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

[alsa-devel] [PATCH 0/6] ASoC: platform cleanup
by Kuninori Morimoto 11 Jan '17

11 Jan '17

Hi Mark, Lars-Peter These removes un-used platform feature, and move last platform feature to component. If these are OK, all platform feature goes to component. Last platform piece is platform_list on soc-core.c Kuninori Morimoto (6): ASoC: remove .delay from snd_soc_platform_driver ASoC: remove .bespoke_trigger from snd_soc_platform_driver ASoC: remove snd_soc_platform_trigger() ASoC: add for_each_component{_safe} ASoC: add Component level struct snd_compr_ops ASoC: add Component level struct snd_pcm_ops include/sound/soc.h | 23 +++-- sound/soc/soc-compress.c | 228 ++++++++++++++++++++++++++++--------------- sound/soc/soc-core.c | 14 +-- sound/soc/soc-pcm.c | 244 +++++++++++++++++++++++++++++++++-------------- 4 files changed, 339 insertions(+), 170 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: ak4642: Replace mdelay function to msleep
by Kuninori Morimoto 11 Jan '17

11 Jan '17

From: Harunobu Kurokawa <harunobu.kurokawa.dn(a)renesas.com> Replace mdelay to msleep to avoid busy loop on ak4642_lout_event(). Otherwise, sometimes playback doesn't work correctly when pulseaudio was used. Signed-off-by: Harunobu Kurokawa <harunobu.kurokawa.dn(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 - based on latest kernel sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2609f95..23ab964 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -189,7 +189,7 @@ static int ak4642_lout_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: case SND_SOC_DAPM_POST_PMD: /* Power save mode OFF */ - mdelay(300); + msleep(300); snd_soc_update_bits(codec, SG_SL2, LOPS, 0); break; } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: mpc5200_psc_ac97: Remove unused DAI ID defines
by Lars-Peter Clausen 10 Jan '17

10 Jan '17

The DAI ID defines are back from the time when DAIs were referenced by a numerical ID. These days a string is used instead and the defines are unused. The last user of these defines was removed in commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support"). So remove the defines as well. This also means the mpc5200_psc_ac97.h file no longer has any content and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/fsl/efika-audio-fabric.c | 1 - sound/soc/fsl/mpc5200_psc_ac97.c | 1 - sound/soc/fsl/mpc5200_psc_ac97.h | 13 ------------- 3 files changed, 15 deletions(-) delete mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c index f200d1c..667f421 100644 --- a/sound/soc/fsl/efika-audio-fabric.c +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -26,7 +26,6 @@ #include <sound/soc.h> #include "mpc5200_dma.h" -#include "mpc5200_psc_ac97.h" #define DRV_NAME "efika-audio-fabric" diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 243700c..07ee355 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -25,7 +25,6 @@ #include <asm/mpc52xx_psc.h> #include "mpc5200_dma.h" -#include "mpc5200_psc_ac97.h" #define DRV_NAME "mpc5200-psc-ac97" diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h deleted file mode 100644 index e881e78..0000000 --- a/sound/soc/fsl/mpc5200_psc_ac97.h +++ /dev/null @@ -1,13 +0,0 @@ -/* - * Freescale MPC5200 PSC in AC97 mode - * ALSA SoC Digital Audio Interface (DAI) driver - * - */ - -#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ -#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ - -#define MPC5200_AC97_NORMAL 0 -#define MPC5200_AC97_SPDIF 1 - -#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */ -- 2.1.4

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[alsa-devel] [PATCH v5] ASoC: omap-mcbsp: Add PM QoS support for McBSP to prevent glitches
by Matt Ranostay 10 Jan '17

10 Jan '17

We can get audio errors if hitting deeper idle states on omaps: [alsa.c:230] error: Fatal problem with alsa output, error -5. [audio.c:614] error: Error in writing audio (Input/output error?)! This seems to happen with off mode idle enabled as power for the whole SoC may get cut off between filling the McBSP fifo using DMA. While active DMA blocks deeper idle states in hardware, McBSP activity does not seem to do so. Basing the QoS latency calculation on the FIFO size, threshold, sample rate, and channels. Based on the original patch by Tony Lindgren Link: https://patchwork.kernel.org/patch/9305867/ Cc: Tony Lindgren <tony(a)atomide.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Matt Ranostay <matt(a)ranostay.consulting> --- Changes from v1: * add calculations for latency per number of FIFO locations Changes from v2: * add missing mcbsp.h header change Changes from v3: * base the latency calculations on threshold, buffer size, sample rate, and channels Changes from v4: * using Peter Ujfalusi's suggestions for restoring a higher latency on audio stream completion, or if not applicable remove the QoS request sound/soc/omap/mcbsp.c | 37 +++++++++++++++++++++++++++++++++++++ sound/soc/omap/mcbsp.h | 3 +++ sound/soc/omap/omap-mcbsp.c | 11 ++++++++++- 3 files changed, 50 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 06fec5699cc8..f079650b71e3 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -25,6 +25,7 @@ #include <linux/io.h> #include <linux/slab.h> #include <linux/pm_runtime.h> +#include <linux/pm_qos.h> #include <linux/platform_data/asoc-ti-mcbsp.h> @@ -640,9 +641,28 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) */ void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) { + struct pm_qos_request *pm_qos_req = &mcbsp->pm_qos_req; + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; int enable_srg = 0; + int latency; u16 w; + /* Prevent omap hardware from hitting off between FIFO fills */ + if (!mcbsp->latency[stream2] || + mcbsp->latency[stream1] < mcbsp->latency[stream2]) + latency = mcbsp->latency[stream1]; + else + latency = mcbsp->latency[stream2]; + + if (latency) { + if (pm_qos_request_active(pm_qos_req)) + pm_qos_update_request(pm_qos_req, latency); + else + pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, + latency); + } + if (mcbsp->st_data) omap_st_start(mcbsp); @@ -731,6 +751,20 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) if (mcbsp->st_data) omap_st_stop(mcbsp); + + if (pm_qos_request_active(&mcbsp->pm_qos_req)) { + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : + SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : + SNDRV_PCM_STREAM_PLAYBACK; + + mcbsp->latency[stream1] = 0; + if (mcbsp->latency[stream2]) + pm_qos_update_request(&mcbsp->pm_qos_req, + mcbsp->latency[stream2]); + else + pm_qos_remove_request(&mcbsp->pm_qos_req); + } } int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) @@ -1098,6 +1132,9 @@ int omap_mcbsp_init(struct platform_device *pdev) void omap_mcbsp_cleanup(struct omap_mcbsp *mcbsp) { + if (pm_qos_request_active(&mcbsp->pm_qos_req)) + pm_qos_remove_request(&mcbsp->pm_qos_req); + if (mcbsp->pdata->buffer_size) sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index 61e93b1c185d..46ae1269a698 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -323,8 +323,11 @@ struct omap_mcbsp { unsigned int fmt; unsigned int in_freq; + unsigned int latency[2]; int clk_div; int wlen; + + struct pm_qos_request pm_qos_req; }; void omap_mcbsp_config(struct omap_mcbsp *mcbsp, diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d018e966e533..7e00938271c6 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -226,6 +226,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, int wlen, channels, wpf; int pkt_size = 0; unsigned int format, div, framesize, master; + unsigned int buffer_size = mcbsp->pdata->buffer_size; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); channels = params_channels(params); @@ -240,7 +241,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (mcbsp->pdata->buffer_size) { + if (buffer_size) { + int latency; + if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) { int period_words, max_thrsh; int divider = 0; @@ -271,6 +274,12 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* Use packet mode for non mono streams */ pkt_size = channels; } + + latency = ((((buffer_size - pkt_size) / channels) * 1000) + / (params->rate_num / params->rate_den)); + + mcbsp->latency[substream->stream] = latency; + omap_mcbsp_set_threshold(substream, pkt_size); } -- 2.10.2

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[alsa-devel] [PATCH] ASoC: tm2_wm5110: include gpio/consumer.h
by Arnd Bergmann 10 Jan '17

10 Jan '17

gpiod_set_value_cansleep is declared in linux/gpio/consumer.h, but that is not always included implicitly, so we have to include both gpio.h and gpio/consumer.h here: sound/soc/samsung/tm2_wm5110.c: In function 'tm2_mic_bias': sound/soc/samsung/tm2_wm5110.c:220:3: error: implicit declaration of function 'gpiod_set_value_cansleep';did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] Fixes: 1bfbc260a5b4 ("ASoC: samsung: Add machine driver for Exynos5433 based TM2 board") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/samsung/tm2_wm5110.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 5cdf7d19b87f..24cc9d63ce87 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -12,6 +12,7 @@ #include <linux/clk.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> #include <sound/pcm_params.h> -- 2.9.0

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[alsa-devel] [PATCH v5 0/3] ASoC: Intel: Skylake: Driver updates
by jeeja.kp＠intel.com 10 Jan '17

10 Jan '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series includes driver updates for the handling scenarios during System S0/S3 transitions when audio stream is stopped during system. o move DMA configuration in DSP Gateway Module widget event handler to ensure DMA is configured after resume. o Enable SSP MLCK using Supply widget. This will ensure the MCLK is enabled when the widget is power on. o Check DMA decouple register state before coupling/decoupling the DMA. o In case of HDMI pass-through pipe, reset only the FE Pipe when stream in in XRUN state. o Resume the stream from HW renderer position. o HDMI Codec pin and converter state need to be restored when stream is resumed. Changes in v2: - Addressed Mark's comment on patch 1, split this into 2 as this had additional changes other than configuring the DMA. - Addressed Takashi's comment on the usage of snd_hdac_updatel() Changes in v3: - Addressed Takashi's comment on superfluous shift used in reading the value of PPCTL register and also cleaned the condition for updating the reg value. Changes in v4: - Acked by Takashi for patch "ALSA: hda: check stream decoupled register state". - Removed compilation warning in patch "ASoC: hdac_hdmi: Enable pin and converter in prepare". - Remove 4 patches which are already applied. Changes in v5: - Removed the SSP MCLK patchs from this series, will rework and posted later based on Marks' comments. Jeeja KP (3): ASoC: Intel: Skylake: Don't reset pass-through pipe in BE prepare ASoC: Intel: Skylake: set the resume point to LPIB ASoC: hdac_hdmi: Enable pin and converter in prepare sound/soc/codecs/hdac_hdmi.c | 37 +++++++++++++++++-------------------- sound/soc/intel/skylake/skl-pcm.c | 6 +++--- 2 files changed, 20 insertions(+), 23 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH] ASoC: ak4642: Replace mdelay function to msleep
by Kuninori Morimoto 10 Jan '17

10 Jan '17

From: Harunobu Kurokawa <harunobu.kurokawa.dn(a)renesas.com> Replace mdelay to msleep to avoid busy loop on ak4642_lout_event(). Otherwise, sometimes playback doesn't work correctly when pulseaudio was used. Signed-off-by: Harunobu Kurokawa <harunobu.kurokawa.dn(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 40500cd..0205ae1 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -186,7 +186,7 @@ static int ak4642_lout_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: /* after widget power up */ /* Power save mode OFF */ - mdelay(popup_wait); + msleep(popup_wait); snd_soc_update_bits(codec, SG_SL2, LOPS, 0); break; case SND_SOC_DAPM_PRE_PMD: /* before widget power down */ -- 2.9.2

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[alsa-devel] [PATCH] Add a quirk for Plantronics BT600
by denk 10 Jan '17

10 Jan '17

Plantronics BT600 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x1" and "cannot get freq at ep 0x82". This patch adds the USB ID of the BT600 to quirks.c and avoids those error messages. Signed-off-by: Dennis Kadioglu <denk(a)post.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index b3fd238..9d747a9 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1136,6 +1136,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */ + case USB_ID(0x047F, 0x02F7): /* Plantronics BT-600 */ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */ ---

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[alsa-devel] [PATCH 1/4] alsa-lib:pcm: returning semop error code for semaphore up/down failures.
by sutar.mounesh＠gmail.com 10 Jan '17

10 Jan '17

From: Mounesh Sutar <sutar.mounesh(a)gmail.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 91e816c..aee38c5 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -263,7 +263,10 @@ static inline int snd_pcm_direct_semaphore_down(snd_pcm_direct_t *dmix, int sem_ { struct sembuf op[2] = { { sem_num, 0, 0 }, { sem_num, 1, SEM_UNDO } }; int err = semop(dmix->semid, op, 2); - if (err == 0) dmix->locked[sem_num]++; + if (err == 0) + dmix->locked[sem_num]++; + else if (err == -1) + err = -errno; return err; } @@ -271,7 +274,10 @@ static inline int snd_pcm_direct_semaphore_up(snd_pcm_direct_t *dmix, int sem_nu { struct sembuf op = { sem_num, -1, SEM_UNDO | IPC_NOWAIT }; int err = semop(dmix->semid, &op, 1); - if (err == 0) dmix->locked[sem_num]--; + if (err == 0) + dmix->locked[sem_num]--; + else if (err == -1) + err = -errno; return err; } -- 2.7.4

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[alsa-devel] [PATCH 0/2] alsa-utils: improve alsa-info.sh
by Pierre-Louis Bossart 10 Jan '17

10 Jan '17

Provide additional DMI information and filter ACPI/HID device status to help with platform quirks Pierre-Louis Bossart (2): alsa-info: provide more DMI information alsa-info: add ACPI device status alsa-info/alsa-info.sh | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) -- 2.7.4

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