June 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] idea: a reserve alsa plugin
by David Henningsson 17 Jul '13

17 Jul '13

Just had an idea which I'll write down here before I forget it again...and I'm not saying I'll implement this anytime soon either, but here goes: There is a device reserve protocol between PulseAudio and JACK2 - when JACK needs the sound card, it'll send a dbus message to PulseAudio and grab a name in D-Bus. However, there are plenty of applications who like to access ALSA directly, without going through JACK2 or PulseAudio. By making a "reserve" plugin, we could have this functionality for those apps too. In practice, if the app usually opens "plughw:0" or "hw:0", it could instead open "reserve:plughw:0" or "reserve:hw:0" to also reserve the device from PulseAudio usage while the device is open. Meanwhile, PulseAudio is free to use other audio devices (which is not the case when using e g pasuspender). How does that sound? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

6 12

[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Let the codec acquire its clock
by Fabio Estevam 15 Jul '13

15 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> On a mx6qsabrelite board the following error happens on probe: sgtl5000: probe of 0-000a failed with error -5 imx-sgtl5000 sound.13: ASoC: CODEC (null) not registered imx-sgtl5000 sound.13: snd_soc_register_card failed (-517) platform sound.13: Driver imx-sgtl5000 requests probe defer Prior to reading the codec ID we need to turn the SYS_MCLK clock, so let's enable the codec clock inside sgtl5000_i2c_probe(). Also remove the codec clock enable/disable functions from the machine driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Shawn, I haven't had a chance to convert mx28 to pass the codec clock via device tree yet. It looks like it is a bit trickier there, as it uses the saif clock. Given that mx28 audio playback is already broken in linux-next, this series does not make things worse on mx28 side. sound/soc/codecs/sgtl5000.c | 34 ++++++++++++++++++++++++++++++---- sound/soc/fsl/imx-sgtl5000.c | 30 +++++++----------------------- 2 files changed, 37 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c8f2afb..2e0227b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -114,6 +114,7 @@ struct sgtl5000_priv { struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; struct ldo_regulator *ldo; struct regmap *regmap; + struct clk *mclk; }; /* @@ -1522,16 +1523,28 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } + sgtl5000->mclk = devm_clk_get(&client->dev, NULL); + if (IS_ERR(sgtl5000->mclk)) { + ret = PTR_ERR(sgtl5000->mclk); + dev_err(&client->dev, "Failed to get mclock: %d\n", ret); + return ret; + } + + ret = clk_prepare_enable(sgtl5000->mclk); + if (ret) + return ret; + /* read chip information */ ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); if (ret) - return ret; + goto disable_clk; if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != SGTL5000_PARTID_PART_ID) { dev_err(&client->dev, "Device with ID register %x is not a sgtl5000\n", reg); - return -ENODEV; + ret = -ENODEV; + goto disable_clk; } rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; @@ -1542,17 +1555,30 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, /* Ensure sgtl5000 will start with sane register values */ ret = sgtl5000_fill_defaults(sgtl5000); if (ret) - return ret; + goto disable_clk; ret = snd_soc_register_codec(&client->dev, &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) + goto disable_clk; + + return 0; + +disable_clk: + clk_disable_unprepare(sgtl5000->mclk); return ret; } static int sgtl5000_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + struct sgtl5000_priv *sgtl5000; + sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), + GFP_KERNEL); + if (!sgtl5000) + return -ENOMEM; + snd_soc_unregister_codec(&client->dev); + clk_disable_unprepare(sgtl5000->mclk); return 0; } diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index a60aaa0..823151b 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -129,20 +129,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } data->codec_clk = clk_get(&codec_dev->dev, NULL); - if (IS_ERR(data->codec_clk)) { - /* assuming clock enabled by default */ - data->codec_clk = NULL; - ret = of_property_read_u32(codec_np, "clock-frequency", - &data->clk_frequency); - if (ret) { - dev_err(&codec_dev->dev, - "clock-frequency missing or invalid\n"); - goto fail; - } - } else { - data->clk_frequency = clk_get_rate(data->codec_clk); - clk_prepare_enable(data->codec_clk); - } + if (IS_ERR(data->codec_clk)) + goto fail; + + data->clk_frequency = clk_get_rate(data->codec_clk); data->dai.name = "HiFi"; data->dai.stream_name = "HiFi"; @@ -157,10 +147,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto clk_fail; + goto fail; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) - goto clk_fail; + goto fail; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; @@ -170,7 +160,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = snd_soc_register_card(&data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto clk_fail; + goto fail; } platform_set_drvdata(pdev, data); @@ -179,8 +169,6 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; -clk_fail: - clk_put(data->codec_clk); fail: if (ssi_np) of_node_put(ssi_np); @@ -194,10 +182,6 @@ static int imx_sgtl5000_remove(struct platform_device *pdev) { struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); - if (data->codec_clk) { - clk_disable_unprepare(data->codec_clk); - clk_put(data->codec_clk); - } snd_soc_unregister_card(&data->card); return 0; -- 1.8.1.2

3 6

[alsa-devel] [PATCH 0/5] ASoC: add simple-card DT support
by Kuninori Morimoto 12 Jul '13

12 Jul '13

Hi Mark, Simon These patches add DT support on simple-card driver. #1 patch adds new snd_soc_of_parse_daifmt() function on soc-core.c and #5 patch will use it. #2 - #5 patches are based on mark/topic/fsi branch >> Simon #2, #3 patches modify sh-arm platform settings. please check it Kuninori Morimoto (5): ASoC: add snd_soc_of_parse_daifmt() ASoC: simple-card: use struct device pointer for dev_xxx() ASoC: simple-card: add asoc_simple_dai_set for initializing ASoC: simple-card: remove pointless struct asoc_simple_dai_init_info ASoC: simple-card: add Device Tree support .../devicetree/bindings/sound/simple-card.txt | 53 ++++++++ arch/arm/mach-shmobile/board-ap4evb.c | 19 +-- arch/arm/mach-shmobile/board-armadillo800eva.c | 18 +-- arch/arm/mach-shmobile/board-kzm9g.c | 12 +- arch/arm/mach-shmobile/board-mackerel.c | 19 +-- arch/sh/boards/mach-ecovec24/setup.c | 11 +- arch/sh/boards/mach-se/7724/setup.c | 13 +- include/sound/simple_card.h | 11 +- include/sound/soc.h | 2 + sound/soc/generic/simple-card.c | 132 ++++++++++++++++---- sound/soc/soc-core.c | 115 +++++++++++++++++ 11 files changed, 307 insertions(+), 98 deletions(-) Best regards --- Kuninori Morimoto

6 110

[alsa-devel] Problem with MX27 ASoC audio
by Randolph Chung 11 Jul '13

11 Jul '13

I have a target that uses iMX27(L) + tlv320aic3x. The kernel tree is HEAD of linux-arm from about a week ago. Simple playback and recording (using aplay and arecord) work fine. However, I've run into a strange problem when both playback and capture streams are opened at the same time. Consider the attached test program. The capture device is setup with a period time of 10ms, and the program reads samples in a loop. The expected result is that I get a sample every 10ms. This works fine if the playback device is *not* opened (more specifically, if the hw_params are not set). However if the playback device is opened (#if 1 code) then I get samples every 20ms, which seems very wrong. I have tried using both DMA and FIQ ssi-pcm interfaces and both give the same strange result. With the FIQ driver, if I instrument the timer callback, the timer triggers every 10ms but there is only enough data every 20ms. The codec is being driven with a 13MHz MCLK (output from MX27) and the BCLK and WCLK come from the codec. They seem to have the correct rates. Any ideas? Can anybody else with MX27 hardware verify if my test program also fails on their target? thanks, randolph /* Test program */ #include <alsa/asoundlib.h> #include <stdio.h> #include <stdlib.h> #include <assert.h> #include <sys/time.h> int main(int argc, char **argv) { int err = -1; char buf[4096]; unsigned long period_size; snd_pcm_t *in_handle = NULL, *out_handle; snd_pcm_hw_params_t *hwparams; if (snd_pcm_open(&in_handle, "hw:0,0", SND_PCM_STREAM_CAPTURE, 0) < 0) { printf("Cannot open capture device\n"); return -1; } snd_pcm_hw_params_malloc(&hwparams); snd_pcm_hw_params_any(in_handle, hwparams); snd_pcm_hw_params_set_access(in_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(in_handle, hwparams, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(in_handle, hwparams, 1); snd_pcm_hw_params_set_rate(in_handle, hwparams, 8000, 0); snd_pcm_hw_params_set_period_time(in_handle, hwparams, 10000, 0); snd_pcm_hw_params_get_period_size(hwparams, &period_size, 0); snd_pcm_hw_params(in_handle, hwparams); snd_pcm_hw_params_free(hwparams); if (snd_pcm_open(&out_handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0) < 0) { printf("Cannot open playback device\n"); return -1; } #if 1 snd_pcm_hw_params_malloc(&hwparams); snd_pcm_hw_params_any(out_handle, hwparams); snd_pcm_hw_params_set_access(out_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(out_handle, hwparams, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(out_handle, hwparams, 1); snd_pcm_hw_params_set_rate(out_handle, hwparams, 8000, 0); snd_pcm_hw_params_set_period_time(out_handle, hwparams, 10000, 0); snd_pcm_hw_params(out_handle, hwparams); snd_pcm_hw_params_free(hwparams); #endif printf("period size = %u\n", period_size); printf("Config done, start loop\n"); while (1) { struct timeval tv; int ret = snd_pcm_readi(in_handle, buf, period_size); if (ret < 0) snd_pcm_prepare(in_handle); gettimeofday(&tv, NULL); printf("[%u.%06u] snd_pcm_readi() return %d\n", tv.tv_sec, tv.tv_usec, ret); } return 0; }

5 11

[alsa-devel] CA0110 analog mic capture flatline
by Ki'Sak 11 Jul '13

11 Jul '13

Good day alsa-devel, I have a "Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG" which I have been unable to capture any content from the analog mic jack. I am a Gentoo Linux user with the 3.8.13 kernel / alsa 1.0.26. This issue has been filed with the wrong devs over at ubuntu's bugtracker and opensuse's forum. Also, I found a report that analog capture was working on this card as recently as September 2009. Thank you for your time, Ki'Sak Note: I have already tried to report this issue, this is a resend without urls due to 2 weeks without a response or it reaching the mailing list.

2 3

[alsa-devel] [PATCH - oss plugin update revised 1/2] oss: Add support of 32-bit and other formats
by alex.d.wiggins＠gmail.com 08 Jul '13

08 Jul '13

From: Alex Wiggins <alex.d.wiggins(a)gmail.com> Added support of 32-bit and other audio formats via 3 changes: - Additional audio format definitions at the top of the file (if required); - Additional audio formats in oss_hw_params (to allow as input to pass to OSS); - Additional audio formats in oss_hw_constraint (to allow to report to ALSA). Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c index d43b44c..d5b5bc4 100644 --- a/oss/pcm_oss.c +++ b/oss/pcm_oss.c @@ -24,6 +24,32 @@ #include <alsa/pcm_external.h> #include <linux/soundcard.h> +/* definitions for additional OSS formats if required */ +#ifndef AFMT_S32_LE +#define AFMT_S32_LE 0x00001000 +#endif +#ifndef AFMT_S32_BE +#define AFMT_S32_BE 0x00002000 +#endif +#ifndef AFMT_S24_LE +#define AFMT_S24_LE 0x00008000 +#endif +#ifndef AFMT_S24_BE +#define AFMT_S24_BE 0x00010000 +#endif +#ifndef AFMT_S24_PACKED +#define AFMT_S24_PACKED 0x00040000 +#endif +#ifndef AFMT_FLOAT +#define AFMT_FLOAT 0x00004000 +#endif +#ifndef AFMT_SPDIF_RAW +#define AFMT_SPDIF_RAW 0x00020000 +#endif +#ifndef AFMT_VORBIS +#define AFMT_VORBIS 0x00000800 +#endif + typedef struct snd_pcm_oss { snd_pcm_ioplug_t io; char *device; @@ -165,6 +191,145 @@ static int oss_hw_params(snd_pcm_ioplug_t *io, case SND_PCM_FORMAT_S16_BE: oss->format = AFMT_S16_BE; break; + /* allow additional data formats (such as 32 bit) as input + to pass to OSS (set in OSS driver) */ + case SND_PCM_FORMAT_S32_LE: + oss->format = AFMT_S32_LE; + break; + case SND_PCM_FORMAT_S32_BE: + oss->format = AFMT_S32_BE; + break; + case SND_PCM_FORMAT_S8: + oss->format = AFMT_S8; + break; + case SND_PCM_FORMAT_U16_LE: + oss->format = AFMT_U16_LE; + break; + case SND_PCM_FORMAT_U16_BE: + oss->format = AFMT_U16_BE; + break; + case SND_PCM_FORMAT_S24_LE: + oss->format = AFMT_S24_LE; + break; + case SND_PCM_FORMAT_S24_BE: + oss->format = AFMT_S24_BE; + break; + case SND_PCM_FORMAT_MU_LAW: + oss->format = AFMT_MU_LAW; + break; + case SND_PCM_FORMAT_A_LAW: + oss->format = AFMT_A_LAW; + break; + case SND_PCM_FORMAT_IMA_ADPCM: + oss->format = AFMT_IMA_ADPCM; + break; + case SND_PCM_FORMAT_MPEG: + oss->format = AFMT_MPEG; + break; + case SND_PCM_FORMAT_S24_3LE: + oss->format = AFMT_S24_PACKED; + break; + case SND_PCM_FORMAT_FLOAT: + oss->format = AFMT_FLOAT; + break; + case SND_PCM_FORMAT_IEC958_SUBFRAME: + oss->format = AFMT_SPDIF_RAW; + break; + /* following formats need defining checking native endian: */ + #if 0 + case SND_PCM_FORMAT_S16: + oss->format = AFMT_S16_NE; + break; + case SND_PCM_FORMAT_U16: + oss->format = AFMT_U16_NE; + break; + case SND_PCM_FORMAT_S24: + oss->format = AFMT_S24_NE; + break; + case SND_PCM_FORMAT_S32: + oss->format = AFMT_S32_NE; + #endif + /* following formats found in ALSA but not in OSS: */ + #if 0 + case SND_PCM_FORMAT_U24_LE: + oss->format = AFMT_U24_LE; + break; + case SND_PCM_FORMAT_U24_BE: + oss->format = AFMT_U24_BE; + break; + case SND_PCM_FORMAT_U32_LE: + oss->format = AFMT_U32_LE; + break; + case SND_PCM_FORMAT_U32_BE: + oss->format = AFMT_U32_BE; + break; + case SND_PCM_FORMAT_FLOAT_LE: + oss->format = AFMT_FLOAT_LE; + break; + case SND_PCM_FORMAT_FLOAT_BE: + oss->format = AFMT_FLOAT_BE; + break; + case SND_PCM_FORMAT_FLOAT64_LE: + oss->format = AFMT_FLOAT64_LE; + break; + case SND_PCM_FORMAT_FLOAT64_BE: + oss->format = AFMT_FLOAT64_BE; + break; + case SND_PCM_FORMAT_IEC958_SUBFRAME_LE: + oss->format = AFMT_SPDIF_RAW_LE; + break; + case SND_PCM_FORMAT_IEC958_SUBFRAME_BE: + oss->format = AFMT_SPDIF_RAW_BE; + break; + case SND_PCM_FORMAT_GSM: + oss->format = AFMT_GSM; + break; + case SND_PCM_FORMAT_SPECIAL: + oss->format = AFMT_SPECIAL; + break; + case SND_PCM_FORMAT_S24_3BE: + oss->format = AFMT_S24_PACKED_BE; + break; + case SND_PCM_FORMAT_U24_3LE: + oss->format = AFMT_U24_PACKED; + break; + case SND_PCM_FORMAT_U24_3BE: + oss->format = AFMT_U24_PACKED_BE; + break; + case SND_PCM_FORMAT_S20_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_S20_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_U20_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_U20_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_S18_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_S18_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_U18_3LE: + oss->format = ; + break; + case SND_PCM_FORMAT_U18_3BE: + oss->format = ; + break; + case SND_PCM_FORMAT_U24: + oss->format = AFMT_U24_NE; + break; + case SND_PCM_FORMAT_U32: + oss->format = AFMT_U32_NE; + break; + case SND_PCM_FORMAT_FLOAT64: + oss->format = AFMT_FLOAT64; + break; + #endif default: fprintf(stderr, "*** OSS: unsupported format %s\n", snd_pcm_format_name(io->format)); return -EINVAL; @@ -239,7 +404,7 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) SND_PCM_ACCESS_MMAP_INTERLEAVED }; unsigned int nformats; - unsigned int format[5]; + unsigned int format[20]; unsigned int nchannels; unsigned int channel[6]; /* period and buffer bytes must be power of two */ @@ -273,6 +438,40 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) format[nformats++] = SND_PCM_FORMAT_S16_BE; if (tmp & AFMT_MU_LAW) format[nformats++] = SND_PCM_FORMAT_MU_LAW; + /* allow additional data formats (such as 32 bit) + to be reported from OSS to ALSA below */ + if (tmp & AFMT_S32_LE) + format[nformats++] = SND_PCM_FORMAT_S32_LE; + if (tmp & AFMT_S32_BE) + format[nformats++] = SND_PCM_FORMAT_S32_BE; + if (tmp & AFMT_A_LAW) + format[nformats++] = SND_PCM_FORMAT_A_LAW; + if (tmp & AFMT_IMA_ADPCM) + format[nformats++] = SND_PCM_FORMAT_IMA_ADPCM; + if (tmp & AFMT_S8) + format[nformats++] = SND_PCM_FORMAT_S8; + if (tmp & AFMT_U16_LE) + format[nformats++] = SND_PCM_FORMAT_U16_LE; + if (tmp & AFMT_U16_BE) + format[nformats++] = SND_PCM_FORMAT_U16_BE ; + if (tmp & AFMT_MPEG) + format[nformats++] = SND_PCM_FORMAT_MPEG; + if (tmp & AFMT_FLOAT) + format[nformats++] = SND_PCM_FORMAT_FLOAT; + if (tmp & AFMT_S24_LE) + format[nformats++] = SND_PCM_FORMAT_S24_LE; + if (tmp & AFMT_S24_BE) + format[nformats++] = SND_PCM_FORMAT_S24_BE; + if (tmp & AFMT_SPDIF_RAW) + format[nformats++] = SND_PCM_FORMAT_IEC958_SUBFRAME; + if (tmp & AFMT_S24_PACKED) + format[nformats++] = SND_PCM_FORMAT_S24_3LE; + /* following formats found in OSS but not in ALSA */ + if (tmp & AFMT_AC3) + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; + if (tmp & AFMT_VORBIS) + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; + /* assume signed 16 bit (native CPU endian) if not found above */ if (! nformats) format[nformats++] = SND_PCM_FORMAT_S16; if ((err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, -- 1.8.3.1

4 5

[alsa-devel] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 08 Jul '13

08 Jul '13

Hi, This series adds DT support for phycore-ac97. Beside ac97 support, the series adds imx-pcm-fiq and generic DMA bindings to fsl-ssi. Version 9 contains some style fixes. Regards, Markus Changes in v9: - Style and compile fixes. - Removed "ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97". This patch is not necessary with the seperated phycore-ac97-dt driver. Changes in v8: - Fix build issues in phycore-ac97. Changes in v7: - Only one soc_ac97_ops struct for both ssi drivers (fsl-ssi and imx-ssi) to make it possible to compile a kernel with both drivers. - Other small bug/style fixes described in the changelogs of the patches. Changes in v6: - Rebased onto 3.10-rc3 with Shawn's pcm cleanup patches - Setup of reset functions moved to phycore-ac97.c Changes in v5: - Pass of_node via parent device to imx-pcm-dma. Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (10): ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl: Move soc_ac97_ops from imx-ssi to fsl_ssi ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ASoC: fsl-ssi: imx ac97 support ARM: imx: Export ac97 reset functions ASoC: Add phycore-ac97-dt driver ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 + .../bindings/sound/phytec,phycore-ac97.txt | 14 + arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 17 +- sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_ssi.c | 451 +++++++++++++++++---- sound/soc/fsl/fsl_ssi.h | 7 + sound/soc/fsl/imx-pcm-dma.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 18 +- sound/soc/fsl/imx-pcm.h | 15 +- sound/soc/fsl/imx-ssi.c | 14 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97-dt.c | 242 +++++++++++ sound/soc/fsl/phycore-ac97.c | 13 +- 15 files changed, 704 insertions(+), 115 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt create mode 100644 sound/soc/fsl/phycore-ac97-dt.c

4 25

[alsa-devel] [RFC] [patch 1/1] imx-sgtl5000: return E_PROBE_DEFER if ssi/codec not found
by Arnaud Patard 04 Jul '13

04 Jul '13

If the ssi or codec drivers are not loaded (for instance, because spi or i2c bus drivers are not loaded), returning -EINVAL will for people to unload and then reload the module to get sound working. Returning E_PROBE_DEFER will mitigate this. Signed-off-by: Arnaud Patard <arnaud.patard(a)rtp-net.org> Index: linux/sound/soc/fsl/imx-sgtl5000.c =================================================================== --- linux.orig/sound/soc/fsl/imx-sgtl5000.c 2013-05-16 09:33:01.000000000 +0200 +++ linux/sound/soc/fsl/imx-sgtl5000.c 2013-05-16 09:35:14.000000000 +0200 @@ -113,13 +113,13 @@ static int imx_sgtl5000_probe(struct pla 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; + ret = -EPROBE_DEFER; goto fail; } codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EINVAL; + return -EPROBE_DEFER; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);

3 4

[alsa-devel] [PATCH] Add -lm to test programs to fix build failure.
by Jordi Mallach 04 Jul '13

04 Jul '13

Add -lm to test programs that use sin(), to fix a build failure with recent binutils. Signed-off-by: Jordi Mallach <jordi(a)debian.org> --- test/Makefile.am | 2 ++ 1 file changed, 2 insertions(+) diff --git a/test/Makefile.am b/test/Makefile.am index 8705402..c1896f8 100644 --- a/test/Makefile.am +++ b/test/Makefile.am @@ -7,8 +7,10 @@ check_PROGRAMS=control pcm pcm_min latency seq \ control_LDADD=../src/libasound.la pcm_LDADD=../src/libasound.la +pcm_LDFLAGS= -lm pcm_min_LDADD=../src/libasound.la latency_LDADD=../src/libasound.la +latency_LDFLAGS= -lm seq_LDADD=../src/libasound.la playmidi1_LDADD=../src/libasound.la timer_LDADD=../src/libasound.la -- 1.8.3.1

1 1

[alsa-devel] [PATCH] ASoC: Add PCM1681 codec driver.
by Marek Belisko 01 Jul '13

01 Jul '13

PCM1681 can be controlled via I2C, SPI or in bootstrap mode. This code add support only for I2C mode. Signed-off-by: Marek Belisko <marek.belisko(a)streamunlimited.com> --- .../devicetree/bindings/sound/ti,pcm1681.txt | 15 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm1681.c | 328 ++++++++++++++++++++ 4 files changed, 349 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 sound/soc/codecs/pcm1681.c diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt new file mode 100644 index 0000000..4df1718 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt @@ -0,0 +1,15 @@ +Texas Instruments PCM1681 8-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,pcm1681". + - reg: The i2c address. Should contain <0x4c>. + +Examples: + + i2c_bus { + pcm1681@4c { + compatible = "ti,pcm1681"; + reg = <0x4c>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fb..e2daf82 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC + select SND_SOC_PCM1681 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C @@ -292,6 +293,9 @@ config SND_SOC_MAX9850 config SND_SOC_HDMI_CODEC tristate +config SND_SOC_PCM1681 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd806..4a068d2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -42,6 +42,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -171,6 +172,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c new file mode 100644 index 0000000..6122fd1 --- /dev/null +++ b/sound/soc/codecs/pcm1681.c @@ -0,0 +1,328 @@ +/* + * PCM1681 ASoC codec driver + * + * Copyright (c) StreamUnlimited GmbH 2013 + * Marek Belisko <marek.belisko(a)streamunlimited.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. + * + * 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. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define PCM1681_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +#define PCM1681_PCM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1681_SOFT_MUTE_ALL 0xff +#define PCM1681_DEEMPH_RATE_MASK 0x18 +#define PCM1681_DEEMPH_MASK 0x01 + +#define PCM1681_ATT_CONTROL(X) (X <= 6 ? X : X + 9) /* Attenuation level */ +#define PCM1681_SOFT_MUTE 0x07 /* Soft mute control register */ +#define PCM1681_DAC_CONTROL 0x08 /* DAC operation control */ +#define PCM1681_FMT_CONTROL 0x09 /* Audio interface data format */ +#define PCM1681_DEEMPH_CONTROL 0x0a /* De-emphasis control */ +#define PCM1681_ZERO_DETECT_STATUS 0x0e /* Zero detect status reg */ + +static const struct reg_default pcm1681_reg_defaults[] = { + { 0x01, 0xff }, + { 0x02, 0xff }, + { 0x03, 0xff }, + { 0x04, 0xff }, + { 0x05, 0xff }, + { 0x06, 0xff }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x06 }, + { 0x0A, 0x00 }, + { 0x0B, 0xff }, + { 0x0C, 0x0f }, + { 0x0D, 0x00 }, + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, +}; + +static bool pcm1681_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x00) || (reg == 0x0f)); +} + +static bool pcm1681_writeable_reg(struct device *dev, unsigned register reg) +{ + return pcm1681_accessible_reg(dev, reg) && + (reg != PCM1681_ZERO_DETECT_STATUS); +} + +struct pcm1681_private { + struct regmap *regmap; + unsigned int format; + /* Current deemphasis status */ + unsigned int deemph; + /* Current rate for deemphasis control */ + unsigned int rate; +}; + +static int pcm1681_deemph[] = { 44100, 48000, 32000 }; + +static int pcm1681_set_deemph(struct snd_soc_codec *codec) +{ + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int i = 0, val = -1, ret; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++) + if (pcm1681_deemph[i] == priv->rate) + val = i; + + /* enable choosen frequency */ + if (val != -1) + regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_RATE_MASK, val); + + /* enable/disable deemphasis functionality */ + ret = regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_MASK, priv->deemph); + + return ret; +} + +static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return pcm1681_set_deemph(codec); +} + +static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The PCM1681 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + priv->format = format; + + return 0; +} + +static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int ret, val = 0; + + if (mute) + val = PCM1681_SOFT_MUTE_ALL; + + ret = regmap_write(priv->regmap, PCM1681_SOFT_MUTE, val); + if (ret < 0) + return ret; + + return 0; +} + +static int pcm1681_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) + val = 0x00; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x03; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x04; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x05; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + regmap_update_bits(priv->regmap, PCM1681_FMT_CONTROL, 0x0f, val); + + return 0; +} + +static const struct snd_soc_dai_ops pcm1681_dai_ops = { + .set_fmt = pcm1681_set_dai_fmt, + .hw_params = pcm1681_hw_params, + .digital_mute = pcm1681_digital_mute, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); + +static const struct snd_kcontrol_new pcm1681_controls[] = { + SOC_DOUBLE_R_TLV("PCM1681 Channel 1/2 Playback Volume", + PCM1681_ATT_CONTROL(1), PCM1681_ATT_CONTROL(2), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("PCM1681 Channel 3/4 Playback Volume", + PCM1681_ATT_CONTROL(3), PCM1681_ATT_CONTROL(4), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("PCM1681 Channel 5/6 Playback Volume", + PCM1681_ATT_CONTROL(5), PCM1681_ATT_CONTROL(6), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("PCM1681 Channel 7/8 Playback Volume", + PCM1681_ATT_CONTROL(7), PCM1681_ATT_CONTROL(8), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_SINGLE_BOOL_EXT("PCM1681 De-emphasis Switch", 0, + pcm1681_get_deemph, pcm1681_put_deemph), +}; + +struct snd_soc_dai_driver pcm1681_dai = { + .name = "pcm1681-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = PCM1681_PCM_RATES, + .formats = PCM1681_PCM_FORMATS, + }, + .ops = &pcm1681_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcm1681_dt_ids[] = { + { .compatible = "ti,pcm1681", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); +#endif + +static const struct regmap_config pcm1681_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(pcm1681_reg_defaults), + .reg_defaults = pcm1681_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), + .writeable_reg = pcm1681_writeable_reg, + .readable_reg = pcm1681_accessible_reg, +}; + +static int pcm1681_probe(struct snd_soc_codec *codec) +{ + return 0; +} + +static int pcm1681_remove(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { + .probe = pcm1681_probe, + .remove = pcm1681_remove, + .controls = pcm1681_controls, + .num_controls = ARRAY_SIZE(pcm1681_controls), +}; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct i2c_device_id pcm1681_i2c_id[] = { + {"pcm1681", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id); + +static int pcm1681_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + int ret; + struct pcm1681_private *priv; + + priv = devm_kzalloc(&client->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(client, &pcm1681_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&client->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(client, priv); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_pcm1681, + &pcm1681_dai, 1); +} + +static int pcm1681_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver pcm1681_i2c_driver = { + .driver = { + .name = "pcm1681", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(pcm1681_dt_ids), + }, + .id_table = pcm1681_i2c_id, + .probe = pcm1681_i2c_probe, + .remove = pcm1681_i2c_remove, +}; + +module_i2c_driver(pcm1681_i2c_driver); + +#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */ + +MODULE_DESCRIPTION("Texas Instruments PCM1681 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Marek Belisko <marek.belisko(a)streamunlimited.com>"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

4 4

[alsa-devel] [PATCH 1/2] ASoC: adau1701: more direct regmap usage
by Daniel Mack 30 Jun '13

30 Jun '13

Replace calls to snd_soc_update_bits() with regmap_update_bits(). Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/adau1701.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index ac8b2c4..ce56065 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -334,7 +334,7 @@ static int adau1701_set_capture_pcm_format(struct snd_soc_codec *codec, mask |= ADAU1701_SEROCTL_MSB_DEALY_MASK; } - snd_soc_update_bits(codec, ADAU1701_SEROCTL, mask, val); + regmap_update_bits(adau1701->regmap, ADAU1701_SEROCTL, mask, val); return 0; } @@ -362,7 +362,7 @@ static int adau1701_set_playback_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_SERICTL, + regmap_update_bits(adau1701->regmap, ADAU1701_SERICTL, ADAU1701_SERICTL_MODE_MASK, val); return 0; @@ -403,7 +403,7 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_DSPCTRL, + regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_SR_MASK, val); format = params_format(params); @@ -489,6 +489,7 @@ static int adau1701_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { unsigned int mask = ADAU1701_AUXNPOW_VBPD | ADAU1701_AUXNPOW_VRPD; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); switch (level) { case SND_SOC_BIAS_ON: @@ -497,11 +498,13 @@ static int adau1701_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* Enable VREF and VREF buffer */ - snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, 0x00); + regmap_update_bits(adau1701->regmap, + ADAU1701_AUXNPOW, mask, 0x00); break; case SND_SOC_BIAS_OFF: /* Disable VREF and VREF buffer */ - snd_soc_update_bits(codec, ADAU1701_AUXNPOW, mask, mask); + regmap_update_bits(adau1701->regmap, + ADAU1701_AUXNPOW, mask, mask); break; } @@ -513,6 +516,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; unsigned int mask = ADAU1701_DSPCTRL_DAM; + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); unsigned int val; if (mute) @@ -520,7 +524,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute) else val = mask; - snd_soc_update_bits(codec, ADAU1701_DSPCTRL, mask, val); + regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, mask, val); return 0; } @@ -542,7 +546,8 @@ static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id, return -EINVAL; } - snd_soc_update_bits(codec, ADAU1701_OSCIPOW, ADAU1701_OSCIPOW_OPD, val); + regmap_update_bits(adau1701->regmap, ADAU1701_OSCIPOW, + ADAU1701_OSCIPOW_OPD, val); adau1701->sysclk = freq; return 0; -- 1.8.1.4

3 4

[alsa-devel] [PATCH - oss plugin update revised 2/2] oss: Add user variables to set period and buffer config
by alex.d.wiggins＠gmail.com 30 Jun '13

30 Jun '13

From: Alex Wiggins <alex.d.wiggins(a)gmail.com> Added 3 user variables to control the OSS period and buffer config: - period_size (integer): if not set then queries OSS for period size (by using SNDCTL_DSP_GETBLKSIZE), if positive then sets the OSS period size (SND_PCM_IOPLUG_HW_PERIOD_BYTES), if negative then allows any power of 2 period size. - periods (integer): sets the number of OSS periods (SND_PCM_IOPLUG_HW_PERIODS); - buffer_size (integer): sets the OSS buffer size (SND_PCM_IOPLUG_HW_BUFFER_BYTES). Note that ALSA performs sanity checks to prevent conflicts. Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c index d5b5bc4..a92c3e9 100644 --- a/oss/pcm_oss.c +++ b/oss/pcm_oss.c @@ -60,6 +60,12 @@ typedef struct snd_pcm_oss { unsigned int period_shift; unsigned int periods; unsigned int frame_bytes; + /* user variables set using ALSA config files */ + /* included here as this snd_pcm_oss_t struct + is stored in io->private_data */ + long int user_period_size; + long int user_periods; + long int user_buffer_size; } snd_pcm_oss_t; static snd_pcm_sframes_t oss_write(snd_pcm_ioplug_t *io, @@ -408,6 +414,7 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) unsigned int nchannels; unsigned int channel[6]; /* period and buffer bytes must be power of two */ + /* minimum: 2^8 = 256; maximum: 2^23 = 8388608 */ static const unsigned int bytes_list[] = { 1U<<8, 1U<<9, 1U<<10, 1U<<11, 1U<<12, 1U<<13, 1U<<14, 1U<<15, 1U<<16, 1U<<17, 1U<<18, 1U<<19, 1U<<20, 1U<<21, 1U<<22, 1U<<23 @@ -500,20 +507,97 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) if (err < 0) return err; - /* period size (in power of two) */ - err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, - ARRAY_SIZE(bytes_list), bytes_list); - if (err < 0) - return err; - /* periods */ - err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIODS, 2, 1024); - if (err < 0) - return err; - /* buffer size (in power of two) */ - err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_BUFFER_BYTES, - ARRAY_SIZE(bytes_list), bytes_list); - if (err < 0) - return err; + /* period and buffer sizes: default behaviour below is to + query period size from OSS */ + + /* consider user set paramaters + (period_size; periods; buffer_size) */ + + /* note: ALSA performs sanity checks + to prevent conflicting parameters */ + + /* period_size */ + if (oss->user_period_size < 0) { + /* period_size set to negative value: + allow any power of 2 period size */ + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, + ARRAY_SIZE(bytes_list), bytes_list); + if (err < 0) + return err; + } else { + if (oss->user_period_size > 0) { + /* period_size set to positive value: + use user set period size */ + const unsigned int user_period_size + = oss->user_period_size; + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, 1, + &user_period_size); + } else { + /* query OSS for period size (called fragment size) */ + unsigned int fragsize; + if (ioctl(oss->fd, SNDCTL_DSP_GETBLKSIZE, &fragsize) + < 0) + fprintf(stderr, +"*** alsa-plugins-oss: error querying SNDCTL_DSP_GETBLKSIZE.\n"); + + /* note: second query of SNDCTL_DSP_GETBLKSIZE + can give different result + (i.e. 4096 instead of 2048) */ + /* in oss_audio_core.c getblksize function, + 4096 is the default return value: + so second time probably not reliable */ + + /* period size (in power of two) */ + unsigned int x = fragsize; + /* check if fragsize is a power of two */ + if ((x != 0) && ((x & (~x + 1)) == x)) + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, + 1, &fragsize); + else + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIOD_BYTES, + ARRAY_SIZE(bytes_list), bytes_list); + if (err < 0) + return err; + } + } + + /* periods (number of periods per buffer) */ + if (oss->user_periods > 0) { + /* force number of periods to user set periods */ + const unsigned int user_periods = oss->user_periods; + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_PERIODS, 1, &user_periods); + if (err < 0) + return err; + } else { + /* allow any number of periods between 2 and 1024 inclusive + (ALSA can choose) */ + err = snd_pcm_ioplug_set_param_minmax(io, + SND_PCM_IOPLUG_HW_PERIODS, 2, 1024); + if (err < 0) + return err; + } + + /* buffer_size */ + if (oss->user_buffer_size > 0) { + /* force buffer size to user set buffer_size */ + const unsigned int user_buffer_size = oss->user_buffer_size; + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_BUFFER_BYTES, 1, &user_buffer_size); + if (err < 0) + return err; + } else { + /* buffer size (in power of two) */ + err = snd_pcm_ioplug_set_param_list(io, + SND_PCM_IOPLUG_HW_BUFFER_BYTES, + ARRAY_SIZE(bytes_list), bytes_list); + if (err < 0) + return err; + } return 0; } @@ -558,6 +642,10 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) const char *device = "/dev/dsp"; int err; snd_pcm_oss_t *oss; + /* default values for user variables*/ + long int user_period_size = 0; + long int user_periods = 0; + long int user_buffer_size = 0; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); @@ -573,6 +661,30 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) } continue; } + + /* user variables */ + if (strcmp(id, "period_size") == 0) { + if (snd_config_get_integer(n, &user_period_size) < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + continue; + } + if (strcmp(id, "periods") == 0) { + if (snd_config_get_integer(n, &user_periods) < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + continue; + } + if (strcmp(id, "buffer_size") == 0) { + if (snd_config_get_integer(n, &user_buffer_size) < 0) { + SNDERR("Invalid type for %s", id); + return -EINVAL; + } + continue; + } + SNDERR("Unknown field %s", id); return -EINVAL; } @@ -597,6 +709,11 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) goto error; } + /* save user variables from ALSA config files */ + oss->user_period_size = user_period_size; + oss->user_periods = user_periods; + oss->user_buffer_size = user_buffer_size; + oss->io.version = SND_PCM_IOPLUG_VERSION; oss->io.name = "ALSA <-> OSS PCM I/O Plugin"; oss->io.poll_fd = oss->fd; -- 1.8.3.1

1 0

Re: [alsa-devel] [PATCH 0/2] ASoc: ac97: fix multi-platform build failures
by Mark Brown 29 Jun '13

29 Jun '13

On Fri, Jun 28, 2013 at 11:17:47AM -0700, Kevin Hilman wrote: > Hi Mark, > > When building next-20130628, I noticed a few build errors for > cm_x300_defconfig that were introduced by the new multi-platform AC'97 > support. Applied both, thanks.

1 0

Re: [alsa-devel] [PATCH 1/2] ASoC: tegra20-ac97: Remove duplicate error message
by Mark Brown 29 Jun '13

29 Jun '13

On Fri, Jun 28, 2013 at 09:53:24PM +0200, Thierry Reding wrote: > devm_ioremap_resource() already outputs an error message when any of the > operations it performs fails, so the duplicate in the caller can be > removed. Applied both, thanks.

1 0

[alsa-devel] ASoC updates for v3.11
by Mark Brown 28 Jun '13

28 Jun '13

The following changes since commit 9e895ace5d82df8929b16f58e9f515f6d54ab82d: Linux 3.10-rc7 (2013-06-22 09:47:31 -1000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-2 for you to fetch changes up to 27516080b21cbcb936440d2a3171867860b9a881: Merge remote-tracking branch 'asoc/topic/x86' into asoc-next (2013-06-28 12:17:07 +0100) The diffstat is a bit odd I think due to the synthetic tree I generated in an effort to cut out the fixes from v3.10-rc7 from it :/ ---------------------------------------------------------------- ASoC: More updates for v3.11 Some more fixes and enhancements, and also a bunch of refectoring for AC'97 support which enables more than one AC'97 controller driver to be built in. ---------------------------------------------------------------- Daniel Mack (10): ASoC: tas5086: open-code I2C transfer routines ASoC: tas5086: add more register defines ASoC: tas5086: add DAPM mux controls ASoC: tas5086: add support for pwm start mode config ASoC: adau1701: move firmware download to adau1701_reset() ASoC: adau1701: allow configuration of PLL mode pins ASoC: adau1701: switch to direct regmap API usage ASoC: adau1701: add support for pin muxing ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size ASoC: tas5086: fix Mid-Z implementation Lars-Peter Clausen (11): ASoC: tlv320aix3x: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm8400: Use SOC_SINGLE_EXT_TLV() instead of open-coding it ASoC: wm8903: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm8904: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm8990: Use SOC_SINGLE_EXT_TLV() instead of open-coding it ASoC: wm8991: Use SOC_SINGLE_EXT_TLV() instead of open-coding it ASoC: wm8994: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm8995: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm_hubs: Use SOC_SINGLE_EXT() instead of open-coding it ASoC: wm_adsp: Use SND_SOC_DAPM_PGA_E() instead of open-coding it ASoC: 88pm860x: Use SND_SOC_DAPM_PGA_E() instead of open-coding it Mark Brown (25): ASoC: stac9766: Remove version number ASoC: wm9705: Remove noisy print on boot ASoC: samsung-ac97: Use devm_clk_get() ASoC: samsung-ac97: Convert to devm_ioremap_resource() ASoC: psc-ac97: Use devm_ioremap_resource() ASoC: psc-ac97: Convert to module_platform_driver() ASoC: ac97c: Use module_platform_driver() ASoC: bf5xx-ac97: Convert to devm_gpio_request_one() ASoC: ep93xx: Remove redundant dev_set_drvdata() calls ASoC: nuc900-ac97: Convert to use devm_ APIs ASoC: tegra20-ac97: Convert to devm_clk_get() ASoC: tegra20-ac97: Convert to devm_ioremap_resource() ASoC: tegra-ac97: Do common and clock init prior to component registration ASoC: txx9aclc_ac97: Convert to devm_ioremap_resource() ASoC: wm9705: Remove noisy print on boot Merge branch 'topic/stac9766' of git://git.kernel.org/.../broonie/sound into asoc-ac97 Merge branch 'topic/wm9705' of git://git.kernel.org/.../broonie/sound into asoc-ac97 ASoC: ac97: Support multi-platform AC'97 ASoC: mfld: Remove unused variable Merge remote-tracking branch 'asoc/topic/adau1701' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/ext' into asoc-next Merge remote-tracking branch 'asoc/topic/tas5086' into asoc-next Merge remote-tracking branch 'asoc/topic/twl6040' into asoc-next Merge remote-tracking branch 'asoc/topic/x86' into asoc-next Peter Ujfalusi (4): ASoC: twl6040: Drop using devm_request_threaded_irq() mfd: twl6040: Update register bit definitions ASoC: twl6040: Assign id for each DAI ASoC: twl6040: Add digital mute support Wei Yongjun (1): ASoC: mid-x86: Convert to use devm_* APIs .../devicetree/bindings/sound/adi,adau1701.txt | 35 ++ .../devicetree/bindings/sound/ti,tas5086.txt | 11 + include/linux/mfd/twl6040.h | 7 + include/sound/soc.h | 4 +- sound/soc/au1x/ac97c.c | 21 +- sound/soc/au1x/psc-ac97.c | 33 +- sound/soc/blackfin/bf5xx-ac97.c | 29 +- sound/soc/cirrus/ep93xx-ac97.c | 12 +- sound/soc/codecs/88pm860x-codec.c | 6 +- sound/soc/codecs/ac97.c | 7 +- sound/soc/codecs/ad1980.c | 12 +- sound/soc/codecs/adau1701.c | 302 ++++++++++++-- sound/soc/codecs/stac9766.c | 26 +- sound/soc/codecs/tas5086.c | 330 ++++++++++++++- sound/soc/codecs/tlv320aic3x.c | 6 +- sound/soc/codecs/twl6040.c | 109 ++++- sound/soc/codecs/wm8400.c | 9 +- sound/soc/codecs/wm8903.c | 6 +- sound/soc/codecs/wm8904.c | 9 +- sound/soc/codecs/wm8990.c | 11 +- sound/soc/codecs/wm8991.h | 9 +- sound/soc/codecs/wm8994.c | 12 +- sound/soc/codecs/wm8995.h | 7 +- sound/soc/codecs/wm9705.c | 16 +- sound/soc/codecs/wm9712.c | 18 +- sound/soc/codecs/wm9713.c | 18 +- sound/soc/codecs/wm_adsp.c | 459 ++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 13 +- sound/soc/codecs/wm_hubs.c | 6 +- sound/soc/fsl/imx-ssi.c | 11 +- sound/soc/fsl/mpc5200_psc_ac97.c | 10 +- sound/soc/mid-x86/mfld_machine.c | 32 +- sound/soc/nuc900/nuc900-ac97.c | 60 +-- sound/soc/pxa/pxa2xx-ac97.c | 8 +- sound/soc/samsung/ac97.c | 42 +- sound/soc/sh/hac.c | 8 +- sound/soc/soc-core.c | 16 + sound/soc/tegra/tegra20_ac97.c | 66 ++- sound/soc/txx9/txx9aclc-ac97.c | 17 +- 39 files changed, 1447 insertions(+), 366 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1701.txt

2 1

[alsa-devel] [PATCH 1/2] ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size
by Daniel Mack 28 Jun '13

28 Jun '13

The TAS5086_CLOCK_CONTROL also has a size of 1 byte. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/tas5086.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index fec9a28..84c3f27 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -130,7 +130,7 @@ static const struct reg_default tas5086_reg_defaults[] = { static int tas5086_register_size(struct device *dev, unsigned int reg) { switch (reg) { - case TAS5086_DEV_ID ... TAS5086_BKNDERR: + case TAS5086_CLOCK_CONTROL ... TAS5086_BKNDERR: return 1; case TAS5086_INPUT_MUX: case TAS5086_PWM_OUTPUT_MUX: -- 1.8.1.4

2 2

[alsa-devel] [PATCH] ASoC: mfld: Remove unused variable
by Mark Brown 28 Jun '13

28 Jun '13

From: Mark Brown <broonie(a)linaro.org> Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/mid-x86/mfld_machine.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index aec29a8..ee36384 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -412,8 +412,6 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) static int snd_mfld_mc_remove(struct platform_device *pdev) { - struct mfld_mc_private *mc_drv_ctx = platform_get_drvdata(pdev); - pr_debug("snd_mfld_mc_remove called\n"); snd_soc_unregister_card(&snd_soc_card_mfld); return 0; -- 1.8.3.1

1 0

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix the max length of control name in generic parser
by Takashi Iwai 28 Jun '13

28 Jun '13

add_control_with_pfx() in hda_generic.c assumes a shorter name string for the control element, and this resulted in the truncation of the long but valid string like "Headphone Surround Switch" in the middle. This patch aligns the max size to the actual limit of snd_ctl_elem_id, 44. Cc: <stable(a)vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 1485d87..6460fc5 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -854,7 +854,7 @@ static int add_control_with_pfx(struct hda_gen_spec *spec, int type, const char *pfx, const char *dir, const char *sfx, int cidx, unsigned long val) { - char name[32]; + char name[44]; snprintf(name, sizeof(name), "%s %s %s", pfx, dir, sfx); if (!add_control(spec, type, name, cidx, val)) return -ENOMEM; -- 1.8.3.1

1 1

[alsa-devel] [PATCH] ALSA: hda - Guess what, it's two more Dell headset mic quirks
by David Henningsson 28 Jun '13

28 Jun '13

Add two more machines that need quirks for headset mics to work. Tested-by: Shawn Wang <shawn.wang(a)canonical.com> BugLink: https://bugs.launchpad.net/bugs/1195636 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d7041d..86c2406 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3506,6 +3506,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05fb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Yet another Dell headset mic quirk
by David Henningsson 28 Jun '13

28 Jun '13

This quirk is needed for the headset mic to work on this Dell machine. BugLink: https://bugs.launchpad.net/bugs/1195597 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ae12111..6d7041d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3509,6 +3509,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- 1.7.9.5

2 1

[alsa-devel] C-Media 6632A unsupported (asus xonar u7)
by Julius von kohout 28 Jun '13

28 Jun '13

Hi ive got a new usb soundcard with the brand new C-Media 6632A chip and want to help with debugging. xonar u7 in usb 2.0 mode C-Media 6632A Ubuntu 13.04 julius@R850:~$ alsactl init 1 Found hardware: "USB-Audio" "USB Mixer" "USB1043:857c" "" "" Hardware is initialized using a generic method julius@R850:~$ alsamixer -c 1 cannot load mixer controls: Broken pipe alsa hardware volume (numid=2,iface=MIXER,name='Speaker Playback Volume') level on start is way to low and can only be adjusted via QasHctl. The volume knob on soundcard works for pulseaudio volume. spdif can only be used via direct alsa access without pulseaudio. alsa-info script results http://pastebin.ca/2409660 julius@R850:~$ amixer -c 1 controls numid=7,iface=MIXER,name='PCM Capture Source' numid=5,iface=MIXER,name='PCM Capture Switch' numid=6,iface=MIXER,name='PCM Capture Switch',index=1 numid=4,iface=MIXER,name='IEC958 In Playback Switch' numid=3,iface=MIXER,name='Input Gain Pad Control' numid=1,iface=MIXER,name='Speaker Playback Switch' numid=2,iface=MIXER,name='Speaker Playback Volume' [ 8219.399637] usb 2-1.2.3: new high-speed USB device number 22 using ehci-pci [ 8219.513315] usb 2-1.2.3: New USB device found, idVendor=1043, idProduct=857c [ 8219.513327] usb 2-1.2.3: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 8219.513333] usb 2-1.2.3: Product: Xonar U7 [ 8219.513338] usb 2-1.2.3: Manufacturer: ASUS [ 8219.609328] input: ASUS Xonar U7 as /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1.2/2-1.2.3/2-1.2.3:1.4/input/input29 [ 8219.609743] hid-generic 0003:1043:857C.0016: input,hidraw4: USB HID v1.00 Device [ASUS Xonar U7] on usb-0000:00:1d.0-1.2.3/input4 [ 8219.676164] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.697280] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.718770] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.738130] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.757999] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.779861] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.800222] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8219.823964] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8520.766440] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 [ 8539.332352] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 julius@R850:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k3.8.0-25-generic. julius@R850:~$ cat /proc/asound/cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xc4820000 irq 50 1 [U7 ]: USB-Audio - Xonar U7 ASUS Xonar U7 at usb-0000:00:1d.0-1.2.3, high speed julius@R850:~$ lsmod | grep snd snd_hda_codec_hdmi 36913 1 snd_hda_codec_realtek 78399 1 snd_usb_audio 140685 0 snd_usbmidi_lib 24938 1 snd_usb_audio snd_hda_intel 39619 3 snd_hda_codec 136453 3 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_intel snd_hwdep 13602 2 snd_usb_audio,snd_hda_codec snd_pcm 97451 4 snd_usb_audio,snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 18710 2 snd_pcm,snd_hda_intel snd_seq_midi 13324 0 snd_seq_midi_event 14899 1 snd_seq_midi snd_rawmidi 30180 2 snd_usbmidi_lib,snd_seq_midi snd_seq 61554 2 snd_seq_midi_event,snd_seq_midi snd_seq_device 14497 3 snd_seq,snd_rawmidi,snd_seq_midi snd_timer 29425 2 snd_pcm,snd_seq snd 68876 18 snd_hda_codec_realtek,snd_usb_audio,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_seq,snd_rawmidi,snd_usbmidi_lib,snd_hda_codec,snd_hda_intel,snd_seq_device soundcore 12680 1 snd julius@R850:~$ lsusb Bus 001 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 002 Device 021: ID 18d1:4ee1 Google Inc. Bus 002 Device 015: ID 0424:2514 Standard Microsystems Corp. USB 2.0 Hub Bus 002 Device 016: ID 046d:c245 Logitech, Inc. Bus 002 Device 017: ID 05a4:9881 Ortek Technology, Inc. Bus 002 Device 022: ID 1043:857c iCreate Technologies Corp. Bus 002 Device 020: ID 046a:0180 Cherry GmbH julius@R850:~$ dpkg -l | grep alsa ii alsa-base 1.0.25+dfsg-0ubuntu4 all ALSA driver configuration files ii alsa-utils 1.0.25-4ubuntu2 amd64 Utilities for configuring and using ALSA ii bluez-alsa:amd64 4.101-0ubuntu8b1 amd64 Bluetooth ALSA support ii gstreamer0.10-alsa:amd64 0.10.36-1.1ubuntu1 amd64 GStreamer plugin for ALSA ii gstreamer1.0-alsa:amd64 1.0.6-1 amd64 GStreamer plugin for ALSA

2 1

[alsa-devel] How to clear the ALSA audio settings remembered by system?
by Lin, Mengdong 27 Jun '13

27 Jun '13

Hi, Could anyone tell me how to clear the ALSA audio settings remembered by system? I change some audio codec settings by amixer. And it seems the ALSA has remembered these settings and apply on next boot. And these will change my test result. How can I let system forgot these changes? Many thanks Mengdong

2 2

Re: [alsa-devel] [PATCH 3/6] sound/pci:trivial: replace numeric with standard PM state macros
by Takashi Iwai 27 Jun '13

27 Jun '13

At Thu, 27 Jun 2013 20:55:11 +0800, Yijing Wang wrote: > > Use standard PM state macros PCI_Dx instead of numeric 0/1/2.. > > Signed-off-by: Yijing Wang <wangyijing(a)huawei.com> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: Jiri Kosina <trivial(a)kernel.org> > Cc: Bill Pemberton <wfp5p(a)virginia.edu> > Cc: alsa-devel(a)alsa-project.org > Cc: linux-kernel(a)vger.kernel.org Thanks, applied. Takashi > --- > sound/pci/cs4281.c | 2 +- > sound/pci/ens1370.c | 2 +- > sound/pci/ymfpci/ymfpci_main.c | 2 +- > 3 files changed, 3 insertions(+), 3 deletions(-) > > diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c > index 6a86950..2f6a4b6 100644 > --- a/sound/pci/cs4281.c > +++ b/sound/pci/cs4281.c > @@ -1312,7 +1312,7 @@ static int snd_cs4281_free(struct cs4281 *chip) > /* Sound System Power Management - Turn Everything OFF */ > snd_cs4281_pokeBA0(chip, BA0_SSPM, 0); > /* PCI interface - D3 state */ > - pci_set_power_state(chip->pci, 3); > + pci_set_power_state(chip->pci, PCI_D3hot); > > if (chip->irq >= 0) > free_irq(chip->irq, chip); > diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c > index db2dc83..b30afd1 100644 > --- a/sound/pci/ens1370.c > +++ b/sound/pci/ens1370.c > @@ -1939,7 +1939,7 @@ static int snd_ensoniq_free(struct ensoniq *ensoniq) > #endif > if (ensoniq->irq >= 0) > synchronize_irq(ensoniq->irq); > - pci_set_power_state(ensoniq->pci, 3); > + pci_set_power_state(ensoniq->pci, PCI_D3hot); > __hw_end: > #ifdef CHIP1370 > if (ensoniq->dma_bug.area) > diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c > index 22056c5..d591c15 100644 > --- a/sound/pci/ymfpci/ymfpci_main.c > +++ b/sound/pci/ymfpci/ymfpci_main.c > @@ -2258,7 +2258,7 @@ static int snd_ymfpci_free(struct snd_ymfpci *chip) > /* FIXME: temporarily disabled, otherwise we cannot fire up > * the chip again unless reboot. ACPI bug? > */ > - pci_set_power_state(chip->pci, 3); > + pci_set_power_state(chip->pci, PCI_D3hot); > #endif > > #ifdef CONFIG_PM_SLEEP > -- > 1.7.1 > >

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Re: [alsa-devel] [PATCH 2/2] ASoC: tegra20-ac97: Convert to devm_ioremap_resource()
by Mark Brown 27 Jun '13

27 Jun '13

On Thu, Jun 27, 2013 at 12:22:12PM +0200, Thierry Reding wrote: > Can you please remove this error message as well, please? A message to > the same effect is already output by devm_ioremap_resource(). Other than > that, looks good: Too late, send an incremental patch please.

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[alsa-devel] [RFC PATCH v2 00/12] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Tomasz Figa 27 Jun '13

27 Jun '13

This is another version of my patches extending support of amba-pl08x DMA engine driver to PL080S DMA engine (PL080 modified by Samsung) found in Samsung S3C64xx SoCs. Changes since RFC v1: - Returned to original way of storing quirks as booleans, as suggested by Russell, Linus and Arnd. - Added reg_config field to pl08x_phy_chan struct, which stores variant-specific address of channel config register, as suggested by Russell. - Simplified handling of extended maximum transfer size of PL080S (no more conditional passing of 0 as length to pl08x_cctl_bits()). - Reworked LLI handling in the driver to stop casting arbitrary memory to a struct and allow different word count of LLI entry, as suggested by Linus. - Removed AMBA ID override from S3C64xx PL080 initialization code. - Fixed brokenness of Samsung DMA wrapper API, which caused cyclic buffers to be queued multiple times when DMA engine is used. - Included patch adding clock aliases for DMA engines (depends on Common Clock Framework driver for S3C64xx). - Fixed several minor stylistic issues. For reference, here is the original description of the series: One of the biggest roadblocks on the way of S3C64xx to DeviceTree support is its DMA driver, which is completely platform-specific and provides private API (s3c-dma), not even saying that its design is completely against multiplatform-awareness. The DMA controller present on this SoC series is a custom variant of ARM PrimeCell PL080 modified by Samsung to add some extra features. It is mostly compatible with original PL080, except: - CH_CONTROL2 register is added between CH_CONTROL and CH_CONFIG, - offset of CH_CONFIG register is different, - transfer size field is moved from CH_CONTROL to CH_CONTROL2, - transfer size field is extended to 24 bits, allowing much bigger single transfer, - LLI consists of one more word, to account for CH_CONTROL2 register. Since all the rest is fully compatible with standard PL080 there is no point in having separate driver just for this single variant, so I decided to look into adding support for it to the amba-pl08x driver. There was already some attempt to achieve this before, but this was before Russel's big rework of the driver to use virtual channels, making the old patches being not much of use. This RFC series is a proof of concept that I managed to make during last days of hacking. Except one patch adding clkdev lookup to clock driver (which is being replaced with a CCF-compliant driver ATM), this is enough to get memcpy and slave transfers to work on S3C64xx. I have tested this on Mini6410 and SMDK6410 boards using dmatest for memcpy and Samsung I2S with madplay/aplay for slave transfers. Unfortunately I do not have access to other platforms with PL08x so I could not test for any regressions introduced on them. Credits for two patches go to Alban Bedel, who made a series fixing this driver to make it usable with audio drivers. I rebased his patches on top of mine and corrected coding style a bit. OK, that's all. Any comments are welcome. Feel free to start throwing eggs and tomatoes if you find this awful, but I won't be upset if I get some Tested-by or Acked-by as well. ;) Alban Bedel (2): dmaengine: PL08x: Fix reading the byte count in cctl dmaengine: PL08x: Add cyclic transfer support Tomasz Figa (10): dmaengine: PL08x: Refactor pl08x_getbytes_chan() to lower indentation dmaengine: PL08x: Add support for different offset of CONFIG register dmaengine: PL08x: Rework LLI handling to be less fragile dmaengine: PL08x: Add support for PL080S variant dmaengine: PL08x: Add support for different maximum transfer size ASoC: Samsung: Do not queue cyclic buffers multiple times clk: samsung: s3c64xx: Add aliases for DMA clocks spi: s3c64xx: Do not require legacy DMA API in case of S3C64XX ASoC: Samsung: Do not require legacy DMA API in case of S3C64XX ARM: s3c64xx: Add support for DMA using generic amba-pl08x driver arch/arm/Kconfig | 1 + arch/arm/mach-s3c64xx/Kconfig | 8 +- arch/arm/mach-s3c64xx/Makefile | 1 + arch/arm/mach-s3c64xx/common.h | 5 + arch/arm/mach-s3c64xx/include/mach/dma.h | 65 +++++ arch/arm/mach-s3c64xx/pl080.c | 244 ++++++++++++++++++ arch/arm/plat-samsung/s3c-dma-ops.c | 13 +- drivers/clk/samsung/clk-s3c64xx.c | 2 + drivers/dma/amba-pl08x.c | 426 ++++++++++++++++++++++--------- drivers/spi/Kconfig | 2 +- include/linux/amba/pl080.h | 1 + sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/dma.c | 7 + 13 files changed, 651 insertions(+), 126 deletions(-) create mode 100644 arch/arm/mach-s3c64xx/pl080.c -- 1.8.2.1

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