November 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 1/2] ALSA: pcm: add SNDRV_PCM_FORMAT_{S, U}20
by Maciej S. Szmigiero 27 Nov '17

27 Nov '17

This format is similar to existing SNDRV_PCM_FORMAT_{S,U}20_3 that keep 20-bit PCM samples in 3 bytes, however i.MX6 platform SSI FIFO does not allow 3-byte accesses (including DMA) so a 4-byte (more conventional) format is needed for it. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Drop "_4" suffix from these formats since they aren't non-standard ones, use empty format slots starting from format number 25 for them, add information that they are LSB justified formats. Corresponding alsa-lib changes will be posted as soon as this patch is merged on the kernel side, to keep alsa-lib and kernel synchronized. include/sound/pcm.h | 8 ++++++++ include/sound/soc-dai.h | 2 ++ include/uapi/sound/asound.h | 9 +++++++++ sound/core/pcm_misc.c | 16 ++++++++++++++++ 4 files changed, 35 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 24febf9e177c..e054c583d3b3 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -169,6 +169,10 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_IMA_ADPCM _SNDRV_PCM_FMTBIT(IMA_ADPCM) #define SNDRV_PCM_FMTBIT_MPEG _SNDRV_PCM_FMTBIT(MPEG) #define SNDRV_PCM_FMTBIT_GSM _SNDRV_PCM_FMTBIT(GSM) +#define SNDRV_PCM_FMTBIT_S20_LE _SNDRV_PCM_FMTBIT(S20_LE) +#define SNDRV_PCM_FMTBIT_U20_LE _SNDRV_PCM_FMTBIT(U20_LE) +#define SNDRV_PCM_FMTBIT_S20_BE _SNDRV_PCM_FMTBIT(S20_BE) +#define SNDRV_PCM_FMTBIT_U20_BE _SNDRV_PCM_FMTBIT(U20_BE) #define SNDRV_PCM_FMTBIT_SPECIAL _SNDRV_PCM_FMTBIT(SPECIAL) #define SNDRV_PCM_FMTBIT_S24_3LE _SNDRV_PCM_FMTBIT(S24_3LE) #define SNDRV_PCM_FMTBIT_U24_3LE _SNDRV_PCM_FMTBIT(U24_3LE) @@ -202,6 +206,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_FLOAT SNDRV_PCM_FMTBIT_FLOAT_LE #define SNDRV_PCM_FMTBIT_FLOAT64 SNDRV_PCM_FMTBIT_FLOAT64_LE #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE +#define SNDRV_PCM_FMTBIT_S20 SNDRV_PCM_FMTBIT_S20_LE +#define SNDRV_PCM_FMTBIT_U20 SNDRV_PCM_FMTBIT_U20_LE #endif #ifdef SNDRV_BIG_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_BE @@ -213,6 +219,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_FLOAT SNDRV_PCM_FMTBIT_FLOAT_BE #define SNDRV_PCM_FMTBIT_FLOAT64 SNDRV_PCM_FMTBIT_FLOAT64_BE #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE +#define SNDRV_PCM_FMTBIT_S20 SNDRV_PCM_FMTBIT_S20_BE +#define SNDRV_PCM_FMTBIT_U20 SNDRV_PCM_FMTBIT_U20_BE #endif struct snd_pcm_file { diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 58acd00cae19..d970879944fc 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -102,6 +102,8 @@ struct snd_compr_stream; SNDRV_PCM_FMTBIT_S16_BE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S20_LE |\ + SNDRV_PCM_FMTBIT_S20_BE |\ SNDRV_PCM_FMTBIT_S24_3LE |\ SNDRV_PCM_FMTBIT_S24_3BE |\ SNDRV_PCM_FMTBIT_S32_LE |\ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index c227ccba60ae..7385024041d2 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -214,6 +214,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_IMA_ADPCM ((__force snd_pcm_format_t) 22) #define SNDRV_PCM_FORMAT_MPEG ((__force snd_pcm_format_t) 23) #define SNDRV_PCM_FORMAT_GSM ((__force snd_pcm_format_t) 24) +#define SNDRV_PCM_FORMAT_S20_LE ((__force snd_pcm_format_t) 25) /* \ */ +#define SNDRV_PCM_FORMAT_S20_BE ((__force snd_pcm_format_t) 26) /* | */ +#define SNDRV_PCM_FORMAT_U20_LE ((__force snd_pcm_format_t) 27) /* | in four bytes, */ +#define SNDRV_PCM_FORMAT_U20_BE ((__force snd_pcm_format_t) 28) /* / LSB justified */ +/* gap in the numbering for a future standard linear format */ #define SNDRV_PCM_FORMAT_SPECIAL ((__force snd_pcm_format_t) 31) #define SNDRV_PCM_FORMAT_S24_3LE ((__force snd_pcm_format_t) 32) /* in three bytes */ #define SNDRV_PCM_FORMAT_S24_3BE ((__force snd_pcm_format_t) 33) /* in three bytes */ @@ -248,6 +253,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_LE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_LE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE +#define SNDRV_PCM_FORMAT_S20 SNDRV_PCM_FORMAT_S20_LE +#define SNDRV_PCM_FORMAT_U20 SNDRV_PCM_FORMAT_U20_LE #endif #ifdef SNDRV_BIG_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_BE @@ -259,6 +266,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_BE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_BE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE +#define SNDRV_PCM_FORMAT_S20 SNDRV_PCM_FORMAT_S20_BE +#define SNDRV_PCM_FORMAT_U20 SNDRV_PCM_FORMAT_U20_BE #endif typedef int __bitwise snd_pcm_subformat_t; diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 9be81025372f..c62bfe27106f 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -170,6 +170,22 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { [SNDRV_PCM_FORMAT_GSM] = { .le = -1, .signd = -1, }, + [SNDRV_PCM_FORMAT_S20_LE] = { + .width = 20, .phys = 32, .le = 1, .signd = 1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_S20_BE] = { + .width = 20, .phys = 32, .le = 0, .signd = 1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_U20_LE] = { + .width = 20, .phys = 32, .le = 1, .signd = 0, + .silence = { 0x00, 0x00, 0x08, 0x00 }, + }, + [SNDRV_PCM_FORMAT_U20_BE] = { + .width = 20, .phys = 32, .le = 0, .signd = 0, + .silence = { 0x00, 0x08, 0x00, 0x00 }, + }, [SNDRV_PCM_FORMAT_SPECIAL] = { .le = -1, .signd = -1, },

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[alsa-devel] [PATCH v3 2/2] ASoC: fsl_ssi: add 20-bit sample format for AC'97 and use it for capture
by Maciej S. Szmigiero 27 Nov '17

27 Nov '17

When testing AC'97 capture on UDOO board (currently the only user of fsl_ssi driver in the AC'97 mode) it become obvious that there is a massive distortion above certain, small input signal. This problem has been traced to silicon errata ERR003778: "In AC97, 16-bit mode, received data is shifted by 4-bit locations" that has "No fix scheduled". This errata suggests a workaround of doing a 4-bit shift back in SDMA script for this specific operation mode, however our SDMA scripts are shared between various SoC peripherals so we can't really modify them. There is a simple way to avoid this problem, however, that is to disallow recording in 16-bit mode and only support it in AC'97-native 20-bit mode. We have to use a 4-byte format for this since SSI FIFOs do not allow 3-byte accesses (and these aren't supported by imx-sdma driver anyway). With this change the capture distortion is gone. We can also add this format as an additional one supported for playback, using this opportunity to make sure that we use CPU-endian-native formats in AC'97 mode as we already do in I2S mode. There is no problem in using different bit widths in playback and capture in AC'97 mode so allow this, too. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Adapt format name to changes in the first patch from this series. Changes from v2: None. sound/soc/fsl/fsl_ssi.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 20ef09e1a395..c350117c8e31 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1278,14 +1278,15 @@ static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S20, }, .capture = { .stream_name = "AC97 Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + /* 16-bit capture is broken (errata ERR003778) */ + .formats = SNDRV_PCM_FMTBIT_S20, }, .ops = &fsl_ssi_dai_ops, }; @@ -1557,11 +1558,12 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - if (!fsl_ssi_is_ac97(ssi_private)) + if (!fsl_ssi_is_ac97(ssi_private)) { ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_samplebits = 1; + } ssi_private->cpu_dai_drv.symmetric_channels = 1; - ssi_private->cpu_dai_drv.symmetric_samplebits = 1; } /* Determine the FIFO depth. */

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[alsa-devel] [PATCH] ASoC: fsl_ssi: Override bit clock rate based on slot number
by Nicolin Chen 26 Nov '17

26 Nov '17

The set_sysclk() now is used to override the output bit clock rate. But this is not a common way to implement a set_dai_sysclk(). And this creates a problem when a general machine driver (simple-card for example) tries to do set_dai_sysclk() by passing an input clock rate for the baud clock instead of setting the bit clock rate as fsl_ssi driver expected. So this patch solves this problem by firstly removing set_sysclk() since the hw_params() can calculate the bit clock rate. Secondly, in order not to break those TDM use cases which previously might have been using set_sysclk() to override the bit clock rate, this patch changes the driver to override it based on the slot number. The patch also removes an obsolete comment of the dir parameter. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_ssi.c | 26 ++++++++------------------ 1 file changed, 8 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 64598d1..3657c88 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -197,12 +197,12 @@ struct fsl_ssi_soc_data { * @use_dma: DMA is used or FIQ with stream filter * @use_dual_fifo: DMA with support for both FIFOs used * @fifo_deph: Depth of the SSI FIFOs + * @slots: number of slots * @rxtx_reg_val: Specific register settings for receive/transmit configuration * * @clk: SSI clock * @baudclk: SSI baud clock for master mode * @baudclk_streams: Active streams that are using baudclk - * @bitclk_freq: bitclock frequency set by .set_dai_sysclk * * @dma_params_tx: DMA transmit parameters * @dma_params_rx: DMA receive parameters @@ -233,12 +233,12 @@ struct fsl_ssi_private { bool use_dual_fifo; bool has_ipg_clk_name; unsigned int fifo_depth; + unsigned int slots; struct fsl_ssi_rxtx_reg_val rxtx_reg_val; struct clk *clk; struct clk *baudclk; unsigned int baudclk_streams; - unsigned int bitclk_freq; /* regcache for volatile regs */ u32 regcache_sfcsr; @@ -700,8 +700,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, * Note: This function can be only called when using SSI as DAI master * * Quick instruction for parameters: - * freq: Output BCLK frequency = samplerate * 32 (fixed) * channels - * dir: SND_SOC_CLOCK_OUT -> TxBCLK, SND_SOC_CLOCK_IN -> RxBCLK. + * freq: Output BCLK frequency = samplerate * 32 (fixed) * slots (or channels) */ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai, @@ -716,9 +715,9 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, unsigned int freq; bool baudclk_is_used; - /* Prefer the explicitly set bitclock frequency */ - if (ssi_private->bitclk_freq) - freq = ssi_private->bitclk_freq; + /* Generate bit clock based on the slot or channel number */ + if (ssi_private->slots) + freq = ssi_private->slots * 32 * params_rate(hw_params); else freq = params_channels(hw_params) * 32 * params_rate(hw_params); @@ -805,16 +804,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, return 0; } -static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); - - ssi_private->bitclk_freq = freq; - - return 0; -} - /** * fsl_ssi_hw_params - program the sample size * @@ -1121,6 +1110,8 @@ static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, regmap_update_bits(regs, CCSR_SSI_SCR, CCSR_SSI_SCR_SSIEN, val); + ssi_private->slots = slots; + return 0; } @@ -1191,7 +1182,6 @@ static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .hw_params = fsl_ssi_hw_params, .hw_free = fsl_ssi_hw_free, .set_fmt = fsl_ssi_set_dai_fmt, - .set_sysclk = fsl_ssi_set_dai_sysclk, .set_tdm_slot = fsl_ssi_set_dai_tdm_slot, .trigger = fsl_ssi_trigger, }; -- 2.7.4

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[alsa-devel] Issue with the Realtek ALC 898 codec on the Clevo P751DM2-G and the ESS Sabre Audio Jack for the headphone output.
by Dennis Mungai 26 Nov '17

26 Nov '17

Hello there, I have Ubuntu 16.04LTS installed on the Clevo P751DM2-G, also marketed as the Schenker XMG U507, the System76's Serval WS (2017 model), the Eurocom Sky X4E2, among others. The sound card is the Realtek ALC898 with the following output configuration: 1. One SPDIF/headphone combo jack (to the right) 2. One Line Out. 3. One Line-In. 4. One Mic-In. Background information: On Windows, one can use the Realtek HD Audio control panel pop-up menu on jack detection to set the output type when an audio device is plugged in to an appropriate jack, combined with Sound Blaster X-Fi MB5 software to enable post-processing effects such as Bass, noise cancellation, etc. What I have tried so far: 1. Setting the model definition at /etc/modprobe.d/sound.conf with the following entries: options snd-hda-intel model=no-primary-hp enable=1 index=0 Using such a line worked before on an MSI GS43VR 6RE Phantom Pro that also has the same (or a similar) ESS Sabre Audio DAC component. Expected behavior: When plugged into the microphone jack, the ESS Sabre DAC should be activated and sound should be routed to the headset. Observed anomaly: Sound comes out through the speakers instead. However, the line-out output works. In that case, when plugged into the line out, Audio works as expected. I was able to report this as a bug to Launchpad, with the information below: https://bugs.launchpad.net/bugs/1731940 ProblemType: Bug DistroRelease: Ubuntu 16.04 Package: alsa-base 1.0.25+dfsg-0ubuntu5 Uname: Linux 4.14.0 x86_64 NonfreeKernelModules: nvidia_uvm nvidia_drm nvidia_modeset nvidia ApportVersion: 2.20.1-0ubuntu2.10 Architecture: amd64 AudioDevicesInUse: USER PID ACCESS COMMAND /dev/snd/controlC1: lin 2199 F.... pulseaudio /dev/snd/controlC0: lin 2199 F.... pulseaudio CurrentDesktop: GNOME Date: Mon Nov 13 17:34:53 2017 InstallationDate: Installed on 2017-07-25 (111 days ago) InstallationMedia: Ubuntu 16.04.2 LTS "Xenial Xerus" - Release amd64 (20170215.2) PackageArchitecture: all SourcePackage: alsa-driver Symptom: audio Symptom_AlsaPlaybackTest: ALSA playback test through plughw:PCH failed Symptom_Card: Built-in Audio - HDA Intel PCH Symptom_DevicesInUse: USER PID ACCESS COMMAND /dev/snd/controlC1: gdm 1812 F.... pulseaudio lin 2199 F.... pulseaudio /dev/snd/controlC0: gdm 1812 F.... pulseaudio lin 2199 F.... pulseaudio Symptom_Jack: Speaker, Internal Symptom_Type: No sound at all Title: [P7xxDM2(-G), Realtek ALC898, Speaker, Internal] No sound at all UpgradeStatus: No upgrade log present (probably fresh install) dmi.bios.date: 12/21/2016 dmi.bios.vendor: American Megatrends Inc. dmi.bios.version: 1.06.04 dmi.board.asset.tag: Tag 12345 dmi.board.name: P7xxDM2(-G) dmi.board.vendor: Notebook dmi.board.version: Not Applicable dmi.chassis.asset.tag: No Asset Tag dmi.chassis.type: 10 dmi.chassis.vendor: No Enclosure dmi.chassis.version: N/A dmi.modalias: dmi:bvnAmericanMegatrendsInc.:bvr1.06.04:bd12/21/2016:svnNotebook:pnP7xxDM2(-G):pvrNotApplicable:rvnNotebook:rnP7xxDM2(-G):rvrNotApplicable:cvnNoEnclosure:ct10:cvrN/A: dmi.product.family: Not Applicable dmi.product.name: P7xxDM2(-G) dmi.product.version: Not Applicable dmi.sys.vendor: Notebook ** Affects: alsa-driver (Ubuntu) Importance: Undecided Status: New Regards, Dennis.

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[alsa-devel] LADSPA ALSA plugin
by Robert Bielik 26 Nov '17

26 Nov '17

Hi all, Started experimenting with using LADSPA plugins with the ALSA LADSPA PCM plugin, and it works nicely. However, it seems only the PCM is interface is exposed from the plugin. Question is if there is a way to expose the LADSPA plugin parameters directly to alsamixer ? Regards /Rob

1 1

[alsa-devel] [PATCH] ASoC: amd: added error checks in dma driver
by Vijendar Mukunda 24 Nov '17

24 Nov '17

added error checks in acp dma driver Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Akshu Agrawal <Akshu.Agrawal(a)amd.com> Signed-off-by: Guenter Roeck <groeck(a)chromium.org> --- sound/soc/amd/acp-pcm-dma.c | 45 ++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 40 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 17d76fa..804e659 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -848,6 +848,9 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + if (!rtd) + return -EINVAL; + buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd->acp_mmio, substream->stream); @@ -873,6 +876,8 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + if (!rtd) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config_acp_dma_channel(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, PLAYBACK_START_DMA_DESCR_CH12, @@ -1066,6 +1071,10 @@ static int acp_audio_probe(struct platform_device *pdev) struct resource *res; const u32 *pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "Missing platform data\n"); + return -ENODEV; + } audio_drv_data = devm_kzalloc(&pdev->dev, sizeof(struct audio_drv_data), GFP_KERNEL); if (audio_drv_data == NULL) @@ -1074,6 +1083,8 @@ static int acp_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); audio_drv_data->acp_mmio = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(audio_drv_data->acp_mmio)) + return PTR_ERR(audio_drv_data->acp_mmio); /* The following members gets populated in device 'open' * function. Till then interrupts are disabled in 'acp_init' * and device doesn't generate any interrupts. @@ -1099,7 +1110,11 @@ static int acp_audio_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, audio_drv_data); /* Initialize the ACP */ - acp_init(audio_drv_data->acp_mmio, audio_drv_data->asic_type); + status = acp_init(audio_drv_data->acp_mmio, audio_drv_data->asic_type); + if (status) { + dev_err(&pdev->dev, "ACP Init failed\n"); + return status; + } status = snd_soc_register_platform(&pdev->dev, &acp_asoc_platform); if (status != 0) { @@ -1116,9 +1131,14 @@ static int acp_audio_probe(struct platform_device *pdev) static int acp_audio_remove(struct platform_device *pdev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(&pdev->dev); - acp_deinit(adata->acp_mmio); + status = acp_deinit(adata->acp_mmio); + if (status) { + dev_err(&pdev->dev, "ACP Deinit failed\n"); + return status; + } snd_soc_unregister_platform(&pdev->dev); pm_runtime_disable(&pdev->dev); @@ -1128,9 +1148,14 @@ static int acp_audio_remove(struct platform_device *pdev) static int acp_pcm_resume(struct device *dev) { u16 bank; + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_init(adata->acp_mmio, adata->asic_type); + status = acp_init(adata->acp_mmio, adata->asic_type); + if (status) { + dev_err(dev, "ACP Init failed\n"); + return status; + } if (adata->play_stream && adata->play_stream->runtime) { /* For Stoney, Memory gating is disabled,i.e SRAM Banks @@ -1162,18 +1187,28 @@ static int acp_pcm_resume(struct device *dev) static int acp_pcm_runtime_suspend(struct device *dev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_deinit(adata->acp_mmio); + status = acp_deinit(adata->acp_mmio); + if (status) { + dev_err(dev, "ACP Deinit failed\n"); + return status; + } acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } static int acp_pcm_runtime_resume(struct device *dev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_init(adata->acp_mmio, adata->asic_type); + status = acp_init(adata->acp_mmio, adata->asic_type); + if (status) { + dev_err(dev, "ACP Init failed\n"); + return status; + } acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } -- 2.7.4

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[alsa-devel] [PATCH] SoC: rt5514: Make sure the DMIC delay will be happened after normal SUPPLY widgets power on
by Oder Chiou 24 Nov '17

24 Nov '17

The patch makes sure the DMIC delay will be happened after normal SUPPLY widgets power on. If there are some platforms that provide the MCLK using the SUPPLY widget, it will make sure the delay time is helpful. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 61ccbc6..198df01 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -567,7 +567,7 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SUPPLY_S("DMIC CLK", 1, SND_SOC_NOPM, 0, 0, rt5514_set_dmic_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("ADC CLK", RT5514_CLK_CTRL1, -- 2.7.4

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[alsa-devel] [PATCH] test: correct emulation for channel-map TLV
by Takashi Sakamoto 24 Nov '17

24 Nov '17

Current implementation of channel-map TLV on test program is not valid. Furthermore, it brings buffer-over-run due to byte counting. This commit fixes it. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- test/user-ctl-element-set.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) diff --git a/test/user-ctl-element-set.c b/test/user-ctl-element-set.c index 668831a8..e94152b9 100644 --- a/test/user-ctl-element-set.c +++ b/test/user-ctl-element-set.c @@ -126,33 +126,39 @@ static void change_int_elem_members(struct elem_set_trial *trial, static int allocate_int_elem_set_tlv(struct elem_set_trial *trial, unsigned int **tlv) { - unsigned int len, pos; + unsigned int count, pos; unsigned int i, j; struct chmap_entry *entry; /* Calculate size of TLV packet for channel-mapping information. */ - len = 0; + count = 0; for (i = 1; i <= 25; ++i) { - len += sizeof(struct chmap_entry); - len += i * sizeof(unsigned int); + count += 2; /* sizeof(struct chmap_entry). */ + count += i; /* struct chmap_entry.maps. */ } - *tlv = malloc(len); - if (*tlv == NULL) + *tlv = malloc((2 + count) * sizeof(unsigned int)); + if (!*tlv) return -ENOMEM; /* * Emulate channel-mapping information in in-kernel implementation. * Here, 25 entries are for each different channel. */ - pos = 0; - for (i = 1; i <= 25 && pos < len; ++i) { + (*tlv)[0] = SNDRV_CTL_TLVT_CONTAINER; + (*tlv)[1] = count * sizeof(unsigned int); + pos = 2; + + for (i = 1; i <= 25 && pos < count; ++i) { entry = (struct chmap_entry *)&(*tlv)[pos]; + entry->type = SNDRV_CTL_TLVT_CHMAP_FIXED; entry->length = i * sizeof(unsigned int); + pos += 2; + for (j = 0; j < i; ++j) entry->maps[j] = SND_CHMAP_MONO + j; - pos += sizeof(struct chmap_entry) + i * sizeof(unsigned int); + pos += i; } return 0; -- 2.14.1

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[alsa-devel] [PATCH v2] ASoC: rt5514: Make sure the DMIC delay will be happened after normal SUPPLY widgets power on
by Oder Chiou 24 Nov '17

24 Nov '17

The patch makes sure the DMIC delay will be happened after normal SUPPLY widgets power on. If there are some platforms that provide the MCLK using the SUPPLY widget, it will make sure the delay time is helpful. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 61ccbc6..198df01 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -567,7 +567,7 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SUPPLY_S("DMIC CLK", 1, SND_SOC_NOPM, 0, 0, rt5514_set_dmic_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("ADC CLK", RT5514_CLK_CTRL1, -- 2.7.4

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Re: [alsa-devel] Focusrite Scarlett 6i6 gen1 - input handling fix
by Jens Verwiebe 23 Nov '17

23 Nov '17

Dammit, sorry, i better add the patch as textfile (?) Formatting looks good here, but ml shows spaces where i expect tabs. New try: The Scarlett 6i6 has no padding on rear inputs 3/4 but a gainstage (Low/Hi). Adding this functionality Signed-off-by: Jens Verwiebe <info at jensverwiebe.de> 1 file changed, 21 insertions(+), 2 deletions(-) -- Jens Verwiebe Allerskehre 44 - 22309 Hamburg Tel.: +49 40 68 78 50 mobile: +49 172 400 49 07 mailto: info(a)jensverwiebe.de web: http://www.jensverwiebe.de

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