcr_dma(soc_card_data->guts, 1, 0, 0); + guts_set_pmuxcr_dma(soc_card_data->guts, 1, 3, 0); + guts_set_pmuxcr_dma(soc_card_data->guts, 0, 3, 0); + + switch (ssi_info->id) { + case 0: + clrsetbits_be32(&soc_card_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI); + break; + case 1: + clrsetbits_be32(&soc_card_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI); + break; + } + + return 0; +} + +/** + * mpc8610_hpcd_startup: program the board with various hardware parameters + * + * This function takes board-specific information, like clock frequencies + * and serial data formats, and passes that information to the codec and + * transport drivers. + */ +static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct mpc8610_hpcd_data *soc_card_data = rtd->soc_card->private_data; + int ret = 0; + + /* Tell the CPU driver what the serial protocol is. */ + ret = snd_soc_dai_set_fmt(cpu_dai, soc_card_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver audio format\n"); + return ret; + } + + /* Tell the codec driver what the serial protocol is. */ + ret = snd_soc_dai_set_fmt(codec_dai, soc_card_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver audio format\n"); + return ret; + } + + /* + * Tell the CPU driver what the clock frequency is, and whether it's a + * slave or master. + */ + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, soc_card_data->clk_frequency, + soc_card_data->cpu_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver clock parameters\n"); + return ret; + } + + /* + * Tell the codec driver what the MCLK frequency is, and whether it's + * a slave or master. + */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, soc_card_data->clk_frequency, + soc_card_data->codec_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver clock params\n"); + return ret; + } + + return 0; +} + +/** + * mpc8610_hpcd_audio_exit: Remove the sound device + * + * This function is called to remove the sound device for one SSI. We + * de-program the DMACR and PMUXCR register. + */ +void mpc8610_hpcd_audio_exit(struct snd_soc_card *soc_card) +{ + struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; + struct snd_soc_dai *cpu_dai = + snd_soc_card_get_dai(soc_card, "fsl,mpc8610-ssi"); + struct fsl_ssi_info *ssi_info = cpu_dai->private_data; + + /* Restore the signal routing */ + + guts_set_dmacr(soc_card_data->guts, + ssi_info->dma_info[0].controller_id, + ssi_info->dma_info[0].channel_id, 0); + guts_set_dmacr(soc_card_data->guts, + ssi_info->dma_info[1].controller_id, + ssi_info->dma_info[1].channel_id, 0); + + switch (ssi_info->id) { + case 0: + clrsetbits_be32(&soc_card_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); + break; + case 1: + clrsetbits_be32(&soc_card_data->guts->pmuxcr, + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); + break; + } +} + +/** + * mpc8610_hpcd_ops: ASoC fabric driver operations + */ +static struct snd_soc_ops mpc8610_hpcd_ops = { + .startup = mpc8610_hpcd_startup, +}; + +static struct snd_soc_pcm_config mpc8610_pcm_config = { + .name = "MPC8610 HPCD", + .codec = "cirrus,cs4270", + .codec_dai = "cirrus,cs4270", + .platform = "fsl_pcm", + .cpu_dai = "fsl,mpc8610-ssi", + .ops = &mpc8610_hpcd_ops, + .playback = 1, + .capture = 1, +}; + +/** + * mpc8610_hpcd_probe: OF probe function for the fabric driver + * + * This function gets called when an SSI node is found in the device tree. + * + * Although this is a fabric driver, the SSI node is the "master" node with + * respect to audio hardware connections. Therefore, we create a new ASoC + * device for each new SSI node that has a codec attached. + */ +static int mpc8610_hpcd_probe(struct of_device *ofdev, + const struct of_device_id *match) +{ + struct snd_soc_card *soc_card = NULL; + struct device_node *guts_np = NULL; + struct mpc8610_hpcd_data *soc_card_data; + int ret = -ENODEV; + + soc_card_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); + if (!soc_card_data) { + dev_err(&ofdev->dev, "could not allocate card structure\n"); + return -ENOMEM; + } + + /* Map the global utilities registers. */ + guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); + if (!guts_np) { + dev_err(&ofdev->dev, "could not obtain address of GUTS\n"); + ret = -EINVAL; + goto error; + } + soc_card_data->guts = of_iomap(guts_np, 0); + of_node_put(guts_np); + if (!soc_card_data->guts) { + dev_err(&ofdev->dev, "could not map GUTS\n"); + ret = -EINVAL; + goto error; + } + + soc_card = snd_soc_card_create("MPC8610", &ofdev->dev, + SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (!soc_card) { + dev_err(&ofdev->dev, "could not create card\n"); + goto error; + } + + soc_card->longname = "MPC8610HPCD"; + soc_card->init = mpc8610_hpcd_audio_init; + soc_card->exit = mpc8610_hpcd_audio_exit; + soc_card->private_data = soc_card_data; + soc_card->dev = &ofdev->dev; + + ret = snd_soc_card_create_pcms(soc_card, &mpc8610_pcm_config, 1); + if (ret) { + dev_err(&ofdev->dev, "could not create PCMs\n"); + goto error; + } + + /* every has been added at this point */ + dev_set_drvdata(&ofdev->dev, soc_card); + ret = snd_soc_card_register(soc_card); + if (ret) { + dev_err(&ofdev->dev, "could not register card\n"); + goto error; + } + + return 0; + +error: + if (soc_card_data->guts) + iounmap(soc_card_data->guts); + + kfree(soc_card_data); + + if (soc_card) + snd_soc_card_free(soc_card); + + return ret; +} + +/** + * mpc8610_hpcd_remove: remove the OF device + * + * This function is called when the OF device is removed. + */ +static int mpc8610_hpcd_remove(struct of_device *ofdev) +{ + struct snd_soc_card *soc_card = dev_get_drvdata(&ofdev->dev); + struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; + + if (soc_card_data->guts) + iounmap(soc_card_data->guts); + + kfree(soc_card_data); + + snd_soc_card_free(soc_card); + + dev_set_drvdata(&ofdev->dev, NULL); + + return 0; +} + +static struct of_device_id mpc8610_hpcd_match[] = { + { + .compatible = "fsl,mpc8610-ssi", + }, + {} +}; +MODULE_DEVICE_TABLE(of, mpc8610_hpcd_match); + +static struct of_platform_driver mpc8610_hpcd_of_driver = { + .owner = THIS_MODULE, + .name = "mpc8610_hpcd", + .match_table = mpc8610_hpcd_match, + .probe = mpc8610_hpcd_probe, + .remove = mpc8610_hpcd_remove, +}; + +/** + * mpc8610_hpcd_init: fabric driver initialization. + * + * This function is called when this module is loaded. + */ +static int __init mpc8610_hpcd_init(void) +{ + int ret; + + printk(KERN_INFO "Freescale MPC8610 HPCD ALSA SoC fabric driver\n"); + + ret = of_register_platform_driver(&mpc8610_hpcd_of_driver); + + if (ret) + printk(KERN_ERR + "mpc8610-hpcd: failed to register platform driver\n"); + + return ret; +} + +/** + * mpc8610_hpcd_exit: fabric driver exit + * + * This function is called when this driver is unloaded. + */ +static void __exit mpc8610_hpcd_exit(void) +{ + of_unregister_platform_driver(&mpc8610_hpcd_of_driver); +} + +module_init(mpc8610_hpcd_init); +module_exit(mpc8610_hpcd_exit); + +MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); +MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC fabric driver"); +MODULE_LICENSE("GPL"); -- 1.5.5

1 0

[alsa-devel] ASoC problem
by Alan Lu 16 May '08

16 May '08

Hi, I built linux-2.6.25.2 kernel with ASoC driver for SMDK2412 board. The following is dmesg, Advanced Linux Sound Architecture Driver Version 1.0.16rc2 (Thu Jan 31 16:40:16 ASoC version 0.13.2 ALSA device list: No soundcards found. I traced the soc_core.c and it never reaches soc_probe() routine. That's why "No soundcards found." Could you give me some light ? Appreciated!

5 10

[alsa-devel] Buffering issues 2
by Lennart Poettering 16 May '08

16 May '08

Heya! In addition to the buffering issues pointed out on http://mailman.alsa-project.org/pipermail/alsa-devel/2008-April/007354.html I now have some more issues with the way buffering works in ALSA. This time I am pretty sure it's a bug in the HDA drivers. PulseAudio basically does this (again, as part of the glitch-free playback model, see http://0pointer.de/blog/projects/pulse-glitch-free.html): for (;;) { for (;;) { snd_pcm_hwsync(); n = snd_pcm_avail_update(); if (!n) break; fill_up(n); } usleep(buffer_time - 20ms); } The buffer time is 370ms, so we usually sleep for 350ms. It's a lot more complex actually, due to deviating timing and some code that we don't enter a busy loop if the CPU is slower then the sound card and so on and so on. However, basically it's this algorithm. This works fine on USB. However on HDA it sometimes happens that snd_pcm_avail_update() is not properly updated at the right time. Instead it seems to return what was current one iteration earlier. I.e. we fill up the hw buffer until _avail_update() returns 0. Then assuming that it is fully filled up we go to sleep for 350ms. When we wake up we query _avail_update() again and it will immediately return 0. We thus don't write anything to the device. Instead we go to sleep for another 350ms. Of course, when we wake up again the buffer will already have underrun (since 700 ms passed since the last write to the buffer and th buffer is only 370ms long) and we have a problem. In short: _avail_update() told us that everything was alright and we trusted it and it lied to us. The happens to be configured to 2 periods. So even if _avail_update() has some kind of period granularity (?) it should always have told us after we come back from the sleeping that at least *one* period is free. But it said 0 frames. Nada. Rien. Nichts. Kaputt. I *am* calling snd_pcm_hwsync(). So I'd assume that _avail_update() would be as up to date as it gets. But it isn't. :-( This happens regardless if I open the device as "front:0" (i.e. with softvol in line) or as "hw:0" (no plugins). This doesn't always happen. It seems to depend on the machine how often this happens. On one machine I can reliably reproduce this after 30 iterations. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

3 5

Re: [alsa-devel] Intel hda ICH7 not worked anymore after using wine (SOLVED!). Problems again
by Xavi de Blas 16 May '08

16 May '08

Hello, we have been talking on this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-July/001939.html because i had a problem with my hda that Takashi said it's really specific to my hardware: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-July/001963.html Now with kernel 2.6.24-16 and alsa 1.0.16, i have no audio again: i tried now using the patch Tobin made: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-July/001934.html and it' doesn't work. In fact, i haven't used the patch, i only added the line: { .id = 0x51d41981, .name = "AD1981", .patch = patch_ad1981 }, after the line: { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, on: /usr/local/src/alsa-driver-1.0.16/alsa-kernel/pci/hda/patch_analog.c ./configure, make, sudo make install on reboot, the error is the same as before: dmesg|grep alsa [ 41.284537] ALSA /build/buildd/linux-ubuntu-modules-2.6.24-2.6.24/debian/build/build-generic/sound/alsa-driver/pci/hda/hda_codec.c:230: hda_codec: invalid dep_range_val 53:1a here is my /proc/asound/card0/codec#0 http://pastebin.ca/1018596 Thanks a lot 2007/7/9 Xavi de Blas <xaviblas(a)gmail.com>: > It worked!!! > > Thanks a lot > > I thought my hardware was broken! > > 2007/7/9, Tobin Davis <tdavis(a)dsl-only.net>: >> >> Patch needs to be applied as follows: >> >> cd alsa-driver-1.0.14/alsa-kernel >> patch -p1 < LenovoX60S.patch >> cd .. >> >> Then you can rerun configure && make && make install. >> >> Tobin >> >> On Mon, 2007-07-09 at 17:25 +0200, Xavi de Blas wrote: >> >> 2007/7/9, Tobin Davis <tdavis(a)dsl-only.net>: >> > Sorry for missing this thread. I just woke up. >> > >> > I looked at your first pastebin output, and the actual audio codec shows >> > up as 0x51d41981, when it should be 0x11d41981. This is why the Analog >> > driver patch isn't detecting it. >> > >> > Can you try this patch against alsa-driver-1.0.14? It should detect >> > this codec. Still not sure why it is misconfigured, unless there is a >> > bios issue. >> > >> >> I'm afraid, probably when i tried the Linuxant codecs for the modem it >> broke the >> bios. Not sure, but sound worked, and continued working some reboots >> later. But when i tried a few things with wine and rebooted, sound >> never worked again >> >> When i aply the patch (probably i don't do it good) there are problems: >> >> >> xavier@corall:/usr/local/src/alsa-src/alsa-driver-1.0.14$ cat >> LenovoX60S.patch | patch -p1 >> patching file pci/hda/patch_analog.c >> Hunk #1 FAILED at 3402. >> 1 out of 1 hunk FAILED -- saving rejects to file >> pci/hda/patch_analog.c.rej >> >> >> xavier@corall:/usr/local/src/alsa-src/alsa-driver-1.0.14/pci/hda$ cat >> patch_analog.c.rej >> *************** struct hda_codec_preset snd_hda_preset_a >> *** 3402,3407 **** >> { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, >> { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, >> { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, >> { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, >> { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 }, >> { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, >> --- 3402,3408 ---- >> { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, >> { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, >> { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, >> + { .id = 0x51d41981, .name = "AD1981", .patch = patch_ad1981 }, >> { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, >> { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 }, >> { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, >> >> >> The patch is ok? >> >> Thanks >> >> (Tobin sorry for sending messge two times) >> >> -- >> Tobin Davis <tdavis(a)dsl-only.net> >

2 3

[alsa-devel] [PATCH] PCI168 snd-azt3328 Linux driver: another huge update
by Andreas Mohr 16 May '08

16 May '08

Hello all (and especially my fellow driver users!), ok, color me crazy, but somehow I really felt like doing another round of Undocumented Hacking (tm) again, with my nice sound card, after an overly long total inactivity. - figured out "Digital(ly) Enhanced Game Port" functionality, implemented support for it (eliminating gameport polling overhead) - removed optional joystick activation, gameport now enabled unconditionally, since we now support it via the PCI I/O space, not via conflict-prone legacy I/O (which I was thus able to DISABLE now)! - fix playback bug (a muted wave output would get unmuted upon start of playback, of course this is not what we want, thus remember mute state) - implement partial power management: when idle, lower clock rate and disable codec (reduced noise!), and disable gameport circuit when unused - instantiate OPL3 timer, too - much better implementation of snd_azf3328_mixer_write_volume_gradually() - slightly optimized interrupt handling - lots of cleanup This time, I also found a way to verify proper OPL3 operation via MIDI file playback (emulation via synth hardware). Patch done and hardware-tested against 2.6.26-rc1, which should hopefully match against current ALSA. Everything checked against checkpatch.pl (lots of corrections there as well). Oh, and please don't ask me to split up everything again, lest I'll kill myself - way too many changes this time, and frankly I don't care whether it's debuggable or not since probably not too many people use this driver anyway and I'll fix any problems if/when they turn up (I'm still quite actively enjoying this card myself, thank you very much ;). Note that the new version has an object size increase of merely about 500 bytes, despite all these improvements! Given these additions, the azt3328 driver should now easily have managed to make the PCI168 the best-supported fully undocumented soundcard in the Linux universe ;) (one could say that this card is "a wee bit too old" already, but it's giving me a very nice peace of mind to know that it's almost fully supported now, and that it's now a prime example of really good Linux hardware support) Signed-off-by: Andreas Mohr <andi(a)lisas.de> --- linux-2.6.26-rc1.orig/sound/pci/azt3328.c 2008-05-03 20:59:44.000000000 +0200 +++ linux-2.6.26-rc1/sound/pci/azt3328.c 2008-05-14 22:47:15.000000000 +0200 @@ -1,6 +1,6 @@ /* * azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168). - * Copyright (C) 2002, 2005, 2006, 2007 by Andreas Mohr <andi AT lisas.de> + * Copyright (C) 2002, 2005 - 2008 by Andreas Mohr <andi AT lisas.de> * * Framework borrowed from Bart Hartgers's als4000.c. * Driver developed on PCI168 AP(W) version (PCI rev. 10, subsystem ID 1801), @@ -35,9 +35,20 @@ * (3 weeks' worth of evenings filled with driver work). * (and no, I did NOT go the easy way: to pick up a SB PCI128 for 9 Euros) * + * It is quite likely that the AZF3328 chip is the PCI cousin of the + * AZF3318 ("azt1020 pnp", "MM Pro 16") ISA chip, given very similar specs. + * * The AZF3328 chip (note: AZF3328, *not* AZT3328, that's just the driver name - * for compatibility reasons) has the following features: + * for compatibility reasons) from Azfin (joint-venture of Aztech and Fincitec, + * Fincitec acquired by National Semiconductor in 2002, together with the + * Fincitec-related company ARSmikro) has the following features: * + * - compatibility & compliance: + * - Microsoft PC 97 ("PC 97 Hardware Design Guide", + * http://www.microsoft.com/whdc/archive/pcguides.mspx) + * - Microsoft PC 98 Baseline Audio + * - MPU401 UART + * - Sound Blaster Emulation (DOS Box) * - builtin AC97 conformant codec (SNR over 80dB) * Note that "conformant" != "compliant"!! this chip's mixer register layout * *differs* from the standard AC97 layout: @@ -48,21 +59,28 @@ * addresses illegally. So far unfortunately it looks like the very flexible * ALSA AC97 support is still not enough to easily compensate for such a * grave layout violation despite all tweaks and quirks mechanisms it offers. - * - builtin genuine OPL3 + * - builtin genuine OPL3 - verified to work fine, 20080506 * - full duplex 16bit playback/record at independent sampling rate - * - MPU401 (+ legacy address support) FIXME: how to enable legacy addr?? + * - MPU401 (+ legacy address support, claimed by one official spec sheet) + * FIXME: how to enable legacy addr?? * - game port (legacy address support) - * - builtin 3D enhancement (said to be YAMAHA Ymersion) * - builtin DirectInput support, helps reduce CPU overhead (interrupt-driven - * features supported) + * features supported). - See common term "Digital Enhanced Game Port"... + * (probably DirectInput 3.0 spec - confirm) + * - builtin 3D enhancement (said to be YAMAHA Ymersion) * - built-in General DirectX timer having a 20 bits counter * with 1us resolution (see below!) - * - I2S serial port for external DAC + * - I2S serial output port for external DAC * - supports 33MHz PCI spec 2.1, PCI power management 1.0, compliant with ACPI * - supports hardware volume control * - single chip low cost solution (128 pin QFP) * - supports programmable Sub-vendor and Sub-system ID * required for Microsoft's logo compliance (FIXME: where?) + * At least the Trident 4D Wave DX has one bit somewhere + * to enable writes to PCI subsystem VID registers, that should be it. + * This might easily be in extended PCI reg space, since PCI168 also has + * some custom data starting at 0x80. What kind of config settings + * are located in our extended PCI space anyway?? * - PCI168 AP(W) card: power amplifier with 4 Watts/channel at 4 Ohms * * Note that this driver now is actually *better* than the Windows driver, @@ -74,6 +92,23 @@ * - "timidity -iAv -B2,8 -Os -EFreverb=0" * - "pmidi -p 128:0 jazz.mid" * + * OPL3 hardware playback testing, try something like: + * cat /proc/asound/hwdep + * and + * aconnect -o + * Then use + * sbiload -Dhw:x,y --opl3 /usr/share/sounds/opl3/std.o3 ......./drums.o3 + * where x,y is the xx-yy number as given in hwdep. + * Then try + * pmidi -p 21:0 jazz.mid + * Oh, and make sure to unmute the FM mixer control (doh!) + * NOTE: power use during OPL3 playback is _VERY_ high (70W --> 90W!) + * despite no CPU activity, possibly due to hindering ACPI idling somehow. + * Shouldn't be a problem of the AZF3328 chip itself, I'd hope. + * Higher PCM / FM mixer levels seem to conflict (causes crackling), + * at least sometimes. Maybe even use with hardware sequencer timer above :) + * adplay/adplug-utils might soon offer hardware-based OPL3 playback, too. + * * Certain PCI versions of this card are susceptible to DMA traffic underruns * in some systems (resulting in sound crackling/clicking/popping), * probably because they don't have a DMA FIFO buffer or so. @@ -87,6 +122,8 @@ * better than a VIA, yet ironically I still get crackling, like many other * people with the same chipset. * Possible remedies: + * - use speaker (amplifier) output instead of headphone output + * (in case crackling is due to overloaded output clipping) * - plug card into a different PCI slot, preferrably one that isn't shared * too much (this helps a lot, but not completely!) * - get rid of PCI VGA card, use AGP instead @@ -94,18 +131,23 @@ * - fiddle with PCI latency settings (setpci -v -s BUSID latency_timer=XX) * Not too helpful. * - Disable ACPI/power management/"Auto Detect RAM/PCI Clk" in BIOS - * + * * BUGS * - full-duplex might *still* be problematic, not fully tested recently * - (non-bug) "Bass/Treble or 3D settings don't work" - they do get evaluated * if you set PCM output switch to "pre 3D" instead of "post 3D". * If this can't be set, then get a mixer application that Isn't Stupid (tm) * (e.g. kmix, gamix) - unfortunately several are!! - * + * - locking is not entirely clean, especially the audio stream activity + * ints --> may be racy + * - an _unconnected_ secondary joystick at the gameport will be reported + * to be "active" (floating values, not precisely -1) due to the way we need + * to read the Digital Enhanced Game Port. Not sure whether it is fixable. + * * TODO * - test MPU401 MIDI playback etc. - * - add some power micro-management (disable various units of the card - * as long as they're unused). However this requires I/O ports which I + * - add more power micro-management (disable various units of the card + * as long as they're unused). However this requires more I/O ports which I * haven't figured out yet and which thus might not even exist... * The standard suspend/resume functionality could probably make use of * some improvement, too... @@ -113,6 +155,7 @@ * - figure out some cleverly evil scheme to possibly make ALSA AC97 code * fully accept our quite incompatible ""AC97"" mixer and thus save some * code (but I'm not too optimistic that doing this is possible at all) + * - use MMIO (memory-mapped I/O)? Slightly faster access, e.g. for gameport. */ #include <asm/io.h> @@ -138,7 +181,7 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE)) -#define SUPPORT_JOYSTICK 1 +#define SUPPORT_GAMEPORT 1 #endif #define DEBUG_MISC 0 @@ -147,13 +190,14 @@ #define DEBUG_PLAY_REC 0 #define DEBUG_IO 0 #define DEBUG_TIMER 0 +#define DEBUG_GAME 0 #define MIXER_TESTING 0 #if DEBUG_MISC #define snd_azf3328_dbgmisc(format, args...) printk(KERN_ERR format, ##args) #else #define snd_azf3328_dbgmisc(format, args...) -#endif +#endif #if DEBUG_CALLS #define snd_azf3328_dbgcalls(format, args...) printk(format, ##args) @@ -163,25 +207,31 @@ #define snd_azf3328_dbgcalls(format, args...) #define snd_azf3328_dbgcallenter() #define snd_azf3328_dbgcallleave() -#endif +#endif #if DEBUG_MIXER #define snd_azf3328_dbgmixer(format, args...) printk(format, ##args) #else #define snd_azf3328_dbgmixer(format, args...) -#endif +#endif #if DEBUG_PLAY_REC #define snd_azf3328_dbgplay(format, args...) printk(KERN_ERR format, ##args) #else #define snd_azf3328_dbgplay(format, args...) -#endif +#endif #if DEBUG_MISC #define snd_azf3328_dbgtimer(format, args...) printk(KERN_ERR format, ##args) #else #define snd_azf3328_dbgtimer(format, args...) -#endif +#endif + +#if DEBUG_GAME +#define snd_azf3328_dbggame(format, args...) printk(KERN_ERR format, ##args) +#else +#define snd_azf3328_dbggame(format, args...) +#endif static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ module_param_array(index, int, NULL, 0444); @@ -195,39 +245,44 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable AZF3328 soundcard."); -#ifdef SUPPORT_JOYSTICK -static int joystick[SNDRV_CARDS]; -module_param_array(joystick, bool, NULL, 0444); -MODULE_PARM_DESC(joystick, "Enable joystick for AZF3328 soundcard."); -#endif - static int seqtimer_scaling = 128; module_param(seqtimer_scaling, int, 0444); MODULE_PARM_DESC(seqtimer_scaling, "Set 1024000Hz sequencer timer scale factor (lockup danger!). Default 128."); +typedef struct { + struct snd_pcm_substream *substream; + int enabled; + int running; + unsigned long portbase; +} snd_azf3328_audio_stream_t; + +typedef enum { + AZF_PLAYBACK = 0, + AZF_CAPTURE = 1, +} snd_azf3328_stream_index_t; + struct snd_azf3328 { /* often-used fields towards beginning, then grouped */ - unsigned long codec_port; - unsigned long io2_port; - unsigned long mpu_port; - unsigned long synth_port; - unsigned long mixer_port; + + unsigned long codec_io; /* usually 0xb000, size 128 */ + unsigned long game_io; /* usually 0xb400, size 8 */ + unsigned long mpu_io; /* usually 0xb800, size 4 */ + unsigned long opl3_io; /* usually 0xbc00, size 8 */ + unsigned long mixer_io; /* usually 0xc000, size 64 */ spinlock_t reg_lock; struct snd_timer *timer; - + struct snd_pcm *pcm; - struct snd_pcm_substream *playback_substream; - struct snd_pcm_substream *capture_substream; - unsigned int is_playing; - unsigned int is_recording; + snd_azf3328_audio_stream_t audio_stream[2]; struct snd_card *card; struct snd_rawmidi *rmidi; -#ifdef SUPPORT_JOYSTICK +#ifdef SUPPORT_GAMEPORT struct gameport *gameport; + int axes[4]; #endif struct pci_dev *pci; @@ -237,9 +292,9 @@ /* register value containers for power management * Note: not always full I/O range preserved (just like Win driver!) */ u16 saved_regs_codec [AZF_IO_SIZE_CODEC_PM / 2]; - u16 saved_regs_io2 [AZF_IO_SIZE_IO2_PM / 2]; + u16 saved_regs_game [AZF_IO_SIZE_GAME_PM / 2]; u16 saved_regs_mpu [AZF_IO_SIZE_MPU_PM / 2]; - u16 saved_regs_synth[AZF_IO_SIZE_SYNTH_PM / 2]; + u16 saved_regs_opl3[AZF_IO_SIZE_OPL3_PM / 2]; u16 saved_regs_mixer[AZF_IO_SIZE_MIXER_PM / 2]; #endif }; @@ -252,126 +307,181 @@ MODULE_DEVICE_TABLE(pci, snd_azf3328_ids); + +static int +snd_azf3328_io_reg_setb(unsigned reg, u8 mask, int do_set) +{ + u8 prev = inb(reg), new; + + new = (do_set) ? prev|mask : prev & ~mask; + /* we need to always write the new value no matter whether it differs or not, + * since some register bits don't indicate their setting */ + outb(new, reg); + if (new != prev) + return 1; + + return 0; +} + +static int +snd_azf3328_io_reg_setw(unsigned reg, u16 mask, int do_set) +{ + u16 prev = inw(reg), new; + + new = (do_set) ? prev|mask : prev & ~mask; + /* we need to always write the new value no matter whether it differs or not, + * since some register bits don't indicate their setting */ + outw(new, reg); + if (new != prev) + return 1; + + return 0; +} + static inline void -snd_azf3328_codec_outb(const struct snd_azf3328 *chip, int reg, u8 value) +snd_azf3328_codec_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) { - outb(value, chip->codec_port + reg); + outb(value, chip->codec_io + reg); } static inline u8 -snd_azf3328_codec_inb(const struct snd_azf3328 *chip, int reg) +snd_azf3328_codec_inb(const struct snd_azf3328 *chip, unsigned reg) { - return inb(chip->codec_port + reg); + return inb(chip->codec_io + reg); } static inline void -snd_azf3328_codec_outw(const struct snd_azf3328 *chip, int reg, u16 value) +snd_azf3328_codec_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) { - outw(value, chip->codec_port + reg); + outw(value, chip->codec_io + reg); } static inline u16 -snd_azf3328_codec_inw(const struct snd_azf3328 *chip, int reg) +snd_azf3328_codec_inw(const struct snd_azf3328 *chip, unsigned reg) +{ + return inw(chip->codec_io + reg); +} + +static inline void +snd_azf3328_codec_outl(const struct snd_azf3328 *chip, unsigned reg, u32 value) { - return inw(chip->codec_port + reg); + outl(value, chip->codec_io + reg); +} + +static inline u32 +snd_azf3328_codec_inl(const struct snd_azf3328 *chip, unsigned reg) +{ + return inl(chip->codec_io + reg); } static inline void -snd_azf3328_codec_outl(const struct snd_azf3328 *chip, int reg, u32 value) +snd_azf3328_game_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) { - outl(value, chip->codec_port + reg); + outb(value, chip->game_io + reg); } static inline void -snd_azf3328_io2_outb(const struct snd_azf3328 *chip, int reg, u8 value) +snd_azf3328_game_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) { - outb(value, chip->io2_port + reg); + outw(value, chip->game_io + reg); } static inline u8 -snd_azf3328_io2_inb(const struct snd_azf3328 *chip, int reg) +snd_azf3328_game_inb(const struct snd_azf3328 *chip, unsigned reg) +{ + return inb(chip->game_io + reg); +} + +static inline u16 +snd_azf3328_game_inw(const struct snd_azf3328 *chip, unsigned reg) { - return inb(chip->io2_port + reg); + return inw(chip->game_io + reg); } static inline void -snd_azf3328_mixer_outw(const struct snd_azf3328 *chip, int reg, u16 value) +snd_azf3328_mixer_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) { - outw(value, chip->mixer_port + reg); + outw(value, chip->mixer_io + reg); } static inline u16 -snd_azf3328_mixer_inw(const struct snd_azf3328 *chip, int reg) +snd_azf3328_mixer_inw(const struct snd_azf3328 *chip, unsigned reg) { - return inw(chip->mixer_port + reg); + return inw(chip->mixer_io + reg); } -static void -snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip, int reg, int do_mute) +#define AZF_MUTE_BIT 0x80 + +static int +snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip, + unsigned reg, int do_mute +) { - unsigned long portbase = chip->mixer_port + reg + 1; - unsigned char oldval; + unsigned long portbase = chip->mixer_io + reg + 1; + int updated; /* the mute bit is on the *second* (i.e. right) register of a * left/right channel setting */ - oldval = inb(portbase); - if (do_mute) - oldval |= 0x80; - else - oldval &= ~0x80; - outb(oldval, portbase); + updated = snd_azf3328_io_reg_setb(portbase, AZF_MUTE_BIT, do_mute); + + /* indicate whether it was muted before */ + return (do_mute) ? !updated : updated; } static void -snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg, unsigned char dst_vol_left, unsigned char dst_vol_right, int chan_sel, int delay) +snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, + unsigned reg, + unsigned char dst_vol_left, + unsigned char dst_vol_right, + int chan_sel, int delay +) { - unsigned long portbase = chip->mixer_port + reg; + unsigned long portbase = chip->mixer_io + reg; unsigned char curr_vol_left = 0, curr_vol_right = 0; - int left_done = 0, right_done = 0; - + int left_change = 0, right_change = 0; + snd_azf3328_dbgcallenter(); - if (chan_sel & SET_CHAN_LEFT) + + if (chan_sel & SET_CHAN_LEFT) { curr_vol_left = inb(portbase + 1); - else - left_done = 1; - if (chan_sel & SET_CHAN_RIGHT) + + /* take care of muting flag contained in left channel */ + if (curr_vol_left & AZF_MUTE_BIT) + dst_vol_left |= AZF_MUTE_BIT; + else + dst_vol_left &= ~AZF_MUTE_BIT; + + left_change = (curr_vol_left > dst_vol_left) ? -1 : 1; + } + + if (chan_sel & SET_CHAN_RIGHT) { curr_vol_right = inb(portbase + 0); - else - right_done = 1; - - /* take care of muting flag (0x80) contained in left channel */ - if (curr_vol_left & 0x80) - dst_vol_left |= 0x80; - else - dst_vol_left &= ~0x80; + + right_change = (curr_vol_right > dst_vol_right) ? -1 : 1; + } do { - if (!left_done) { - if (curr_vol_left > dst_vol_left) - curr_vol_left--; - else - if (curr_vol_left < dst_vol_left) - curr_vol_left++; - else - left_done = 1; - outb(curr_vol_left, portbase + 1); + if (left_change) { + if (curr_vol_left != dst_vol_left) { + curr_vol_left += left_change; + outb(curr_vol_left, portbase + 1); + } else + left_change = 0; } - if (!right_done) { - if (curr_vol_right > dst_vol_right) - curr_vol_right--; - else - if (curr_vol_right < dst_vol_right) - curr_vol_right++; - else - right_done = 1; + if (right_change) { + if (curr_vol_right != dst_vol_right) { + curr_vol_right += right_change; + /* during volume change, the right channel is crackling * somewhat more than the left channel, unfortunately. * This seems to be a hardware issue. */ - outb(curr_vol_right, portbase + 0); + outb(curr_vol_right, portbase + 0); + } else + right_change = 0; } if (delay) mdelay(delay); - } while ((!left_done) || (!right_done)); + } while ((left_change) || (right_change)); snd_azf3328_dbgcallleave(); } @@ -379,7 +489,7 @@ * general mixer element */ struct azf3328_mixer_reg { - unsigned int reg; + unsigned reg; unsigned int lchan_shift, rchan_shift; unsigned int mask; unsigned int invert: 1; @@ -544,13 +654,14 @@ "Mix", "Mic" }; static const char * const texts3[] = { - "Mic", "CD", "Video", "Aux", + "Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone" }; static const char * const texts4[] = { "pre 3D", "post 3D" }; struct azf3328_mixer_reg reg; + const char *p = NULL; snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -561,18 +672,20 @@ if (reg.reg == IDX_MIXER_ADVCTL2) { switch(reg.lchan_shift) { case 8: /* modem out sel */ - strcpy(uinfo->value.enumerated.name, texts1[uinfo->value.enumerated.item]); + p = texts1[uinfo->value.enumerated.item]; break; case 9: /* mono sel source */ - strcpy(uinfo->value.enumerated.name, texts2[uinfo->value.enumerated.item]); + p = texts2[uinfo->value.enumerated.item]; break; case 15: /* PCM Out Path */ - strcpy(uinfo->value.enumerated.name, texts4[uinfo->value.enumerated.item]); + p = texts4[uinfo->value.enumerated.item]; break; } } else - strcpy(uinfo->value.enumerated.name, texts3[uinfo->value.enumerated.item] -); + if (reg.reg == IDX_MIXER_REC_SELECT) + p = texts3[uinfo->value.enumerated.item]; + + strcpy(uinfo->value.enumerated.name, p); return 0; } @@ -583,7 +696,7 @@ struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol); struct azf3328_mixer_reg reg; unsigned short val; - + snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value); val = snd_azf3328_mixer_inw(chip, reg.reg); if (reg.reg == IDX_MIXER_REC_SELECT) { @@ -605,7 +718,7 @@ struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol); struct azf3328_mixer_reg reg; unsigned int oreg, nreg, val; - + snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value); oreg = snd_azf3328_mixer_inw(chip, reg.reg); val = oreg; @@ -717,15 +830,16 @@ snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000); /* mute and zero volume channels */ - for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_init_values); idx++) { + for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_init_values); ++idx) { snd_azf3328_mixer_outw(chip, snd_azf3328_init_values[idx][0], snd_azf3328_init_values[idx][1]); } - + /* add mixer controls */ sw = snd_azf3328_mixer_controls; - for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_mixer_controls); idx++, sw++) { + for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_mixer_controls); + ++idx, ++sw) { if ((err = snd_ctl_add(chip->card, snd_ctl_new1(sw, chip))) < 0) return err; } @@ -757,8 +871,8 @@ } static void -snd_azf3328_setfmt(struct snd_azf3328 *chip, - unsigned int reg, +snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, + unsigned reg, unsigned int bitrate, unsigned int format_width, unsigned int channels @@ -769,24 +883,25 @@ snd_azf3328_dbgcallenter(); switch (bitrate) { - case 4000: val |= SOUNDFORMAT_FREQ_SUSPECTED_4000; break; - case 4800: val |= SOUNDFORMAT_FREQ_SUSPECTED_4800; break; - case 5512: val |= SOUNDFORMAT_FREQ_5510; break; /* the AZF3328 names it "5510" for some strange reason */ - case 6620: val |= SOUNDFORMAT_FREQ_6620; break; - case 8000: val |= SOUNDFORMAT_FREQ_8000; break; - case 9600: val |= SOUNDFORMAT_FREQ_9600; break; - case 11025: val |= SOUNDFORMAT_FREQ_11025; break; - case 13240: val |= SOUNDFORMAT_FREQ_SUSPECTED_13240; break; - case 16000: val |= SOUNDFORMAT_FREQ_16000; break; - case 22050: val |= SOUNDFORMAT_FREQ_22050; break; - case 32000: val |= SOUNDFORMAT_FREQ_32000; break; - case 44100: val |= SOUNDFORMAT_FREQ_44100; break; - case 48000: val |= SOUNDFORMAT_FREQ_48000; break; - case 66200: val |= SOUNDFORMAT_FREQ_SUSPECTED_66200; break; + case AZF_FREQ_4000: val |= SOUNDFORMAT_FREQ_SUSPECTED_4000; break; + case AZF_FREQ_4800: val |= SOUNDFORMAT_FREQ_SUSPECTED_4800; break; + case AZF_FREQ_5512: + /* the AZF3328 names it "5510" for some strange reason */ + val |= SOUNDFORMAT_FREQ_5510; break; + case AZF_FREQ_6620: val |= SOUNDFORMAT_FREQ_6620; break; + case AZF_FREQ_8000: val |= SOUNDFORMAT_FREQ_8000; break; + case AZF_FREQ_9600: val |= SOUNDFORMAT_FREQ_9600; break; + case AZF_FREQ_11025: val |= SOUNDFORMAT_FREQ_11025; break; + case AZF_FREQ_13240: val |= SOUNDFORMAT_FREQ_SUSPECTED_13240; break; + case AZF_FREQ_16000: val |= SOUNDFORMAT_FREQ_16000; break; + case AZF_FREQ_22050: val |= SOUNDFORMAT_FREQ_22050; break; + case AZF_FREQ_32000: val |= SOUNDFORMAT_FREQ_32000; break; default: snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate); - val |= SOUNDFORMAT_FREQ_44100; - break; + /* fall-through */ + case AZF_FREQ_44100: val |= SOUNDFORMAT_FREQ_44100; break; + case AZF_FREQ_48000: val |= SOUNDFORMAT_FREQ_48000; break; + case AZF_FREQ_66200: val |= SOUNDFORMAT_FREQ_SUSPECTED_66200; break; } /* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */ /* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */ @@ -805,10 +920,10 @@ val |= SOUNDFORMAT_FLAG_16BIT; spin_lock_irqsave(&chip->reg_lock, flags); - + /* set bitrate/format */ snd_azf3328_codec_outw(chip, reg, val); - + /* changing the bitrate/format settings switches off the * audio output with an annoying click in case of 8/16bit format change * (maybe shutting down DAC/ADC?), thus immediately @@ -830,31 +945,81 @@ snd_azf3328_dbgcallleave(); } +static inline void +snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328 *chip, + unsigned reg +) +{ + /* choose lowest frequency for low power consumption. + * While this will cause louder noise due to rather coarse frequency, + * it should never matter since output should always + * get disabled properly when idle anyway. */ + snd_azf3328_codec_setfmt(chip, reg, AZF_FREQ_4000, 8, 1); +} + +static inline void +snd_azf3328_codec_enable(struct snd_azf3328 *chip, int enable) +{ + /* no idea what exactly is being done here, but I strongly assume it's + * PM related */ + snd_azf3328_io_reg_setw( + chip->codec_io+IDX_IO_6AH, + IO_6A_PAUSE_PLAYBACK_BIT8, + !enable + ); +} + +static void +snd_azf3328_codec_activity(struct snd_azf3328 *chip, + snd_azf3328_stream_index_t stream_type, + int enable +) +{ + int need_change = (chip->audio_stream[stream_type].running != enable); + + snd_azf3328_dbgplay( + "codec_activity: type %d, enable %d, need_change %d\n", + stream_type, enable, need_change + ); + if (need_change) { + snd_azf3328_stream_index_t other = + (stream_type == AZF_PLAYBACK) ? + AZF_CAPTURE : AZF_PLAYBACK; + /* small check to prevent shutting down the other party + * in case it's active */ + if (!((!enable) && (chip->audio_stream[other].running))) + snd_azf3328_codec_enable(chip, enable); + + /* ...and adjust clock, too + * (reduce noise and power consumption) */ + if (!enable) + snd_azf3328_codec_setfmt_lowpower( + chip, + chip->audio_stream[stream_type].portbase + + IDX_IO_PLAY_SOUNDFORMAT + ); + } + chip->audio_stream[stream_type].running = enable; +} + static void snd_azf3328_setdmaa(struct snd_azf3328 *chip, long unsigned int addr, unsigned int count, unsigned int size, - int do_recording) + snd_azf3328_stream_index_t stream_type +) { - unsigned long flags, portbase; - unsigned int is_running; - snd_azf3328_dbgcallenter(); - if (do_recording) { - /* access capture registers, i.e. skip playback reg section */ - portbase = chip->codec_port + 0x20; - is_running = chip->is_recording; - } else { - /* access the playback register section */ - portbase = chip->codec_port + 0x00; - is_running = chip->is_playing; - } + if (!chip->audio_stream[stream_type].running) { + /* AZF3328 uses a two buffer pointer DMA playback approach */ + + unsigned long flags, portbase, addr_area2; + + /* width 32bit (prevent overflow): */ + unsigned long count_areas, count_tmp; - /* AZF3328 uses a two buffer pointer DMA playback approach */ - if (!is_running) { - unsigned long addr_area2; - unsigned long count_areas, count_tmp; /* width 32bit -- overflow!! */ + portbase = chip->audio_stream[stream_type].portbase; count_areas = size/2; addr_area2 = addr+count_areas; count_areas--; /* max. index */ @@ -884,11 +1049,11 @@ snd_azf3328_dbgcallenter(); #if 0 - snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); - snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, 0); + snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_PLAYBACK); #endif snd_azf3328_dbgcallleave(); return 0; @@ -906,11 +1071,11 @@ snd_azf3328_dbgcallenter(); #if 0 - snd_azf3328_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); - snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, 1); + snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_CAPTURE); #endif snd_azf3328_dbgcallleave(); return 0; @@ -923,6 +1088,7 @@ struct snd_pcm_runtime *runtime = substream->runtime; int result = 0; unsigned int status1; + int previously_muted; snd_azf3328_dbgcalls("snd_azf3328_playback_trigger cmd %d\n", cmd); @@ -930,20 +1096,22 @@ case SNDRV_PCM_TRIGGER_START: snd_azf3328_dbgplay("START PLAYBACK\n"); - /* mute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + /* mute WaveOut (avoid clicking during setup) */ + previously_muted = + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); - snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); spin_lock(&chip->reg_lock); - /* stop playback */ status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS); + + /* stop playback */ status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); - + /* FIXME: clear interrupts or what??? */ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_IRQTYPE, 0xffff); spin_unlock(&chip->reg_lock); @@ -951,7 +1119,7 @@ snd_azf3328_setdmaa(chip, runtime->dma_addr, snd_pcm_lib_period_bytes(substream), snd_pcm_lib_buffer_bytes(substream), - 0); + AZF_PLAYBACK); spin_lock(&chip->reg_lock); #ifdef WIN9X @@ -978,30 +1146,34 @@ DMA_SOMETHING_ELSE); #endif spin_unlock(&chip->reg_lock); + snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 1); /* now unmute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); + if (!previously_muted) + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); - chip->is_playing = 1; snd_azf3328_dbgplay("STARTED PLAYBACK\n"); break; case SNDRV_PCM_TRIGGER_RESUME: snd_azf3328_dbgplay("RESUME PLAYBACK\n"); /* resume playback if we were active */ - if (chip->is_playing) + spin_lock(&chip->reg_lock); + if (chip->audio_stream[AZF_PLAYBACK].running) snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) | DMA_RESUME); + spin_unlock(&chip->reg_lock); break; case SNDRV_PCM_TRIGGER_STOP: snd_azf3328_dbgplay("STOP PLAYBACK\n"); - /* mute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + /* mute WaveOut (avoid clicking during setup) */ + previously_muted = + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); spin_lock(&chip->reg_lock); - /* stop playback */ status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS); + /* stop playback */ status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); @@ -1013,10 +1185,12 @@ status1 &= ~DMA_PLAY_SOMETHING1; snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); spin_unlock(&chip->reg_lock); - + snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0); + /* now unmute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); - chip->is_playing = 0; + if (!previously_muted) + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); + snd_azf3328_dbgplay("STOPPED PLAYBACK\n"); break; case SNDRV_PCM_TRIGGER_SUSPEND: @@ -1035,7 +1209,7 @@ printk(KERN_ERR "FIXME: unknown trigger mode!\n"); return -EINVAL; } - + snd_azf3328_dbgcallleave(); return result; } @@ -1057,7 +1231,7 @@ snd_azf3328_dbgplay("START CAPTURE\n"); - snd_azf3328_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); @@ -1067,7 +1241,7 @@ status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS); status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); - + /* FIXME: clear interrupts or what??? */ snd_azf3328_codec_outw(chip, IDX_IO_REC_IRQTYPE, 0xffff); spin_unlock(&chip->reg_lock); @@ -1075,7 +1249,7 @@ snd_azf3328_setdmaa(chip, runtime->dma_addr, snd_pcm_lib_period_bytes(substream), snd_pcm_lib_buffer_bytes(substream), - 1); + AZF_CAPTURE); spin_lock(&chip->reg_lock); #ifdef WIN9X @@ -1102,16 +1276,18 @@ DMA_SOMETHING_ELSE); #endif spin_unlock(&chip->reg_lock); + snd_azf3328_codec_activity(chip, AZF_CAPTURE, 1); - chip->is_recording = 1; snd_azf3328_dbgplay("STARTED CAPTURE\n"); break; case SNDRV_PCM_TRIGGER_RESUME: snd_azf3328_dbgplay("RESUME CAPTURE\n"); /* resume recording if we were active */ - if (chip->is_recording) + spin_lock(&chip->reg_lock); + if (chip->audio_stream[AZF_CAPTURE].running) snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) | DMA_RESUME); + spin_unlock(&chip->reg_lock); break; case SNDRV_PCM_TRIGGER_STOP: snd_azf3328_dbgplay("STOP CAPTURE\n"); @@ -1129,8 +1305,8 @@ status1 &= ~DMA_PLAY_SOMETHING1; snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); spin_unlock(&chip->reg_lock); - - chip->is_recording = 0; + snd_azf3328_codec_activity(chip, AZF_CAPTURE, 0); + snd_azf3328_dbgplay("STOPPED CAPTURE\n"); break; case SNDRV_PCM_TRIGGER_SUSPEND: @@ -1149,7 +1325,7 @@ printk(KERN_ERR "FIXME: unknown trigger mode!\n"); return -EINVAL; } - + snd_azf3328_dbgcallleave(); return result; } @@ -1162,11 +1338,11 @@ snd_pcm_uframes_t frmres; #ifdef QUERY_HARDWARE - bufptr = inl(chip->codec_port+IDX_IO_PLAY_DMA_START_1); + bufptr = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_START_1); #else bufptr = substream->runtime->dma_addr; #endif - result = inl(chip->codec_port+IDX_IO_PLAY_DMA_CURRPOS); + result = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_CURRPOS); /* calculate offset */ result -= bufptr; @@ -1183,11 +1359,11 @@ snd_pcm_uframes_t frmres; #ifdef QUERY_HARDWARE - bufptr = inl(chip->codec_port+IDX_IO_REC_DMA_START_1); + bufptr = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_START_1); #else bufptr = substream->runtime->dma_addr; #endif - result = inl(chip->codec_port+IDX_IO_REC_DMA_CURRPOS); + result = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_CURRPOS); /* calculate offset */ result -= bufptr; @@ -1196,27 +1372,232 @@ return frmres; } +/******************************************************************/ + +#ifdef SUPPORT_GAMEPORT +static inline void +snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip, int enable) +{ + snd_azf3328_io_reg_setb( + chip->game_io+IDX_GAME_HWCONFIG, + GAME_HWCFG_IRQ_ENABLE, + enable + ); +} + +static inline void +snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, int enable) +{ + snd_azf3328_io_reg_setb( + chip->game_io+IDX_GAME_HWCONFIG, + GAME_HWCFG_LEGACY_ADDRESS_ENABLE, + enable + ); +} + +static inline void +snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, int enable) +{ + snd_azf3328_io_reg_setw( + chip->codec_io+IDX_IO_6AH, + IO_6A_SOMETHING2_GAMEPORT, + !enable + ); +} + +static inline void +snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip) +{ + /* + * skeleton handler only + * (we do not want axis reading in interrupt handler - too much load!) + */ + snd_azf3328_dbggame("gameport irq\n"); + + /* this should ACK the gameport IRQ properly, hopefully. */ + snd_azf3328_game_inw(chip, IDX_GAME_AXIS_VALUE); +} + +static int +snd_azf3328_gameport_open(struct gameport *gameport, int mode) +{ + struct snd_azf3328 *chip = gameport_get_port_data(gameport); + int res; + + snd_azf3328_dbggame("gameport_open, mode %d\n", mode); + switch (mode) { + case GAMEPORT_MODE_COOKED: + case GAMEPORT_MODE_RAW: + res = 0; + break; + default: + res = -1; + break; + } + + snd_azf3328_gameport_axis_circuit_enable(chip, (res == 0)); + + return res; +} + +static void +snd_azf3328_gameport_close(struct gameport *gameport) +{ + struct snd_azf3328 *chip = gameport_get_port_data(gameport); + + snd_azf3328_dbggame("gameport_close\n"); + snd_azf3328_gameport_axis_circuit_enable(chip, 0); +} + +static int +snd_azf3328_gameport_cooked_read(struct gameport *gameport, + int *axes, + int *buttons +) +{ + struct snd_azf3328 *chip = gameport_get_port_data(gameport); + int i; + u8 val; + + snd_assert(chip, return 0); + + spin_lock(&chip->reg_lock); + val = snd_azf3328_game_inb(chip, IDX_GAME_LEGACY_COMPATIBLE); + *buttons = (~(val) >> 4) & 0xf; + + /* ok, this one is a bit dirty: cooked_read is being polled by a timer, + * thus we're atomic and cannot actively wait in here + * (which would be useful for us since it probably would be better + * to trigger a measurement in here, then wait a short amount of + * time until it's finished, then read values of _this_ measurement). + * + * Thus we simply resort to reading values if they're available already + * and trigger the next measurement. + */ + + val = snd_azf3328_game_inb(chip, IDX_GAME_AXES_CONFIG); + if (val & GAME_AXES_SAMPLING_READY) { + for (i = 0; i < 4; ++i) { + /* configure the axis to read */ + val = (i << 4) | 0x0f; + snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val); + + chip->axes[i] = snd_azf3328_game_inw( + chip, IDX_GAME_AXIS_VALUE + ); + } + } + + /* trigger next axes sampling, to be evaluated the next time we + * enter this function */ + + /* for some very, very strange reason we cannot enable + * Measurement Ready monitoring for all axes here, + * at least not when only one joystick connected */ + val = 0x03; /* we're able to monitor axes 1 and 2 only */ + snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val); + + snd_azf3328_game_outw(chip, IDX_GAME_AXIS_VALUE, 0xffff); + spin_unlock(&chip->reg_lock); + + for (i = 0; i < 4; i++) { + axes[i] = chip->axes[i]; + if (axes[i] == 0xffff) + axes[i] = -1; + } + + snd_azf3328_dbggame("cooked_read: axes %d %d %d %d buttons %d\n", + axes[0], axes[1], axes[2], axes[3], *buttons + ); + + return 0; +} + +static int __devinit +snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) +{ + struct gameport *gp; + + int io_port = chip->game_io; + + chip->gameport = gp = gameport_allocate_port(); + if (!gp) { + printk(KERN_ERR "azt3328: cannot alloc memory for gameport\n"); + return -ENOMEM; + } + + gameport_set_name(gp, "AZF3328 Gameport"); + gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci)); + gameport_set_dev_parent(gp, &chip->pci->dev); + gp->io = io_port; + gameport_set_port_data(gp, chip); + + gp->open = snd_azf3328_gameport_open; + gp->close = snd_azf3328_gameport_close; + gp->fuzz = 16; /* seems ok */ + gp->cooked_read = snd_azf3328_gameport_cooked_read; + + /* DISABLE legacy address: we don't need it! */ + snd_azf3328_gameport_legacy_address_enable(chip, 0); + + snd_azf3328_gameport_axis_circuit_enable(chip, 0); + + gameport_register_port(chip->gameport); + + return 0; +} + +static void +snd_azf3328_gameport_free(struct snd_azf3328 *chip) +{ + if (chip->gameport) { + gameport_unregister_port(chip->gameport); + chip->gameport = NULL; + } + snd_azf3328_gameport_irq_enable(chip, 0); +} +#else +static inline int +snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) { return -ENOSYS; } +static inline void +snd_azf3328_gameport_free(struct snd_azf3328 *chip) { } +static inline void +snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip) +{ + printk(KERN_WARNING "huh, game port IRQ occurred!?\n"); +} +#endif /* SUPPORT_GAMEPORT */ + +/******************************************************************/ + static irqreturn_t snd_azf3328_interrupt(int irq, void *dev_id) { struct snd_azf3328 *chip = dev_id; u8 status, which; +#if DEBUG_PLAY_REC static unsigned long irq_count; +#endif status = snd_azf3328_codec_inb(chip, IDX_IO_IRQSTATUS); /* fast path out, to ease interrupt sharing */ - if (!(status & (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_MPU401|IRQ_TIMER))) + if (!(status & + (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_GAMEPORT|IRQ_MPU401|IRQ_TIMER) + )) return IRQ_NONE; /* must be interrupt for another device */ snd_azf3328_dbgplay("Interrupt %ld!\nIDX_IO_PLAY_FLAGS %04x, IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", - irq_count, + irq_count++, snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), status); - + if (status & IRQ_TIMER) { - /* snd_azf3328_dbgplay("timer %ld\n", inl(chip->codec_port+IDX_IO_TIMER_VALUE) & TIMER_VALUE_MASK); */ + /* snd_azf3328_dbgplay("timer %ld\n", + snd_azf3328_codec_inl(chip, IDX_IO_TIMER_VALUE) + & TIMER_VALUE_MASK + ); */ if (chip->timer) snd_timer_interrupt(chip->timer, chip->timer->sticks); /* ACK timer */ @@ -1232,11 +1613,16 @@ snd_azf3328_codec_outb(chip, IDX_IO_PLAY_IRQTYPE, which); spin_unlock(&chip->reg_lock); - if (chip->pcm && chip->playback_substream) { - snd_pcm_period_elapsed(chip->playback_substream); + if (chip->pcm && chip->audio_stream[AZF_PLAYBACK].substream) { + snd_pcm_period_elapsed( + chip->audio_stream[AZF_PLAYBACK].substream + ); snd_azf3328_dbgplay("PLAY period done (#%x), @ %x\n", which, - inl(chip->codec_port+IDX_IO_PLAY_DMA_CURRPOS)); + snd_azf3328_codec_inl( + chip, IDX_IO_PLAY_DMA_CURRPOS + ) + ); } else snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); if (which & IRQ_PLAY_SOMETHING) @@ -1249,16 +1635,23 @@ snd_azf3328_codec_outb(chip, IDX_IO_REC_IRQTYPE, which); spin_unlock(&chip->reg_lock); - if (chip->pcm && chip->capture_substream) { - snd_pcm_period_elapsed(chip->capture_substream); + if (chip->pcm && chip->audio_stream[AZF_CAPTURE].substream) { + snd_pcm_period_elapsed( + chip->audio_stream[AZF_CAPTURE].substream + ); snd_azf3328_dbgplay("REC period done (#%x), @ %x\n", which, - inl(chip->codec_port+IDX_IO_REC_DMA_CURRPOS)); + snd_azf3328_codec_inl( + chip, IDX_IO_REC_DMA_CURRPOS + ) + ); } else snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); if (which & IRQ_REC_SOMETHING) snd_azf3328_dbgplay("azt3328: unknown rec IRQ type occurred, please report!\n"); } + if (status & IRQ_GAMEPORT) + snd_azf3328_gameport_interrupt(chip); /* MPU401 has less critical IRQ requirements * than timer and playback/recording, right? */ if (status & IRQ_MPU401) { @@ -1268,7 +1661,6 @@ * If so, then I don't know how... */ snd_azf3328_dbgplay("azt3328: MPU401 IRQ\n"); } - irq_count++; return IRQ_HANDLED; } @@ -1287,8 +1679,8 @@ .rates = SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT, - .rate_min = 4000, - .rate_max = 66200, + .rate_min = AZF_FREQ_4000, + .rate_max = AZF_FREQ_66200, .channels_min = 1, .channels_max = 2, .buffer_bytes_max = 65536, @@ -1315,8 +1707,8 @@ .rates = SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT, - .rate_min = 4000, - .rate_max = 66200, + .rate_min = AZF_FREQ_4000, + .rate_max = AZF_FREQ_66200, .channels_min = 1, .channels_max = 2, .buffer_bytes_max = 65536, @@ -1329,10 +1721,24 @@ static unsigned int snd_azf3328_fixed_rates[] = { - 4000, 4800, 5512, 6620, 8000, 9600, 11025, 13240, 16000, 22050, 32000, - 44100, 48000, 66200 }; + AZF_FREQ_4000, + AZF_FREQ_4800, + AZF_FREQ_5512, + AZF_FREQ_6620, + AZF_FREQ_8000, + AZF_FREQ_9600, + AZF_FREQ_11025, + AZF_FREQ_13240, + AZF_FREQ_16000, + AZF_FREQ_22050, + AZF_FREQ_32000, + AZF_FREQ_44100, + AZF_FREQ_48000, + AZF_FREQ_66200 +}; + static struct snd_pcm_hw_constraint_list snd_azf3328_hw_constraints_rates = { - .count = ARRAY_SIZE(snd_azf3328_fixed_rates), + .count = ARRAY_SIZE(snd_azf3328_fixed_rates), .list = snd_azf3328_fixed_rates, .mask = 0, }; @@ -1346,7 +1752,7 @@ struct snd_pcm_runtime *runtime = substream->runtime; snd_azf3328_dbgcallenter(); - chip->playback_substream = substream; + chip->audio_stream[AZF_PLAYBACK].substream = substream; runtime->hw = snd_azf3328_playback; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_azf3328_hw_constraints_rates); @@ -1361,7 +1767,7 @@ struct snd_pcm_runtime *runtime = substream->runtime; snd_azf3328_dbgcallenter(); - chip->capture_substream = substream; + chip->audio_stream[AZF_CAPTURE].substream = substream; runtime->hw = snd_azf3328_capture; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_azf3328_hw_constraints_rates); @@ -1375,7 +1781,7 @@ struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); snd_azf3328_dbgcallenter(); - chip->playback_substream = NULL; + chip->audio_stream[AZF_PLAYBACK].substream = NULL; snd_azf3328_dbgcallleave(); return 0; } @@ -1386,7 +1792,7 @@ struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); snd_azf3328_dbgcallenter(); - chip->capture_substream = NULL; + chip->audio_stream[AZF_CAPTURE].substream = NULL; snd_azf3328_dbgcallleave(); return 0; } @@ -1441,102 +1847,8 @@ /******************************************************************/ -#ifdef SUPPORT_JOYSTICK -static int __devinit -snd_azf3328_config_joystick(struct snd_azf3328 *chip, int dev) -{ - struct gameport *gp; - struct resource *r; - - if (!joystick[dev]) - return -ENODEV; - - if (!(r = request_region(0x200, 8, "AZF3328 gameport"))) { - printk(KERN_WARNING "azt3328: cannot reserve joystick ports\n"); - return -EBUSY; - } - - chip->gameport = gp = gameport_allocate_port(); - if (!gp) { - printk(KERN_ERR "azt3328: cannot allocate memory for gameport\n"); - release_and_free_resource(r); - return -ENOMEM; - } - - gameport_set_name(gp, "AZF3328 Gameport"); - gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci)); - gameport_set_dev_parent(gp, &chip->pci->dev); - gp->io = 0x200; - gameport_set_port_data(gp, r); - - snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR, - snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) | LEGACY_JOY); - - gameport_register_port(chip->gameport); - - return 0; -} - -static void -snd_azf3328_free_joystick(struct snd_azf3328 *chip) -{ - if (chip->gameport) { - struct resource *r = gameport_get_port_data(chip->gameport); - - gameport_unregister_port(chip->gameport); - chip->gameport = NULL; - /* disable gameport */ - snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR, - snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) & ~LEGACY_JOY); - release_and_free_resource(r); - } -} -#else -static inline int -snd_azf3328_config_joystick(struct snd_azf3328 *chip, int dev) { return -ENOSYS; } -static inline void -snd_azf3328_free_joystick(struct snd_azf3328 *chip) { } -#endif - -/******************************************************************/ - -static int -snd_azf3328_free(struct snd_azf3328 *chip) -{ - if (chip->irq < 0) - goto __end_hw; - - /* reset (close) mixer */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); /* first mute master volume */ - snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000); - - /* interrupt setup - mask everything (FIXME!) */ - /* well, at least we know how to disable the timer IRQ */ - snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x00); - - if (chip->irq >= 0) - synchronize_irq(chip->irq); -__end_hw: - snd_azf3328_free_joystick(chip); - if (chip->irq >= 0) - free_irq(chip->irq, chip); - pci_release_regions(chip->pci); - pci_disable_device(chip->pci); - - kfree(chip); - return 0; -} - -static int -snd_azf3328_dev_free(struct snd_device *device) -{ - struct snd_azf3328 *chip = device->device_data; - return snd_azf3328_free(chip); -} - -/******************************************************************/ - -/*** NOTE: the physical timer resolution actually is 1024000 ticks per second, +/*** NOTE: the physical timer resolution actually is 1024000 ticks per second + *** (probably derived from main crystal via a divider of 24), *** but announcing those attributes to user-space would make programs *** configure the timer to a 1 tick value, resulting in an absolutely fatal *** timer IRQ storm. @@ -1564,7 +1876,7 @@ delay = 49; /* minimum time is 49 ticks */ } snd_azf3328_dbgtimer("setting timer countdown value %d, add COUNTDOWN|IRQ\n", delay); - delay |= TIMER_ENABLE_COUNTDOWN | TIMER_ENABLE_IRQ; + delay |= TIMER_COUNTDOWN_ENABLE | TIMER_IRQ_ENABLE; spin_lock_irqsave(&chip->reg_lock, flags); snd_azf3328_codec_outl(chip, IDX_IO_TIMER_VALUE, delay); spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -1582,7 +1894,7 @@ chip = snd_timer_chip(timer); spin_lock_irqsave(&chip->reg_lock, flags); /* disable timer countdown and interrupt */ - /* FIXME: should we write TIMER_ACK_IRQ here? */ + /* FIXME: should we write TIMER_IRQ_ACK here? */ snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_azf3328_dbgcallleave(); @@ -1626,9 +1938,10 @@ snd_azf3328_timer_hw.resolution *= seqtimer_scaling; snd_azf3328_timer_hw.ticks /= seqtimer_scaling; - if ((err = snd_timer_new(chip->card, "AZF3328", &tid, &timer)) < 0) { + + err = snd_timer_new(chip->card, "AZF3328", &tid, &timer); + if (err < 0) goto out; - } strcpy(timer->name, "AZF3328 timer"); timer->private_data = chip; @@ -1636,6 +1949,8 @@ chip->timer = timer; + snd_azf3328_timer_stop(timer); + err = 0; out: @@ -1645,10 +1960,44 @@ /******************************************************************/ +static int +snd_azf3328_free(struct snd_azf3328 *chip) +{ + if (chip->irq < 0) + goto __end_hw; + + /* reset (close) mixer: + * first mute master volume, then reset + */ + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); + snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000); + + snd_azf3328_timer_stop(chip->timer); + snd_azf3328_gameport_free(chip); + + if (chip->irq >= 0) + synchronize_irq(chip->irq); +__end_hw: + if (chip->irq >= 0) + free_irq(chip->irq, chip); + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + + kfree(chip); + return 0; +} + +static int +snd_azf3328_dev_free(struct snd_device *device) +{ + struct snd_azf3328 *chip = device->device_data; + return snd_azf3328_free(chip); +} + #if 0 /* check whether a bit can be modified */ static void -snd_azf3328_test_bit(unsigned int reg, int bit) +snd_azf3328_test_bit(unsigned unsigned reg, int bit) { unsigned char val, valoff, valon; @@ -1659,42 +2008,74 @@ outb(val|(1 << bit), reg); valon = inb(reg); - + outb(val, reg); - printk(KERN_ERR "reg %04x bit %d: %02x %02x %02x\n", reg, bit, val, valoff, valon); + printk(KERN_ERR "reg %04x bit %d: %02x %02x %02x\n", + reg, bit, val, valoff, valon + ); } #endif -#if DEBUG_MISC -static void +static inline void snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip) { +#if DEBUG_MISC u16 tmp; - snd_azf3328_dbgmisc("codec_port 0x%lx, io2_port 0x%lx, mpu_port 0x%lx, synth_port 0x%lx, mixer_port 0x%lx, irq %d\n", chip->codec_port, chip->io2_port, chip->mpu_port, chip->synth_port, chip->mixer_port, chip->irq); - - snd_azf3328_dbgmisc("io2 %02x %02x %02x %02x %02x %02x\n", snd_azf3328_io2_inb(chip, 0), snd_azf3328_io2_inb(chip, 1), snd_azf3328_io2_inb(chip, 2), snd_azf3328_io2_inb(chip, 3), snd_azf3328_io2_inb(chip, 4), snd_azf3328_io2_inb(chip, 5)); - - for (tmp=0; tmp <= 0x01; tmp += 1) - snd_azf3328_dbgmisc("0x%02x: opl 0x%04x, mpu300 0x%04x, mpu310 0x%04x, mpu320 0x%04x, mpu330 0x%04x\n", tmp, inb(0x388 + tmp), inb(0x300 + tmp), inb(0x310 + tmp), inb(0x320 + tmp), inb(0x330 + tmp)); + snd_azf3328_dbgmisc( + "codec_io 0x%lx, game_io 0x%lx, mpu_io 0x%lx, " + "opl3_io 0x%lx, mixer_io 0x%lx, irq %d\n", + chip->codec_io, chip->game_io, chip->mpu_io, + chip->opl3_io, chip->mixer_io, chip->irq + ); + + snd_azf3328_dbgmisc("game %02x %02x %02x %02x %02x %02x\n", + snd_azf3328_game_inb(chip, 0), + snd_azf3328_game_inb(chip, 1), + snd_azf3328_game_inb(chip, 2), + snd_azf3328_game_inb(chip, 3), + snd_azf3328_game_inb(chip, 4), + snd_azf3328_game_inb(chip, 5) + ); + + for (tmp = 0; tmp < 0x07; tmp += 1) + snd_azf3328_dbgmisc("mpu_io 0x%04x\n", inb(chip->mpu_io + tmp)); + + for (tmp = 0; tmp <= 0x07; tmp += 1) + snd_azf3328_dbgmisc("0x%02x: game200 0x%04x, game208 0x%04x\n", + tmp, inb(0x200 + tmp), inb(0x208 + tmp)); + + for (tmp = 0; tmp <= 0x01; tmp += 1) + snd_azf3328_dbgmisc( + "0x%02x: mpu300 0x%04x, mpu310 0x%04x, mpu320 0x%04x, " + "mpu330 0x%04x opl388 0x%04x opl38c 0x%04x\n", + tmp, + inb(0x300 + tmp), + inb(0x310 + tmp), + inb(0x320 + tmp), + inb(0x330 + tmp), + inb(0x388 + tmp), + inb(0x38c + tmp) + ); for (tmp = 0; tmp < AZF_IO_SIZE_CODEC; tmp += 2) - snd_azf3328_dbgmisc("codec 0x%02x: 0x%04x\n", tmp, snd_azf3328_codec_inw(chip, tmp)); + snd_azf3328_dbgmisc("codec 0x%02x: 0x%04x\n", + tmp, snd_azf3328_codec_inw(chip, tmp) + ); for (tmp = 0; tmp < AZF_IO_SIZE_MIXER; tmp += 2) - snd_azf3328_dbgmisc("mixer 0x%02x: 0x%04x\n", tmp, snd_azf3328_mixer_inw(chip, tmp)); + snd_azf3328_dbgmisc("mixer 0x%02x: 0x%04x\n", + tmp, snd_azf3328_mixer_inw(chip, tmp) + ); +#endif /* DEBUG_MISC */ } -#else -static inline void -snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip) {} -#endif static int __devinit snd_azf3328_create(struct snd_card *card, - struct pci_dev *pci, - unsigned long device_type, - struct snd_azf3328 ** rchip) + struct pci_dev *pci, + unsigned long device_type, + struct snd_azf3328 **rchip) { struct snd_azf3328 *chip; int err; @@ -1705,7 +2086,8 @@ *rchip = NULL; - if ((err = pci_enable_device(pci)) < 0) + err = pci_enable_device(pci); + if (err < 0) return err; chip = kzalloc(sizeof(*chip), GFP_KERNEL); @@ -1721,20 +2103,25 @@ /* check if we can restrict PCI DMA transfers to 24 bits */ if (pci_set_dma_mask(pci, DMA_24BIT_MASK) < 0 || pci_set_consistent_dma_mask(pci, DMA_24BIT_MASK) < 0) { - snd_printk(KERN_ERR "architecture does not support 24bit PCI busmaster DMA\n"); + snd_printk(KERN_ERR "architecture does not support " + "24bit PCI busmaster DMA\n" + ); err = -ENXIO; goto out_err; } - if ((err = pci_request_regions(pci, "Aztech AZF3328")) < 0) { + err = pci_request_regions(pci, "Aztech AZF3328"); + if (err < 0) goto out_err; - } - chip->codec_port = pci_resource_start(pci, 0); - chip->io2_port = pci_resource_start(pci, 1); - chip->mpu_port = pci_resource_start(pci, 2); - chip->synth_port = pci_resource_start(pci, 3); - chip->mixer_port = pci_resource_start(pci, 4); + chip->codec_io = pci_resource_start(pci, 0); + chip->game_io = pci_resource_start(pci, 1); + chip->mpu_io = pci_resource_start(pci, 2); + chip->opl3_io = pci_resource_start(pci, 3); + chip->mixer_io = pci_resource_start(pci, 4); + + chip->audio_stream[AZF_PLAYBACK].portbase = chip->codec_io + 0x00; + chip->audio_stream[AZF_CAPTURE].portbase = chip->codec_io + 0x20; if (request_irq(pci->irq, snd_azf3328_interrupt, IRQF_SHARED, card->shortname, chip)) { @@ -1747,29 +2134,29 @@ synchronize_irq(chip->irq); snd_azf3328_debug_show_ports(chip); - - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) goto out_err; - } /* create mixer interface & switches */ - if ((err = snd_azf3328_mixer_new(chip)) < 0) + err = snd_azf3328_mixer_new(chip); + if (err < 0) goto out_err; -#if 0 - /* set very low bitrate to reduce noise and power consumption? */ - snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, 5512, 8, 1); -#endif + /* shutdown codecs to save power */ + /* have snd_azf3328_codec_activity() act properly */ + chip->audio_stream[AZF_PLAYBACK].running = 1; + snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0); /* standard chip init stuff */ - /* default IRQ init value */ + /* default IRQ init value */ tmp = DMA_PLAY_SOMETHING2|DMA_EPILOGUE_SOMETHING|DMA_SOMETHING_ELSE; spin_lock_irq(&chip->reg_lock); snd_azf3328_codec_outb(chip, IDX_IO_PLAY_FLAGS, tmp); snd_azf3328_codec_outb(chip, IDX_IO_REC_FLAGS, tmp); snd_azf3328_codec_outb(chip, IDX_IO_SOMETHING_FLAGS, tmp); - snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x00); /* disable timer */ spin_unlock_irq(&chip->reg_lock); snd_card_set_dev(card, &pci->dev); @@ -1805,52 +2192,61 @@ return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0 ); + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); if (card == NULL) return -ENOMEM; strcpy(card->driver, "AZF3328"); strcpy(card->shortname, "Aztech AZF3328 (PCI168)"); - if ((err = snd_azf3328_create(card, pci, pci_id->driver_data, &chip)) < 0) { + err = snd_azf3328_create(card, pci, pci_id->driver_data, &chip); + if (err < 0) goto out_err; - } card->private_data = chip; - if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_MPU401, - chip->mpu_port, MPU401_INFO_INTEGRATED, - pci->irq, 0, &chip->rmidi)) < 0) { - snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", chip->mpu_port); + err = snd_mpu401_uart_new( + card, 0, MPU401_HW_MPU401, chip->mpu_io, MPU401_INFO_INTEGRATED, + pci->irq, 0, &chip->rmidi + ); + if (err < 0) { + snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", + chip->mpu_io + ); goto out_err; } - if ((err = snd_azf3328_timer(chip, 0)) < 0) { + err = snd_azf3328_timer(chip, 0); + if (err < 0) goto out_err; - } - if ((err = snd_azf3328_pcm(chip, 0)) < 0) { + err = snd_azf3328_pcm(chip, 0); + if (err < 0) goto out_err; - } - if (snd_opl3_create(card, chip->synth_port, chip->synth_port+2, + if (snd_opl3_create(card, chip->opl3_io, chip->opl3_io+2, OPL3_HW_AUTO, 1, &opl3) < 0) { snd_printk(KERN_ERR "azf3328: no OPL3 device at 0x%lx-0x%lx?\n", - chip->synth_port, chip->synth_port+2 ); + chip->opl3_io, chip->opl3_io+2 + ); } else { - if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) { + /* need to use IDs 1, 2 since ID 0 is snd_azf3328_timer above */ + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + goto out_err; + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) goto out_err; - } } opl3->private_data = chip; sprintf(card->longname, "%s at 0x%lx, irq %i", - card->shortname, chip->codec_port, chip->irq); + card->shortname, chip->codec_io, chip->irq); - if ((err = snd_card_register(card)) < 0) { + err = snd_card_register(card); + if (err < 0) goto out_err; - } #ifdef MODULE printk( @@ -1861,19 +2257,18 @@ 1024000 / seqtimer_scaling, seqtimer_scaling); #endif - if (snd_azf3328_config_joystick(chip, dev) < 0) - snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR, - snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) & ~LEGACY_JOY); + snd_azf3328_gameport(chip, dev); pci_set_drvdata(pci, card); dev++; err = 0; goto out; - + out_err: + snd_printk(KERN_ERR "azf3328: something failed, exiting\n"); snd_card_free(card); - + out: snd_azf3328_dbgcallleave(); return err; @@ -1894,27 +2289,27 @@ { struct snd_card *card = pci_get_drvdata(pci); struct snd_azf3328 *chip = card->private_data; - int reg; + unsigned reg; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - + snd_pcm_suspend_all(chip->pcm); - for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++) - chip->saved_regs_mixer[reg] = inw(chip->mixer_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg) + chip->saved_regs_mixer[reg] = inw(chip->mixer_io + reg * 2); /* make sure to disable master volume etc. to prevent looping sound */ snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); - - for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++) - chip->saved_regs_codec[reg] = inw(chip->codec_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++) - chip->saved_regs_io2[reg] = inw(chip->io2_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++) - chip->saved_regs_mpu[reg] = inw(chip->mpu_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++) - chip->saved_regs_synth[reg] = inw(chip->synth_port + reg * 2); + + for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) + chip->saved_regs_codec[reg] = inw(chip->codec_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) + chip->saved_regs_game[reg] = inw(chip->game_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg) + chip->saved_regs_mpu[reg] = inw(chip->mpu_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg) + chip->saved_regs_opl3[reg] = inw(chip->opl3_io + reg * 2); pci_disable_device(pci); pci_save_state(pci); @@ -1927,7 +2322,7 @@ { struct snd_card *card = pci_get_drvdata(pci); struct snd_azf3328 *chip = card->private_data; - int reg; + unsigned reg; pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); @@ -1939,23 +2334,21 @@ } pci_set_master(pci); - for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++) - outw(chip->saved_regs_io2[reg], chip->io2_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++) - outw(chip->saved_regs_mpu[reg], chip->mpu_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++) - outw(chip->saved_regs_synth[reg], chip->synth_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++) - outw(chip->saved_regs_mixer[reg], chip->mixer_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++) - outw(chip->saved_regs_codec[reg], chip->codec_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) + outw(chip->saved_regs_game[reg], chip->game_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg) + outw(chip->saved_regs_mpu[reg], chip->mpu_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg) + outw(chip->saved_regs_opl3[reg], chip->opl3_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg) + outw(chip->saved_regs_mixer[reg], chip->mixer_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) + outw(chip->saved_regs_codec[reg], chip->codec_io + reg * 2); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif - - +#endif /* CONFIG_PM */ static struct pci_driver driver = { --- linux-2.6.26-rc1.orig/sound/pci/azt3328.h 2008-05-03 20:59:44.000000000 +0200 +++ linux-2.6.26-rc1/sound/pci/azt3328.h 2008-05-14 22:47:12.000000000 +0200 @@ -54,7 +54,10 @@ #define SOUNDFORMAT_XTAL1 0x00 #define SOUNDFORMAT_XTAL2 0x01 /* all _SUSPECTED_ values are not used by Windows drivers, so we don't - * have any hard facts, only rough measurements */ + * have any hard facts, only rough measurements. + * All we know is that the crystal used on the board has 24.576MHz, + * like many soundcards (which results in the frequencies below when + * using certain divider values selected by the values below) */ #define SOUNDFORMAT_FREQ_SUSPECTED_4000 0x0c | SOUNDFORMAT_XTAL1 #define SOUNDFORMAT_FREQ_SUSPECTED_4800 0x0a | SOUNDFORMAT_XTAL1 #define SOUNDFORMAT_FREQ_5510 0x0c | SOUNDFORMAT_XTAL2 @@ -72,6 +75,26 @@ #define SOUNDFORMAT_FLAG_16BIT 0x0010 #define SOUNDFORMAT_FLAG_2CHANNELS 0x0020 +/* define frequency helpers, for maximum value safety */ +enum { +#define AZF_FREQ(rate) AZF_FREQ_##rate = rate + AZF_FREQ(4000), + AZF_FREQ(4800), + AZF_FREQ(5512), + AZF_FREQ(6620), + AZF_FREQ(8000), + AZF_FREQ(9600), + AZF_FREQ(11025), + AZF_FREQ(13240), + AZF_FREQ(16000), + AZF_FREQ(22050), + AZF_FREQ(32000), + AZF_FREQ(44100), + AZF_FREQ(48000), + AZF_FREQ(66200), +#undef AZF_FREQ +} AZF_FREQUENCIES; + /** recording area (see also: playback bit flag definitions) **/ #define IDX_IO_REC_FLAGS 0x20 /* ??, PU:0x0000 */ #define IDX_IO_REC_IRQTYPE 0x22 /* ??, PU:0x0000 */ @@ -97,40 +120,164 @@ /** DirectX timer, main interrupt area (FIXME: and something else?) **/ #define IDX_IO_TIMER_VALUE 0x60 /* found this timer area by pure luck :-) */ - #define TIMER_VALUE_MASK 0x000fffffUL /* timer countdown value; triggers IRQ when timer is finished */ - #define TIMER_ENABLE_COUNTDOWN 0x01000000UL /* activate the timer countdown */ - #define TIMER_ENABLE_IRQ 0x02000000UL /* trigger timer IRQ on zero transition */ - #define TIMER_ACK_IRQ 0x04000000UL /* being set in IRQ handler in case port 0x00 (hmm, not port 0x64!?!?) had 0x0020 set upon IRQ handler */ + /* timer countdown value; triggers IRQ when timer is finished */ + #define TIMER_VALUE_MASK 0x000fffffUL + /* activate timer countdown */ + #define TIMER_COUNTDOWN_ENABLE 0x01000000UL + /* trigger timer IRQ on zero transition */ + #define TIMER_IRQ_ENABLE 0x02000000UL + /* being set in IRQ handler in case port 0x00 (hmm, not port 0x64!?!?) + * had 0x0020 set upon IRQ handler */ + #define TIMER_IRQ_ACK 0x04000000UL #define IDX_IO_IRQSTATUS 0x64 - #define IRQ_PLAYBACK 0x0001 - #define IRQ_RECORDING 0x0002 - #define IRQ_MPU401 0x0010 - #define IRQ_TIMER 0x0020 /* DirectX timer */ - #define IRQ_UNKNOWN1 0x0040 /* probably unused, or possibly I2S port? or gameport IRQ? */ - #define IRQ_UNKNOWN2 0x0080 /* probably unused, or possibly I2S port? or gameport IRQ? */ + /* some IRQ bit in here might also be used to signal a power-management timer + * timeout, to request shutdown of the chip (e.g. AD1815JS has such a thing). + * Some OPL3 hardware (e.g. in LM4560) has some special timer hardware which + * can trigger an OPL3 timer IRQ, so maybe there's such a thing as well... */ + + #define IRQ_PLAYBACK 0x0001 + #define IRQ_RECORDING 0x0002 + #define IRQ_UNKNOWN1 0x0004 /* most probably I2S port */ + #define IRQ_GAMEPORT 0x0008 /* Interrupt of Digital(ly) Enhanced Game Port */ + #define IRQ_MPU401 0x0010 + #define IRQ_TIMER 0x0020 /* DirectX timer */ + #define IRQ_UNKNOWN2 0x0040 /* probably unused, or possibly I2S port? */ + #define IRQ_UNKNOWN3 0x0080 /* probably unused, or possibly I2S port? */ #define IDX_IO_66H 0x66 /* writing 0xffff returns 0x0000 */ -#define IDX_IO_SOME_VALUE 0x68 /* this is set to e.g. 0x3ff or 0x300, and writable; maybe some buffer limit, but I couldn't find out more, PU:0x00ff */ -#define IDX_IO_6AH 0x6A /* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); actually inhibits PCM playback!!! maybe power management?? */ - #define IO_6A_PAUSE_PLAYBACK 0x0200 /* bit 9; sure, this pauses playback, but what the heck is this really about?? */ -#define IDX_IO_6CH 0x6C -#define IDX_IO_6EH 0x6E /* writing 0xffff returns 0x83fe */ -/* further I/O indices not saved/restored, so probably not used */ + /* this is set to e.g. 0x3ff or 0x300, and writable; + * maybe some buffer limit, but I couldn't find out more, PU:0x00ff: */ +#define IDX_IO_SOME_VALUE 0x68 + #define IO_68_RANDOM_TOGGLE1 0x0100 /* toggles randomly */ + #define IO_68_RANDOM_TOGGLE2 0x0200 /* toggles randomly */ + /* umm, nope, behaviour of these bits changes depending on what we wrote + * to 0x6b!! */ + +/* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); + * actually inhibits PCM playback!!! maybe power management??: */ +#define IDX_IO_6AH 0x6A + /* bit 5: enabling this will activate permanent counting of bytes 2/3 + * at gameport I/O (0xb402/3) (equal values each) and cause + * gameport legacy I/O at 0x0200 to be _DISABLED_! + * Is this Digital Enhanced Game Port Enable??? Or maybe it's Testmode + * for Enhanced Digital Gameport (see 4D Wave DX card): */ + #define IO_6A_SOMETHING1_GAMEPORT 0x0020 + /* bit 8; sure, this _pauses_ playback (later resumes at same spot!), + * but what the heck is this really about??: */ + #define IO_6A_PAUSE_PLAYBACK_BIT8 0x0100 + /* bit 9; sure, this _pauses_ playback (later resumes at same spot!), + * but what the heck is this really about??: */ + #define IO_6A_PAUSE_PLAYBACK_BIT9 0x0200 + /* BIT8 and BIT9 are _NOT_ able to affect OPL3 MIDI playback, + * thus it suggests influence on PCM only!! + * However OTOH there seems to be no bit anywhere around here + * which is able to disable OPL3... */ + /* bit 10: enabling this actually changes values at legacy gameport + * I/O address (0x200); is this enabling of the Digital Enhanced Game Port??? + * Or maybe this simply switches off the NE558 circuit, since enabling this + * still lets us evaluate button states, but not axis states */ + #define IO_6A_SOMETHING2_GAMEPORT 0x0400 + /* writing 0x0300: causes quite some crackling during + * PC activity such as switching windows (PCI traffic?? + * --> FIFO/timing settings???) */ + /* writing 0x0100 plus/or 0x0200 inhibits playback */ + /* since the Windows .INF file has Flag_Enable_JoyStick and + * Flag_Enable_SB_DOS_Emulation directly together, it stands to reason + * that some other bit in this same register might be responsible + * for SB DOS Emulation activation (note that the file did NOT define + * a switch for OPL3!) */ +#define IDX_IO_6CH 0x6C /* unknown; fully read-writable */ +#define IDX_IO_6EH 0x6E + /* writing 0xffff returns 0x83fe (or 0x03fe only). + * writing 0x83 (and only 0x83!!) to 0x6f will cause 0x6c to switch + * from 0000 to ffff. */ + +/* further I/O indices not saved/restored and not readable after writing, + * so probably not used */ -/*** I/O 2 area port indices ***/ +/*** Gameport area port indices ***/ /* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */ -#define AZF_IO_SIZE_IO2 0x08 -#define AZF_IO_SIZE_IO2_PM 0x06 +#define AZF_IO_SIZE_GAME 0x08 +#define AZF_IO_SIZE_GAME_PM 0x06 -#define IDX_IO2_LEGACY_ADDR 0x04 - #define LEGACY_SOMETHING 0x01 /* OPL3?? */ - #define LEGACY_JOY 0x08 +enum { + AZF_GAME_LEGACY_IO_PORT = 0x200 +} AZF_GAME_CONFIGS; + +#define IDX_GAME_LEGACY_COMPATIBLE 0x00 + /* in some operation mode, writing anything to this port + * triggers an interrupt: + * yup, that's in case IDX_GAME_01H has one of the + * axis measurement bits enabled + * (and of course one needs to have GAME_HWCFG_IRQ_ENABLE, too) */ + +#define IDX_GAME_AXES_CONFIG 0x01 + /* NOTE: layout of this register awfully similar (read: "identical??") + * to AD1815JS.pdf (p.29) */ + + /* enables axis 1 (X axis) measurement: */ + #define GAME_AXES_ENABLE_1 0x01 + /* enables axis 2 (Y axis) measurement: */ + #define GAME_AXES_ENABLE_2 0x02 + /* enables axis 3 (X axis) measurement: */ + #define GAME_AXES_ENABLE_3 0x04 + /* enables axis 4 (Y axis) measurement: */ + #define GAME_AXES_ENABLE_4 0x08 + /* selects the current axis to read the measured value of + * (at IDX_GAME_AXIS_VALUE): + * 00 = axis 1, 01 = axis 2, 10 = axis 3, 11 = axis 4: */ + #define GAME_AXES_READ_MASK 0x30 + /* enable to have the latch continuously accept ADC values + * (and continuously cause interrupts in case interrupts are enabled); + * AD1815JS.pdf says it's ~16ms interval there: */ + #define GAME_AXES_LATCH_ENABLE 0x40 + /* joystick data (measured axes) ready for reading: */ + #define GAME_AXES_SAMPLING_READY 0x80 + + /* NOTE: other card specs (SiS960 and others!) state that the + * game position latches should be frozen when reading and be freed + * (== reset?) after reading!!! + * Freezing most likely means disabling 0x40 (GAME_AXES_LATCH_ENABLE), + * but how to free the value? */ + /* An internet search for "gameport latch ADC" should provide some insight + * into how to program such a gameport system. */ + + /* writing 0xf0 to 01H once reset both counters to 0, in some special mode!? + * yup, in case 6AH 0x20 is not enabled + * (and 0x40 is sufficient, 0xf0 is not needed) */ + +#define IDX_GAME_AXIS_VALUE 0x02 + /* R: value of currently configured axis (word value!); + * W: trigger axis measurement */ + +#define IDX_GAME_HWCONFIG 0x04 + /* note: bits 4 to 7 are never set (== 0) when reading! + * --> reserved bits? */ + /* enables IRQ notification upon axes measurement ready: */ + #define GAME_HWCFG_IRQ_ENABLE 0x01 + /* these bits choose a different frequency for the + * internal ADC counter increment. + * hmm, seems to be a combo of bits: + * 00 --> standard frequency + * 10 --> 1/2 + * 01 --> 1/20 + * 11 --> 1/200: */ + #define GAME_HWCFG_ADC_COUNTER_FREQ_MASK 0x06 + + /* enable gameport legacy I/O address (0x200) + * I was unable to locate any configurability for a different address: */ + #define GAME_HWCFG_LEGACY_ADDRESS_ENABLE 0x08 +/*** MPU401 ***/ #define AZF_IO_SIZE_MPU 0x04 #define AZF_IO_SIZE_MPU_PM 0x04 -#define AZF_IO_SIZE_SYNTH 0x08 -#define AZF_IO_SIZE_SYNTH_PM 0x06 +/*** OPL3 synth ***/ +#define AZF_IO_SIZE_OPL3 0x08 +#define AZF_IO_SIZE_OPL3_PM 0x06 +/* hmm, given that a standard OPL3 has 4 registers only, + * there might be some enhanced functionality lurking at the end + * (especially since register 0x04 has a "non-empty" value 0xfe) */ /*** mixer I/O area port indices ***/ /* (only 0x22 of 0x40 bytes saved/restored by Windows driver)

2 3

Re: [alsa-devel] ioctl request code for specific alsa control
by Clemens Ladisch 16 May '08

16 May '08

Peter Wurmsdobler wrote: > What ioctl request code would I have to pass in order to reach this > control's put/get methods? snd_ctl_elem_write() (alsa-lib/src/control/control.c) snd_ctl_hw_elem_write() (alsa-lib/src/control/control_hw.c) ioctl(SNDRV_CTL_IOCTL_ELEM_WRITE) snd_ctl_ioctl() (linux/sound/core/control.c) snd_ctl_elem_write_user() snd_ctl_elem_write() kctl->put() HTH Clemens

2 1

[alsa-devel] Alsa: Get Playback Volume fails if set by outside source
by Ashwin Chandra 16 May '08

16 May '08

Hello, I am having an issue with the Alsa mixer. Currenty I have a function that gets the volume via snd_mixer_selem_get_playback_volume. This function works fine to get the last-set volume if my application is setting the volume. However if I change the volume using an outside source, such as alsamixer or gnome-volume-control and then I run the above function, the old volume that my application had set last is returned. Now I edited amixer's source code to put a while(1) around the same functions (the snd_mixer_selem_get_playback_volume), and it too does not get the correct volume if set from an outside source. Alsamixer on the other hand seems to work find, but I am not quite sure how they do it. I thought maybe I had to reinitialize the alsa interface (ie. snd_mixer_close...and then open, attach, register, etc), but it doesn't seem that alsamixer does that. Does anyone have any suggestions about what I can do to get the correct volume if set by an outside source?? Thanks!

2 1

[alsa-devel] pcm output timing question
by Paul Adolph 15 May '08

15 May '08

ALSA newbie warning... I have a funny application that does av sync by using the timing of a callback to set the audio timebase. In other words, I need to invoke that callback "as close as possible" to the time a previously-sent buffer of samples is presented, i.e. exits the speakers. The application uses the clock time of the arrival of this callback and does some math to figure out whether to repeat or skip video frames. I do not like this arrangement, but unfortunately I am stuck with it. My job is to hook this app to ALSA audio output. My current design uses a producer-consumer model with a thread that writes fixed-size application buffers to the ALSA ring buffer, driven by a message queue fed by the application. This is a little redundant I think, but I do not own the buffers sent by the app so this is what I came up with. The application needs a regular callback with the timing described above each time one of their buffers is played, and it is assumed they are played in sequence. I tried setting the ALSA callback to trigger when there are an application-buffer-size amount of frames available, using snd_pcm_sw_params_set_avail_min(). My idea was that if I primed the ring buffer to be full before it starts transfers with snd_pcm_sw_params_set_start_threshold(), I would get the ALSA callback each time that number of frames exited the ring buffer. But this did not work: I seem to be getting callbacks much more frequently that I wanted, probably on ALSA period boundaries. So this means I do not understand how snd_pcm_sw_params_set_start_threshold() is supposed to work I guess. Will somebody please comment on the viability of and/or problems with my current design, or suggest an alternative way to configure ALSA that will give me the output timing I need? Currently I am testing on an x86 ubuntu box, kernel 2.6.22-14, ALSA 1.0.14, gcc 4.1.3, glibc 2.6.1, but will also cross-compile for several embedded platforms in the future. Thanks in advance.

1 0

[alsa-devel] Oh, BTW... (OPL3 PM issue)
by Andreas Mohr 15 May '08

15 May '08

Hi, unfortunately I forgot to mention this: when I tested OPL3 functionality of my driver, I of course also tested this to do proper power management, and... well... it didn't! The result of doing an S2D was that the formerly perfectly playing emulated MIDI synth ended up with distorted (sort of Micky-Mouse'ish) voices directly after resume. I really don't think that it's my driver's power management which is at fault here, and I'd strongly guesstimate that ALSA does its own generic OPL3 power management handling for standard OPL3 hardware registration. So??? (aka "what the...!?") Thanks, Andreas Mohr

2 1

[alsa-devel] [PATCH asoc-dev (take #2)] ASoC: Convert N810 machine driver to use gpiolib
by Jarkko Nikula 15 May '08

15 May '08

Use gpiolib since it is now available for OMAPs. Change also references to HW version RX44 to product name N810. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/n810.c | 25 ++++++++++++++----------- 1 files changed, 14 insertions(+), 11 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c2afffb..c32487b 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -30,15 +30,15 @@ #include <asm/mach-types.h> #include <asm/arch/hardware.h> -#include <asm/arch/gpio.h> +#include <linux/gpio.h> #include <asm/arch/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" #include "../codecs/tlv320aic3x.h" -#define RX44_HEADSET_AMP_GPIO 10 -#define RX44_SPEAKER_AMP_GPIO 101 +#define N810_HEADSET_AMP_GPIO 10 +#define N810_SPEAKER_AMP_GPIO 101 static struct clk *sys_clkout2; static struct clk *sys_clkout2_src; @@ -154,9 +154,9 @@ static int n810_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 1); + gpio_set_value(N810_SPEAKER_AMP_GPIO, 1); else - omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 0); + gpio_set_value(N810_SPEAKER_AMP_GPIO, 0); return 0; } @@ -165,9 +165,9 @@ static int n810_jack_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 1); + gpio_set_value(N810_HEADSET_AMP_GPIO, 1); else - omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 0); + gpio_set_value(N810_HEADSET_AMP_GPIO, 0); return 0; } @@ -303,12 +303,12 @@ static int __init n810_soc_init(void) clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); - if (omap_request_gpio(RX44_HEADSET_AMP_GPIO) < 0) + if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) BUG(); - if (omap_request_gpio(RX44_SPEAKER_AMP_GPIO) < 0) + if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0) BUG(); - omap_set_gpio_direction(RX44_HEADSET_AMP_GPIO, 0); - omap_set_gpio_direction(RX44_SPEAKER_AMP_GPIO, 0); + gpio_direction_output(N810_HEADSET_AMP_GPIO, 0); + gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0); return 0; err2: @@ -323,6 +323,9 @@ err1: static void __exit n810_soc_exit(void) { + gpio_free(N810_SPEAKER_AMP_GPIO); + gpio_free(N810_HEADSET_AMP_GPIO); + platform_device_unregister(n810_snd_device); } -- 1.5.5.1

2 1

[alsa-devel] [PATCH asoc-dev] ASoC: Convert N810 machine driver to use gpiolib
by Jarkko Nikula 15 May '08

15 May '08

Use gpiolib since it is now available for OMAPs. Change also references to HW version RX44 to product name N810. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/n810.c | 25 ++++++++++++++----------- 1 files changed, 14 insertions(+), 11 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c2afffb..2d982b1 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -30,15 +30,15 @@ #include <asm/mach-types.h> #include <asm/arch/hardware.h> -#include <asm/arch/gpio.h> +#include <asm/gpio.h> #include <asm/arch/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" #include "../codecs/tlv320aic3x.h" -#define RX44_HEADSET_AMP_GPIO 10 -#define RX44_SPEAKER_AMP_GPIO 101 +#define N810_HEADSET_AMP_GPIO 10 +#define N810_SPEAKER_AMP_GPIO 101 static struct clk *sys_clkout2; static struct clk *sys_clkout2_src; @@ -154,9 +154,9 @@ static int n810_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 1); + gpio_set_value(N810_SPEAKER_AMP_GPIO, 1); else - omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 0); + gpio_set_value(N810_SPEAKER_AMP_GPIO, 0); return 0; } @@ -165,9 +165,9 @@ static int n810_jack_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 1); + gpio_set_value(N810_HEADSET_AMP_GPIO, 1); else - omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 0); + gpio_set_value(N810_HEADSET_AMP_GPIO, 0); return 0; } @@ -303,12 +303,12 @@ static int __init n810_soc_init(void) clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); - if (omap_request_gpio(RX44_HEADSET_AMP_GPIO) < 0) + if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) BUG(); - if (omap_request_gpio(RX44_SPEAKER_AMP_GPIO) < 0) + if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0) BUG(); - omap_set_gpio_direction(RX44_HEADSET_AMP_GPIO, 0); - omap_set_gpio_direction(RX44_SPEAKER_AMP_GPIO, 0); + gpio_direction_output(N810_HEADSET_AMP_GPIO, 0); + gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0); return 0; err2: @@ -323,6 +323,9 @@ err1: static void __exit n810_soc_exit(void) { + gpio_free(N810_SPEAKER_AMP_GPIO); + gpio_free(N810_HEADSET_AMP_GPIO); + platform_device_unregister(n810_snd_device); } -- 1.5.5.1

2 2

[alsa-devel] pcm output timing question
by Paul Adolph 14 May '08

14 May '08

ALSA newbie warning... I have a funny application that does av sync by using the timing of a callback to set the audio timebase. In other words, I need to invoke that callback "as close as possible" to the time a previously-sent buffer of samples is presented, i.e. exits the speakers. The application uses the clock time of the arrival of this callback and does some math to figure out whether to repeat or skip video frames. I do not like this arrangement, but unfortunately I am stuck with it. My job is to hook this app to ALSA audio output. My current design uses a producer-consumer model with a thread that writes fixed-size application buffers to the ALSA ring buffer, driven by a message queue fed by the application. This is a little redundant I think, but I do not own the buffers sent by the app so this is what I came up with. The application needs a regular callback with the timing described above each time one of their buffers is played, and it is assumed they are played in sequence. I tried setting the ALSA callback to trigger when there are an application-buffer-size amount of frames available, using snd_pcm_sw_params_set_avail_min(). My idea was that if I primed the ring buffer to be full before it starts transfers with snd_pcm_sw_params_set_start_threshold(), I would get the ALSA callback each time that number of frames exited the ring buffer. But this did not work: I seem to be getting callbacks much more frequently that I wanted, probably on ALSA period boundaries. So this means I do not understand how snd_pcm_sw_params_set_start_threshold() is supposed to work I guess. Will somebody please comment on the viability of and/or problems with my current design, or suggest an alternative way to configure ALSA that will give me the output timing I need? Currently I am testing on an x86 ubuntu box, kernel 2.6.22-14, ALSA 1.0.14, gcc 4.1.3, glibc 2.6.1, but will also cross-compile for several embedded platforms in the future. Thanks in advance.

1 0

[alsa-devel] [Via8237] - VT1618 data sheet
by Goran Sterjov 14 May '08

14 May '08

Hello. i tried out the VT1618 patch from john utz but i failed to get the "Smart 5.1" surround sound working. so i figured that i'd go ahead and experiment with the driver to make it work on my via epia en15000 motherboard. i tried requesting the data sheet from VIA ten ga-gillion times but they only responded once with a "the datasheet is a high level engineering document. visit the technical support forums for help". i tried explaining that i needed the information to further develop an ALSA driver for the linux system but it seems like i wont get my hands on this god forsaken document. Is there anyone out there who has any documentation on the vt1618..? via support is shocking. Nay, beyond shocking. in John's patch he mentions various registers but unfortunately they dont help me without the docs.. Thanks Goran

1 0

[alsa-devel] ioctl request code for specific alsa control
by Peter Wurmsdobler 14 May '08

14 May '08

Hello, The driver for our audio hardware provides 6 snd_kcontrol_new_t structures, with their proper callbacks, members and name. One is called pmic_control_pb_vol (.name = "Master Playback Volume") in the driver, but appears somehow as "Master" in amixer and can be successfully written to using the OSS interface and ioctl as: int mixerFd = open("/dev/mixer", O_RDWR); if (mixerFd >= 0) { unsigned int leftright; ioctl(mixerFd, SOUND_MIXER_WRITE_VOLUME, &leftright); close(mixerFd); } It is a big enigma for me that the ioctl with the request code used above arrives automagically in the particular audio driver and calls pmic_pb_volume_put() defined in pmic_control_pb_vol. Another control is called pmic_control_op_sw (.name = "Master Output Playback Volume") and appears in amixer as "Master Output" (in fact it is not a volume control but a switch). It is possible to set the output device using amixer. My questions are: What is the ioctl request code for a particular snd_kcontrol_new_t defined in the driver? Alternatively, how does the OSS layer in ALSA map the different ALSA controls to the predefined and fixed OSS controls? The hard way (and what I am currently trying as last resort) is to write to all of them and see what happens. Many thanks for your help, peter

2 1

[alsa-devel] From snd_kcontrol_new_t to amixer
by Peter Wurmsdobler 14 May '08

14 May '08

Hello, I am puzzled by an observation that runs against my common sense. The pmic-alsa-mixer.c provides 6 (six) snd_kcontrol_new_t controls with their callback functions, name and other members, then registers themusing snd_ctl_add(...). However, amixer shows only 5 (five), and all with different names. What is the process of generating these names if not from the kernel structures? By looking through the mixer related code in alsa-utils and alsa-libs, I see that somehow the library loads the mixer controls and possibly retrieves all the information provided by them, including their names. Either some piece of code shortens the names defined in the driver, or it maps them to some generic controls. Any suggestions? The system I am working on uses a Freescale imx27 in conjunction with an MC13783 power management IC, also responsible for audio (it provides a voice codec and a stereo dac). The kernel used is a 2.6.19.2 as part of ltib from Freescale. amixer tells me: mx27# amixer Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined cvolume Playback channels: Mono Capture channels: Mono Limits: Playback 0 - 100 Capture 0 - 100 Mono: Playback 0 [0%] Capture 0 [0%] Simple mixer control 'Master Balance',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 100 Mono: Playback 50 [50%] Simple mixer control 'Master Input',0 Capabilities: cvolume Capture channels: Mono Limits: Capture 0 - 7 Mono: Capture 1 [14%] Simple mixer control 'Master Monoconfig',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 3 Mono: Playback 0 [0%] Simple mixer control 'Master Output',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 15 Mono: Playback 12 [80%] snd_kcontrol_new_t pmic_control_pb_vol __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .index = 0x00, .info = pmic_pb_volume_info, .get = pmic_pb_volume_get, .put = pmic_pb_volume_put, .private_value = 0xffab1, }; snd_kcontrol_new_t pmic_control_pb_bal __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Balance Playback Volume", .index = 0x00, .info = pmic_pb_balance_info, .get = pmic_pb_balance_get, .put = pmic_pb_balance_put, .private_value = 0xffab2, }; snd_kcontrol_new_t pmic_control_pb_mono __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Monoconfig Playback Volume", .index = 0x00, .info = pmic_pb_monoconfig_info, .get = pmic_pb_monoconfig_get, .put = pmic_pb_monoconfig_put, .private_value = 0xffab2, }; snd_kcontrol_new_t pmic_control_op_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Output Playback Volume", .index = 0x00, .info = pmic_mixer_output_info, .get = pmic_mixer_output_get, .put = pmic_mixer_output_put, .private_value = 0xffab4, }; snd_kcontrol_new_t pmic_control_cap_vol __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Capture Volume", .index = 0x00, .info = pmic_cap_volume_info, .get = pmic_cap_volume_get, .put = pmic_cap_volume_put, .private_value = 0xffab5, }; snd_kcontrol_new_t pmic_control_ip_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Input Capture Volume", .index = 0x00, .info = pmic_cap_input_info, .get = pmic_cap_input_get, .put = pmic_cap_input_put, .private_value = 0xffab5, }; int mxc_alsa_create_ctl(snd_card_t * card, void *p_value) { int err = 0; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_op_sw, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_pb_vol, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_pb_mono, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_pb_bal, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_cap_vol, p_value))) < 0) return err; if ((err = snd_ctl_add(card, snd_ctl_new1(&pmic_control_ip_sw, p_value))) < 0) return err; return 0; } Cheers, peter

2 1

Re: [alsa-devel] Problem with aloop
by Benjamin van den Hout 14 May '08

14 May '08

Gustavo da Silva Serra wrote: I had some problems until I could figure it out how to configure alsa > properly in /etc/asound.conf. Maybe this could be your problem? What is > your alsa version? I heard that aloop didn't work well until 1.0.15, or > such (alsa guys: is this true?). > My alsa is 1.0.14 but I'm using the 'HG' version of the alsa-drivers and alsa-kernel packages so that basically gets me the latest-and-greatest drivers from the source repository if I understand correctly. I saw some patches coming by in February and they are already in there so I presume I'm working with the most current source. I'm intrigued by the asound.conf, is there any specific configuration you needed to make the aloop device work properly? I'm using the distro-configured alsa (ubuntu 7.10) and there is no such file somehwere in /etc. Even on my debian server (running Etch), I can't find it. Kind regards, Benjamin

2 2

[alsa-devel] STB TV PCI FM
by Michael Papet 14 May '08

14 May '08

6000704 2.6.24 To: alsa-devel(a)alsa-project.org MIME-Version: 1.0 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: 8bit Message-ID: <934014.92322.qm(a)web62006.mail.re1.yahoo.com> Hi, I'm trying to finish installing an old STB TV PCI FM 6000704 tv capture card working and there's no sound. I'm on Debian Lenny/Testing kernel version 2.6.24. The card itself does not detect or load snd-bt87x when booting. I force it to load with the load_all kernel option when I modprobe. It doesn't seem to use msp3400 in any way. In theory,/proc/asound/card1/pcm0c is the stereo capture, but I can't figure out how to make it capture in stereo. I can set the digital_rate as needed if that will get stereo going. I've tried 36000 and 44100 with no success, but it could be because .asoundrc isn't right. Check http://pastebin.ca/1017578 for alsa-info.sh output. Any advice, or perhaps an .asoundrc would be great. Best regards, Michael modprobe -v snd-bt87x dmesg>> "unknown card 0x878-0x10b4:0x2636 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> bt87x0: Using board 0, analog, digital (rate 32000 Hz"

1 0

[alsa-devel] Auzen X-Fi Prelude
by Kamil Kaminski 14 May '08

14 May '08

Hi, When will be there a working driver for Auzen Prelude card?

1 0

Re: [alsa-devel] Alsa-devel Digest, Vol 15, Issue 30
by Benjamin van den Hout 13 May '08

13 May '08

Gustavo da Silva Serra wrote: I tryied to use the HG versions, but they didn't work well for me. Maybe > you should give the 1.0.16 a try, just to make sure. > I need an asound.conf since my network application uses a very low > latency, that is not default. I am attaching my asound.conf. > Another possible source of problems are the artsd and esd daemons, are > they running? > > Hi Gustavo, I just downloaded the 10.0.16 version and the code for aloop-kernel.c looked quite different from the one in the HG version. I was under the understanding that the HG version is a current development snapshot that eventually gets tagged into an official 1.0.1xx version. That's not the case then? Anyway, I will try with the 10.0.16 version, I see some stuff about a #define called SND_CARD_LOOPBACK_START_SYNC, that sounds promising perhaps. I've also looked at the config file you attached, the stuff about period_time and buffer_size also looks quite interesting and is perhaps something I should play with then :) I'll let you know how things work out! Thanks for your continued help, it is greatly appreciated... Kind regards, Benjamin

1 0

Re: [alsa-devel] [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver
by Mark Brown 13 May '08

13 May '08

On Mon, May 12, 2008 at 06:45:12PM +0800, Bryan Wu wrote: > From: Cliff Cai <cliff.cai(a)analog.com> > > Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> > Signed-off-by: Bryan Wu <cooloney(a)kernel.org> Thanks, I've applied this to the ASoC git tree. CCing in alsa-devel(a)alsa-project.org - ALSA patches should go via there. > +static int ad1980_soc_suspend(struct platform_device *pdev, > + pm_message_t state) > +{ > + return 0; > +} > + > +static int ad1980_soc_resume(struct platform_device *pdev) > +{ > + return 0; > +} Are you sure about these? I would expect the suspend and resume functions to either do some register writes or be omitted if they don't do anything. Standard AC97 codecs would have some power management via register 0x26 if they were doing anything.

3 5

[alsa-devel] [PATCH 01/10] ASoC: build fix for snd_soc_info_bool_ext
by Mark Brown 13 May '08

13 May '08

From: Philipp Zabel <philipp.zabel(a)gmail.com> I suspect that snd_ctl_boolean_mono should have been snd_ctl_boolean_mono_info instead. This fixes the build for magician. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e6ea6f7..d3c8c03 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -238,7 +238,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -#define snd_soc_info_bool_ext snd_ctl_boolean_mono +#define snd_soc_info_bool_ext snd_ctl_boolean_mono_info int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, -- 1.5.5.1

3 18

[alsa-devel] Acer Aspire 8920g / snd-hda-intel / ALC889
by Andy Green 13 May '08

13 May '08

Hi folks - I have a new 8920g on Fedora development running 2.6.25 which has $ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.16rc2 (Thu Jan 31 16:40:16 2008 UTC). I get a decent looking set of alsa things in /proc/asound $ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc300000 irq 22 and dmesg says that model=acer was selected reasonably enough. I can play audio down the logical sound devices, including cat /dev/urandom > /dev/dsp and the interrupt count goes up in /proc/interrupts. Alsamixer shows a bunch of things and I twiddled them all to no avail. There has never been even a click from the speakers or headphone jack on this machine. At least three funnies on this machine: 1) It is confused if it has ALC889 or ALC883 in there $ grep ALC /proc/asound/* -R /proc/asound/card0/codec#0:Codec: Realtek ALC889 <==== 889 /proc/asound/card0/pcm2c/sub0/info:id: ALC883 Analog <=== 883... /proc/asound/card0/pcm2c/sub0/info:name: ALC883 Analog /proc/asound/card0/pcm2c/info:id: ALC883 Analog /proc/asound/card0/pcm2c/info:name: ALC883 Analog /proc/asound/card0/pcm0c/sub0/info:id: ALC883 Analog /proc/asound/card0/pcm0c/sub0/info:name: ALC883 Analog /proc/asound/card0/pcm0c/info:id: ALC883 Analog /proc/asound/card0/pcm0c/info:name: ALC883 Analog /proc/asound/card0/pcm0p/sub0/info:id: ALC883 Analog /proc/asound/card0/pcm0p/sub0/info:name: ALC883 Analog /proc/asound/card0/pcm0p/info:id: ALC883 Analog /proc/asound/card0/pcm0p/info:name: ALC883 Analog /proc/asound/Intel/codec#0:Codec: Realtek ALC889 <=== 889 /proc/asound/Intel/pcm2c/sub0/info:id: ALC883 Analog /proc/asound/Intel/pcm2c/sub0/info:name: ALC883 Analog /proc/asound/Intel/pcm2c/info:id: ALC883 Analog /proc/asound/Intel/pcm2c/info:name: ALC883 Analog /proc/asound/Intel/pcm0c/sub0/info:id: ALC883 Analog /proc/asound/Intel/pcm0c/sub0/info:name: ALC883 Analog /proc/asound/Intel/pcm0c/info:id: ALC883 Analog /proc/asound/Intel/pcm0c/info:name: ALC883 Analog /proc/asound/Intel/pcm0p/sub0/info:id: ALC883 Analog /proc/asound/Intel/pcm0p/sub0/info:name: ALC883 Analog /proc/asound/Intel/pcm0p/info:id: ALC883 Analog /proc/asound/Intel/pcm0p/info:name: ALC883 Analog /proc/asound/oss/sndstat:0: ALC883 Analog (DUPLEX) /proc/asound/oss/sndstat:0: Realtek ALC889 /proc/asound/pcm:00-02: ALC883 Analog : ALC883 Analog : capture 1 /proc/asound/pcm:00-00: ALC883 Analog : ALC883 Analog : playback 1 : capture 1 2) from dmesg ALSA sound/pci/hda/hda_intel.c:1810: chipset global capabilities = 0x4401 ALSA sound/pci/hda/hda_intel.c:749: codec_mask = 0x3 ALSA sound/pci/hda/hda_codec.c:2211: hda_codec: model 'acer' is selected ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Headphone Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Line-Out Playback Volume, skipped ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Speaker Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave Mono Playback Switch, skipped ALSA sound/pci/hda/hda_codec.c:1073: Cannot find slave IEC958 Playback Switch, skipped 3) there is a funny embedded capacitive volume control thing on this device, whether it gates output by itself or is just a bunch of special keyboard keys, I dunno. Any ideas, more information needed? -Andy

2 14

[alsa-devel] basic question on .pointer function
by Harsha priya gupta 13 May '08

13 May '08

Hi, It would be really helpful if someone can help me understand why do we need to implement .pointer and what does it need to do? I am facing an issue to write a driver in this .pointer function. Please help Also if someone can point me to a document where all the playback_ops are defined, it would be really helpful -- -Harsha

2 3

[alsa-devel] Pulseaudio alsa configure hook
by Sjoerd Simons 13 May '08

13 May '08

Hi, The attached patch extends the alsa pulse plugin set with a alsa configuration hook. Allowing one to specify some configuration parameters that only come into effect when pulseaudio is running. For example a configution file like: @hooks [ { func on_pulse_is_running pcm.!default { type pulse } ctl.!default { type pulse } } ] will redirect the default alsa pcm and ctl to pulse iff pulse is running. (Assuming you defined the hook function correctly ofcourse) This is usefull for distributions that don't want to force their users to switch completely to pulseaudio, but have things a bit more dynamic :) The solutions isn't optimal though. It will mean that every program loading accessing alsa will try to make an (extra) connection to pulse to decide what to do. But i think it's the best we can do for now (or at least that i can do with my minimal knowledge of alsa). A nicer solution would be a way to always specify the pulse plugin as default and have a sort of fallback for when that fails. Comments on the patch appreciated :) Sjoerd PS: I'm not on alsa-devel, so please keep me in the CC -- GREAT MOMENTS IN HISTORY (#7): April 2, 1751 Issac Newton becomes discouraged when he falls up a flight of stairs.

2 3

[alsa-devel] [PATCH 2.6.26-rc2] ASoC: Fix wrong enum count for jack_function in N810 machine driver
by Jarkko Nikula 13 May '08

13 May '08

Fix this typo and avoid similar errors by using ARRAY_SIZE macro. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/n810.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 83b1eb4..6533563 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -188,8 +188,8 @@ static const char *audio_map[][3] = { static const char *spk_function[] = {"Off", "On"}; static const char *jack_function[] = {"Off", "Headphone"}; static const struct soc_enum n810_enum[] = { - SOC_ENUM_SINGLE_EXT(2, spk_function), - SOC_ENUM_SINGLE_EXT(3, jack_function), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function), }; static const struct snd_kcontrol_new aic33_n810_controls[] = { -- 1.5.5.1

3 2

[alsa-devel] ARM S3C2410: Trigger start problems
by ing. Davide Rizzo 13 May '08

13 May '08

I experienced a curious problem. My platform is a S3C2410 with a PCM3006 codec connected through I2S. The soc trigger function calls 3 other triggers functions: - first the codec function - then the platform function (that starts dma) - last the the i2s trigger function (that starts i2s peripheral). The dma start function programs and starts the dma peripheral, but the transfer doesn't start because the i2s peripheral isn't started yet. The s3c2410 specific dma start programs the 1st transfer, but before programming the 2nd start it waits about 30ms (2^18 count) in a closed loop for 1st transfer to start. After that delay anyway it fails to program the 2nd transfer. After that, when the i2s trigger function is called the 1st samples' group starts, but nobody preprograms the 2nd sample group in the dma peripheral. When the 1st group ends the dma interrupt service routine is called , but it's too late and the 1st samples' group is repeated twice. I tested 2 solutions: - In the soc trigger function, call the i2s trigger function BEFORE the platform trigger function. I don't know if this solution could broke other platforms, but it's the more correct way to solve the problem. - (platform specific): In arch/arm/plat-s3c24xx-dma.c, at the end of s3c2410_dma_getposition() function, insert a call to s3c2410_dma_started(). That function is periodically called by alsa-lib when playing, and permits dma to program the 2nd buffer before (but it's not guaranteed) the 1st group is completed. With this solution, the programming of DMA till waits for the closed loop. I think this is a patch, but not the correct solution. I wonder if other architectures that use DMA suffer the same problem, so that the 1st proposed solution could help them, too. Davide

2 1

Re: [alsa-devel] Problem with aloop
by Benjamin van den Hout 13 May '08

13 May '08

Gustavo da Silva Serra wrote: Well, I discovered that only xinelib has this problem. Xmms and > alsaplayer, for example, don't have this issue. I am testing VLC and it > seems to work fine. I am not surprised, since the xine source code that > deals with alsa is huge and ugly, it must be very hard to mantain that. > Thanks for your reply! I did some more testing and I can reproduce it as well with the 8bit mode, it occurrs less frequently. When I switch back to 16 bit it takes only a few tries with aplay to get the hiccups. So it seems it is a bit more tricky than I initially thought. Xine can be a strange beast some times, so that's why I used aplay as well. Anybody have some more ideas how to tackle this problem or point me in the right direction so I can solve it myself? It seems I am so close to a working solution... Kind regards, Benjamin

2 1

[alsa-devel] [PATCH] ASoC: build fix for snd_soc_info_bool_ext
by Philipp Zabel 13 May '08

13 May '08

I suspect that snd_ctl_boolean_mono should have been snd_ctl_boolean_mono_info instead. This fixes the build for magician. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- include/sound/soc.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e6ea6f7..d3c8c03 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -238,7 +238,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -#define snd_soc_info_bool_ext snd_ctl_boolean_mono +#define snd_soc_info_bool_ext snd_ctl_boolean_mono_info int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, -- 1.5.5.1

3 2

Re: [alsa-devel] [PATCH 1/1] [ASOC]: AD1980 audio codec ASOC driver
by Takashi Iwai 13 May '08

13 May '08

At Mon, 12 May 2008 18:45:12 +0800, Bryan Wu wrote: > diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c (snip) > +static int ad1980_add_controls(struct snd_soc_codec *codec) > +{ > + int err, i; > + > + for (i = 0; i < ARRAY_SIZE(ad1980_snd_ac97_controls); i++) { > + err = snd_ctl_add(codec->card, snd_soc_cnew( \ The backslash isn't needed. > +static unsigned int ac97_read(struct snd_soc_codec *codec, > + unsigned int reg) > +{ > + u16 *cache = codec->reg_cache; > + > + if (reg == AC97_RESET || reg == AC97_INT_PAGING || \ > + reg == AC97_POWERDOWN || reg == AC97_EXTENDED_STATUS \ > + || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2) Ditto. Maybe a switch is a better choice here. > + return soc_ac97_ops.read(codec->ac97, reg); > + else { > + reg = reg >> 1; > + > + if (reg > (ARRAY_SIZE(ad1980_reg))) Isn't it reg >= ARRAY_SIZE(ad1980_reg) ?? > +static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, > + unsigned int val) > +{ > + u16 *cache = codec->reg_cache; > + > + soc_ac97_ops.write(codec->ac97, reg, val); > + reg = reg >> 1; > + if (reg <= (ARRAY_SIZE(ad1980_reg))) And reg < ARRAY_SIZE(ad1980_reg) thanks, Takashi

1 0

[alsa-devel] PCM plugin transfer callback size issue
by Ben C 13 May '08

13 May '08

Hello, I'm writing a PCM external filter type plugin. However, due to the processing I must handle fixed size each time. I can only handle 256 samples at a time. In the transfer callback, there would be no problem if the size is say 1024 each time, I just loop it. But the problem is that sometimes it comes back with other numbers. One instance, it came back with 1114. So I change my code to ONLY process 1024, and return 1024 at the conclusion of the transfer callback, to see if ALSA is smart enough to handle the fact that I didn't process everything. Well, ALSA is smart enough, except the next transfer call, it only sent me the 90 samples that I didn't process. But since I can't process 90 samples, it doesn't help me either; I needed ALSA to combine that 90 samples with the next block of incoming PCM data. My questions: 1) Is there some sort of configuration I can do to constraint the number of samples? 2) I could even manage the extra samples myself with an internal buffer to combine with the next block. But how do I tell ALSA to give me the next block? If I only processed 1024 samples out of 1114, I could store the 90 samples for next time, but if I lie to ALSA and say I did all 1114, wouldn't it try to process 1114 samples instead of 1024? Thanks, Ben ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

1 0

[alsa-devel] Impossible to enter bugs on bugtracker for ALSA firmware
by Tim Jackson 13 May '08

13 May '08

It's impossible to enter bugs on the bugtracker for ALSA firmware - the bugtracker says that the "Category" field is not completed - but there are no options for that dropdown! I think this is a bug in Mantis 1.1.0 as I have seen it on other installations. Workaround is to create a category "Default"...

1 0

[alsa-devel] Mono streams
by Nagar, Mrigendra 12 May '08

12 May '08

Hi, I have recently started working on ALSA libraries. I am facing following problems. It would be great help If some one can help me out. 1) Mono streams: If I try to play A MONO streams. It plays Faster. My driver configuration is i) Blocking mode ii) Access mode is interleaved iii) No of channels are set to 1 2) Buffer under run while playing Mono stream. i) Blocking mode i) Hardware buffer size is 65536 ii) Periods is set to 64 I always encounter buffer under run If I try to decode and play a stream frame by frame. Frame size is constant and it is 2048. If I club two frames and Try to play it. It plays well. In second case buffer size becomes 4096. My question is: Is there any optimal buffer size. Which one should set to have smooth payout? Regards, Mrigen

2 1

[alsa-devel] Embedded I2S / 192khz
by Andrew Paprocki 11 May '08

11 May '08

Is anyone aware of a Linux/ALSA capable chip that can drive 192khz I2S? I poked through the ASoC code and could only see 96khz capability. Thanks, -Andrew

4 6

[alsa-devel] ARM S3C2410: Trigger stop problems
by ing. Davide Rizzo 11 May '08

11 May '08

My platform is a S3C2410 with a PCM3006 codec connected through I2S. I have a problem on trigger stop: some part of the sound is replayed at the sound end. I noticed that the s3c2410-pcm enqueue function at the end of the sound reprograms the dma at initial values, is this wanted ? Why ? Who should call the trigger stop function to stop the sound at the exact final sample ? Davide

2 1

[alsa-devel] [PATCH, RFC] ASoC: uda1380 cleanup
by Philipp Zabel 10 May '08

10 May '08

This patch mostly consists of (mostly checkpatch) cleanups, removal of the custom dbg macro and uda1380_snd_soc_cnew, which doesn't seem to be needed anymore. I also removed the hardcoded WinCE/magician register values from uda1380_pcm_prepare. Still, a number of FIXMEs remain. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/codecs/uda1380.c | 252 ++++++++++++++++++++++---------------------- 1 files changed, 128 insertions(+), 124 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 25e3490..e79d842 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -25,7 +25,6 @@ #include <linux/ioctl.h> #include <linux/delay.h> #include <linux/i2c.h> -#include <sound/driver.h> #include <sound/core.h> #include <sound/control.h> #include <sound/initval.h> @@ -36,20 +35,8 @@ #include "uda1380.h" -#define UDA1380_VERSION "0.5" +#define UDA1380_VERSION "0.6" #define AUDIO_NAME "uda1380" -/* - * Debug - */ - -#define UDA1380_DEBUG 0 - -#ifdef UDA1380_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif /* * uda1380 register cache @@ -109,21 +96,22 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = (value & 0xff00) >> 8; data[2] = value & 0x00ff; - uda1380_write_reg_cache (codec, reg, value); + uda1380_write_reg_cache(codec, reg, value); /* the interpolator & decimator regs must only be written when the * codec DAI is active. */ if (!codec->active && (reg >= UDA1380_MVOL)) return 0; - dbg("uda hw write %x val %x", reg, value); + pr_debug("uda1380: hw write %x val %x\n", reg, value); if (codec->hw_write(codec->control_data, data, 3) == 3) { unsigned int val; i2c_master_send(codec->control_data, data, 1); i2c_master_recv(codec->control_data, data, 2); val = (data[0]<<8) | data[1]; if (val != value) { - dbg("READ BACK VAL %x", (data[0]<<8) | data[1]); + pr_debug("uda1380: READ BACK VAL %x\n", + (data[0]<<8) | data[1]); return -EIO; } return 0; @@ -134,17 +122,54 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, #define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0) /* declarations of ALSA reg_elem_REAL controls */ -static const char *uda1380_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz", - "96kHz"}; -static const char *uda1380_input_sel[] = { "Line", "Mic + Line R", "Line L", "Mic" }; -static const char *uda1380_output_sel[] = {"DAC", "Analog Mixer"}; -static const char *uda1380_spf_mode[] = {"Flat", "Minimum1", "Minimum2", - "Maximum"}; -static const char *uda1380_capture_sel[] = {"ADC", "Digital Mixer"}; -static const char *uda1380_sel_ns[] = { "3rd-order", "5th-order" }; -static const char *uda1380_mix_control[] = { "off", "PCM only", "before sound processing", "after sound processing" }; -static const char *uda1380_sdet_setting[] = { "3200", "4800", "9600", "19200" }; -static const char *uda1380_os_setting[] = { "single-speed", "double-speed (no mixing)", "quad-speed (no mixing)" }; +static const char *uda1380_deemp[] = { + "None", + "32kHz", + "44.1kHz", + "48kHz", + "96kHz", +}; +static const char *uda1380_input_sel[] = { + "Line", + "Mic + Line R", + "Line L", + "Mic", +}; +static const char *uda1380_output_sel[] = { + "DAC", + "Analog Mixer", +}; +static const char *uda1380_spf_mode[] = { + "Flat", + "Minimum1", + "Minimum2", + "Maximum" +}; +static const char *uda1380_capture_sel[] = { + "ADC", + "Digital Mixer" +}; +static const char *uda1380_sel_ns[] = { + "3rd-order", + "5th-order" +}; +static const char *uda1380_mix_control[] = { + "off", + "PCM only", + "before sound processing", + "after sound processing" +}; +static const char *uda1380_sdet_setting[] = { + "3200", + "4800", + "9600", + "19200" +}; +static const char *uda1380_os_setting[] = { + "single-speed", + "double-speed (no mixing)", + "quad-speed (no mixing)" +}; static const struct soc_enum uda1380_deemp_enum[] = { SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp), @@ -179,15 +204,13 @@ static const struct soc_enum uda1380_os_enum = int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - int max = (kcontrol->private_value >> 16) & 0xff; - int min = (kcontrol->private_value >> 24) & 0xff; - - /* 00000000 (0) ...(-0.25 dB)... 11111000 (-78 dB), 11111100 -INF */ + int max = (signed char)((kcontrol->private_value >> 16) & 0xff); + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; - uinfo->value.integer.min = min; - uinfo->value.integer.max = max; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = max-min; return 0; } @@ -205,14 +228,13 @@ int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); int reg = kcontrol->private_value & 0xff; - int max = (kcontrol->private_value >> 16) & 0xff; - int min = (kcontrol->private_value >> 24) & 0xff; + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + int val = snd_soc_read(codec, reg); ucontrol->value.integer.value[0] = - (signed char)(snd_soc_read(codec, reg) & 0xff); + ((signed char)(val & 0xff))-min; ucontrol->value.integer.value[1] = - (signed char)((snd_soc_read(codec, reg) >> 8) & 0xff); - + ((signed char)((val >> 8) & 0xff))-min; return 0; } @@ -230,56 +252,65 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); int reg = kcontrol->private_value & 0xff; - int max = (kcontrol->private_value >> 16) & 0xff; - int min = (kcontrol->private_value >> 24) & 0xff; - int err; - unsigned short val, val2, val_mask; - - val = (signed char)ucontrol->value.integer.value[0] & 0xff; - val2 = (signed char)ucontrol->value.integer.value[1] & 0xff; - val |= val2<<8; - err = snd_soc_update_bits(codec, reg, val_mask, val); - return err; + int min = (signed char)((kcontrol->private_value >> 24) & 0xff); + unsigned short val; + + val = (ucontrol->value.integer.value[0]+min) & 0xff; + val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; + + return snd_soc_update_bits(codec, reg, 0xffff, val); } -DECLARE_TLV_DB_SCALE(amix_tlv, -4950, 150, 1); /* from -48 dB in 1.5 dB steps (mute instead of -49.5 dB) */ +/* + * from -48 dB in 1.5 dB steps (mute instead of -49.5 dB) + */ +DECLARE_TLV_DB_SCALE(amix_tlv, -4950, 150, 1); +/* + * from -78 dB in 1 dB steps (3 dB steps, really. LSB are ignored), + * from -66 dB in 0.5 dB steps (2 dB steps, really) and + * from -52 dB in 0.25 dB steps + */ static const unsigned int mvol_tlv[] = { TLV_DB_RANGE_HEAD(3), - 0,15, TLV_DB_SCALE_ITEM(-8200, 100, 1), /* from -78 dB in 1 dB steps (3 dB steps, really) */ - 16,43, TLV_DB_SCALE_ITEM(-6600, 50, 0), /* from -66 dB in 0.5 dB steps (2 dB steps, really) */ - 44,252, TLV_DB_SCALE_ITEM(-5200, 25, 0), /* from -52 dB in 0.25 dB steps */ + 0, 15, TLV_DB_SCALE_ITEM(-8200, 100, 1), + 16, 43, TLV_DB_SCALE_ITEM(-6600, 50, 0), + 44, 252, TLV_DB_SCALE_ITEM(-5200, 25, 0), }; +/* + * from -72 dB in 1.5 dB steps (6 dB steps really), + * from -66 dB in 0.75 dB steps (3 dB steps really), + * from -60 dB in 0.5 dB steps (2 dB steps really) and + * from -46 dB in 0.25 dB steps + */ static const unsigned int vc_tlv[] = { TLV_DB_RANGE_HEAD(4), - 0,7, TLV_DB_SCALE_ITEM(-7800, 150, 1), /* from -72 dB in 1.5 dB steps (6 dB steps really) */ - 8,15, TLV_DB_SCALE_ITEM(-6600, 75, 0), /* from -66 dB in 0.75 dB steps (3 dB steps really) */ - 16,43, TLV_DB_SCALE_ITEM(-6000, 50, 0), /* from -60 dB in 0.5 dB steps (2 dB steps really) */ - 44,228, TLV_DB_SCALE_ITEM(-4600, 25, 0),/* from -46 dB in 0.25 dB steps */ -}; - -DECLARE_TLV_DB_SCALE(tr_tlv, 0, 200, 0); /* from 0 dB to 6 dB in 2 dB steps, if SPF mode != flat */ -DECLARE_TLV_DB_SCALE(bb_tlv, 0, 200, 0); /* from 0 dB to 24 dB in 2 dB steps, if SPF mode == maximum */ - /* (SPF mode == flat cuts off at 18 dB max */ - -static const unsigned int dec_tlv[] = { - TLV_DB_RANGE_HEAD(1), - -128,48, TLV_DB_SCALE_ITEM(-6350, 50, 1), - /* 0011000 (48 dB) ...(0.5 dB)... 10000001 (-63 dB) 10000000 -INF */ + 0, 7, TLV_DB_SCALE_ITEM(-7800, 150, 1), + 8, 15, TLV_DB_SCALE_ITEM(-6600, 75, 0), + 16, 43, TLV_DB_SCALE_ITEM(-6000, 50, 0), + 44, 228, TLV_DB_SCALE_ITEM(-4600, 25, 0), }; -DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); /* from 0 dB to 24 dB in 3 dB steps */ -DECLARE_TLV_DB_SCALE(vga_tlv, 0, 200, 0); /* from 0 dB to 30 dB in 2 dB steps */ - +DECLARE_TLV_DB_SCALE(tr_tlv, 0, 200, 0); /* from 0 to 6 dB in 2 dB steps + if SPF mode != flat */ +DECLARE_TLV_DB_SCALE(bb_tlv, 0, 200, 0); /* from 0 to 24 dB in 2 dB steps, + if SPF mode == maximum, otherwise + cuts off at 18 dB max) */ +DECLARE_TLV_DB_SCALE(dec_tlv, -6400, 50, 1); /* from -63 to 24 dB in 0.5 dB + steps (-128...48) */ +DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); /* from 0 to 24 dB in 3 dB steps */ +DECLARE_TLV_DB_SCALE(vga_tlv, 0, 200, 0); /* from 0 to 30 dB in 2 dB steps */ #define SOC_DOUBLE_S8_TLV(xname, reg, min, max, tlv_array) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ - .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ|SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .tlv.p = (tlv_array), \ - .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ - .put = snd_soc_put_volsw_s8, \ - .private_value = (reg) | (((signed char)max) << 16) | (((signed char)min) << 24) } +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ + .put = snd_soc_put_volsw_s8, \ + .private_value = (reg) | (((signed char)max) << 16) | \ + (((signed char)min) << 24) } static const struct snd_kcontrol_new uda1380_snd_controls[] = { SOC_DOUBLE_TLV("Analog Mixer Volume", UDA1380_AMIX, 0, 8, 44, 1, amix_tlv), /* AVCR, AVCL */ @@ -314,24 +345,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = { SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0), }; -/** - * uda1380_snd_soc_cnew - * - * temporary copy of snd_soc_cnew that doesn't overwrite .access - */ -struct snd_kcontrol *uda1380_snd_soc_cnew(const struct snd_kcontrol_new *_template, - void *data, char *long_name) -{ - struct snd_kcontrol_new template; - - memcpy(&template, _template, sizeof(template)); - if (long_name) - template.name = long_name; - template.index = 0; - - return snd_ctl_new1(&template, data); -} - /* add non dapm controls */ static int uda1380_add_controls(struct snd_soc_codec *codec) { @@ -339,7 +352,7 @@ static int uda1380_add_controls(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) { err = snd_ctl_add(codec->card, - uda1380_snd_soc_cnew(&uda1380_snd_controls[i],codec, NULL)); + snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL)); if (err < 0) return err; } @@ -447,7 +460,7 @@ static int uda1380_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, int iface; /* set up DAI based upon fmt */ - iface = uda1380_read_reg_cache (codec, UDA1380_IFACE); + iface = uda1380_read_reg_cache(codec, UDA1380_IFACE); iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK); /* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */ @@ -489,22 +502,16 @@ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) } else { reg_start = UDA1380_DEC; reg_end = UDA1380_AGC; - /* FIXME - wince values */ -// uda1380_write(codec, UDA1380_PM, uda1380_read_reg_cache (codec, UDA1380_PM) | R02_PON_LNA | R02_PON_DAC); - uda1380_write(codec, UDA1380_PM, 0x851f); -// uda1380_write_reg_cache(codec, UDA1380_DEC, 0x0000); -// uda1380_write_reg_cache(codec, UDA1380_PGA, uda1380_read_reg_cache (codec, UDA1380_PM) & ~R21_MT_ADC); /* unmute */ -// uda1380_write_reg_cache(codec, UDA1380_ADC, 0x0001/*0x090d*/); - // ADC_POL_INV=0 VGA_CTRL=9(1001:18dB) gain SEL_LNA=1 SEL_MIC=1 EN_DCFIL=1 } /* FIXME disable DAC_CLK */ - clk = uda1380_read_reg_cache (codec, 00); + clk = uda1380_read_reg_cache(codec, UDA1380_CLK); uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK); - for (reg = reg_start; reg <= reg_end; reg++ ) { - dbg("flush reg %x val %x:",reg, uda1380_read_reg_cache (codec, reg)); - uda1380_write(codec, reg, uda1380_read_reg_cache (codec, reg)); + for (reg = reg_start; reg <= reg_end; reg++) { + pr_debug("uda1380: flush reg %x val %x:", reg, + uda1380_read_reg_cache(codec, reg)); + uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg)); } /* FIXME enable DAC_CLK */ @@ -644,7 +651,7 @@ struct snd_soc_codec_dai uda1380_dai[] = { .set_fmt = uda1380_set_dai_fmt, }, }, -{/* playback only - dual interface */ +{ /* playback only - dual interface */ .name = "UDA1380", .playback = { .stream_name = "Playback", @@ -714,7 +721,7 @@ static int uda1380_resume(struct platform_device *pdev) /* * initialise the UDA1380 driver - * register the mixer and dsp interfaces with the kernel + * register mixer and dsp interfaces with the kernel */ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) { @@ -728,18 +735,17 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) codec->dapm_event = uda1380_dapm_event; codec->dai = uda1380_dai; codec->num_dai = ARRAY_SIZE(uda1380_dai); - codec->reg_cache_size = sizeof(uda1380_reg); codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg), GFP_KERNEL); - if (codec->reg_cache == NULL) return -ENOMEM; - + codec->reg_cache_size = sizeof(uda1380_reg); + codec->reg_cache_step = 2; uda1380_reset(codec); /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if(ret < 0) { - printk(KERN_ERR "uda1380: failed to create pcms\n"); + if (ret < 0) { + pr_err("uda1380: failed to create pcms\n"); goto pcm_err; } @@ -760,7 +766,7 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) uda1380_add_widgets(codec); ret = snd_soc_register_card(socdev); if (ret < 0) { - printk(KERN_ERR "uda1380: failed to register card\n"); + pr_err("uda1380: failed to register card\n"); goto card_err; } @@ -815,13 +821,13 @@ static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind) ret = i2c_attach_client(i2c); if (ret < 0) { - printk(KERN_ERR "failed to attach codec at addr %x\n", addr); + pr_err("uda1380: failed to attach codec at addr %x\n", addr); goto err; } ret = uda1380_init(socdev, setup->dac_clk); if (ret < 0) { - printk(KERN_ERR "failed to initialise UDA1380\n"); + pr_err("uda1380: failed to initialise UDA1380\n"); goto err; } return ret; @@ -834,7 +840,7 @@ err: static int uda1380_i2c_detach(struct i2c_client *client) { - struct snd_soc_codec* codec = i2c_get_clientdata(client); + struct snd_soc_codec *codec = i2c_get_clientdata(client); i2c_detach_client(client); kfree(codec->reg_cache); kfree(client); @@ -846,7 +852,6 @@ static int uda1380_i2c_attach(struct i2c_adapter *adap) return i2c_probe(adap, &addr_data, uda1380_codec_probe); } -/* corgi i2c codec control layer */ static struct i2c_driver uda1380_i2c_driver = { .driver = { .name = "UDA1380 I2C Codec", @@ -868,10 +873,10 @@ static int uda1380_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct uda1380_setup_data *setup; - struct snd_soc_codec* codec; + struct snd_soc_codec *codec; int ret = 0; - printk(KERN_INFO "UDA1380 Audio Codec %s", UDA1380_VERSION); + pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION); setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); @@ -902,14 +907,14 @@ static int uda1380_probe(struct platform_device *pdev) static int uda1380_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec* codec = socdev->codec; + struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3cold); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&uda1380_i2c_driver); #endif kfree(codec); @@ -923,7 +928,6 @@ struct snd_soc_codec_device soc_codec_dev_uda1380 = { .suspend = uda1380_suspend, .resume = uda1380_resume, }; - EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); MODULE_AUTHOR("Giorgio Padrin"); -- 1.5.5.1

2 1

[alsa-devel] [PATCH, RFC] ASoC: update pxa2xx-ssp to work with current mainline
by Philipp Zabel 10 May '08

10 May '08

This patch against the ASoC dev branch updates pxa2xx-ssp to work with current mainline. I only made minimal changes to clock/register access and some dbg->pr_debug cleanup, I guess the DMA setup should be reworked properly, but I'm not familiar enough with that. Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/pxa2xx-ssp.c | 266 +++++++++++++++++++++++++------------------- 1 files changed, 151 insertions(+), 115 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ssp.c b/sound/soc/pxa/pxa2xx-ssp.c index 25118cc..6b1baee 100644 --- a/sound/soc/pxa/pxa2xx-ssp.c +++ b/sound/soc/pxa/pxa2xx-ssp.c @@ -20,8 +20,9 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/clk.h> +#include <linux/io.h> -#include <sound/driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> @@ -30,21 +31,13 @@ #include <asm/hardware.h> #include <asm/arch/pxa-regs.h> +#include <asm/arch/regs-ssp.h> #include <asm/arch/audio.h> #include <asm/arch/ssp.h> #include "pxa2xx-pcm.h" #include "pxa2xx-ssp.h" -#define PXA_SSP_DEBUG 0 - -#if PXA_SSP_DEBUG -#define dbg(format, arg...) \ - printk(KERN_DEBUG format "\n" , ## arg) -#else -#define dbg(format, arg...) do {} while (0) -#endif - /* * SSP audio private data */ @@ -58,9 +51,13 @@ static struct ssp_dev ssp[3]; static struct ssp_state ssp_state[3]; #endif +#define PXA2xx_SSP1_BASE 0x41000000 +#define PXA27x_SSP2_BASE 0x41700000 +#define PXA27x_SSP3_BASE 0x41900000 + static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_mono_out = { .name = "SSP1 PCM Mono out", - .dev_addr = __PREG(SSDR_P1), + .dev_addr = PXA2xx_SSP1_BASE + SSDR, .drcmr = &DRCMRTXSSDR, .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST16 | DCMD_WIDTH2, @@ -68,7 +65,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_mono_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_mono_in = { .name = "SSP1 PCM Mono in", - .dev_addr = __PREG(SSDR_P1), + .dev_addr = PXA2xx_SSP1_BASE + SSDR, .drcmr = &DRCMRRXSSDR, .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST16 | DCMD_WIDTH2, @@ -76,7 +73,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_mono_in = { static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_stereo_out = { .name = "SSP1 PCM Stereo out", - .dev_addr = __PREG(SSDR_P1), + .dev_addr = PXA2xx_SSP1_BASE + SSDR, .drcmr = &DRCMRTXSSDR, .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST16 | DCMD_WIDTH4, @@ -84,7 +81,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_stereo_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_stereo_in = { .name = "SSP1 PCM Stereo in", - .dev_addr = __PREG(SSDR_P1), + .dev_addr = PXA2xx_SSP1_BASE + SSDR, .drcmr = &DRCMRRXSSDR, .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST16 | DCMD_WIDTH4, @@ -92,7 +89,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_stereo_in = { static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_mono_out = { .name = "SSP2 PCM Mono out", - .dev_addr = __PREG(SSDR_P2), + .dev_addr = PXA27x_SSP2_BASE + SSDR, .drcmr = &DRCMRTXSS2DR, .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST16 | DCMD_WIDTH2, @@ -100,7 +97,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_mono_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_mono_in = { .name = "SSP2 PCM Mono in", - .dev_addr = __PREG(SSDR_P2), + .dev_addr = PXA27x_SSP2_BASE + SSDR, .drcmr = &DRCMRRXSS2DR, .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST16 | DCMD_WIDTH2, @@ -108,7 +105,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_mono_in = { static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_stereo_out = { .name = "SSP2 PCM Stereo out", - .dev_addr = __PREG(SSDR_P2), + .dev_addr = PXA27x_SSP2_BASE + SSDR, .drcmr = &DRCMRTXSS2DR, .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST16 | DCMD_WIDTH4, @@ -116,7 +113,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_stereo_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_stereo_in = { .name = "SSP2 PCM Stereo in", - .dev_addr = __PREG(SSDR_P2), + .dev_addr = PXA27x_SSP2_BASE + SSDR, .drcmr = &DRCMRRXSS2DR, .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST16 | DCMD_WIDTH4, @@ -124,7 +121,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_stereo_in = { static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_mono_out = { .name = "SSP3 PCM Mono out", - .dev_addr = __PREG(SSDR_P3), + .dev_addr = PXA27x_SSP3_BASE + SSDR, .drcmr = &DRCMRTXSS3DR, .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST16 | DCMD_WIDTH2, @@ -132,7 +129,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_mono_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_mono_in = { .name = "SSP3 PCM Mono in", - .dev_addr = __PREG(SSDR_P3), + .dev_addr = PXA27x_SSP3_BASE + SSDR, .drcmr = &DRCMRRXSS3DR, .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST16 | DCMD_WIDTH2, @@ -140,7 +137,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_mono_in = { static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_stereo_out = { .name = "SSP3 PCM Stereo out", - .dev_addr = __PREG(SSDR_P3), + .dev_addr = PXA27x_SSP3_BASE + SSDR, .drcmr = &DRCMRTXSS3DR, .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST16 | DCMD_WIDTH4, @@ -148,7 +145,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_stereo_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_stereo_in = { .name = "SSP3 PCM Stereo in", - .dev_addr = __PREG(SSDR_P3), + .dev_addr = PXA27x_SSP3_BASE + SSDR, .drcmr = &DRCMRRXSS3DR, .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST16 | DCMD_WIDTH4, @@ -156,11 +153,11 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_stereo_in = { static struct pxa2xx_pcm_dma_params *ssp_dma_params[3][4] = { {&pxa2xx_ssp1_pcm_mono_out, &pxa2xx_ssp1_pcm_mono_in, - &pxa2xx_ssp1_pcm_stereo_out,&pxa2xx_ssp1_pcm_stereo_in,}, + &pxa2xx_ssp1_pcm_stereo_out, &pxa2xx_ssp1_pcm_stereo_in,}, {&pxa2xx_ssp2_pcm_mono_out, &pxa2xx_ssp2_pcm_mono_in, &pxa2xx_ssp2_pcm_stereo_out, &pxa2xx_ssp2_pcm_stereo_in,}, {&pxa2xx_ssp3_pcm_mono_out, &pxa2xx_ssp3_pcm_mono_in, - &pxa2xx_ssp3_pcm_stereo_out,&pxa2xx_ssp3_pcm_stereo_in,}, + &pxa2xx_ssp3_pcm_stereo_out, &pxa2xx_ssp3_pcm_stereo_in,}, }; static int pxa2xx_ssp_startup(struct snd_pcm_substream *substream) @@ -170,7 +167,7 @@ static int pxa2xx_ssp_startup(struct snd_pcm_substream *substream) int ret = 0; if (!rtd->dai->cpu_dai->active) { - ret = ssp_init (&ssp[cpu_dai->id], cpu_dai->id + 1, + ret = ssp_init(&ssp[cpu_dai->id], cpu_dai->id + 1, SSP_NO_IRQ); if (ret < 0) return ret; @@ -190,12 +187,6 @@ static void pxa2xx_ssp_shutdown(struct snd_pcm_substream *substream) } } -#if defined (CONFIG_PXA27x) -static int cken[3] = {CKEN_SSP1, CKEN_SSP2, CKEN_SSP3}; -#else -static int cken[3] = {CKEN_SSP, CKEN_NSSP, CKEN_ASSP}; -#endif - #ifdef CONFIG_PM static int pxa2xx_ssp_suspend(struct platform_device *pdev, @@ -205,7 +196,7 @@ static int pxa2xx_ssp_suspend(struct platform_device *pdev, return 0; ssp_save_state(&ssp[dai->id], &ssp_state[dai->id]); - pxa_set_cken(cken[dai->id], 0); + clk_disable(ssp[dai->id].ssp->clk); return 0; } @@ -215,7 +206,7 @@ static int pxa2xx_ssp_resume(struct platform_device *pdev, if (!dai->active) return 0; - pxa_set_cken(cken[dai->id], 1); + clk_enable(ssp[dai->id].ssp->clk); ssp_restore_state(&ssp[dai->id], &ssp_state[dai->id]); ssp_enable(&ssp[dai->id]); @@ -227,17 +218,31 @@ static int pxa2xx_ssp_resume(struct platform_device *pdev, #define pxa2xx_ssp_resume NULL #endif +/** + * ssp_set_clkdiv - set SSP clock divider + * @div: serial clock rate divider + */ +void ssp_set_scr(struct ssp_dev *dev, u32 div) +{ + struct ssp_device *ssp = dev->ssp; + u32 sscr0 = __raw_readl(ssp->mmio_base + SSCR0) & ~SSCR0_SCR; + + __raw_writel(sscr0 | SSCR0_SerClkDiv(div), ssp->mmio_base + SSCR0); +} + /* * Set the SSP ports SYSCLK. */ static int pxa2xx_ssp_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { - int port = cpu_dai->id + 1; - u32 sscr0 = SSCR0_P(port) & + void __iomem *mmio_base = ssp[cpu_dai->id].ssp->mmio_base; + int val; + + u32 sscr0 = __raw_readl(mmio_base + SSCR0) & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC); - dbg("pxa2xx_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d", + pr_debug("pxa2xx_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d", cpu_dai->id, clk_id, freq); switch (clk_id) { @@ -245,11 +250,10 @@ static int pxa2xx_ssp_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, sscr0 |= SSCR0_MOD; case PXA2XX_SSP_CLK_PLL: /* Internal PLL is fixed on pxa25x and pxa27x */ -#ifdef CONFIG_PXA27x - ssp_clk[cpu_dai->id].sysclk = 13000000; -#else - ssp_clk[cpu_dai->id].sysclk = 1843200; -#endif + if (cpu_is_pxa25x()) + ssp_clk[cpu_dai->id].sysclk = 1843200; + else + ssp_clk[cpu_dai->id].sysclk = 13000000; break; case PXA2XX_SSP_CLK_EXT: ssp_clk[cpu_dai->id].sysclk = freq; @@ -261,7 +265,7 @@ static int pxa2xx_ssp_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, break; case PXA2XX_SSP_CLK_AUDIO: ssp_clk[cpu_dai->id].sysclk = 0; - SSCR0_P(port) |= SSCR0_SerClkDiv(1); + ssp_set_scr(&ssp[cpu_dai->id], 1); sscr0 |= SSCR0_ADC; break; default: @@ -269,9 +273,10 @@ static int pxa2xx_ssp_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, } /* the SSP CKEN clock must be disabled when changing SSP clock mode */ - pxa_set_cken(cken[cpu_dai->id], 0); - SSCR0_P(port) |= sscr0; - pxa_set_cken(cken[cpu_dai->id], 1); + clk_disable(ssp[cpu_dai->id].ssp->clk); + val = __raw_readl(mmio_base + SSCR0) | sscr0; + __raw_writel(val, mmio_base + SSCR0); + clk_enable(ssp[cpu_dai->id].ssp->clk); return 0; } @@ -281,21 +286,22 @@ static int pxa2xx_ssp_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, static int pxa2xx_ssp_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, int div_id, int div) { - int port = cpu_dai->id + 1; + void __iomem *mmio_base = ssp[cpu_dai->id].ssp->mmio_base; + int val; switch (div_id) { case PXA2XX_SSP_AUDIO_DIV_ACDS: - SSACD_P(port) &= ~ 0x7; - SSACD_P(port) |= SSACD_ACDS(div); + val = (__raw_readl(mmio_base + SSACD) & ~0x7) | SSACD_ACDS(div); + __raw_writel(val, mmio_base + SSACD); break; case PXA2XX_SSP_AUDIO_DIV_SCDB: - SSACD_P(port) &= ~0x8; + val = __raw_readl(mmio_base + SSACD) & ~0x8; if (div == PXA2XX_SSP_CLK_SCDB_1) - SSACD_P(port) |= SSACD_SCDB; + val |= SSACD_SCDB; + __raw_writel(val, mmio_base + SSACD); break; case PXA2XX_SSP_DIV_SCR: - SSCR0_P(port) &= ~SSCR0_SCR; - SSCR0_P(port) |= SSCR0_SerClkDiv(div); + ssp_set_scr(&ssp[cpu_dai->id], div); break; default: return -ENODEV; @@ -310,28 +316,30 @@ static int pxa2xx_ssp_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, static int pxa2xx_ssp_set_dai_pll(struct snd_soc_cpu_dai *cpu_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { - int port = cpu_dai->id + 1; + void __iomem *mmio_base = ssp[cpu_dai->id].ssp->mmio_base; + u32 ssacd; - SSACD_P(port) &= ~0x70; + ssacd = __raw_readl(mmio_base + SSACD) & ~0x70; switch (freq_out) { case 5622000: break; case 11345000: - SSACD_P(port) |= (0x1 << 4); + ssacd |= (0x1 << 4); break; case 12235000: - SSACD_P(port) |= (0x2 << 4); + ssacd |= (0x2 << 4); break; case 14857000: - SSACD_P(port) |= (0x3 << 4); + ssacd |= (0x3 << 4); break; case 32842000: - SSACD_P(port) |= (0x4 << 4); + ssacd |= (0x4 << 4); break; case 48000000: - SSACD_P(port) |= (0x5 << 4); + ssacd |= (0x5 << 4); break; } + __raw_writel(ssacd, mmio_base + SSACD); return 0; } @@ -341,16 +349,18 @@ static int pxa2xx_ssp_set_dai_pll(struct snd_soc_cpu_dai *cpu_dai, static int pxa2xx_ssp_set_dai_tdm_slot(struct snd_soc_cpu_dai *cpu_dai, unsigned int mask, int slots) { - int port = cpu_dai->id + 1; + void __iomem *mmio_base = ssp[cpu_dai->id].ssp->mmio_base; + u32 sscr0; - SSCR0_P(port) &= ~SSCR0_SlotsPerFrm(7); + sscr0 = __raw_readl(mmio_base + SSCR0) & ~SSCR0_SlotsPerFrm(7); /* set number of active slots */ - SSCR0_P(port) |= SSCR0_SlotsPerFrm(slots); + sscr0 |= SSCR0_SlotsPerFrm(slots); + __raw_writel(sscr0, mmio_base + SSCR0); /* set active slot mask */ - SSTSA_P(port) = mask; - SSRSA_P(port) = mask; + __raw_writel(mask, mmio_base + SSTSA); + __raw_writel(mask, mmio_base + SSRSA); return 0; } @@ -360,12 +370,15 @@ static int pxa2xx_ssp_set_dai_tdm_slot(struct snd_soc_cpu_dai *cpu_dai, static int pxa2xx_ssp_set_dai_tristate(struct snd_soc_cpu_dai *cpu_dai, int tristate) { - int port = cpu_dai->id + 1; + void __iomem *mmio_base = ssp[cpu_dai->id].ssp->mmio_base; + u32 sscr1; + sscr1 = __raw_readl(mmio_base + SSCR1); if (tristate) - SSCR1_P(port) &= ~SSCR1_TTE; + sscr1 &= ~SSCR1_TTE; else - SSCR1_P(port) |= SSCR1_TTE; + sscr1 |= SSCR1_TTE; + __raw_writel(sscr1, mmio_base + SSCR1); return 0; } @@ -378,22 +391,20 @@ static int pxa2xx_ssp_set_dai_tristate(struct snd_soc_cpu_dai *cpu_dai, static int pxa2xx_ssp_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int fmt) { - int port = cpu_dai->id + 1; - - /* we can only change the settings if the port is not in use */ - if (SSCR0_P(port) & SSCR0_SSE) - return 0; + void __iomem *mmio_base = ssp[cpu_dai->id].ssp->mmio_base; + int val; /* reset port settings */ - SSCR0_P(port) = 0; - SSCR1_P(port) = 0; - SSPSP_P(port) = 0; + __raw_writel(0, mmio_base + SSCR0); + __raw_writel(0, mmio_base + SSCR1); + __raw_writel(0, mmio_base + SSPSP); /* NOTE: I2S emulation is still very much work in progress here */ /* FIXME: this is what wince uses for msb */ if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_MSB) { - SSCR0_P(port) = SSCR0_EDSS | SSCR0_TISSP | SSCR0_DataSize(16); + __raw_writel(SSCR0_EDSS | SSCR0_TISSP | SSCR0_DataSize(16), + mmio_base + SSCR0); goto master; } @@ -402,42 +413,50 @@ static int pxa2xx_ssp_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, /* 8.4.11 */ /* Only SSCR0[NCS] or SSCR0[ECS] bit fields settings are optional */ - SSCR0_P(port) = SSCR0_EDSS | SSCR0_PSP | SSCR0_DataSize(16); + __raw_writel(SSCR0_EDSS | SSCR0_PSP | SSCR0_DataSize(16), + mmio_base + SSCR0); /* set FIFO thresholds */ - SSCR1_P(port) = SSCR1_RxTresh(14) | SSCR1_TxTresh(1); + __raw_writel(SSCR1_RxTresh(14) | SSCR1_TxTresh(1), + mmio_base + SSCR1); /* normal: */ /* all bit fields must be cleared except: FSRT = 1 and * SFRMWDTH = 16, DMYSTART=0,1) */ - SSPSP_P(port) = SSPSP_FSRT | SSPSP_SFRMWDTH(16) | SSPSP_DMYSTRT(0); + __raw_writel(SSPSP_FSRT | SSPSP_SFRMWDTH(16) | SSPSP_DMYSTRT(0), + mmio_base + SSPSP); goto master; } - SSCR0_P(port) |= SSCR0_PSP; - SSCR1_P(port) = SSCR1_RxTresh(14) | SSCR1_TxTresh(1) | - SSCR1_TRAIL | SSCR1_RWOT; + val = __raw_readl(mmio_base + SSCR0) | SSCR0_PSP; + __raw_writel(val, mmio_base + SSCR0); + __raw_writel(SSCR1_RxTresh(14) | SSCR1_TxTresh(1) | + SSCR1_TRAIL | SSCR1_RWOT, + mmio_base + SSCR1); master: - switch(fmt & SND_SOC_DAIFMT_MASTER_MASK) { + val = __raw_readl(mmio_base + SSCR1); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - SSCR1_P(port) |= (SSCR1_SCLKDIR | SSCR1_SFRMDIR); + val |= (SSCR1_SCLKDIR | SSCR1_SFRMDIR); break; case SND_SOC_DAIFMT_CBM_CFS: - SSCR1_P(port) |= SSCR1_SCLKDIR; + val |= SSCR1_SCLKDIR; break; case SND_SOC_DAIFMT_CBS_CFM: - SSCR1_P(port) |= SSCR1_SFRMDIR; + val |= SSCR1_SFRMDIR; break; case SND_SOC_DAIFMT_CBS_CFS: break; default: return -EINVAL; } + __raw_writel(val, mmio_base + SSCR1); switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: - SSPSP_P(port) |= SSPSP_SFRMP | SSPSP_FSRT; + val = __raw_readl(mmio_base + SSPSP) | SSPSP_SFRMP | SSPSP_FSRT; + __raw_writel(val, mmio_base + SSPSP); break; case SND_SOC_DAIFMT_IB_IF: break; @@ -445,11 +464,12 @@ master: return -EINVAL; } + val = __raw_readl(mmio_base + SSPSP); switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - SSPSP_P(port) |= SSPSP_DMYSTRT(1); + val |= SSPSP_DMYSTRT(1); case SND_SOC_DAIFMT_DSP_B: - SSPSP_P(port) |= SSPSP_SCMODE(2); + val |= SSPSP_SCMODE(2); break; case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_MSB: @@ -458,6 +478,7 @@ master: default: return -EINVAL; } + __raw_writel(val, mmio_base + SSPSP); return 0; } @@ -472,7 +493,8 @@ static int pxa2xx_ssp_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; int dma = 0, chn = params_channels(params); - int port = cpu_dai->id + 1; + void __iomem *mmio_base = ssp[cpu_dai->id].ssp->mmio_base; + u32 sscr0; /* select correct DMA params */ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) @@ -481,34 +503,37 @@ static int pxa2xx_ssp_hw_params(struct snd_pcm_substream *substream, dma += 2; /* stereo DMA offset is 2, mono is 0 */ cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma]; - dbg("pxa2xx_ssp_hw_params: dma %d", dma); + pr_debug("pxa2xx_ssp_hw_params: dma %d", dma); /* we can only change the settings if the port is not in use */ - if (SSCR0_P(port) & SSCR0_SSE) + if (__raw_readl(mmio_base + SSCR0) & SSCR0_SSE) return 0; /* clear selected SSP bits */ - SSCR0_P(port) &= ~(SSCR0_DSS | SSCR0_EDSS); + sscr0 = __raw_readl(mmio_base + SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); + __raw_writel(sscr0, mmio_base + SSCR0); /* bit size */ - switch(params_format(params)) { + sscr0 = __raw_readl(mmio_base + SSCR0); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - SSCR0_P(port) |= SSCR0_DataSize(16); + sscr0 |= SSCR0_DataSize(16); break; case SNDRV_PCM_FORMAT_S24_LE: - SSCR0_P(port) |=(SSCR0_EDSS | SSCR0_DataSize(8)); + sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); /* we must be in network mode (2 slots) for 24 bit stereo */ break; case SNDRV_PCM_FORMAT_S32_LE: - SSCR0_P(port) |= (SSCR0_EDSS | SSCR0_DataSize(16)); + sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); /* we must be in network mode (2 slots) for 32 bit stereo */ break; } + __raw_writel(sscr0, mmio_base + SSCR0); - dbg("SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x", - SSCR0_P(port), SSCR1_P(port), - SSTO_P(port), SSPSP_P(port), - SSSR_P(port), SSACD_P(port)); + pr_debug("SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x", + __raw_readl(mmio_base + SSCR0), __raw_readl(mmio_base + SSCR1), + __raw_readl(mmio_base + SSTO), __raw_readl(mmio_base + SSPSP), + __raw_readl(mmio_base + SSSR), __raw_readl(mmio_base + SSACD)); return 0; } @@ -518,50 +543,61 @@ static int pxa2xx_ssp_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; int ret = 0; - int port = cpu_dai->id + 1; + void __iomem *mmio_base = ssp[cpu_dai->id].ssp->mmio_base; + int val; switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ssp_enable(&ssp[cpu_dai->id]); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = __raw_readl(mmio_base + SSCR1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - SSCR1_P(port) |= SSCR1_TSRE; + val |= SSCR1_TSRE; else - SSCR1_P(port) |= SSCR1_RSRE; - SSSR_P(port) |= SSSR_P(port); + val |= SSCR1_RSRE; + __raw_writel(val, mmio_base + SSCR1); + /* SSSR_P(port) |= SSSR_P(port); */ + val = __raw_readl(mmio_base + SSSR); + __raw_writel(val, mmio_base + SSSR); break; case SNDRV_PCM_TRIGGER_START: + val = __raw_readl(mmio_base + SSCR1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - SSCR1_P(port) |= SSCR1_TSRE; + val |= SSCR1_TSRE; else - SSCR1_P(port) |= SSCR1_RSRE; + val |= SSCR1_RSRE; + __raw_writel(val, mmio_base + SSCR1); ssp_enable(&ssp[cpu_dai->id]); break; case SNDRV_PCM_TRIGGER_STOP: + val = __raw_readl(mmio_base + SSCR1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - SSCR1_P(port) &= ~SSCR1_TSRE; + val &= ~SSCR1_TSRE; else - SSCR1_P(port) &= ~SSCR1_RSRE; + val &= ~SSCR1_RSRE; + __raw_writel(val, mmio_base + SSCR1); break; case SNDRV_PCM_TRIGGER_SUSPEND: ssp_disable(&ssp[cpu_dai->id]); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + val = __raw_readl(mmio_base + SSCR1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - SSCR1_P(port) &= ~SSCR1_TSRE; + val &= ~SSCR1_TSRE; else - SSCR1_P(port) &= ~SSCR1_RSRE; + val &= ~SSCR1_RSRE; + __raw_writel(val, mmio_base + SSCR1); break; default: ret = -EINVAL; } - dbg("SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x SSPSP 0x%08x SSSR 0x%08x", - SSCR0_P(port), SSCR1_P(port), - SSTO_P(port), SSPSP_P(port), - SSSR_P(port)); + pr_debug("SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x SSPSP 0x%08x SSSR 0x%08x", + __raw_readl(mmio_base + SSCR0), __raw_readl(mmio_base + SSCR1), + __raw_readl(mmio_base + SSTO), __raw_readl(mmio_base + SSPSP), + __raw_readl(mmio_base + SSSR)); return ret; } -- 1.5.5.1

2 1

[alsa-devel] No sound on nVidia MCP61 - snd-hda-intel
by Ash Willis 10 May '08

10 May '08

Hi everyone, I have a motherboard with an integrated MCP61. The device is detected and I have mixer controls but I get no sound output whatsoever. I'm using alsa-hg and I have a little experience with writing drivers for ALSA but I won't pretend to know anything about the hda driver. If someone could give me a little direction, I'd be happy to help get this problem sorted. I don't mind hacking on the driver. I just figured it was best to ask for advice first, seeing as it might be an easy problem for someone else to diagnose. alsa-info.sh output can be found here: http://pastebin.ca/1013173 Thanks, Ash -- See Exclusive Video: 10th Annual Young Hollywood Awards http://www.hollywoodlife.net/younghollywoodawards2008/

1 0

Re: [alsa-devel] Request for Realtek patch - Nvidia MCP51 HDA / ALC883 (10de:026c) Notebook component
by Björn van der Meer 09 May '08

09 May '08

> At Fri, 9 May 2008 15:35:50 +0200, > > Björn van der Meer wrote: > > > At Mon, 5 May 2008 08:43:16 +0200, > > > > > > Björn van der Meer wrote: > > > > Hello Everyone, > > > > > > > > I have above component in my laptop and lack the expertise to add it > > > > to to patch_realtek.c myself. > > > > > > > > I think I have collected all the relevant information if one of the > > > > developers wants to give it a try. > > > > > > > > My Notebook: Medion 98300 (a.k.a. WAM2030) > > > > (Please note: Medion is not an OEM. The board is nVidia based, OEM > > > > thought to be Wistron. Apparently rarely used sound component?) > > > > > > > > Internal speakers > > > > Internal mic - unknown > > > > three jacks: headphone, mic, SPDIF(dual use jack). > > > > Jack Sensing under Windows. > > > > 5.1 surround (sounds like the setup should be supported by > > > > 3stack-6ch-dig) > > > > > > > > lspci: > > > > 00:10.1 Audio device [0403]: nVidia Corporation MCP51 High Definition > > > > Audio [10de:026c] (rev a2) > > > > Subsystem: Wistron Corp. Unknown device [17c0:4079] > > > > Flags: 66MHz, fast devsel, IRQ 20 > > > > Memory at c0000000 (32-bit, non-prefetchable) [size=16K] > > > > Capabilities: [44] Power Management version 2 > > > > Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ > > > > Queue=0/0 Enable- > > > > Capabilities: [6c] HyperTransport: MSI Mapping > > > > > > > > hwinfo: > > > > 22: PCI 10.1: 0403 Audio device > > > > [Created at pci.296] > > > > UDI: /org/freedesktop/Hal/devices/pci_10de_26c > > > > Unique ID: wRyD.X7XJ3rTjrR3 > > > > SysFS ID: /devices/pci0000:00/0000:00:10.1 > > > > SysFS BusID: 0000:00:10.1 > > > > Hardware Class: sound > > > > Model: "Wistron MCP51 High Definition Audio" > > > > Vendor: pci 0x10de "nVidia Corporation" > > > > Device: pci 0x026c "MCP51 High Definition Audio" > > > > SubVendor: pci 0x17c0 "Wistron Corp." > > > > SubDevice: pci 0x4079 > > > > Revision: 0xa2 > > > > Driver: "HDA Intel" > > > > Driver Modules: "snd_hda_intel" > > > > Memory Range: 0xc0000000-0xc0003fff (rw,non-prefetchable) > > > > IRQ: 20 (195693 events) > > > > Module Alias: > > > > "pci:v000010DEd0000026Csv000017C0sd00004079bc04sc03i00" Driver Info > > > > #0: > > > > Driver Status: snd_hda_intel is active > > > > Driver Activation Cmd: "modprobe snd_hda_intel" > > > > Config Status: cfg=new, avail=yes, need=no, active=unknown > > > > > > > > alsa-info.sh output is at http://pastebin.com/f49962c92. > > > > > > > > > > > > > > > > Currently the only thing working is output via the internal speakers. > > > > > > And doesn't this still work with the latest ALSA HG driver? > > > > > > > > > Takashi > > > > Thanks for your answer. Latest HG does not compile on my system, throws > > memalloc.c errors. (Could be me doing something wrong though - out on a > > limp) > > > > CC [M] /home/bvdm/alsa-driver/acore/memalloc.o > > /home/bvdm/alsa-driver/acore/memalloc.c: In function ‘snd_mem_init’: > > /home/bvdm/alsa-driver/acore/memalloc.c:729: error: implicit declaration > > of function ‘proc_create’ > > /home/bvdm/alsa-driver/acore/memalloc.c:730: warning: assignment makes > > pointer from integer without a cast > > make[3]: *** [/home/bvdm/alsa-driver/acore/memalloc.o] Error 1 > > make[2]: *** [/home/bvdm/alsa-driver/acore] Error 2 > > make[1]: *** [_module_/home/bvdm/alsa-driver] Error 2 > > make[1]: Leaving directory `/usr/src/linux-headers-2.6.24-16-generic' > > > > > > > > I see references to similar errors on the list, had to do with processor > > identification? > > > > > > Configure detects this: > > checking for processor type... i586 > > checking for i386 machine type... default > > > > /proc/cpuinfo starts with: > > [...] > > vendor_id : AuthenticAMD > > cpu family : 15 > > model : 72 > > model name : AMD Turion(tm) 64 X2 Mobile Technology TL-50 > > [...] > > It's a problem in alsa-driver build stub. I fixed it right now on HG > tree. > > > I have also just lifted patch_realtek.c from the current hg and compiled > > it in a working tarball (actually labelled alsa-driver-hg20080502.tar > > from the realtek site), which compiles, looks a bit better. > > Gives me: > > - Different set of mixer controls (not 5.1 though) > > OK, I'll need a newer alsa-info.sh with the latest driver. New alsa-info.sh is at http://pastebin.ca/1012744 > > > - headphone output in the strange dual-use SPDIF jack only (headphone > > jack only goes 2/3 of the way in) > > Please elaborate? I don't understand your description. hehe, yes. I didn't believe it either. This laptop has three jacks. - light blue, symbol "headphone", normal 3.5mm jack - pink, symbol "microphone", normal 3.5mm jack - black, symbol line out, normal 3.5mm (that only goes in 2/3 of the way) jack AND SPDIF (yes, now glowing red) Actually. Seeing is believing. www.chronovault.net/static/medion98300jacks.jpg > > - but with jack-sensing working in that jack, with the boxes turning off > > when I insert the headphone into the SPDIF. > This sounds good -- do you mean that you have another headphone jacks? > > > Takashi Thank you very much Just hg'ed again compile runs fine. This is without any "options snd-hda-intel model=XXXX" line in /etc/modprobe.d/alsa-base - I now put my headphone into the black jack and get sound, boxes turn off. - light blue jack is without function now. - mixer option "headphone" is without function. - mixer now has master, headphone,pcm,front,frontmic,line,mic,micboost,digital - only master,pcm and front work, all regulate master volume on SPDIF-used-as-headphone-jack and the internal boxes - dont have a mic right here to try out Should I try options there now? Is /etc/init.d/alsasound stop / start a good way to cycle through options? -- Björn van der Meer bvdm(a)chronovault.net

2 1

[alsa-devel] My issues with the ALSA API
by Lennart Poettering 09 May '08

09 May '08

Heya! A while back I compiled a list of issues I found with the ALSA API while I was hacking PulseAudio. I kept it up to date ever since. It has the title "Why I hate ALSA", but this is merely a catchy title, I am not trying to offend anyone, nor do I think that the ALSA API is particularly bad. Back at FOMS in January this year, I showed this list to James. He acknowledged most of those issues. But I am sure that the other ALSA devs might be interested too: http://pulseaudio.org/wiki/WhyIHateALSA Comments welcome! Thank you, Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

4 9

[alsa-devel] Request for Realtek patch - Nvidia MCP51 HDA / ALC883 (10de:026c) Notebook component
by Björn van der Meer 09 May '08

09 May '08

Hello Everyone, I have above component in my laptop and lack the expertise to add it to to patch_realtek.c myself. I think I have collected all the relevant information if one of the developers wants to give it a try. My Notebook: Medion 98300 (a.k.a. WAM2030) (Please note: Medion is not an OEM. The board is nVidia based, OEM thought to be Wistron. Apparently rarely used sound component?) Internal speakers Internal mic - unknown three jacks: headphone, mic, SPDIF(dual use jack). Jack Sensing under Windows. 5.1 surround (sounds like the setup should be supported by 3stack-6ch-dig) lspci: 00:10.1 Audio device [0403]: nVidia Corporation MCP51 High Definition Audio [10de:026c] (rev a2) Subsystem: Wistron Corp. Unknown device [17c0:4079] Flags: 66MHz, fast devsel, IRQ 20 Memory at c0000000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Queue=0/0 Enable- Capabilities: [6c] HyperTransport: MSI Mapping hwinfo: 22: PCI 10.1: 0403 Audio device [Created at pci.296] UDI: /org/freedesktop/Hal/devices/pci_10de_26c Unique ID: wRyD.X7XJ3rTjrR3 SysFS ID: /devices/pci0000:00/0000:00:10.1 SysFS BusID: 0000:00:10.1 Hardware Class: sound Model: "Wistron MCP51 High Definition Audio" Vendor: pci 0x10de "nVidia Corporation" Device: pci 0x026c "MCP51 High Definition Audio" SubVendor: pci 0x17c0 "Wistron Corp." SubDevice: pci 0x4079 Revision: 0xa2 Driver: "HDA Intel" Driver Modules: "snd_hda_intel" Memory Range: 0xc0000000-0xc0003fff (rw,non-prefetchable) IRQ: 20 (195693 events) Module Alias: "pci:v000010DEd0000026Csv000017C0sd00004079bc04sc03i00" Driver Info #0: Driver Status: snd_hda_intel is active Driver Activation Cmd: "modprobe snd_hda_intel" Config Status: cfg=new, avail=yes, need=no, active=unknown alsa-info.sh output is at http://pastebin.com/f49962c92. Currently the only thing working is output via the internal speakers. Thank you very much, B. -- Björn van der Meer bvdm(a)chronovault.net

2 3

Re: [alsa-devel] Problem with aloop
by Benjamin van den Hout 09 May '08

09 May '08

Gustavo da Silva Serra wrote: > I am developing an application that captures sound and send it > through a network, just like a sound server. To test it I am using > kaffeine with alsa output plugin. After some time playing continuously, > or if I keep moving the position slider backward and forward, I get > choppy sound. If I close and reopen the capture stream opened by my > sound server, the sound is restored to good quality. Sometimes I have to > reopen many times before I get a good sound. When I use kaffeine oss > output plugin I don't have this issue. I don't have an audio sound card, > so I don't know if this is an *aloop* issue. > Here are some details: >- RedHat EL 4 kernel 2.6.9.55-ELsmp 64bit >- Alsa 1.0.16 >- Xinelib 1.1.12 Hi all, I'm experiencing the same problem as Gustavo. In my case I have a SIP client calling to another PC and I'm injecting audio with aplay (or amarok, no difference there) into that SIP call. About 8/10 times everything goes well, the other two times the audio that comes out at the other PC sounds really choppy. Both PC's are on the same local network so that is probably not the problem. What I found out is that if I patch aloop-kernel.c in the 'static struct snd_pcm_hardware snd_card_loopback_info' to have ".formats = (SNDRV_PCM_FMTBIT_S8 )," this problem goes away. But then I'm stuck with 8 bit audio. It sounds a bit like the aloop device sometimes doesn't properly synchronise the 16 bit (or 32bit, tried that as well) samples on both PCM devices and things get mis-aligned. I'm quite new to ALSA so please don't laugh too hard if I'm missing something here :) Any input is greatly appreciated! Kind regards, Benjamin van den Hout

2 1

[alsa-devel] [PATCH v2.6.26-rc1] ASoC: Fix TLV320AIC3X mono line output interconnect
by Mark Brown 09 May '08

09 May '08

From: Jarkko Nikula <jarkko.nikula(a)nokia.com> There is no endpoint called MONOLOUT but MONO_LOUT. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic3x.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6e4bc69..957996e 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -553,8 +553,8 @@ static const char *intercon[][3] = { {"HPRCOM", NULL, "Right HP Com"}, /* Mono Output */ - {"MONOLOUT", NULL, "Mono Out"}, - {"MONOLOUT", NULL, "Mono Out"}, + {"MONO_LOUT", NULL, "Mono Out"}, + {"MONO_LOUT", NULL, "Mono Out"}, /* Left Input */ {"Left Line1L Mux", "single-ended", "LINE1L"}, -- 1.5.5.1

2 2

[alsa-devel] [PATCH 2.6.26-rc1] ASoC: Fix TLV320AIC3X mono line output interconnect
by Jarkko Nikula 09 May '08

09 May '08

There is no endpoint called MONOLOUT but MONO_LOUT. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/codecs/tlv320aic3x.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 630684f..09b1661 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -539,8 +539,8 @@ static const char *intercon[][3] = { {"HPRCOM", NULL, "Right HP Com"}, /* Mono Output */ - {"MONOLOUT", NULL, "Mono Out"}, - {"MONOLOUT", NULL, "Mono Out"}, + {"MONO_LOUT", NULL, "Mono Out"}, + {"MONO_LOUT", NULL, "Mono Out"}, /* Left Input */ {"Left Line1L Mux", "single-ended", "LINE1L"}, -- 1.5.4.4

2 1

[alsa-devel] Patch - ESI Maya44 driver
by Rainer Zimmermann 09 May '08

09 May '08

Hi Takashi, here's the maya44 patch. This is a patch against the current alsa-kernel tree. I did add 2 definitions to ice1712.h (snd_ice1724_get_route_val and snd_ice1724_put_route_val), which allowed me to move some of the maya44 specific code to maya44.c . There are still a few maya44 special cases in ice1724.c, because maya44 uses the "con" channel and routing switches differently. This is not really clean style I think, but I would suggest sorting this out later, as it would again require larger changes to ice1724.c and testing with other hardware. Hope everything is ok with you, for now. This patch does *not* limit the sampling rate to 96kHz for capture, as discussed before. best regards, -Rainer -- Lightshed IT Services Löfflerstr. 27, 22765 Hamburg, Germany * mail(a)lightshed.de * www.lightshed.de

3 6

[alsa-devel] [PATCH] soc - fsl_ssi.c fix "BUG: scheduling while atomic"
by Mark Brown 09 May '08

09 May '08

From: Anton Vorontsov <avorontsov(a)ru.mvista.com> This patch fixes following bug caught with PREEMPT enabled: root@b1:~# cat /dev/dsp > /dev/null BUG: scheduling while atomic: cat/965/0x00000003 Call Trace: [df165ce0] [c0008e84] show_stack+0x4c/0x1ac (unreliable) [df165d20] [c001c18c] __schedule_bug+0x64/0x78 [df165d30] [c02b3344] schedule+0x2d8/0x334 [df165d70] [c02b3674] schedule_timeout+0x64/0xe4 [df165db0] [c002c05c] msleep+0x1c/0x34 [df165dc0] [c01f2fe0] fsl_ssi_trigger+0x130/0x144 [df165dd0] [c01ece54] soc_pcm_trigger+0x94/0xb8 [df165df0] [c01da764] snd_pcm_do_start+0x48/0x60 [df165e00] [c01da630] snd_pcm_action_single+0x4c/0xb4 [df165e20] [c01e0f50] snd_pcm_lib_read1+0x2a0/0x2d4 [df165e70] [c01ec274] snd_pcm_oss_read3+0xf0/0x13c [df165eb0] [c01ec2e4] snd_pcm_oss_read2+0x24/0x4c [df165ec0] [c01ec4ac] snd_pcm_oss_read+0x1a0/0x1f0 [df165ef0] [c0076478] vfs_read+0xb4/0x108 [df165f10] [c00768cc] sys_read+0x4c/0x90 [df165f40] [c00117a4] ret_from_syscall+0x0/0x38 Acked-by: Timur Tabi <timur(a)freescale.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b2a11b0..f588545 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -416,7 +416,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) * to put data into its FIFO. Without it, ALSA starts * to complain about overruns. */ - msleep(1); + mdelay(1); } break; -- 1.5.5.1

2 1

[alsa-devel] [PATCH] emux midi synthesizer doesn't honor SOFT_PEDAL-release event
by maximilian attems 09 May '08

09 May '08

When the hardware wavetable synthesizer of an Creative SB Audigy or SB Live! card (with emu10k chip) receives the MIDI SOFT_PEADAL-press event (?? 67 127) the appropriate voice is attenuted. Unfortunately when the pedal is released (event ?? 67 0) the voice does not get it's original volume again. Boolean MIDI controls should interpret 0..63 as false and 64..127 as true. Thanks to Clemens Ladisch for review and correction. Original patch from "Uwe KrÃ#ger" <uwe_debbug(a)arcor.de> Submitted to http://bugs.debian.org/474312 Signed-off-by: maximilian attems <max(a)stro.at> Cc: uwe_debbug(a)arcor.de Cc: Clemens Ladisch <clemens(a)ladisch.de> --- sound/synth/emux/emux_synth.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/synth/emux/emux_synth.c b/sound/synth/emux/emux_synth.c index 478369b..b343818 100644 --- a/sound/synth/emux/emux_synth.c +++ b/sound/synth/emux/emux_synth.c @@ -341,8 +341,12 @@ snd_emux_control(void *p, int type, struct snd_midi_channel *chan) case MIDI_CTL_SOFT_PEDAL: #ifdef SNDRV_EMUX_USE_RAW_EFFECT /* FIXME: this is an emulation */ - snd_emux_send_effect(port, chan, EMUX_FX_CUTOFF, -160, + if (chan->control[type] >= 64) + snd_emux_send_effect(port, chan, EMUX_FX_CUTOFF, -160, EMUX_FX_FLAG_ADD); + else + snd_emux_send_effect(port, chan, EMUX_FX_CUTOFF, 0, + EMUX_FX_FLAG_OFF); #endif break; -- 1.5.5.1

2 1

[alsa-devel] [PATCH] emux midi synthesizer doesn't honor SOFT_PEDAL-release event
by maximilian attems 09 May '08

09 May '08

When the hardware wavetable synthesizer of an Creative SB Audigy or SB Live! card (with emu10k chip) receives the MIDI SOFT_PEADAL-press event (?? 67 127) the appropriate voice is attenuted. Unfortunately when the pedal is released (event ?? 67 0) the voice does not get it's original volume again. The attached patch fixes this problem by analysing the value (0 or 127) of the midi control event and resetting the effect register in case of a release event. I'm not 100% sure if the code to reset the register is correct but at least it works. Original patch from "Uwe KrÃ#ger" <uwe_debbug(a)arcor.de> Submitted to http://bugs.debian.org/474312 Cc: uwe_debbug(a)arcor.de Signed-off-by: maximilian attems <max(a)stro.at> --- sound/synth/emux/emux_synth.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/synth/emux/emux_synth.c b/sound/synth/emux/emux_synth.c index 478369b..f81e081 100644 --- a/sound/synth/emux/emux_synth.c +++ b/sound/synth/emux/emux_synth.c @@ -341,8 +341,12 @@ snd_emux_control(void *p, int type, struct snd_midi_channel *chan) case MIDI_CTL_SOFT_PEDAL: #ifdef SNDRV_EMUX_USE_RAW_EFFECT /* FIXME: this is an emulation */ - snd_emux_send_effect(port, chan, EMUX_FX_CUTOFF, -160, + if (chan->control[type]) + snd_emux_send_effect(port, chan, EMUX_FX_CUTOFF, -160, EMUX_FX_FLAG_ADD); + else + snd_emux_send_effect(port, chan, EMUX_FX_CUTOFF, 0, + EMUX_FX_FLAG_OFF); #endif break; -- 1.5.5.1

2 1

[alsa-devel] [patch] snd-pcsp: relax dependancy on CONFIG_INPUT
by Stas Sergeev 09 May '08

09 May '08

Hello. The attached patch makes an input driver part of snd-pcsp optional. Would it be possible to apply that?

2 6

[alsa-devel] [PATCH] [ALSA] soc - fsl_ssi.c fix "BUG: scheduling while atomic"
by Anton Vorontsov 08 May '08

08 May '08

This patch fixes following bug caught with PREEMPT enabled: root@b1:~# cat /dev/dsp > /dev/null BUG: scheduling while atomic: cat/965/0x00000003 Call Trace: [df165ce0] [c0008e84] show_stack+0x4c/0x1ac (unreliable) [df165d20] [c001c18c] __schedule_bug+0x64/0x78 [df165d30] [c02b3344] schedule+0x2d8/0x334 [df165d70] [c02b3674] schedule_timeout+0x64/0xe4 [df165db0] [c002c05c] msleep+0x1c/0x34 [df165dc0] [c01f2fe0] fsl_ssi_trigger+0x130/0x144 [df165dd0] [c01ece54] soc_pcm_trigger+0x94/0xb8 [df165df0] [c01da764] snd_pcm_do_start+0x48/0x60 [df165e00] [c01da630] snd_pcm_action_single+0x4c/0xb4 [df165e20] [c01e0f50] snd_pcm_lib_read1+0x2a0/0x2d4 [df165e70] [c01ec274] snd_pcm_oss_read3+0xf0/0x13c [df165eb0] [c01ec2e4] snd_pcm_oss_read2+0x24/0x4c [df165ec0] [c01ec4ac] snd_pcm_oss_read+0x1a0/0x1f0 [df165ef0] [c0076478] vfs_read+0xb4/0x108 [df165f10] [c00768cc] sys_read+0x4c/0x90 [df165f40] [c00117a4] ret_from_syscall+0x0/0x38 --- Exception: c01 at 0xfe9a8d4 LR = 0x10002b24 We must not sleep with irqs disabled, so replace msleep with mdelay. Signed-off-by: Anton Vorontsov <avorontsov(a)ru.mvista.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b2a11b0..f588545 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -416,7 +416,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) * to put data into its FIFO. Without it, ALSA starts * to complain about overruns. */ - msleep(1); + mdelay(1); } break; -- 1.5.5.1

2 3

[alsa-devel] softvol plugin question
by Will Wagner 08 May '08

08 May '08

Hi, I have successfully set up alsa for my custom hardware and am using sofvol plugin to do volume control as my hardware does not support it. The problem I am having is that the softvol control does not appear to amixer until I have first used the softvol plugin with aplay. Is there a simple way of 'loading' the softvol plugin at startup so that the control gets created immediately? Thanks, Will. -- ------------------------------------------------------------------------ Will Wagner will_wagner(a)carallon.com Senior Project Engineer Office Tel: +44 (0)20 7371 2032 Carallon Ltd, Studio G20, Shepherds Building, Rockley Rd, London W14 0DA ------------------------------------------------------------------------

3 3

[alsa-devel] ALSA vs. non coherent DMA
by Benjamin Herrenschmidt 08 May '08

08 May '08

Hi Takashi ! I'm bringing up an old thread as I'm just discovering that the problem still hasn't been fixed. There seem to be a few issues with ALSA current usage of mmap vs. non cache coherent architecture, such as embedded PowerPC's. I can see at least two with a quick look to pcm-native.c, one I don't understand and one I think I do: - The control/status mapping. Can you elaborate a bit on what this is actually doing and why it shouldn't be done on "non coherent" architectures ? Currently this -is- done on all powerpc's, whether they are coherent or not and I want to understand what the underlying issue is. - The mmap of DMA pages. Here, the problem appears two fold: * Use of virt_to_page() on virtual addresses returned by dma_alloc_coherent(). * No using the appropriate page protection for a DMA coherent mapping to userspace. It seems like you have solved that in part with implementing a generic dma_mmap_coherent() in the past that for some reason you never merged upstream (I can track that to about 2 years ago). Is there a reason ? I think we need to at least apply a band-aid today as it's becoming a nasty issue for several non-coherent powerpc platforms. It could be in the form of implementing dma_mmap_coherent() and changing Alsa to use it with the appropriate ifdef, or just adding an ifdef CONFIG_PPC with the right code in there for now until a better solution is found. It should be trivial though. Getting the PFN from the DMA address is easy if we have the dma handle and the virtual address, though that -is- definitely platform specific. I can implement a function for that if you need. As for the pgprot, we can come up with something like pgprot_mmap_dma(). Either that or I can fold it all in a powerpc wide implementation of a dma_mmap_coherent() like we envisioned initially. Let me know what approach is preferred here and I'll come up with patches ASAP. As far as I'm concerned, this is a bug and thus must be fixed now for .26 and possibly backported to stable even if we can come up with a non invasive solution). I'm annoyed because it represents a trivial amount of code, this problem should have been fixed a long time ago. Cheers, Ben.

4 5

[alsa-devel] [PATCH] snd_usb_caiaq: add support for "Session I/O" interface
by Daniel Mack 08 May '08

08 May '08

Hi, this patch adds support for "Session I/O". As it certainly won't break anything it could still go for 2.6.26, right? Thanks, Daniel

2 2

[alsa-devel] [PATCH 1/2] soc - at91-pcm - Fix line wrapping
by Mark Brown 08 May '08

08 May '08

There's more checkpatch stuff to fix in the driver, this just fixes the minimum required for the following patch to be clean. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/at91/at91-pcm.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c index 67c88e3..e36533d 100644 --- a/sound/soc/at91/at91-pcm.c +++ b/sound/soc/at91/at91-pcm.c @@ -103,7 +103,8 @@ static void at91_pcm_dma_irq(u32 ssc_sr, if (prtd->period_ptr >= prtd->dma_buffer_end) { prtd->period_ptr = prtd->dma_buffer; } - at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr); + at91_ssc_write(params->ssc_base + params->pdc->xnpr, + prtd->period_ptr); at91_ssc_write(params->ssc_base + params->pdc->xncr, prtd->period_size / params->pdc_xfer_size); } @@ -191,10 +192,12 @@ static int at91_pcm_trigger(struct snd_pcm_substream *substream, at91_ssc_write(params->ssc_base + AT91_SSC_IER, params->mask->ssc_endx | params->mask->ssc_endbuf); - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); + at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, + params->mask->pdc_enable); - DBG("sr=%lx imr=%lx\n", at91_ssc_read(params->ssc_base + AT91_SSC_SR), - at91_ssc_read(params->ssc_base + AT91_SSC_IER)); + DBG("sr=%lx imr=%lx\n", + at91_ssc_read(params->ssc_base + AT91_SSC_SR), + at91_ssc_read(params->ssc_base + AT91_SSC_IER)); break; case SNDRV_PCM_TRIGGER_STOP: -- 1.5.5.1

2 2

[alsa-devel] SoC sound support for imx27
by Juergen Beisert 08 May '08

08 May '08

Hi, I'm going to adapt Freescale's i.MX27 CPU to the SoC sound support via I2S. I found the imx21 already in the repository (may it will help for my job). Is anybody else working on i.MX27 support (to avoid to reinvention everything)? Regards Juergen -- Dipl.-Ing. Juergen Beisert | http://www.pengutronix.de Pengutronix - Linux Solutions for Science and Industry Handelsregister: Amtsgericht Hildesheim, HRA 2686 Vertretung Sued/Muenchen, Germany Phone: +49-8766-939 228 | Fax: +49-5121-206917-9

5 16

[alsa-devel] [PATCH] soc at91 minor bug fixes
by Patrik Sevallius 08 May '08

08 May '08

Found these two bugs while browsing through the code. The first one is a cut-n-paste bug, instead of disabling the clock when request_irq() fails, it enabled it once more. The second one fixes a debug printout, AT91_SSC_IER is write only, AT91_SSC_IMR is readable (the printed string actually says imr). Frank Mandarino was busy so he asked me to send these to this list. /Patrik Signed-off-by: Patrik Sevallius <patrik.sevallius(a)enea.com> Signed-off-by: Frank Mandarino <fmandarino(a)endrelia.com> ================================================================================ --- sound/soc/at91/at91-ssc.c.orig 2008-05-07 09:33:49.000000000 +0200 +++ sound/soc/at91/at91-ssc.c 2008-05-07 09:34:18.000000000 +0200 @@ -590,7 +590,7 @@ printk(KERN_WARNING "at91-ssc: request_irq failure\n"); DBG("Stopping pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid); + at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid); return ret; } --- sound/soc/at91/at91-pcm.c.orig 2008-05-07 13:23:12.000000000 +0200 +++ sound/soc/at91/at91-pcm.c 2008-05-07 13:26:40.000000000 +0200 @@ -194,7 +194,7 @@ at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); DBG("sr=%lx imr=%lx\n", at91_ssc_read(params->ssc_base + AT91_SSC_SR), - at91_ssc_read(params->ssc_base + AT91_SSC_IER)); + at91_ssc_read(params->ssc_base + AT91_SSC_IMR)); break; case SNDRV_PCM_TRIGGER_STOP:

2 1

Re: [alsa-devel] [pulseaudio-discuss] alsa pulse bugs
by Lennart Poettering 06 May '08

06 May '08

On Mon, 05.05.08 09:58, Colin Guthrie (gmane(a)colin.guthr.ie) wrote: > > tom(a)dbservice.com wrote: > > I reported three bugs to the alsa bugtracker: > > > > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3942 > > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3943 > > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3944 > > > > It's three bugs after all. Once these are fixed I expect Wine to work. > > *fingers crossed* > > Ace Tom :) > > I've cross posted this to the alsa devel list. Hopefully Takashi (who's > committed some pulse-related stuff recently) will have some insight here. > > Re the snd_pcm_delay() including network latency (#3945), this clearly > makes sense for network streams. Does you proposed fix include this > delay (albeit with the improvement that it also will drop to 0 if there > are no samples queued)? > > Also re #3942, I believe (but not sure) that the max buffer in the > glitch free branch has been increased significantly (as it now needs to > keep some degree of past sound as well as future buffer to allow the > rewinds to work properly (this is no doubt an inaccurate description of > why it's needed.... :p)) I'm not sure how this would affect this > solution, but Lennart will be able to advise better. Yes, with glitch-free maxlength becomes mostly irrelevant. It should default to "0" know, to let the server decide. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] Auzentech and Creative X-FI
by Sebastian Schneider 06 May '08

06 May '08

Hi, I was wondering if the X-FI drivers will be working with the X-FI from Auzentech or just with the cards from Creative. Is there something known about this? And how far is the development? Is somewhere a cvs/svn/git system where to get the live files? Thanks in advance. -- Ist Ihr Browser Vista-kompatibel? Jetzt die neuesten Browser-Versionen downloaden: http://www.gmx.net/de/go/browser

2 1

[alsa-devel] [RFC] virtual master control for HD-audio
by Takashi Iwai 06 May '08

06 May '08

Hi, another thing I'd like to push into the next kernel is the virtual master volume control. As I posted in earlier posts, it adds virtual controls that have several slave controls with the same types, e.g. Front, Surround, Center, LFE, etc. Then these are adjusted simultaneously via the master control. It'd be appreciated if some one can test the patches with HD-audio h/w that has no master control yet (e.g. some STAC codecs). Note that this won't add the master volumes if there are no real volume controls. Some codecs have really no volume control, and this won't help for such devices. Two (and one for driver) patches will follow after this. Takashi

5 27