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

[alsa-devel] [PATCH v2] add native DSD support for XMOS based DACs
by Jurgen Kramer 05 Sep '14

05 Sep '14

Add quirks for XMOS based DACs for native DSD playback support using the new DSD_U32_LE sample format. This version adds native DSD support for: - iFi Audio micro iDSD/nano iDSD (they use the same prod. id) - DIYINHK USB to I2S/DSD converter Changes from v1: - use specific product id and alt setting per XMOS based device Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 19a921e..5ae0536 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1174,5 +1174,20 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, } } + /* XMOS based USB DACs */ + if (le16_to_cpu(chip->dev->descriptor.idVendor) == 0x20b1) { + switch (le16_to_cpu(chip->dev->descriptor.idProduct)) { + /* iFi Audio micro/nano iDSD */ + case 0x3008: + if (fp->altsetting == 2) + return SNDRV_PCM_FMTBIT_DSD_U32_LE; + /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */ + case 0x2009: + if (fp->altsetting == 3) + return SNDRV_PCM_FMTBIT_DSD_U32_LE; + default: + return 0; + } + } return 0; } -- 1.9.3

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[alsa-devel] [PATCH] add native DSD support for XMOS based DACs.
by Jurgen Kramer 05 Sep '14

05 Sep '14

Add quirk for XMOS based DACs for native DSD playback support using the new DSD_U32_LE sample format. Works for all DAC vendors which use XMOS as idVendor. E.g. iFi Audio with their micro iDSD and nano iDSD. Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 19a921e..e0efffb 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1174,5 +1174,15 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, } } + /* XMOS based DACs, e.g. iFi Audio micro/nano iDSD */ + if (le16_to_cpu(chip->dev->descriptor.idVendor) == 0x20b1) { + switch (fp->altsetting) { + case 2: + return SNDRV_PCM_FMTBIT_DSD_U32_LE; + default: + return 0; + } + } + return 0; } -- 1.9.3

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[alsa-devel] [PATCH] pcm: add new DSD sampleformat for native DSD playback on XMOS based devices
by Jurgen Kramer 05 Sep '14

05 Sep '14

XMOS based USB DACs with native DSD support expose this feature via a USB alternate setting. The audio format is either 32-bit raw or a 32-bit PCM format. To utilize this feature on linux this patch introduces a new 32-bit DSD sampleformat DSD_U32_LE. A follow up patch will add a quirk for XMOS based devices to utilize the new format. Further patches will add support to alsa-lib. Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- include/sound/pcm.h | 1 + include/uapi/sound/asound.h | 3 ++- sound/core/pcm.c | 1 + sound/core/pcm_misc.c | 4 ++++ 4 files changed, 8 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 6f3e10c..faefd83 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -183,6 +183,7 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_G723_40_1B _SNDRV_PCM_FMTBIT(G723_40_1B) #define SNDRV_PCM_FMTBIT_DSD_U8 _SNDRV_PCM_FMTBIT(DSD_U8) #define SNDRV_PCM_FMTBIT_DSD_U16_LE _SNDRV_PCM_FMTBIT(DSD_U16_LE) +#define SNDRV_PCM_FMTBIT_DSD_U32_LE _SNDRV_PCM_FMTBIT(DSD_U32_LE) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 32168f7..6ee5867 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -219,7 +219,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ #define SNDRV_PCM_FORMAT_DSD_U8 ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD (x8) */ #define SNDRV_PCM_FORMAT_DSD_U16_LE ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD (x16), little endian */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U16_LE +#define SNDRV_PCM_FORMAT_DSD_U32_LE ((__force snd_pcm_format_t) 50) /* DSD, 4-byte samples DSD (x32), little endian */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U32_LE #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 43932e8..72eb20a 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -215,6 +215,7 @@ static char *snd_pcm_format_names[] = { FORMAT(G723_40_1B), FORMAT(DSD_U8), FORMAT(DSD_U16_LE), + FORMAT(DSD_U32_LE), }; const char *snd_pcm_format_name(snd_pcm_format_t format) diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 2c6fd80..ae7a0fe 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -148,6 +148,10 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { .width = 16, .phys = 16, .le = 1, .signd = 0, .silence = { 0x69, 0x69 }, }, + [SNDRV_PCM_FORMAT_DSD_U32_LE] = { + .width = 32, .phys = 32, .le = 1, .signd = 0, + .silence = { 0x69, 0x69, 0x69, 0x69 }, + }, /* FIXME: the following three formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { .le = -1, .signd = -1, -- 1.9.3

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[alsa-devel] [PATCH 0/9] ASoC: probe/remove and suspend/resume manual bias level transitions cleanup
by Lars-Peter Clausen 04 Sep '14

04 Sep '14

The goal of this series is to allow to cut back the number of manual bias level transitions that drivers have to do. We do have a lot of drivers which manually go to SND_SOC_BIAS_OFF on suspend and go back to SND_SOC_BIAS_STANDBY on resume. Often this is even the only thing done in the suspend and resume handlers. This leads to a lot of drivers with identical suspend and resume code. In this series a new flag called suspend_bias_off is introduced. When this flag is set for a driver the ASoC core will automatically switch to SND_SOC_BIAS_OFF during suspend. This allows to remove the manual transitions (and often the suspend and resume handlers altogether). Similarly a lot of drivers also go to SND_SOC_BIAS_STANDBY at the end of their probe() callback and a lot of drivers go to SND_SOC_BIAS_OFF at the beginning in their remove() handler. The first one is already unnecessary as things are today since the core will make sure to set the DAPM context to SND_SOC_BIAS_STANDBY when the first snd_soc_dapm_sync() is executed during card instantiation after all components have been probed. To be able to remove the later from individual drivers this series introduces a call to snd_soc_dapm_shutdown() when the card is removed. This not only makes sure that the bias level of all components is at SND_SOC_BIAS_OFF, but also makes sure that all widgets are properly powered down. This series only converts a small set of drivers for now to keep small and clear and to be able to focus on the core changes, but I do have patches for cleaning up most of the drivers which in total removes over a 1000 lines of redundant code. The series depends on topic/suspend. - Lars Lars-Peter Clausen (9): ASoC: Set card->instantiated to false when removing the card ASoC: Shutdown DAPM contexts when removing a card ASoC: Add support for automatically going to BIAS_OFF on suspend ASoC: Always run default suspend/resume code ASoC: adau1373: Cleanup manual bias level transitions ASoC: adau17x1: Cleanup manual bias level transitions ASoC: adav80x: Cleanup manual bias level transitions ASoC: ssm2518: Cleanup manual bias level transitions ASoC: ssm2602: Cleanup manual bias level transitions include/sound/soc-dapm.h | 3 ++- include/sound/soc.h | 1 + sound/soc/codecs/adau1373.c | 7 ------- sound/soc/codecs/adau1761.c | 2 +- sound/soc/codecs/adau1781.c | 2 +- sound/soc/codecs/adau17x1.c | 8 -------- sound/soc/codecs/adau17x1.h | 1 - sound/soc/codecs/adav80x.c | 23 ++--------------------- sound/soc/codecs/ssm2518.c | 13 ------------- sound/soc/codecs/ssm2602.c | 24 ++---------------------- sound/soc/soc-core.c | 17 ++++++++++++----- sound/soc/soc-dapm.c | 20 ++++++++++++++++++-- 12 files changed, 39 insertions(+), 82 deletions(-) -- 1.8.0

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[alsa-devel] [PATCH v2] ASoC: rt5670: add clock source selection controls
by bardliao＠realtek.com 04 Sep '14

04 Sep '14

From: Bard Liao <bardliao(a)realtek.com> We can select the clock source of some digital widgets in rt5670. This patch adds the controls of those selections. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Mark, I write the changelog at the beginning of rt5670.c. But I am not sure if it is a regular way to make a changelog. --- sound/soc/codecs/rt5670.c | 121 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 121 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index ba9d9b4..87131a1 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -7,6 +7,10 @@ * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. + * + * Changelog: + * + * - Add clock source selections for some digital widgets. */ #include <linux/module.h> @@ -430,6 +434,105 @@ static SOC_ENUM_SINGLE_DECL(rt5670_if2_dac_enum, RT5670_DIG_INF1_DATA, static SOC_ENUM_SINGLE_DECL(rt5670_if2_adc_enum, RT5670_DIG_INF1_DATA, RT5670_IF2_ADC_SEL_SFT, rt5670_data_select); +static const char * const rt5670_asrc_clk_source[] = { + "clk_sysy_div_out", "clk_i2s1_track", "clk_i2s2_track", + "clk_i2s3_track", "clk_i2s4_track", "clk_sys2", "clk_sys3", + "clk_sys4", "clk_sys5" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5670_da_sto_asrc_enum, RT5670_ASRC_2, + 12, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_da_monol_asrc_enum, RT5670_ASRC_2, + 8, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_da_monor_asrc_enum, RT5670_ASRC_2, + 4, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_ad_sto1_asrc_enum, RT5670_ASRC_2, + 0, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_up_filter_asrc_enum, RT5670_ASRC_3, + 12, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_down_filter_asrc_enum, RT5670_ASRC_3, + 8, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_ad_monol_asrc_enum, RT5670_ASRC_3, + 4, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_ad_monor_asrc_enum, RT5670_ASRC_3, + 0, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_ad_sto2_asrc_enum, RT5670_ASRC_5, + 12, rt5670_asrc_clk_source); + +static const SOC_ENUM_SINGLE_DECL(rt5670_dsp_asrc_enum, RT5670_DSP_CLK, + 0, rt5670_asrc_clk_source); + +static int rt5670_clk_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + unsigned int u_bit = 0, p_bit = 0; + struct soc_enum *em = + (struct soc_enum *)kcontrol->private_value; + + switch (em->reg) { + case RT5670_ASRC_2: + switch (em->shift_l) { + case 0: + u_bit = 0x8; + p_bit = RT5670_PWR_ADC_S1F; + break; + case 4: + u_bit = 0x100; + p_bit = RT5670_PWR_DAC_MF_R; + break; + case 8: + u_bit = 0x200; + p_bit = RT5670_PWR_DAC_MF_L; + break; + case 12: + u_bit = 0x400; + p_bit = RT5670_PWR_DAC_S1F; + break; + } + break; + case RT5670_ASRC_3: + switch (em->shift_l) { + case 0: + u_bit = 0x1; + p_bit = RT5670_PWR_ADC_MF_R; + break; + case 4: + u_bit = 0x2; + p_bit = RT5670_PWR_ADC_MF_L; + break; + } + break; + case RT5670_ASRC_5: + u_bit = 0x4; + p_bit = RT5670_PWR_ADC_S2F; + break; + } + + if (u_bit || p_bit) { + switch (ucontrol->value.integer.value[0]) { + case 1 ... 4: /*enable*/ + if (snd_soc_read(codec, RT5670_PWR_DIG2) & p_bit) + snd_soc_update_bits(codec, + RT5670_ASRC_1, u_bit, u_bit); + break; + default: /*disable*/ + snd_soc_update_bits(codec, RT5670_ASRC_1, u_bit, 0); + break; + } + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new rt5670_snd_controls[] = { /* Headphone Output Volume */ SOC_DOUBLE("HP Playback Switch", RT5670_HP_VOL, @@ -482,6 +585,24 @@ static const struct snd_kcontrol_new rt5670_snd_controls[] = { SOC_ENUM("ADC IF2 Data Switch", rt5670_if2_adc_enum), SOC_ENUM("DAC IF2 Data Switch", rt5670_if2_dac_enum), + + SOC_ENUM_EXT("DA STO Clk Sel", rt5670_da_sto_asrc_enum, + snd_soc_get_enum_double, rt5670_clk_sel_put), + SOC_ENUM_EXT("DA MONOL Clk Sel", rt5670_da_monol_asrc_enum, + snd_soc_get_enum_double, rt5670_clk_sel_put), + SOC_ENUM_EXT("DA MONOR Clk Sel", rt5670_da_monor_asrc_enum, + snd_soc_get_enum_double, rt5670_clk_sel_put), + SOC_ENUM_EXT("AD STO1 Clk Sel", rt5670_ad_sto1_asrc_enum, + snd_soc_get_enum_double, rt5670_clk_sel_put), + SOC_ENUM_EXT("AD MONOL Clk Sel", rt5670_ad_monol_asrc_enum, + snd_soc_get_enum_double, rt5670_clk_sel_put), + SOC_ENUM_EXT("AD MONOR Clk Sel", rt5670_ad_monor_asrc_enum, + snd_soc_get_enum_double, rt5670_clk_sel_put), + SOC_ENUM("UP Clk Sel", rt5670_up_filter_asrc_enum), + SOC_ENUM("DOWN Clk Sel", rt5670_down_filter_asrc_enum), + SOC_ENUM_EXT("AD STO2 Clk Sel", rt5670_ad_sto2_asrc_enum, + snd_soc_get_enum_double, rt5670_clk_sel_put), + SOC_ENUM("DSP Clk Sel", rt5670_dsp_asrc_enum), }; /** -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 0/4] Fixup hda-emu so David's test suite does not fail
by David Henningsson 04 Sep '14

04 Sep '14

So; from what I can see, all newly discovered errors looked like errors in hda-emu rather than errors in the codec. But I don't mind having a second confirmation on that, because I'm not totally sure. With these patches, my test suite is again back to 0 errors. :-) David Henningsson (4): hda-emu: Enable CONFIG_SND_DYNAMIC_MINORS hda-emu: Add verbs for old Nvidia HDMI 8ch 7x hda-emu: Allow some more codec IDs to use ATI HDMI special verbs hda-emu/tester: Fix calculation of machines hda-spec.c | 17 +++++++++++++++++ include/codec_config.h | 2 ++ tester/summary.py | 4 +++- 3 files changed, 22 insertions(+), 1 deletion(-) -- 1.9.1

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[alsa-devel] [PATCH 0/5] hda-emu: Test all codecs, not just the first one
by David Henningsson 04 Sep '14

04 Sep '14

I've improved my test suite to test all the codecs of an alsa-info file, instead of just testing the first one. However, this means that some errors started to show up in my test suite again, so it also means more work to do...but hopefully it's all leading better quality HDA driver in the end! David Henningsson (5): hda-emu: Deal with two cards with one codec each hda-emu: Quit early if requested codec index is above limit hda-emu: Improve error message for modem codecs hda-emu/tester: Add codec index parameter to hda-emu-tester hda-emu/tester: Test all codecs, not just one hda-parse.c | 29 +++++++++++++++------- tester/hda-emu-tester.py | 2 ++ tester/runner.py | 14 +++++++++-- tester/summary.py | 62 ++++++++++++++++++++++++++++++++++++++---------- 4 files changed, 83 insertions(+), 24 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Enable support for S24_LE format
by Peter Ujfalusi 04 Sep '14

04 Sep '14

S24_LE is the same on the bus as S24_3LE, which means the codec can support it. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 52ed57c69dfa..fe16c34607bb 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -18,7 +18,8 @@ #define AIC31XX_RATES SNDRV_PCM_RATE_8000_192000 #define AIC31XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ - | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) #define AIC31XX_STEREO_CLASS_D_BIT 0x1 -- 2.1.0

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[alsa-devel] ALSA Core: Sound Card registration (CPU --> CODEC_1 --> CODEC_2)
by phaniu＠codeaurora.org 03 Sep '14

03 Sep '14

Hi, We have a question regarding how sound card can be registered for the following scenario: CPU connecting to Audio CODEC_1 and then CODEC_1 connecting to another Audio CODEC_2 (over some bus like i2c). CODEC_1 can be represented as a codec_dai and can be registered to ASoC framework (with snd_soc_register_codec()) during system boot up. However, for end-to-end audio to work on this system, we also need CODEC_2 somehow registered to ASoC framework so that audio path on CODEC_2 can configured. Could you please let us know if CODEC_2 can also registered as another dai and if yes, how it can be achieved ? Or is there any other way to probe_defer sound card registration till CODEC_2 is enumerated on the bus. In this case, to what component of the sound card dailink structure, the Audio CODEC_2 can be represented ? Please share your thoughts on this. Best Regards, Phani The Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation

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Re: [alsa-devel] [v3 04/11] ASoC: Intel: sst: Add IPC handling
by Mark Brown 03 Sep '14

03 Sep '14

On Mon, Sep 01, 2014 at 07:27:07PM +0530, Subhransu S. Prusty wrote: > On Mon, Sep 01, 2014 at 01:51:14PM +0100, Mark Brown wrote: > > On Mon, Sep 01, 2014 at 05:47:53PM +0530, Subhransu S. Prusty wrote: > > > > I'd expect much louder complaints if we try to free something that's not > > > > allocated - what happens if we end up reallocating something quickly and > > > > then double freeing? Better to complain if we hit such a code path. > > > > > "freed" is a block which is passed by the caller to be freed up. Will add a > > > comment. > > How would that address the problem? Obviously the caller is trying to > > free what they're passing in. > sst_create_block() which allocates the memory and sst_free_block() which > frees the memory are called in a synchronous way. A single thread who is > allocating waits till a response arrives, if that response is valid then > after processing the response the sst_free_block() is called to free up the > memory. So the double freeing will not happen. Does this address your concern? No. You've described what happens when things are working and everything is operating correctly and there are no bugs in the kernel, the goal with error checking is to provide robustness against the possibility that one of those things isn't true so we can tell what went wrong more easily than if we get memory corruption.

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