October 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH][fix] ALSA: bebob: fix wrong decoding of clock information for Terratec PHASE 88 Rack FW
by Takashi Sakamoto 27 Oct '14

27 Oct '14

Terratec PHASE 88 rack fw has two registers for source of clock, one is for internal/external, and another is for wordclock/spdif for external. When clock source is internal, information in another register has no meaning. Thus it must be ignored, but current implementation decodes it. This causes over-indexing reference to labels. Reported-by: András Murányi <muranyia(a)gmail.com> Tested-by: András Murányi <muranyia(a)gmail.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_terratec.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/firewire/bebob/bebob_terratec.c b/sound/firewire/bebob/bebob_terratec.c index 0e4c0bf..9940611 100644 --- a/sound/firewire/bebob/bebob_terratec.c +++ b/sound/firewire/bebob/bebob_terratec.c @@ -24,7 +24,12 @@ phase88_rack_clk_src_get(struct snd_bebob *bebob, unsigned int *id) if (err < 0) goto end; - *id = (enable_ext & 0x01) | ((enable_word & 0x01) << 1); + if (enable_ext == 0) + *id = 0; + else if (enable_word == 0) + *id = 1; + else + *id = 2; end: return err; } -- 1.9.1

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Re: [alsa-devel] ALSA - AD1893
by Raymond Yau 27 Oct '14

27 Oct '14

> I am working on Chromium and trying to get a machine with AD1983 to > function correctly. > > I'm on 3.14 kernel. My issue is that I need a way for CRAS (no pulse) > to tell HDA driver to switch capture from Front mic to rear Line/Mic. > The way alsamixer does it, but I want to add that functionality to Cras. > Could you point my to which IOCTL to call or alsa API to look at? You have to post the outpout of alsa-info.sh It depend on the pin default of the rear jack is line in or mic https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… - line_in_auto_switch (bool): enable/disable the line-in auto-switch feature; default false - add_stereo_mix_input (bool): add the stereo mix (analog-loopback mix) to the input mux if available - add_jack_modes (bool): add "xxx Jack Mode" enum controls to each I/O jack for allowing to change the headphone amp and mic bias VREF capabilities https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Stereo mix was already added to those desktop using VIA or ADI codec when auto-mic is not enabled Stereo mix also has input from DAC, line in and mic node 0x14 Record Selector can also select input from mono mixer node 0xf (mono mix) For those HDA codec with one ADC and most AC97 codecs amixer -c0 cset "Capture Source" "Line" Any reason to use codec specific ioctl since it won't work on other hda codec

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[alsa-devel] [RFC][PATCH 00/15 v4] OXFW driver, a succesor to firewire-speakers
by Takashi Sakamoto 26 Oct '14

26 Oct '14

This patchset updates a part of my previous series: [alsa-devel] [RFC v3] [PATCH 00/52] Enhancement for support of firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… This patchset is based on this series. (This series is under reviewing.) [alsa-devel] [PATCH 00/49 v4] Enhancement for support of Firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-April/075841.html If you try this patchset, please apply the series in advance and apply this patchset. Or just use backport codes in my github repository: https://github.com/takaswie/snd-firewire-improve Or if you use Ubuntu 14.04, DKMS package is available from David's PPA. (Special thanks to David Henningsson.) https://launchpad.net/~diwic/+archive/snd-firewire-improve If you have Griffin FireWave, I would strongly request to test. Please see 'An issue for FireWave' in this message. == Background Currently, in ALSA, 'firewire-speakers' driver supports OXFW970/971 chipset which Oxford Semiconductor produced. But this driver has some restrictions: - Don't support MIDI substreams - Don't support substreams for PCM/MIDI capturing There are several devices which are based on OXFW970/971. Especially, devices which Behringer and Mackie produced support playback/capture PCM samples/MIDI messages. So firewire-speakers needs re-work to support these devices. This patchset renames 'firewire-speakers' to 'oxfw'. Maintainers already agreed to rename firewire-speakers to suitable name, in my first RFC in Jan 2014. == Supported devices Currently these devices are supported: * Griffin Firewave * LaCie Firewire Speakers * Behringer F-Control Audio 202 * Mackie(Loud) Onyx-i series (former models) * Mackie(Loud) Onyx Satellite * Mackie(Loud) Tapco Link.Firewire These devices can be supported but its IDs are unknown: * Mackie(Loud), d.2 pro * Mackie(Loud), d.4 pro * Mackie(Loud), U.420 * Mackie(Loud), U.420d When you have unknown devices, please inform me output of this command: $ hexdump -C /sys/bus/firewire/devices/fwX/config_rom (X is node id for your device. Tipically '1'.) I tested this driver with Behringer F-Control Audio 202. == Features - Capturing/Playbacking PCM samples/MIDI messages at supported sampling rates via ALSA PCM/MIDI interfaces - Retrieving firewire node information via ALSA hwdep interface - Locking/Unlocking kernel streaming via ALSA hwdep interface - Receiving notification at starting/stopping kernel streaming via ALSA hwdep interface == Solutions for reported regressions Two regressions were reported for my previous series: 1.Some names in card structure is trimmed. http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073128.h… 2.The number of PCM channels are not detected correctly. http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073145.h… The cause of issue 1 is that the device has too long names in its config rom. To solve this issue, for the driver, this patchset uses the same way which firewire-speakers uses, propper alias in device-specific structure. The cause of issue 2 is that the device has several format entries for the same sampling rate: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073561.h… To solve this issue, this patchset includes improved mechanism to detect stream formation. Additionally, this patchset adds new device entry for Mackie(Loud) Tapco Link.Firewire, refering to FFADO bug ticket 236: http://subversion.ffado.org/ticket/236 == An issue for FireWave I have an issue for Griffin Firewave. This device can handle AV/C Stream Format Information command and driver can get to know stream formations via the command. With Stefan's help, responses from FireWave against this command are clear. http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073561.h… According to these responses, supported formations are: Input Stream to device: Rate PCM MIDI 32000 6 0 44100 6 0 48000 6 0 96000 6 0 32000 2 0 44100 2 0 48000 2 0 0 0 0 0 0 0 0 0 0 (I also expect to get this output from /proc/asound/FireWave/firewire/formation. I note it's order is not important.) But current 'firewire-speakers' has PCM constraints below: - 6ch at 32/44.1/48/96 kHz - 2ch at 48/96 kHz http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073562.h… So this OXFW driver may lost a support for 2ch/96.0kHz for FireWave and gain 2ch/32.0kHz and 2ch/44.1kHz. I don't know exactly the reason of these gaps. So I strongly hope testers to check these two items: - FireWave really can handle 2ch/96kHz with firewire-speakers driver. - FireWave really can handle 2ch/32.0kHz and 2ch/44.1kHz with oxfw driver. I think a test with aplay is easy to execute: $ aplay -D hw:FireWave -f S32_LE -c 2 -r 96000 /dev/urandom $ aplay -D hw:FireWave -f S32_LE -c 2 -r 48000 /dev/urandom $ aplay -D hw:FireWave -f S32_LE -c 2 -r 44100 /dev/urandom $ aplay -D hw:FireWave -f S32_LE -c 2 -r 32000 /dev/urandom If 2ch/96kHz is apparently available, I'll change codes as a quirk. FYI, when I tested unavailable number of PCM channels with a hack, FCA202 playback PCM substreams but becomes freezed after stopping streaming. $ cat /var/log/syslog ... kernel: [10098.670390] snd_oxfw fw1.0: transaction failed: timeout ... I believe the other OXFW970/971 based devices have a similar behaviour. ================ Takashi Sakamoto (15): speakers: Rename to oxfw and rename some members oxfw: Move to its own directory oxfw: Split stream functionality to a new file and add a header file oxfw: Split PCM functionality to a new file oxfw: Split control functionality to a new file oxfw: Change the way to name card oxfw: Change the way to make PCM rules/constraints oxfw: Add proc interface for debugging purpose oxfw: Change the way to start stream oxfw: Add support for AV/C stream format command to get supported stream formation oxfw: Add a quirk for Griffin FireWave oxfw: Add support for Behringer/Mackie devices oxfw: Add support AMDTP in-stream and PCM capture oxfw: Add support for capture/playback MIDI messages oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 18 +- sound/firewire/Makefile | 3 +- sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw.c | 284 +++++++++++++ sound/firewire/oxfw/oxfw.h | 141 +++++++ sound/firewire/oxfw/oxfw_command.c | 113 ++++++ sound/firewire/oxfw/oxfw_control.c | 283 +++++++++++++ sound/firewire/oxfw/oxfw_hwdep.c | 190 +++++++++ sound/firewire/oxfw/oxfw_midi.c | 168 ++++++++ sound/firewire/oxfw/oxfw_pcm.c | 367 +++++++++++++++++ sound/firewire/oxfw/oxfw_proc.c | 83 ++++ sound/firewire/oxfw/oxfw_stream.c | 658 ++++++++++++++++++++++++++++++ sound/firewire/speakers.c | 792 ------------------------------------- 15 files changed, 2308 insertions(+), 801 deletions(-) create mode 100644 sound/firewire/oxfw/Makefile create mode 100644 sound/firewire/oxfw/oxfw.c create mode 100644 sound/firewire/oxfw/oxfw.h create mode 100644 sound/firewire/oxfw/oxfw_command.c create mode 100644 sound/firewire/oxfw/oxfw_control.c create mode 100644 sound/firewire/oxfw/oxfw_hwdep.c create mode 100644 sound/firewire/oxfw/oxfw_midi.c create mode 100644 sound/firewire/oxfw/oxfw_pcm.c create mode 100644 sound/firewire/oxfw/oxfw_proc.c create mode 100644 sound/firewire/oxfw/oxfw_stream.c delete mode 100644 sound/firewire/speakers.c -- 1.8.3.2

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[alsa-devel] Request for comments for SND_SOC_IMX_WM8731
by Jonathan Bennett 26 Oct '14

26 Oct '14

This patch is a continuation of my work on supporting the Utilite devices on the vanilla kernel. I've modified the code slightly to conform to established practice. Namely instead of using src-port and ext-port, we use mux-int-port and mux-ext-port. This driver has support for putting the codec in either master mode or slave mode. So far I've only had success with using slave mode. For trying to submit for kernel inclusion, should I strip out the code for codec master mode? As a kernel dev noob, any other comments are welcome. Thanks, Jonathan Bennett diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 081e406..09c923f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,17 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8731 + tristate "SoC Audio support for i.MX boards with wm8731 + depends on OF && I2C + select SND_SOC_WM8731 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Soc Audio support for i.MX with wm8731 + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..4292fac 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8731-objs := imx-wm8731.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -65,5 +66,6 @@ obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o +obj-$(CONFIG_SND_SOC_IMX_WM8731) += snd-soc-imx-wm8731.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8731.c b/sound/soc/fsl/imx-wm8731.c new file mode 100644 index 0000000..8ce4e9f --- /dev/null +++ b/sound/soc/fsl/imx-wm8731.c @@ -0,0 +1,687 @@ +/* + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright (C) 2012 Freescale Semiconductor, Inc. + * Copyright (C) 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> + +#include "../codecs/wm8731.h" +#include "imx-audmux.h" +#include "imx-ssi.h" + +#define DAI_NAME_SIZE 32 +#define WM8731_MCLK_FREQ (24000000 / 2) + +struct imx_wm8731_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct i2c_client *codec_dev; + /* audio_clocking_data */ + struct clk *pll; + struct clk *clock_root; + long sysclk; + long current_rate; + /* apis */ + int (*clock_enable)(int enable,struct imx_wm8731_data *data); +}; + +static int imx_wm8731_init(struct snd_soc_pcm_runtime *rtd); +static int imx_hifi_hw_params_slv_mode(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params); +static void imx_hifi_shutdown(struct snd_pcm_substream *substream); + +struct imx_priv { + struct platform_device *pdev; + struct imx_wm8731_data *data; +}; + +static struct imx_priv card_priv; + +static struct snd_soc_ops imx_hifi_ops = { + .shutdown = imx_hifi_shutdown, +}; + +/* imx card dapm widgets */ +static const struct snd_soc_dapm_widget imx_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_LINE("Line Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +/* imx machine connections to the codec pins */ +static const struct snd_soc_dapm_route audio_map[] = { + { "Headphone Jack", NULL, "LHPOUT" }, + { "Headphone Jack", NULL, "RHPOUT" }, + + { "Ext Spk", NULL, "LOUT" }, + { "Ext Spk", NULL, "ROUT" }, + + { "LLINEIN", NULL, "Line Jack" }, + { "RLINEIN", NULL, "Line Jack" }, + + { "MICIN", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "Mic Jack"}, +}; + +static int wm8731_slv_mode_init(struct imx_wm8731_data *data) +{ + struct clk *new_parent; + struct clk *ssi_clk; + struct i2c_client *codec_dev = data->codec_dev; + + new_parent = devm_clk_get(&codec_dev->dev, "pll4"); + if (IS_ERR(new_parent)) { + pr_err("Could not get \"pll4\" clock \n"); + return PTR_ERR(new_parent); + } + + ssi_clk = devm_clk_get(&codec_dev->dev, "imx-ssi.1"); + if (IS_ERR(ssi_clk)) { + pr_err("Could not get \"imx-ssi.1\" clock \n"); + return PTR_ERR(ssi_clk); + } + + clk_set_parent(ssi_clk, new_parent); + + data->pll = new_parent; + data->clock_root = ssi_clk; + data->current_rate = 0; + + data->sysclk = 0; + + return 0; +} + +static int wm8731_slv_mode_clock_enable(int enable, struct imx_wm8731_data *data) +{ + long pll_rate; + long rate_req; + long rate_avail; + + if (!enable) + return 0; + + if (data->sysclk == data->current_rate) + return 0; + + switch (data->sysclk) { + case 11289600: + pll_rate = 632217600; + break; + + case 12288000: + pll_rate = 688128000; + break; + + default: + return -EINVAL; + } + + rate_req = pll_rate; + rate_avail = clk_round_rate(data->pll, rate_req); + clk_set_rate(data->pll, rate_avail); + + rate_req = data->sysclk; + rate_avail = clk_round_rate(data->clock_root, + rate_req); + clk_set_rate(data->clock_root, rate_avail); + + pr_info("%s: \"imx-ssi.1\" rate = %ld (= %ld)\n", + __func__, rate_avail, rate_req); + + data->current_rate = data->sysclk; + + return 0; +} + +static int imx_hifi_startup_slv_mode(struct snd_pcm_substream *substream) +{ + /* + * As SSI's sys clock rate depends on sampling rate, + * the clock enabling code is moved to imx_hifi_hw_params(). + */ + return 0; +} + +static int wm8731_mst_mode_init(struct imx_wm8731_data *data) +{ + long rate; + struct clk *new_parent; + struct clk *ssi_clk; + struct i2c_client *codec_dev = data->codec_dev; + + new_parent = devm_clk_get(&codec_dev->dev, "cko2"); + if (IS_ERR(new_parent)) { + pr_err("Could not get \"cko2\" clock \n"); + return PTR_ERR(new_parent); + } + + ssi_clk = devm_clk_get(&codec_dev->dev, "cko"); + if (IS_ERR(ssi_clk)) { + pr_err("Could not get \"cko\" clock \n"); + return PTR_ERR(ssi_clk); + } + + rate = clk_round_rate(new_parent, WM8731_MCLK_FREQ); + clk_set_rate(new_parent, rate); + + clk_set_parent(ssi_clk, new_parent); + + rate = clk_round_rate(ssi_clk, WM8731_MCLK_FREQ); + clk_set_rate(ssi_clk, rate); + + pr_info("%s: \"CLKO\" rate = %ld (= %d)\n", + __func__, rate, WM8731_MCLK_FREQ); + + data->pll = new_parent; + data->clock_root = ssi_clk; + data->sysclk = rate; + + return 0; +} + +static int wm8731_mst_mode_clock_enable(int enable, struct imx_wm8731_data *data) +{ + struct clk *clko = data->clock_root; + + if (enable) + clk_enable(clko); + else + clk_disable(clko); + + return 0; +} + +static int imx_hifi_startup_mst_mode(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_card *card = codec_dai->card; + struct imx_wm8731_data *data = snd_soc_card_get_drvdata(card); + + if (!codec_dai->active) + data->clock_enable(1,data); + + return 0; +} + + +static int imx_hifi_hw_params_slv_mode(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = codec_dai->card; + struct imx_wm8731_data *data = snd_soc_card_get_drvdata(card); + + u32 dai_format; + snd_pcm_format_t sample_format; + unsigned int channels; + unsigned int tx_mask, rx_mask; + unsigned int sampling_rate; + unsigned int div_2, div_psr, div_pm; + int ret; + + sampling_rate = params_rate(params); + sample_format = params_format(params); + + channels = params_channels(params); + printk("%s:%s sampling rate = %u channels = %u \n", __FUNCTION__, + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? "Playback" : "Capture"), + sampling_rate, channels); + + /* set CPU DAI configuration */ + switch (sampling_rate) { + case 8000: + case 32000: + case 48000: + case 96000: + data->sysclk = 12288000; + break; + + case 44100: + case 88200: + data->sysclk = 11289600; + break; + + default: + return -EINVAL; + } + + wm8731_slv_mode_clock_enable(1,data); + + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS; + + ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); + if (ret < 0) + return ret; + + /* set i.MX active slot mask */ + /* S[TR]CCR:DC */ + tx_mask = ~((1 << channels) - 1); + rx_mask = tx_mask; + snd_soc_dai_set_tdm_slot(cpu_dai, tx_mask, rx_mask, 2, 32); + + /* + * SSI sysclk divider: + * div_2: /1 or /2 + * div_psr: /1 or /8 + * div_pm: /1 .. /256 + */ + div_2 = 0; + div_psr = 0; + switch (sampling_rate) { + case 8000: + // 1x1x12 + div_pm = 11; + break; + case 32000: + // 1x1x3 + div_pm = 2; + break; + case 48000: + // 1x1x2 + div_pm = 1; + break; + case 96000: + // 1x1x1 + div_pm = 0; + break; + case 44100: + // 1x1x2 + div_pm = 1; + break; + case 88200: + // 1x1x1 + div_pm = 0; + break; + default: + return -EINVAL; + } + + /* sync mode: a single clock controls both playback and capture */ + snd_soc_dai_set_clkdiv(cpu_dai, IMX_SSI_TX_DIV_2, (div_2 ? SSI_STCCR_DIV2 : 0)); + snd_soc_dai_set_clkdiv(cpu_dai, IMX_SSI_TX_DIV_PSR, (div_psr ? SSI_STCCR_PSR : 0)); + snd_soc_dai_set_clkdiv(cpu_dai, IMX_SSI_TX_DIV_PM, div_pm); + + /* set codec DAI configuration */ + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + ret = snd_soc_dai_set_fmt(codec_dai, dai_format); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8731_SYSCLK_MCLK, + data->sysclk, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + pr_err("Failed to set codec master clock to %u: %d \n", + data->sysclk, ret); + return ret; + } + + return 0; +} + +static int imx_hifi_hw_params_mst_mode(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = codec_dai->card; + struct imx_wm8731_data *data = snd_soc_card_get_drvdata(card); + u32 dai_format; + unsigned int channels; + unsigned int tx_mask, rx_mask; + unsigned int sampling_rate; + int ret; + + + sampling_rate = params_rate(params); + channels = params_channels(params); + pr_debug("%s:%s sampling rate = %u channels = %u \n", __FUNCTION__, + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? "Playback" : "Capture"), + sampling_rate, channels); + + /* set cpu DAI configuration */ + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM; + + ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); + if (ret < 0) + return ret; + + /* set i.MX active slot mask */ + /* S[TR]CCR:DC */ + tx_mask = ~((1 << channels) - 1); + rx_mask = tx_mask; + snd_soc_dai_set_tdm_slot(cpu_dai, tx_mask, rx_mask, 2, 32); + + /* set codec DAI configuration */ + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + ret = snd_soc_dai_set_fmt(codec_dai, dai_format); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8731_SYSCLK_MCLK, + data->sysclk, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + pr_err("Failed to set codec master clock to %u: %d \n", + data->sysclk, ret); + return ret; + } + + return 0; +} + +static void imx_hifi_shutdown(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_card *card = codec_dai->card; + struct imx_wm8731_data *data = snd_soc_card_get_drvdata(card); + + if (!codec_dai->active) + data->clock_enable(0,data); + + return; +} + +static int imx_wm8731_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret = 0; + struct snd_soc_codec *codec = rtd->codec; + + /* Add imx specific widgets */ + ret = snd_soc_dapm_new_controls(&codec->dapm, imx_dapm_widgets, + ARRAY_SIZE(imx_dapm_widgets)); + if (ret) + goto out_retcode; + + /* Set up imx specific audio path audio_map */ + ret = snd_soc_dapm_add_routes(&codec->dapm, audio_map, ARRAY_SIZE(audio_map)); + if (ret) + goto out_retcode; + + ret = snd_soc_dapm_enable_pin(&codec->dapm, "Headphone Jack"); + if (ret) + goto out_retcode; + + ret = snd_soc_dapm_nc_pin(&codec->dapm, "Ext Spk"); + if (ret) + goto out_retcode; + +out_retcode: + + if (ret) + pr_err("%s: failed with error code: %d \n", __FUNCTION__, ret); + else + pr_info("%s: success \n", __FUNCTION__); + + return ret; +} + +/** + * Configure AUDMUX interconnection between + * _slave (CPU side) and _master (codec size) + * + * When SSI operates in master mode, 5-wire interconnect with + * audio codec is required: + * TXC - BCLK + * TXD - DAC data + * RXD - ADC data + * TXFS - {DAC|ADC}LRC, i.e. word clock + * RXC - MCLK, i.e. oversampling clock + * Audmux is operated in asynchronous mode to enable 6-wire + * interface (as opposed to 4-wire interface in sync mode). + */ +static int imx_audmux_config_slv_mode(int _slave, int _master) +{ + unsigned int ptcr, pdcr; + int slave = _slave - 1; + int master = _master - 1; + + ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(slave) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_RCSEL(slave | 0x8) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(slave); + + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(slave); + imx_audmux_v2_configure_port(master, ptcr, pdcr); + ptcr = ptcr & ~IMX_AUDMUX_V2_PTCR_SYN; + imx_audmux_v2_configure_port(master, ptcr, pdcr); + + ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_RCSEL(master | 0x8) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(master); + + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(master); + imx_audmux_v2_configure_port(slave, ptcr, pdcr); + ptcr = ptcr & ~IMX_AUDMUX_V2_PTCR_SYN; + imx_audmux_v2_configure_port(slave, ptcr, pdcr); + + return 0; +} + +static int imx_audmux_config_mst_mode(int _slave, int _master) +{ + unsigned int ptcr, pdcr; + int slave = _slave - 1; + int master = _master - 1; + + ptcr = IMX_AUDMUX_V2_PTCR_SYN; + ptcr |= IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(master) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(master); + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(master); + imx_audmux_v2_configure_port(slave, ptcr, pdcr); + + ptcr = IMX_AUDMUX_V2_PTCR_SYN; + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(slave); + imx_audmux_v2_configure_port(master, ptcr, pdcr); + + return 0; +} + +static int imx_wm8731_probe(struct platform_device *pdev) +{ + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8731_data *data; + unsigned int src_port, ext_port; + unsigned int ssi_mode; + const char *ssi_mode_str; + + int ret; + + priv->pdev = pdev; + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + card_priv.data = data; + + data->codec_dev = codec_dev; + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8731-hifi"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops; + data->dai.init = &imx_wm8731_init; + + ret = of_property_read_u32(pdev->dev.of_node, "mux-int-port", &src_port); + if (ret) { + dev_err(&pdev->dev, "failed to get \"mux-int-port\" value\n"); + ret = -EINVAL; + goto fail; + } + + ret = of_property_read_u32(pdev->dev.of_node, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "failed to get \"mux-ext-port\" value\n"); + ret = -EINVAL; + goto fail; + } + + ret = of_property_read_string(ssi_np, "fsl,mode", &ssi_mode_str); + if (ret) { + dev_err(&pdev->dev, "failed to get \"fsl,mode\" value\n"); + ret = -EINVAL; + goto fail; + } + + ssi_mode = strcmp(ssi_mode_str, "i2s-master"); + + if (ssi_mode) { + /* Master Mode */ + imx_audmux_config_mst_mode(src_port, ext_port); + wm8731_mst_mode_init(data); + data->clock_enable = wm8731_mst_mode_clock_enable; + imx_hifi_ops.hw_params = imx_hifi_hw_params_mst_mode; + imx_hifi_ops.startup = imx_hifi_startup_mst_mode; + } else { + /* Slave Mode */ + imx_audmux_config_slv_mode(src_port, ext_port); + wm8731_slv_mode_init(data); + data->clock_enable = wm8731_slv_mode_clock_enable; + imx_hifi_ops.hw_params = imx_hifi_hw_params_slv_mode; + imx_hifi_ops.startup = imx_hifi_startup_slv_mode; + } + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + data->card.num_links = 1; + data->card.dai_link = &data->dai; + + data->card.dapm_widgets = imx_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_dapm_widgets); + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } +/* of_node_put(ssi_np); + of_node_put(codec_np); */ + return 0; +fail: + + if (ssi_np) + of_node_put(ssi_np); + + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8731_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + snd_soc_unregister_card(card); + + return 0; +} + +static const struct of_device_id imx_wm8731_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8731", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8731_dt_ids); + +static struct platform_driver imx_wm8731_driver = { + .driver = { + .name = "imx-wm8731", + .owner = THIS_MODULE, + .of_match_table = imx_wm8731_dt_ids, + }, + .probe = imx_wm8731_probe, + .remove = imx_wm8731_remove, +}; +module_platform_driver(imx_wm8731_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8731 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8731");

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[alsa-devel] [PATCH] ASoC: rt5645:Mark RT5645_TDM_CTRL_3 as readable
by yang.a.fang＠intel.com 26 Oct '14

26 Oct '14

From: "Fang, Yang A" <yang.a.fang(a)intel.com> amixer query fails due to it is not readable reigster Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3fb83bf..d16331e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -139,6 +139,7 @@ static const struct reg_default rt5645_reg[] = { { 0x76, 0x000a }, { 0x77, 0x0c00 }, { 0x78, 0x0000 }, + { 0x79, 0x0123 }, { 0x80, 0x0000 }, { 0x81, 0x0000 }, { 0x82, 0x0000 }, @@ -334,6 +335,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_DMIC_CTRL2: case RT5645_TDM_CTRL_1: case RT5645_TDM_CTRL_2: + case RT5645_TDM_CTRL_3: case RT5645_GLB_CLK: case RT5645_PLL_CTRL1: case RT5645_PLL_CTRL2: -- 1.7.9.5

1 0

[alsa-devel] [PATCH 0/2] Disable the ThinkPad HW mute/level control when possible
by Andy Lutomirski 25 Oct '14

25 Oct '14

This series is being resurrected after several years, and it's quite different from before, so I'm restarting the version number at 1 :) ThinkPad volume and mute controls are a mess. For whatever reason, ThinkPads have mute buttons that send KEY_MUTE *and* control an invisible-to-ALSA mute switch. Some of them have volume controls that interact with this switch as well. This is a perennial source of problems. On most ThinkPads, if you press mute and then unmute using GUI controls, you have no sound, because userspace and ALSA state gets out of sync with the hardware switch. There's a separate "sound card" that exposes the hardware switch, but userspace code generally doesn't understand that. There are already a few _OSI(Linux) overrides to turn all the hardware buttons into regular buttons. Rather than quirking ACPI everywhere, just teach thinkpad-acpi to program the buttons for full software control and to disable hardware controls. That allows us to remove the ACPI quirks and have normal mute controls. This approach should be much simpler than adding even more kludgey ALSA integration for questionable gain. I haven't tested on my X220 (which has a mute light in addition to a mute button), since that laptop's SATA controller died last week. It should work, though, and if someone else could test this on a ThinkPad with a mute light, that would be great. Andy Lutomirski (2): thinkpad-acpi: Try to use full software mute control acpi: Remove _OSI(Linux) for ThinkPads drivers/acpi/blacklist.c | 54 ------------------ drivers/platform/x86/thinkpad_acpi.c | 107 +++++++++++++++++++++++++++++++---- 2 files changed, 97 insertions(+), 64 deletions(-) -- 1.9.3

2 10

[alsa-devel] Recording with asus xonar essence st card
by Vasily Postnicov 25 Oct '14

25 Oct '14

Hello. This is not related to alsa, but I decided to write this message here, because as far as I know alsa has the best open source driver for asus xonar series. I am using DragonFlyBSD and a bit modified driver originally written for FreeBSD by Alexander Polakov https://github.com/shamazmazum/xonar-dragonfly It lacks support for recording. I am trying to add this functionality from alsa/OSS drivers, but I have some unanswered questions. There are they here: 1) How do I initialize my soundcard for recording properly? I think I must enable CS5381 ADC using GPIO registers, but what are OXYGEN_REC_ROUTING, OXYGEN_ADC_MONITOR and OXYGEN_A_MONITOR_ROUTING for? Does MONITOR mean the same thing as in the user manual? If ROUTING registers are just switches for source of a signal to be recorded, is it enough to use OXYGEN_REC_B_ROUTE_I2S_ADC_2 for ST card to record from line-in/mic jack? 2) How do I control volume of sound while recording? Have Xonar Essence cards any controls for this? I cannot find it out looking at oxygen_mixer.c

2 1

[alsa-devel] [PATCH] ASoC: rt5645:Mark RT5645_TDM_CTRL_3 as readable
by yang.a.fang＠intel.com 25 Oct '14

25 Oct '14

From: "Fang, Yang A" <yang.a.fang(a)intel.com> amixer query fails due to it is not readable reigster Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/rt5645.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3fb83bf..d16331e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -139,6 +139,7 @@ static const struct reg_default rt5645_reg[] = { { 0x76, 0x000a }, { 0x77, 0x0c00 }, { 0x78, 0x0000 }, + { 0x79, 0x0123 }, { 0x80, 0x0000 }, { 0x81, 0x0000 }, { 0x82, 0x0000 }, @@ -334,6 +335,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_DMIC_CTRL2: case RT5645_TDM_CTRL_1: case RT5645_TDM_CTRL_2: + case RT5645_TDM_CTRL_3: case RT5645_GLB_CLK: case RT5645_PLL_CTRL1: case RT5645_PLL_CTRL2: -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ALSA: pcm: Avoid mmap warnings on x86
by Takashi Iwai 24 Oct '14

24 Oct '14

On x86, using dma_mmap_coherent() for the pages allocated via dma_alloc_coherent() results in a warning like: aplay:32536 map pfn RAM range req uncached-minus for [mem 0x21d500000-0x21d51ffff], got write-back Until the issue is addressed in the core side, take back to the old good way in PCM code only for x86. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a915ed07aee1..2f7ad10ee7c3 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3324,6 +3324,7 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, area->vm_end - area->vm_start, area->vm_page_prot); } #endif /* CONFIG_GENERIC_ALLOCATOR */ +#ifndef CONFIG_X86 /* for avoiding warnings arch/x86/mm/pat.c */ if (!substream->ops->page && substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) return dma_mmap_coherent(substream->dma_buffer.dev.dev, @@ -3331,6 +3332,7 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, substream->runtime->dma_area, substream->runtime->dma_addr, area->vm_end - area->vm_start); +#endif /* CONFIG_X86 */ /* mmap with fault handler */ area->vm_ops = &snd_pcm_vm_ops_data_fault; return 0; -- 2.1.2

1 0

Re: [alsa-devel] Default value change for restore default in initial state
by Takashi Iwai 24 Oct '14

24 Oct '14

At Fri, 24 Oct 2014 08:35:12 +0000, Kailang wrote: > > Hi Takashi, > > We have some changed for restore default for ALC282 and ALC283. Could you give a bit more information, what these updates actually fix? thanks, Takashi

1 0