September 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] fix memory leak in snd_pcm_set_chmap
by GitHub pull_request - opened 16 Sep '19

16 Sep '19

alsa-project/alsa-lib pull request #11 was opened from conradjones: memory leak in snd_pcm_set_chmap : memory is allocated in the function pointed at by : pcm->ops->get_chmap called in: snd_pcm_chmap_t *snd_pcm_get_chmap(snd_pcm_t *pcm) Request URL : https://github.com/alsa-project/alsa-lib/pull/11 Patch URL : https://github.com/alsa-project/alsa-lib/pull/11.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] fix memory leak in snd_pcm_set_chmap
by GitHub pull_request - edited 15 Sep '19

15 Sep '19

alsa-project/alsa-lib pull request #10 was edited from conradjones: ignore Request URL : https://github.com/alsa-project/alsa-lib/pull/10 Patch URL : https://github.com/alsa-project/alsa-lib/pull/10.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] fix memory leak in snd_pcm_set_chmap
by GitHub pull_request - opened 15 Sep '19

15 Sep '19

alsa-project/alsa-lib pull request #10 was opened from conradjones: asan detects this as a memory leak. snd_pcm_get_chmap clearly states that the caller is required to free the returned value, i cannot see it stored anywhere so cannot see how it could be freed. Request URL : https://github.com/alsa-project/alsa-lib/pull/10 Patch URL : https://github.com/alsa-project/alsa-lib/pull/10.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[alsa-devel] [PATCH] ALSA: usb-audio: Add Hiby device family to quirks for native DSD support
by ilya.pshonkin＠netforce.ua 15 Sep '19

15 Sep '19

From: Sudokamikaze <sudokamikaze(a)protonmail.com> This patch adds quirk VID ID for Hiby portable players family with native DSD playback support Signed-off-by: Sudokamikaze <sudokamikaze(a)protonmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 78858918cbc1..64a8d73972e3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1655,6 +1655,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x152a: /* Thesycon devices */ case 0x25ce: /* Mytek devices */ case 0x2ab6: /* T+A devices */ + case 0xc502: /* HiBy devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.23.0

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[alsa-devel] Jack detection in Lenovo OneLink+ and other docks
by Benjamin Berg 15 Sep '19

15 Sep '19

Hi, I have a Lenovo OneLink+ and TB3 docks here. The USB audio adapter on these docks work fine. However, Jack detection is not functional unfortunately which is somewhat annoying. There are HID devices available on the docking stations which seem to report the jack sensing information. Assuming we can decode this information correctly, does anyone have an idea how this can be hooked up so that the information will be available on the audio device itself? Benjamin The OneLink+ dock exposes the following interfaces: |__ Port 1: Dev 90, If 0, Class=Hub, Driver=hub/4p, 480M |__ Port 3: Dev 92, If 1, Class=CDC Data, Driver=cdc_ether, 480M |__ Port 3: Dev 92, If 0, Class=Communications, Driver=cdc_ether, 480M |__ Port 4: Dev 94, If 2, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 4: Dev 94, If 0, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 4: Dev 94, If 3, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 4: Dev 94, If 1, Class=Audio, Driver=snd-usb-audio, 12M Similar on the TB 3 Dock from Lenovo: /: Bus 05.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/4p, 480M |__ Port 1: Dev 2, If 0, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 2, If 1, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 2, If 2, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 2, If 3, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 4: Dev 3, If 0, Class=Human Interface Device, Driver=usbhid, 12M On each of these devices, plugging/unplugging will fire various HID events, seemingly with some sensing information. On the OneLink Plus: D: 0 R: 69 05 0c 09 01 a1 01 85 01 15 00 25 01 09 e9 09 ea 75 01 95 02 81 42 09 e2 95 01 81 06 06 01 ff 09 01 95 09 81 06 05 0c 09 cd 09 b7 09 b5 09 b6 95 04 81 06 06 07 ff 09 01 75 08 95 0d 81 06 09 00 95 0f 91 02 c0 N: C-Media Electronics Inc. ThinkPad OneLink Plus Dock Audio P: usb-0000:00:14.0-4.1.4/input3 I: 3 17ef 3055 # Plug Headphone D: 0 E: 0.000000 16 01 00 06 00 00 20 83 ab 08 00 08 c0 10 00 00 00 E: 3.103810 16 01 00 0a 00 00 20 83 ab 08 00 03 80 10 00 00 00 # Unplug headphone E: 9.951613 16 01 00 08 00 00 20 83 ab 08 00 03 80 00 00 00 00 E: 10.016073 16 01 00 04 00 00 20 83 ab 08 00 03 80 10 00 00 00 # Plug Microphone E: 17.824186 16 01 00 06 00 00 20 83 ab 08 00 03 80 00 00 00 00 E: 20.928234 16 01 00 0a 00 00 20 83 ab 08 00 03 80 10 00 00 00 # Unplug Microphone E: 26.734048 16 01 00 08 00 00 20 83 ab 08 00 08 c0 10 00 00 00 E: 26.784007 16 01 00 04 00 00 20 83 ab 08 00 03 80 10 00 00 00 # Plug headset E: 44.160216 16 01 00 06 00 00 20 83 ab 08 00 03 80 00 00 00 00 # Unplug headset E: 56.687367 16 01 00 04 00 00 20 83 ab 08 00 08 c0 10 00 00 00 On the TB 3 dock: D: 0 R: 120 05 0c 09 01 a1 01 85 01 15 00 25 01 09 e9 09 ea 75 01 95 02 81 02 09 e2 95 01 81 06 09 00 95 02 81 03 09 36 a1 02 05 09 19 01 29 02 75 02 95 01 15 01 25 02 81 40 c0 05 0c 09 00 15 00 25 01 75 01 95 01 81 03 85 02 05 0c 09 00 95 10 81 02 85 03 09 00 91 02 85 04 09 00 75 08 95 26 91 02 85 05 09 00 95 22 81 02 85 06 09 00 95 24 91 02 85 07 09 00 95 20 81 02 c0 N: Lenovo ThinkPad Thunderbolt 3 Dock USB Audio P: usb-0000:0a:00.0-1/input3 I: 3 17ef 306a # Plug in headphone D: 0 E: 0.000000 3 02 00 80 # Unplug headphone E: 16.480011 3 02 00 00 # Plug in microphone E: 34.224331 3 02 00 80 # Unplug microphone E: 46.960128 3 02 00 00 # Plug in headset E: 62.592182 3 02 00 80 E: 63.120449 3 02 00 c0 # Unplug headset E: 72.608250 3 02 00 80 E: 72.624253 3 02 00 00

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[alsa-devel] Applied "ASoC: sdm845: remove unneeded semicolon" to the asoc tree
by Mark Brown 15 Sep '19

15 Sep '19

The patch ASoC: sdm845: remove unneeded semicolon has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fca11622d600228bec405456f41590155b3a3eca Mon Sep 17 00:00:00 2001 From: Saiyam Doshi <saiyamdoshi.in(a)gmail.com> Date: Sat, 14 Sep 2019 08:41:33 +0530 Subject: [PATCH] ASoC: sdm845: remove unneeded semicolon Remove excess semicolon after closing parenthesis. Signed-off-by: Saiyam Doshi <saiyamdoshi.in(a)gmail.com> Link: https://lore.kernel.org/r/20190914031133.GA28447@SD Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/sdm845.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 882f52ed8231..28f3cef696e6 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -319,7 +319,7 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, 0, SNDRV_PCM_STREAM_PLAYBACK); - }; + } break; case SECONDARY_MI2S_TX: -- 2.20.1

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[alsa-devel] [PATCH] ASoC: Intel: kbl_rt5663_rt5514_max98927: Add dmic format constraint
by Yu-Hsuan Hsu 14 Sep '19

14 Sep '19

24 bits recording from DMIC is not supported for KBL platform because the TDM slot between PCH and codec is 16 bits only. We should add a constraint to remove that unsupported format. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda8784f1a01..67b276a65a8d2d 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -400,6 +400,9 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, dmic_constraints); + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } -- 2.23.0.162.g0b9fbb3734-goog

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[alsa-devel] [PATCH] ASoC: fsl_sai: Implement set_bclk_ratio
by Daniel Baluta 13 Sep '19

13 Sep '19

From: Viorel Suman <viorel.suman(a)nxp.com> This is to allow machine drivers to set a certain bitclk rate which might not be exactly rate * frame size. Cc: NXP Linux Team <linux-imx(a)nxp.com> Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 21 +++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 20 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index fe126029f4e3..e896b577b1f7 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -137,6 +137,16 @@ static int fsl_sai_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, return 0; } +static int fsl_sai_set_dai_bclk_ratio(struct snd_soc_dai *dai, + unsigned int ratio) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai); + + sai->bclk_ratio = ratio; + + return 0; +} + static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { @@ -423,8 +433,14 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, slot_width = sai->slot_width; if (!sai->is_slave_mode) { - ret = fsl_sai_set_bclk(cpu_dai, tx, - slots * slot_width * params_rate(params)); + if (sai->bclk_ratio) + ret = fsl_sai_set_bclk(cpu_dai, tx, + sai->bclk_ratio * + params_rate(params)); + else + ret = fsl_sai_set_bclk(cpu_dai, tx, + slots * slot_width * + params_rate(params)); if (ret) return ret; @@ -640,6 +656,7 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { + .set_bclk_ratio = fsl_sai_set_dai_bclk_ratio, .set_sysclk = fsl_sai_set_dai_sysclk, .set_fmt = fsl_sai_set_dai_fmt, .set_tdm_slot = fsl_sai_set_dai_tdm_slot, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 3a3f6f8e5595..f96f8d97489d 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -177,6 +177,7 @@ struct fsl_sai { unsigned int mclk_streams; unsigned int slots; unsigned int slot_width; + unsigned int bclk_ratio; const struct fsl_sai_soc_data *soc_data; struct snd_dmaengine_dai_dma_data dma_params_rx; -- 2.17.1

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[alsa-devel] [PATCH v3] drm: bridge/dw_hdmi: add audio sample channel status setting
by Cheng-Yi Chiang 13 Sep '19

13 Sep '19

From: Yakir Yang <ykk(a)rock-chips.com> When transmitting IEC60985 linear PCM audio, we configure the Aduio Sample Channel Status information in the IEC60958 frame. The status bit is already available in iec.status of hdmi_codec_params. This fix the issue that audio does not come out on some monitors (e.g. LG 22CV241) Note that these registers are only for interfaces: I2S audio interface, General Purpose Audio (GPA), or AHB audio DMA (AHBAUDDMA). For S/PDIF interface this information comes from the stream. Currently this function dw_hdmi_set_channel_status is only called from dw-hdmi-i2s-audio in I2S setup. Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Change from v2 to v3: 1. Reuse what is already set in iec.status in hw_param. 2. Remove all useless definition of registers and values. 3. Note that the original sampling frequency is not written to the channel status as we reuse create_iec958_consumer in pcm_iec958.c. Without that it can still play audio fine. .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 1 + drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 20 +++++++++++++++++++ drivers/gpu/drm/bridge/synopsys/dw-hdmi.h | 2 ++ include/drm/bridge/dw_hdmi.h | 1 + 4 files changed, 24 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index 34d8e837555f..20f4f92dd866 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -102,6 +102,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, } dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); + dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); dw_hdmi_set_channel_allocation(hdmi, hparms->cea.channel_allocation); diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index bd65d0479683..aa7efd4da1c8 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -582,6 +582,26 @@ static unsigned int hdmi_compute_n(unsigned int freq, unsigned long pixel_clk) return n; } +/* + * When transmitting IEC60958 linear PCM audio, these registers allow to + * configure the channel status information of all the channel status + * bits in the IEC60958 frame. For the moment this configuration is only + * used when the I2S audio interface, General Purpose Audio (GPA), + * or AHB audio DMA (AHBAUDDMA) interface is active + * (for S/PDIF interface this information comes from the stream). + */ +void dw_hdmi_set_channel_status(struct dw_hdmi *hdmi, + u8 *channel_status) +{ + /* + * Set channel status register for frequency and word length. + * Use default values for other registers. + */ + hdmi_writeb(hdmi, channel_status[3], HDMI_FC_AUDSCHNLS7); + hdmi_writeb(hdmi, channel_status[4], HDMI_FC_AUDSCHNLS8); +} +EXPORT_SYMBOL_GPL(dw_hdmi_set_channel_status); + static void hdmi_set_clk_regenerator(struct dw_hdmi *hdmi, unsigned long pixel_clk, unsigned int sample_rate) { diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h index 6988f12d89d9..fcff5059db24 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h @@ -158,6 +158,8 @@ #define HDMI_FC_SPDDEVICEINF 0x1062 #define HDMI_FC_AUDSCONF 0x1063 #define HDMI_FC_AUDSSTAT 0x1064 +#define HDMI_FC_AUDSCHNLS7 0x106e +#define HDMI_FC_AUDSCHNLS8 0x106f #define HDMI_FC_DATACH0FILL 0x1070 #define HDMI_FC_DATACH1FILL 0x1071 #define HDMI_FC_DATACH2FILL 0x1072 diff --git a/include/drm/bridge/dw_hdmi.h b/include/drm/bridge/dw_hdmi.h index cf528c289857..4b3e863c4f8a 100644 --- a/include/drm/bridge/dw_hdmi.h +++ b/include/drm/bridge/dw_hdmi.h @@ -156,6 +156,7 @@ void dw_hdmi_setup_rx_sense(struct dw_hdmi *hdmi, bool hpd, bool rx_sense); void dw_hdmi_set_sample_rate(struct dw_hdmi *hdmi, unsigned int rate); void dw_hdmi_set_channel_count(struct dw_hdmi *hdmi, unsigned int cnt); +void dw_hdmi_set_channel_status(struct dw_hdmi *hdmi, u8 *channel_status); void dw_hdmi_set_channel_allocation(struct dw_hdmi *hdmi, unsigned int ca); void dw_hdmi_audio_enable(struct dw_hdmi *hdmi); void dw_hdmi_audio_disable(struct dw_hdmi *hdmi); -- 2.23.0.162.g0b9fbb3734-goog

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[alsa-devel] [PATCH 0/3] update supported sample format
by Shengjiu Wang 13 Sep '19

13 Sep '19

update supported sample format Shengjiu Wang (3): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++++--------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 93 +++++++++++++++++++++++++++++++++--- 3 files changed, 135 insertions(+), 30 deletions(-) -- 2.21.0

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