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[alsa-devel] are freebob firewire cards likely to be supported by alsa in the future?
by andy baxter 05 Sep '08

05 Sep '08

Hello, Just wondering if there are any plans to develop alsa support for firewire freebob cards like the M-Audio firewire solo (which I have). I have this working fine under jackd using the freebob driver, but as far as I know there isn't an alsa driver for it. I assume that since there is already open source code to drive the chip, it should be possible to write an alsa driver? This isn't an urgent thing for me - for the moment I'm quite happy to use my onboard card for playing music / movies etc, and keep the firewire card for recording music etc, but it might be useful sometimes to have an alsa driver for this card as well. andy

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[alsa-devel] [ANNOUNCE] Git User's Survey 2008
by Jakub Narebski 04 Sep '08

04 Sep '08

Hi all, We would like to ask you a few questions about your use of the Git version control system. This survey is mainly to understand who is using Git, how and why. The results will be published on the Git Wiki http://git.or.cz/gitwiki/GitSurvey2008 and discussed on the git mailing list. The survey would be open from 1 September till 10 October. Please devote a bit of your time to fill this simple questionnaire; it will help a lot the Git community to understand your needs, what you like of Git, and of course what you don't like of it. The survey can be found here: http://www.survs.com/survey?id=M3PIVU72&channel=2WXE4BVTW8 http://tinyurl.com/GitSurvey2008 -- Git Development Community

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Re: [alsa-devel] snd-soc-cs4270: Convert to a new-style i2c driver (work in progress)
by Jean Delvare 04 Sep '08

04 Sep '08

Hi Timur, On Sun, 31 Aug 2008 11:28:08 -0500, Timur Tabi wrote: > On Sun, Aug 31, 2008 at 9:18 AM, Jean Delvare <khali(a)linux-fr.org> wrote: > > Hi Tibur, > > It's "Timur" Oops, apparently I mixed your first name and last name. Sorry about that. > > I am in the process of converting your cs4270 codec driver from the > > legacy i2c model to the new (standard) one. > > It has already been converted. > > http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=commit;h=0d… Ah, excellent. One less thing for me to do :) Thanks! Two comments about your patch: * Your driver now lacks a remove method. Unless I miss something, if the snd-soc-cs4270 driver is unloaded, you will leave dangling resources behind (codec->reg_cache in particular.) * I2C_DRIVERID_CS4270 must be removed from i2c-id.h. > I don't know why this commit hasn't been pushed upstream, though. As I understand it, without that patch the mpc8610_hpcd doesn't work, as the I2C address of the sound codec will be made busy by the platform code and thus the snd-soc-cs4270 driver won't be able to attach to it. If I am correct then I suggest that you ask Takashi to push the patch to Linus now to fix that. > > This is work in progress. > > The patch below converts the cs4270 driver itself. However we also need > > to convert its users. As far as I can see there's only one user at this > > point: mpc8610_hpcd. > > That's correct. I've never seen any interest in the CS4270 outside of > Freescale. > > > The problem is that this driver doesn't look like the other codec > > drivers I have already converted. So, we need to add code to > > instantiate the cs4270 i2c device, but I don't know where this should > > happen. Given that the mpc8610_hpcd is apparently based on Open > > Firmware, I guess that the i2c device should be instantiated directly > > by the platform code. I see that the device is declared in > > mpc8610_hpcd.dts, so maybe it's already done and my patch should work > > already? What do you think? > > I think you need to use the right git repository for your development. :-) Very good point indeed. I'll make sure to check Takashi's sound-2.6 tree before attempting to convert any other SoC codec driver. Do you know off the top of your head if other drivers have already been converted? Thanks, -- Jean Delvare

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[alsa-devel] Pull request for alsa-plugins
by Lennart Poettering 04 Sep '08

04 Sep '08

Takashi, Jaroslav, please pull and merge a series of 26 patches for alsa-plugins' PulseAudio driver which I prepared in this repository of mine: git://git.0pointer.de/alsa-plugins.git gitweb is here: http://git.0pointer.de/?p=alsa-plugins.git There are quite a few bug fixes and feature enhancements among those patches. It also includes some clean-ups in the coding style. i.e. previously the indenting of the PA driver was pure chaos, every single patch seemed to introduce a new indentation width. The first patch in this series simply reindents the driver to Linux style. All other patches are on top of that one. Here's the shortlog: Lennart Poettering (26): Reindent to Linux kernel style Add Emacs-style /*-*- linux-c -*-*/ header comment Make pulse_new() a proper C function Don't modify the SIGPIPE handler Call pa_context_disconnect() explicitly use SNDERR instead of fprintf to print error messages Support S32 sample types Add trailing NUL character to snprintf output Get rid of pulse_poll_revents() Add more error checking Remove fix for bug 0003470 Rework hardware parameter selection A bag of minor clean ups for ctl_pulse.c Make pulse_ext_callback const Drop our own implementation of the poll() callbacks A bag of minor clean-ups for pulse.c Split out O_NONBLOCK setting into seperate function Save a byte of memory Adjust buffering metrics to match what PA internally uses Make sure we always have a sensible channel mapping Use PA_STREAM_EARLY_REQUESTS if available Use S32/FLOAT32 only where available in the PA libs Add const to our snd_pcm_ioplug_callback_t instances Don't implement our own poll handlers, we can use the default ones Remove our own poll handler implementation entirely A bag of clean-ups for pcm_ctl.c Those patches only touch the pulse/ subdir. The tree is freshly rebased against current alsa-plugins master. May I ask you to make me the "semi-official" maintainer of this driver? I.e. I'd like to be consulted (as in 'Signed-Off-by') before any patches for it are merged? If requested I can also mail this as series of 26 seperate patches. Thank you, Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] DaVinci ASoC DMA stalls
by Mark Lokowich 04 Sep '08

04 Sep '08

I've been using the DaVinci ASoC for a few months and have recently upgraded to the 2.6.26 based DaVinci git kernel. Now my audio DMA stalls more readily after stopping an active stream. I can manually trigger the event by poking the ESR to reactivate the stalled stream, suggesting the problem is in the ASP-to-DMA XEVT interface. This problem is less prevalent in the 2.6.25 based kernel. Any help? Regards, Mark Lokowich Advanced Communication Design

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[alsa-devel] how to convert stereo capture to mono stream (hardware incompatibility)
by brunal 04 Sep '08

04 Sep '08

Hi all, I've already asked my question in the alsa-user list, but as nobody was able to answer, I'm trying here if I have more chance. I'm using a USB M-Audio fasttrack pro sound card under linux 2.6.24.rt, and I want to plug it into a voice recognition software named Julius. There is an implementation of alsa in Julius, but I can't make it working with my USB sound card, because when alsa tries to set channel capture to 1, I get an error from alsa. Here is the code : /* set number of channels */ if ((err = snd_pcm_hw_params_set_channels(handle, hwparams, 1)) < 0) { jlog("Error: adin_alsa: cannot set PCM channel to %d (%s)\n", 1, snd_strerror(err)); return(FALSE); } Which always return FALSE. The thing is that when I change this line to set stereo channels, Julius recognition engine complains when it runs, and strip lots of audio packets, which seems to broke recognition. So my question is : how can I by-pass this problem? I've tried to do it with the .asoundrc file, but it seems that plugin or dmix stuff only works with playback, not recording. Does someone have an idea how to make a recording plugin that accept stereo stream from hardware and output only the left channel? I didn't understand how stereo stream are written in the audio buffer gave by alsa. Can someone explain me how to separate the two stream once it is written in the buffer? I really thanks all who would take little attention to my questions, it's been 5 days I'm digging on the web but I only get code working for setreo applications, which doesn't help me so much. And I know I might have an answer if I look into the audacity code, but it seems to me it's like using a bazooka for fishing redfish.... Best Regards, Bruno.

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[alsa-devel] [PATCH 1/1] ASoC codec: SSM2602 audio codec driver (v2)
by Bryan Wu 04 Sep '08

04 Sep '08

From: Cliff Cai <cliff.cai(a)analog.com> v1-v2: - coding style fixing - use pr_xxx macros to replace printk(KERN_XXX...) - use new-style i2c interface - update to use latest ASoC API Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2602.c | 773 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ssm2602.h | 131 ++++++++ 4 files changed, 909 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ssm2602.c create mode 100644 sound/soc/codecs/ssm2602.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5d77dc3..8ef5ce6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -92,3 +92,6 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC3X tristate depends on I2C + +config SND_SOC_SSM2602 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 35daaa9..0cd55ee 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-ssm2602-objs := ssm2602.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o @@ -33,3 +34,4 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c new file mode 100644 index 0000000..feb462e --- /dev/null +++ b/sound/soc/codecs/ssm2602.c @@ -0,0 +1,773 @@ +/* + * File: sound/soc/codecs/ssm2602.c + * Author: Cliff Cai <Cliff.Cai(a)analog.com> + * + * Created: Tue June 06 2008 + * Description: Driver for ssm2602 sound chip built in ADSP-BF52xC + * + * Modified: + * Copyright 2008 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.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 "ssm2602.h" + +#define AUDIO_NAME "ssm2602" +#define SSM2602_VERSION "0.1" + +struct snd_soc_codec_device soc_codec_dev_ssm2602; + +/* codec private data */ +struct ssm2602_priv { + unsigned int sysclk; + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; +}; + +/* + * ssm2602 register cache + * We can't read the ssm2602 register space when we are + * using 2 wire for device control, so we cache them instead. + * There is no point in caching the reset register + */ +static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = { + 0x0017, 0x0017, 0x0079, 0x0079, + 0x0000, 0x0000, 0x0000, 0x000a, + 0x0000, 0x0000 +}; + +/* + * read ssm2602 register cache + */ +static inline unsigned int ssm2602_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == SSM2602_RESET) + return 0; + if (reg >= SSM2602_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write ssm2602 register cache + */ +static inline void ssm2602_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= SSM2602_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the ssm2602 register space + */ +static int ssm2602_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 ssm2602 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + ssm2602_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define ssm2602_reset(c) ssm2602_write(c, SSM2602_RESET, 0) +/*Appending several "None"s just for OSS mixer use*/ +static const char *ssm2602_input_select[] = { + "Line", "Mic", "None", "None", "None", + "None", "None", "None", +}; + +static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; + +static const struct soc_enum ssm2602_enum[] = { + SOC_ENUM_SINGLE(SSM2602_APANA, 2, 2, ssm2602_input_select), + SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, 4, ssm2602_deemph), +}; + +static const struct snd_kcontrol_new ssm2602_snd_controls[] = { + +SOC_DOUBLE_R("Master Playback Volume", SSM2602_LOUT1V, SSM2602_ROUT1V, + 0, 127, 0), +SOC_DOUBLE_R("Master Playback ZC Switch", SSM2602_LOUT1V, SSM2602_ROUT1V, + 7, 1, 0), + +SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0), +SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1), + +SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0), +SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1), + +SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1), + +SOC_SINGLE("ADC High Pass Filter Switch", SSM2602_APDIGI, 0, 1, 1), +SOC_SINGLE("Store DC Offset Switch", SSM2602_APDIGI, 4, 1, 0), + +SOC_ENUM("Capture Source", ssm2602_enum[0]), + +SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]), +}; + +/* add non dapm controls */ +static int ssm2602_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Output Mixer */ +static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0), +SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0), +SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 1, 0), +}; + +/* Input mux */ +static const struct snd_kcontrol_new ssm2602_input_mux_controls = +SOC_DAPM_ENUM("Input Select", ssm2602_enum[0]); + +static const struct snd_soc_dapm_widget ssm2602_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1, + &ssm2602_output_mixer_controls[0], + ARRAY_SIZE(ssm2602_output_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("LHPOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_OUTPUT("RHPOUT"), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1), +SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls), +SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1), +SND_SOC_DAPM_INPUT("MICIN"), +SND_SOC_DAPM_INPUT("RLINEIN"), +SND_SOC_DAPM_INPUT("LLINEIN"), +}; + +static const struct snd_soc_dapm_route audio_conn[] = { + /* output mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "HiFi Playback Switch", "DAC"}, + {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, + + /* outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + + /* input mux */ + {"Input Mux", "Line", "Line Input"}, + {"Input Mux", "Mic", "Mic Bias"}, + {"ADC", NULL, "Input Mux"}, + + /* inputs */ + {"Line Input", NULL, "LLINEIN"}, + {"Line Input", NULL, "RLINEIN"}, + {"Mic Bias", NULL, "MICIN"}, + + /* terminator */ + {NULL, NULL, NULL}, +}; + +static int ssm2602_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ssm2602_dapm_widgets, + ARRAY_SIZE(ssm2602_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +struct _coeff_div { + u32 mclk; + u32 rate; + u16 fs; + u8 sr:4; + u8 bosr:1; + u8 usb:1; +}; + +/* codec mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 48k */ + {12288000, 48000, 256, 0x0, 0x0, 0x0}, + {18432000, 48000, 384, 0x0, 0x1, 0x0}, + {12000000, 48000, 250, 0x0, 0x0, 0x1}, + + /* 32k */ + {12288000, 32000, 384, 0x6, 0x0, 0x0}, + {18432000, 32000, 576, 0x6, 0x1, 0x0}, + {12000000, 32000, 375, 0x6, 0x0, 0x1}, + + /* 8k */ + {12288000, 8000, 1536, 0x3, 0x0, 0x0}, + {18432000, 8000, 2304, 0x3, 0x1, 0x0}, + {11289600, 8000, 1408, 0xb, 0x0, 0x0}, + {16934400, 8000, 2112, 0xb, 0x1, 0x0}, + {12000000, 8000, 1500, 0x3, 0x0, 0x1}, + + /* 96k */ + {12288000, 96000, 128, 0x7, 0x0, 0x0}, + {18432000, 96000, 192, 0x7, 0x1, 0x0}, + {12000000, 96000, 125, 0x7, 0x0, 0x1}, + + /* 44.1k */ + {11289600, 44100, 256, 0x8, 0x0, 0x0}, + {16934400, 44100, 384, 0x8, 0x1, 0x0}, + {12000000, 44100, 272, 0x8, 0x1, 0x1}, + + /* 88.2k */ + {11289600, 88200, 128, 0xf, 0x0, 0x0}, + {16934400, 88200, 192, 0xf, 0x1, 0x0}, + {12000000, 88200, 136, 0xf, 0x1, 0x1}, +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return 0; +} + +static int ssm2602_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; + struct ssm2602_priv *ssm2602 = codec->private_data; + u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; + int i = get_coeff(ssm2602->sysclk, params_rate(params)); + u16 srate = (coeff_div[i].sr << 2) | + (coeff_div[i].bosr << 1) | coeff_div[i].usb; + + ssm2602_write(codec, SSM2602_ACTIVE, 0); + ssm2602_write(codec, SSM2602_SRATE, srate); + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x000c; + break; + } + ssm2602_write(codec, SSM2602_IFACE, iface); + ssm2602_write(codec, SSM2602_ACTIVE, ACTIVATE_CODEC); + return 0; +} + +static int ssm2602_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; + struct snd_pcm_runtime *master_runtime; + + /* The DAI has shared clocks so if we already have a playback or + * capture going then constrain this substream to match it. + */ + if (ssm2602->master_substream) { + master_runtime = ssm2602->master_substream->runtime; + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + ssm2602->slave_substream = substream; + } else + ssm2602->master_substream = substream; + + return 0; +} + +static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + /* set active */ + ssm2602_write(codec, SSM2602_ACTIVE, ACTIVATE_CODEC); + + return 0; +} + +static void ssm2602_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + /* deactivate */ + if (!codec->active) { + udelay(50); + ssm2602_write(codec, SSM2602_ACTIVE, 0); + } +} + +static int ssm2602_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = ssm2602_read_reg_cache(codec, SSM2602_APDIGI) & 0xfff7; + if (mute) + ssm2602_write(codec, SSM2602_APDIGI, + mute_reg | ENABLE_DAC_MUTE); + else + ssm2602_write(codec, SSM2602_APDIGI, mute_reg); + return 0; +} + +static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + ssm2602->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface |= 0x0040; + 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 |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + 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 |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + /* set iface */ + ssm2602_write(codec, SSM2602_IFACE, iface); + return 0; +} + +static int ssm2602_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = ssm2602_read_reg_cache(codec, SSM2602_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + /* vref/mid, osc on, dac unmute */ + ssm2602_write(codec, SSM2602_PWR, 0); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + ssm2602_write(codec, SSM2602_PWR, reg | CLK_OUT_PDN); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + ssm2602_write(codec, SSM2602_ACTIVE, 0); + ssm2602_write(codec, SSM2602_PWR, 0xffff); + break; + + } + codec->bias_level = level; + return 0; +} + +#define SSM2602_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) + +struct snd_soc_dai ssm2602_dai = { + .name = "SSM2602", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SSM2602_RATES, + .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SSM2602_RATES, + .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .ops = { + .startup = ssm2602_startup, + .prepare = ssm2602_pcm_prepare, + .hw_params = ssm2602_hw_params, + .shutdown = ssm2602_shutdown, + }, + .dai_ops = { + .digital_mute = ssm2602_mute, + .set_sysclk = ssm2602_set_dai_sysclk, + .set_fmt = ssm2602_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(ssm2602_dai); + +static int ssm2602_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; + + ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ssm2602_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(ssm2602_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + ssm2602_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the ssm2602 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ssm2602_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int reg, ret = 0; + + codec->name = "SSM2602"; + codec->owner = THIS_MODULE; + codec->read = ssm2602_read_reg_cache; + codec->write = ssm2602_write; + codec->set_bias_level = ssm2602_set_bias_level; + codec->dai = &ssm2602_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(ssm2602_reg); + codec->reg_cache = kmemdup(ssm2602_reg, sizeof(ssm2602_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + ssm2602_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + pr_err("ssm2602: failed to create pcms\n"); + goto pcm_err; + } + /*power on device*/ + ssm2602_write(codec, SSM2602_ACTIVE, 0); + /* set the update bits */ + reg = ssm2602_read_reg_cache(codec, SSM2602_LINVOL); + ssm2602_write(codec, SSM2602_LINVOL, reg | LRIN_BOTH); + reg = ssm2602_read_reg_cache(codec, SSM2602_RINVOL); + ssm2602_write(codec, SSM2602_RINVOL, reg | RLIN_BOTH); + reg = ssm2602_read_reg_cache(codec, SSM2602_LOUT1V); + ssm2602_write(codec, SSM2602_LOUT1V, reg | LRHP_BOTH); + reg = ssm2602_read_reg_cache(codec, SSM2602_ROUT1V); + ssm2602_write(codec, SSM2602_ROUT1V, reg | RLHP_BOTH); + /*select Line in as default input*/ + ssm2602_write(codec, SSM2602_APANA, + ENABLE_MIC_BOOST2 | SELECT_DAC | ENABLE_MIC_BOOST); + ssm2602_write(codec, SSM2602_PWR, 0); + + ssm2602_add_controls(codec); + ssm2602_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + pr_err("ssm2602: 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 *ssm2602_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * ssm2602 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int ssm2602_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) +{ + struct snd_soc_device *socdev = ssm2602_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = ssm2602_init(socdev); + if (ret < 0) + pr_err("failed to initialise SSM2602\n"); + + return ret; +} + +static int ssm2602_i2c_remove(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + kfree(codec->reg_cache); + return 0; +} + +static const struct i2c_device_id ssm2602_i2c_id[] = { + { "ssm2602", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); +/* corgi i2c codec control layer */ +static struct i2c_driver ssm2602_i2c_driver = { + .driver = { + .name = "SSM2602 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = ssm2602_i2c_probe, + .remove = ssm2602_i2c_remove, + .id_table = ssm2602_i2c_id, +}; + +static int ssm2602_add_i2c_device(struct platform_device *pdev, + const struct ssm2602_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&ssm2602_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "ssm2602", I2C_NAME_SIZE); + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + return 0; +err_driver: + i2c_del_driver(&ssm2602_i2c_driver); + return -ENODEV; +} + +#endif + +static int ssm2602_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct ssm2602_setup_data *setup; + struct snd_soc_codec *codec; + struct ssm2602_priv *ssm2602; + int ret = 0; + + pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + ssm2602 = kzalloc(sizeof(struct ssm2602_priv), GFP_KERNEL); + if (ssm2602 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = ssm2602; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ssm2602_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + codec->hw_write = (hw_write_t)i2c_master_send; + ret = ssm2602_add_i2c_device(pdev, setup); + } +#else + /* other interfaces */ +#endif + return ret; +} + +/* remove everything here */ +static int ssm2602_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) + ssm2602_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_unregister_device(codec->control_data); + i2c_del_driver(&ssm2602_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ssm2602 = { + .probe = ssm2602_probe, + .remove = ssm2602_remove, + .suspend = ssm2602_suspend, + .resume = ssm2602_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602); + +MODULE_DESCRIPTION("ASoC ssm2602 driver"); +MODULE_AUTHOR("Cliff Cai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2602.h b/sound/soc/codecs/ssm2602.h new file mode 100644 index 0000000..ec5e604 --- /dev/null +++ b/sound/soc/codecs/ssm2602.h @@ -0,0 +1,131 @@ +/* + * File: sound/soc/codecs/ssm2602.h + * Author: Cliff Cai <Cliff.Cai(a)analog.com> + * + * Created: Tue June 06 2008 + * Description: Driver for SSM2602 sound chip built in ADSP-BF52xC + * + * Modified: + * Copyright 2008 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef _SSM2602_H +#define _SSM2602_H + +/* SSM2602 Codec Register definitions */ + +#define SSM2602_LINVOL 0x00 +#define SSM2602_RINVOL 0x01 +#define SSM2602_LOUT1V 0x02 +#define SSM2602_ROUT1V 0x03 +#define SSM2602_APANA 0x04 +#define SSM2602_APDIGI 0x05 +#define SSM2602_PWR 0x06 +#define SSM2602_IFACE 0x07 +#define SSM2602_SRATE 0x08 +#define SSM2602_ACTIVE 0x09 +#define SSM2602_RESET 0x0f + +/*SSM2602 Codec Register Field definitions + *(Mask value to extract the corresponding Register field) + */ + +/*Left ADC Volume Control (SSM2602_REG_LEFT_ADC_VOL)*/ +#define LIN_VOL 0x01F /* Left Channel PGA Volume control */ +#define LIN_ENABLE_MUTE 0x080 /* Left Channel Input Mute */ +#define LRIN_BOTH 0x100 /* Left Channel Line Input Volume update */ + +/*Right ADC Volume Control (SSM2602_REG_RIGHT_ADC_VOL)*/ +#define RIN_VOL 0x01F /* Right Channel PGA Volume control */ +#define RIN_ENABLE_MUTE 0x080 /* Right Channel Input Mute */ +#define RLIN_BOTH 0x100 /* Right Channel Line Input Volume update */ + +/*Left DAC Volume Control (SSM2602_REG_LEFT_DAC_VOL)*/ +#define LHP_VOL 0x07F /* Left Channel Headphone volume control */ +#define ENABLE_LZC 0x080 /* Left Channel Zero cross detect enable */ +#define LRHP_BOTH 0x100 /* Left Channel Headphone volume update */ + +/*Right DAC Volume Control (SSM2602_REG_RIGHT_DAC_VOL)*/ +#define RHP_VOL 0x07F /* Right Channel Headphone volume control */ +#define ENABLE_RZC 0x080 /* Right Channel Zero cross detect enable */ +#define RLHP_BOTH 0x100 /* Right Channel Headphone volume update */ + +/*Analogue Audio Path Control (SSM2602_REG_ANALOGUE_PATH)*/ +#define ENABLE_MIC_BOOST 0x001 /* Primary Microphone Amplifier gain booster control */ +#define ENABLE_MIC_MUTE 0x002 /* Microphone Mute Control */ +#define ADC_IN_SELECT 0x004 /* Microphone/Line IN select to ADC (1=MIC, 0=Line In) */ +#define ENABLE_BYPASS 0x008 /* Line input bypass to line output */ +#define SELECT_DAC 0x010 /* Select DAC (1=Select DAC, 0=Don't Select DAC) */ +#define ENABLE_SIDETONE 0x020 /* Enable/Disable Side Tone */ +#define SIDETONE_ATTN 0x0C0 /* Side Tone Attenuation */ +#define ENABLE_MIC_BOOST2 0x100 /* Secondary Microphone Amplifier gain booster control */ + +/*Digital Audio Path Control (SSM2602_REG_DIGITAL_PATH)*/ +#define ENABLE_ADC_HPF 0x001 /* Enable/Disable ADC Highpass Filter */ +#define DE_EMPHASIS 0x006 /* De-Emphasis Control */ +#define ENABLE_DAC_MUTE 0x008 /* DAC Mute Control */ +#define STORE_OFFSET 0x010 /* Store/Clear DC offset when HPF is disabled */ + +/*Power Down Control (SSM2602_REG_POWER) + *(1=Enable PowerDown, 0=Disable PowerDown) + */ +#define LINE_IN_PDN 0x001 /* Line Input Power Down */ +#define MIC_PDN 0x002 /* Microphone Input & Bias Power Down */ +#define ADC_PDN 0x004 /* ADC Power Down */ +#define DAC_PDN 0x008 /* DAC Power Down */ +#define OUT_PDN 0x010 /* Outputs Power Down */ +#define OSC_PDN 0x020 /* Oscillator Power Down */ +#define CLK_OUT_PDN 0x040 /* CLKOUT Power Down */ +#define POWER_OFF 0x080 /* POWEROFF Mode */ + +/*Digital Audio Interface Format (SSM2602_REG_DIGITAL_IFACE)*/ +#define IFACE_FORMAT 0x003 /* Digital Audio input format control */ +#define AUDIO_DATA_LEN 0x00C /* Audio Data word length control */ +#define DAC_LR_POLARITY 0x010 /* Polarity Control for clocks in RJ,LJ and I2S modes */ +#define DAC_LR_SWAP 0x020 /* Swap DAC data control */ +#define ENABLE_MASTER 0x040 /* Enable/Disable Master Mode */ +#define BCLK_INVERT 0x080 /* Bit Clock Inversion control */ + +/*Sampling Control (SSM2602_REG_SAMPLING_CTRL)*/ +#define ENABLE_USB_MODE 0x001 /* Enable/Disable USB Mode */ +#define BOS_RATE 0x002 /* Base Over-Sampling rate */ +#define SAMPLE_RATE 0x03C /* Clock setting condition (Sampling rate control) */ +#define CORECLK_DIV2 0x040 /* Core Clock divider select */ +#define CLKOUT_DIV2 0x080 /* Clock Out divider select */ + +/*Active Control (SSM2602_REG_ACTIVE_CTRL)*/ +#define ACTIVATE_CODEC 0x001 /* Activate Codec Digital Audio Interface */ + +/*********************************************************************/ + +#define SSM2602_CACHEREGNUM 10 + +#define SSM2602_SYSCLK 0 +#define SSM2602_DAI 0 + +struct ssm2602_setup_data { + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai ssm2602_dai; +extern struct snd_soc_codec_device soc_codec_dev_ssm2602; + +#endif -- 1.5.6

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[alsa-devel] [PATCH 1/1] ASoC codec: fix compiling error in ad1980 driver after ASoC API changed
by Bryan Wu 04 Sep '08

04 Sep '08

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> --- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ad1980.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index bfbab3d..4e09c1f 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -141,7 +141,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } -struct snd_soc_codec_dai ad1980_dai = { +struct snd_soc_dai ad1980_dai = { .name = "AC97", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/ad1980.h b/sound/soc/codecs/ad1980.h index 5d4710d..db6c850 100644 --- a/sound/soc/codecs/ad1980.h +++ b/sound/soc/codecs/ad1980.h @@ -17,7 +17,7 @@ #define PR5 0x2000 #define PR6 0x4000 -extern struct snd_soc_codec_dai ad1980_dai; +extern struct snd_soc_dai ad1980_dai; extern struct snd_soc_codec_device soc_codec_dev_ad1980; #endif -- 1.5.6

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[alsa-devel] Does Gravis Classic driver work?
by Krzysztof Helt 04 Sep '08

04 Sep '08

Hi, I have acquired an unknown clone of the GF1 classic card. I load the snd-gusclassic driver with modprobe snd-gusclassic port=0x220 and the /proc/asound/cards says 0 [G37 ]: GUS - Gravis UltraSound Classic (3.7) Gravis UltraSound Classic (3.7) at 0x220, irq 5, dma 5&5 The alsamixer works. I set maximum volume for the PCM and master controls. The aplay -l says: **** List of PLAYBACK Hardware Devices **** card 0: G37 [Gravis UltraSound Classic (3.7)], device 0: GF1 [GF1 (synth)] Subdevices: 1/1 Subdevice #0: subdevice #0 but the aplay -L says nothing. the aplay with the .wav file does not play. aplay /usr/share/sounds/alsa/Front_Center.wav ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:545: audio open error: No such file or directory I have tried to follow the alsa wiki but it has not helped (btw, there is no driver the "gusclassic" listed, only the "gus"). Could you point me to any piece of information how to set up the card? The card works with DOS game with Miles (?) drivers. Regards, Krzysztof ---------------------------------------------------------------------- Teraz taniej na zagraniczne komorki! Sprawdz >> http://link.interia.pl/f1ef7

2 1

[alsa-devel] noise and consume pcm faster
by Nie Jun 03 Sep '08

03 Sep '08

Hi, I found a strange issue that ALSA consume pcm data faster than normal. Does any one had come across it? When I sychronize video frame against audio, the video thread will check how much pcm data has been written to ALSA API but has not been fetched by DMA. This query causes ALSA consume pcm data faster than normal, noise appears too. A stream that is 131 seconds will playback just 112 seconds. It only appears when audio is stereo and sampling rate is directly supported by hardware. This bug can be reproduced with aplay. To reproduce it with aplay, you just need to add a thread and query the ALSA latency 20~30times every second. My system: XSCALE PXA300/zylonite, Linux2.6.25, alsa-lib-1.0.13(I tried alsa-lib-1.0.17, bugs too), codec: wm9713. Linux2.6.14 has this issue too, but I did not find it in 2.6.21 and 2.6.24. Does anyone has idea about it? Thanks! Jun Nie

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