March 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Allow complete shutdown of McASP when not in use
by Peter Ujfalusi 05 Mar '15

05 Mar '15

Rearrange the pm_runtime_get/put_sync calls so the IP will be turned off when it is not in use. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 23 ++++++++++++++--------- 1 file changed, 14 insertions(+), 9 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 031c1fb44ae7..a51197ac0b92 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -528,6 +528,7 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + pm_runtime_get_sync(mcasp->dev); switch (div_id) { case 0: /* MCLK divider */ mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, @@ -553,6 +554,7 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, return -EINVAL; } + pm_runtime_put_sync(mcasp->dev); return 0; } @@ -567,6 +569,7 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + pm_runtime_get_sync(mcasp->dev); if (dir == SND_SOC_CLOCK_OUT) { mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); @@ -579,6 +582,7 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, mcasp->sysclk_freq = freq; + pm_runtime_put_sync(mcasp->dev); return 0; } @@ -1078,6 +1082,9 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) u32 reg; int i; + if (!dai->active) + pm_runtime_get_sync(mcasp->dev); + for (i = 0; i < ARRAY_SIZE(context_regs); i++) context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]); @@ -1094,6 +1101,8 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai) context->xrsr_regs[i] = mcasp_get_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i)); + pm_runtime_put_sync(mcasp->dev); + return 0; } @@ -1104,6 +1113,8 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai) u32 reg; int i; + pm_runtime_get_sync(mcasp->dev); + for (i = 0; i < ARRAY_SIZE(context_regs); i++) mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]); @@ -1120,6 +1131,9 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai) mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i), context->xrsr_regs[i]); + if (!dai->active) + pm_runtime_put_sync(mcasp->dev); + return 0; } #else @@ -1427,13 +1441,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); - ret = pm_runtime_get_sync(&pdev->dev); - if (IS_ERR_VALUE(ret)) { - dev_err(&pdev->dev, "pm_runtime_get_sync() failed\n"); - pm_runtime_disable(&pdev->dev); - return ret; - } - mcasp->base = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); if (!mcasp->base) { dev_err(&pdev->dev, "ioremap failed\n"); @@ -1633,14 +1640,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return 0; err: - pm_runtime_put_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); return ret; } static int davinci_mcasp_remove(struct platform_device *pdev) { - pm_runtime_put_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); return 0; -- 2.3.0

2 7

[alsa-devel] Using simple-audio-card with imx6sl-evk
by Fabio Estevam 05 Mar '15

05 Mar '15

Hi, I am trying to get imx6sl-evk to work with simple-audio-card with the following changes: arch/arm/boot/dts/imx6sl-evk.dts | 52 ++++++++++++++++++++++++++++++---------- 1 file changed, 40 insertions(+), 12 deletions(-) diff --git a/arch/arm/boot/dts/imx6sl-evk.dts b/arch/arm/boot/dts/imx6sl-evk.dts index fda4932..24b9849 100644 --- a/arch/arm/boot/dts/imx6sl-evk.dts +++ b/arch/arm/boot/dts/imx6sl-evk.dts @@ -10,6 +10,7 @@ #include <dt-bindings/gpio/gpio.h> #include <dt-bindings/input/input.h> +#include <dt-bindings/sound/fsl-imx-audmux.h> #include "imx6sl.dtsi" / { @@ -92,19 +93,25 @@ }; sound { - compatible = "fsl,imx6sl-evk-wm8962", "fsl,imx-audio-wm8962"; - model = "wm8962-audio"; - ssi-controller = <&ssi2>; - audio-codec = <&codec>; - audio-routing = + compatible = "simple-audio-card"; + simple-audio-card,name = "wm8962-audio"; + simple-audio-card,format = "i2s"; + simple-audio-card,bitclock-master = <&dailink_master>; + simple-audio-card,frame-master = <&dailink_master>; + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = "Headphone Jack", "HPOUTL", - "Headphone Jack", "HPOUTR", - "Ext Spk", "SPKOUTL", - "Ext Spk", "SPKOUTR", - "AMIC", "MICBIAS", - "IN3R", "AMIC"; - mux-int-port = <2>; - mux-ext-port = <3>; + "Headphone Jack", "HPOUTR"; + + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + + dailink_master: simple-audio-card,codec { + sound-dai = <&codec>; + }; }; }; @@ -112,6 +119,26 @@ pinctrl-names = "default"; pinctrl-0 = <&pinctrl_audmux3>; status = "okay"; + + ssi2 { + fsl,audmux-port = <1>; + fsl,port-config = < + (IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(2) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(2)) + IMX_AUDMUX_V2_PDCR_RXDSEL(2) + >; + }; + + audmuxpins3 { + fsl,audmux-port = <2>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN + IMX_AUDMUX_V2_PDCR_RXDSEL(1) + >; + }; }; &ecspi1 { @@ -255,6 +282,7 @@ compatible = "wlf,wm8962"; reg = <0x1a>; clocks = <&clks IMX6SL_CLK_EXTERN_AUDIO>; + #sound-dai-cells = <0>; DCVDD-supply = <&vgen3_reg>; DBVDD-supply = <&reg_aud3v>; AVDD-supply = <&vgen3_reg>; Then I get the following probe error: [ 2.030249] input: WM8962 Beep Generator as /devices/soc0/soc/2100000.aips-bus/21a4000.i2c/i2c-1/1-001a/input/input1 [ 2.042388] wm8962 1-001a: simple-card: set_sysclk error [ 2.047805] asoc-simple-card sound: ASoC: failed to init 202c000.ssi-wm8962: -22 [ 2.055242] asoc-simple-card sound: ASoC: failed to instantiate card -22 [ 2.065117] asoc-simple-card: probe of sound failed with error -22 I compared how imx-wm8962 calls snd_soc_dai_set_sysclk and did the same on simple card (quick hack): --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -134,7 +134,7 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, } if (set->sysclk) { - ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); + ret = snd_soc_dai_set_sysclk(dai, 1, set->sysclk, 0); if (ret && ret != -ENOTSUPP) { dev_err(dai->dev, "simple-card: set_sysclk error\n"); goto err; ,and this allows the probe to succeed. When I try to play I get: root@freescale /home$ aplay clarinet.wav [ 16.424340] wm8962 1-001a: Unsupported sysclk ratio 680 Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono and the track is played with the wrong frequency and wrong sysclk ratio. Any suggestions as to how to use simple-card on this board? Thanks, Fabio Estevam

2 1

[alsa-devel] [PATCH 0/5] ASoC: davinci: Use edma-pcm and remove davinci-pcm
by Peter Ujfalusi 05 Mar '15

05 Mar '15

Hi, This series will deprecate and removes the davinci-pcm platform driver and converts the daVinci DAI drivers to use the edma-pcm. The main feature of davinci-pcm has been the so called ping-pong mode, which can help in situations when the McASP/McBSP/ASP/VCIF experiences underrun or overrun in playback or capture. This was due to the fact the davinci-pcm w/o ping-pong needed to reprogram the paRAM slot after each period, this could cause delay in DMA operation, which leads to starvation. The edm-pcm does not need reprogramming and it seams to be working as good as the davinci-pcm with ping-pong. I have tested this series on OMAP-L138 EVM with McASP and McBSP. VCIF has been only compile tested since I do not have access to the HW. The edma-pcm is already used by AM335x and AM437x and it has been tested on those platforms extensively. After this series we no longer have legacy eDMA users which clears the path for the eDMA driver stack cleanup. Regards, Peter --- Peter Ujfalusi (5): ASoC: davinci: Select SND_EDMA_SOC when SND_DAVINCI_SOC is enabled ASoC: davinci-i2s: Convert to use edma-pcm ASoC: davinci-vcif: Convert to use edma-pcm ASoC: davinci-mcasp: Deprecate the use of davinci-pcm in favor of edma-pcm ASoC: davinci: Remove unused davinci-pcm platform driver sound/soc/davinci/Kconfig | 18 +- sound/soc/davinci/Makefile | 2 - sound/soc/davinci/davinci-i2s.c | 67 ++- sound/soc/davinci/davinci-mcasp.c | 87 +--- sound/soc/davinci/davinci-pcm.c | 861 -------------------------------------- sound/soc/davinci/davinci-pcm.h | 41 -- sound/soc/davinci/davinci-vcif.c | 55 ++- 7 files changed, 79 insertions(+), 1052 deletions(-) delete mode 100644 sound/soc/davinci/davinci-pcm.c delete mode 100644 sound/soc/davinci/davinci-pcm.h -- 2.3.0

3 13

[alsa-devel] [ALSA-LIB][PATCH] Add support for compress offload API
by Qais Yousef 05 Mar '15

05 Mar '15

Merge tinycompress library into alsa-lib using LGPL license. Only function names were modified to match the coding style in alsa-lib, prepending snd_compr to function names and structs. Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- configure.ac | 9 + include/Makefile.am | 4 + include/compress.h | 245 ++++++++++++++++ include/sound/compress_offload.h | 191 +++++++++++++ include/sound/compress_params.h | 404 ++++++++++++++++++++++++++ src/Makefile.am | 7 + src/compress/Makefile.am | 8 + src/compress/compress.c | 599 +++++++++++++++++++++++++++++++++++++++ 8 files changed, 1467 insertions(+) create mode 100644 include/compress.h create mode 100644 include/sound/compress_offload.h create mode 100644 include/sound/compress_params.h create mode 100644 src/compress/Makefile.am create mode 100644 src/compress/compress.c diff --git a/configure.ac b/configure.ac index f0995e3ae787..a768730781e0 100644 --- a/configure.ac +++ b/configure.ac @@ -368,6 +368,9 @@ AC_ARG_ENABLE(mixer, AC_ARG_ENABLE(pcm, AS_HELP_STRING([--disable-pcm], [disable the PCM component]), [build_pcm="$enableval"], [build_pcm="yes"]) +AC_ARG_ENABLE(compress, + AS_HELP_STRING([--disable-compress], [disable the compress component]), + [build_compress="$enableval"], [build_compress="yes"]) AC_ARG_ENABLE(rawmidi, AS_HELP_STRING([--disable-rawmidi], [disable the raw MIDI component]), [build_rawmidi="$enableval"], [build_rawmidi="yes"]) @@ -418,6 +421,7 @@ AC_SUBST(PYTHON_INCLUDES) AM_CONDITIONAL([BUILD_MIXER], [test x$build_mixer = xyes]) AM_CONDITIONAL([BUILD_PCM], [test x$build_pcm = xyes]) +AM_CONDITIONAL([BUILD_COMPRESS], [test x$build_compress = xyes]) AM_CONDITIONAL([BUILD_RAWMIDI], [test x$build_rawmidi = xyes]) AM_CONDITIONAL([BUILD_HWDEP], [test x$build_hwdep = xyes]) AM_CONDITIONAL([BUILD_SEQ], [test x$build_seq = xyes]) @@ -431,6 +435,9 @@ fi if test "$build_pcm" = "yes"; then AC_DEFINE([BUILD_PCM], "1", [Build PCM component]) fi +if test "$build_compress" = "yes"; then + AC_DEFINE([BUILD_COMPRESS], "1", [Build compress component]) +fi if test "$build_rawmidi" = "yes"; then AC_DEFINE([BUILD_RAWMIDI], "1", [Build raw MIDI component]) fi @@ -641,6 +648,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ include/Makefile include/sound/Makefile src/Versions src/Makefile \ src/control/Makefile src/mixer/Makefile \ src/pcm/Makefile src/pcm/scopes/Makefile \ + src/compress/Makefile \ src/rawmidi/Makefile src/timer/Makefile \ src/hwdep/Makefile src/seq/Makefile src/ucm/Makefile \ src/alisp/Makefile \ @@ -693,6 +701,7 @@ cat >> include/asoundlib.h <<EOF #include <alsa/conf.h> EOF test "$build_pcm" = "yes" && echo "#include <alsa/pcm.h>" >> include/asoundlib.h +test "$build_compress" = "yes" && echo "#include <alsa/compress.h>" >> include/asoundlib.h test "$build_rawmidi" = "yes" && echo "#include <alsa/rawmidi.h>" >> include/asoundlib.h test "$build_pcm" = "yes" && echo "#include <alsa/timer.h>" >> include/asoundlib.h test "$build_hwdep" = "yes" && echo "#include <alsa/hwdep.h>" >> include/asoundlib.h diff --git a/include/Makefile.am b/include/Makefile.am index 4baa03af69e1..395a2ed60d70 100644 --- a/include/Makefile.am +++ b/include/Makefile.am @@ -30,6 +30,10 @@ alsainclude_HEADERS += pcm_ioplug.h endif endif +if BUILD_COMPRESS +alsainclude_HEADERS += compress.h +endif + if BUILD_RAWMIDI alsainclude_HEADERS += rawmidi.h endif diff --git a/include/compress.h b/include/compress.h new file mode 100644 index 000000000000..250ce0c3f7c4 --- /dev/null +++ b/include/compress.h @@ -0,0 +1,245 @@ +/* + * tinycompress library for compress audio offload in alsa + * Copyright (c) 2011-2012, Intel Corporation. + * + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + + +#ifndef __ALSA_COMPRESS_H +#define __ALSA_COMPRESS_H + +#include <stdbool.h> +#include <linux/types.h> + +#if defined(__cplusplus) +extern "C" { +#endif +/* + * struct compr_config: config structure, needs to be filled by app + * If fragment_size or fragments are zero, this means "don't care" + * and tinycompress will choose values that the driver supports + * + * @fragment_size: size of fragment requested, in bytes + * @fragments: number of fragments + * @codec: codec type and parameters requested + */ +struct snd_compr_config { + __u32 fragment_size; + __u32 fragments; + struct snd_codec *codec; +}; + +struct snd_compr_gapless_mdata { + __u32 encoder_delay; + __u32 encoder_padding; +}; + +#define COMPRESS_OUT 0x20000000 +#define COMPRESS_IN 0x10000000 + +struct snd_compr; +struct snd_compr_tstamp; + +/* + * snd_compr_open: open a new compress stream + * returns the valid struct snd_compr on success, NULL on failure + * If config does not specify a requested fragment size, on return + * it will be updated with the size and number of fragments that + * were configured + * + * @card: sound card number + * @device: device number + * @flags: device flags can be COMPRESS_OUT or COMPRESS_IN + * @config: stream config requested. Returns actual fragment config + */ +struct snd_compr *snd_compr_open(unsigned int card, unsigned int device, + unsigned int flags, struct snd_compr_config *config); + +/* + * snd_compr_close: close the compress stream + * + * @compr: compress stream to be closed + */ +void snd_compr_close(struct snd_compr *compr); + +/* + * snd_compr_get_hpointer: get the hw timestamp + * return 0 on success, negative on error + * + * @compr: compress stream on which query is made + * @avail: buffer availble for write/read, in bytes + * @tstamp: hw time + */ +int snd_compr_get_hpointer(struct snd_compr *compr, + unsigned int *avail, struct timespec *tstamp); + + +/* + * snd_compr_get_tstamp: get the raw hw timestamp + * return 0 on success, negative on error + * + * @compr: compress stream on which query is made + * @samples: number of decoded samples played + * @sampling_rate: sampling rate of decoded samples + */ +int snd_compr_get_tstamp(struct snd_compr *compr, + unsigned int *samples, unsigned int *sampling_rate); + +/* + * snd_compr_write: write data to the compress stream + * return bytes written on success, negative on error + * By default this is a blocking call and will not return + * until all bytes have been written or there was a + * write error. + * If non-blocking mode has been enabled with snd_compr_nonblock(), + * this function will write all bytes that can be written without + * blocking and will then return the number of bytes successfully + * written. If the return value is not an error and is < size + * the caller can use snd_compr_wait() to block until the driver + * is ready for more data. + * + * @compr: compress stream to be written to + * @buf: pointer to data + * @size: number of bytes to be written + */ +int snd_compr_write(struct snd_compr *compr, const void *buf, unsigned int size); + +/* + * snd_compr_read: read data from the compress stream + * return bytes read on success, negative on error + * By default this is a blocking call and will block until + * size bytes have been written or there was a read error. + * If non-blocking mode was enabled using snd_compr_nonblock() + * the behaviour will change to read only as many bytes as + * are currently available (if no bytes are available it + * will return immediately). The caller can then use + * snd_compr_wait() to block until more bytes are available. + * + * @compr: compress stream from where data is to be read + * @buf: pointer to data buffer + * @size: size of given buffer + */ +int snd_compr_read(struct snd_compr *compr, void *buf, unsigned int size); + +/* + * snd_compr_start: start the compress stream + * return 0 on success, negative on error + * + * @compr: compress stream to be started + */ +int snd_compr_start(struct snd_compr *compr); + +/* + * snd_compr_stop: stop the compress stream + * return 0 on success, negative on error + * + * @compr: compress stream to be stopped + */ +int snd_compr_stop(struct snd_compr *compr); + +/* + * snd_compr_pause: pause the compress stream + * return 0 on success, negative on error + * + * @compr: compress stream to be paused + */ +int snd_compr_pause(struct snd_compr *compr); + +/* + * snd_compr_resume: resume the compress stream + * return 0 on success, negative on error + * + * @compr: compress stream to be resumed + */ +int snd_compr_resume(struct snd_compr *compr); + +/* + * snd_compr_drain: drain the compress stream + * return 0 on success, negative on error + * + * @compr: compress stream to be drain + */ +int snd_compr_drain(struct snd_compr *compr); + +/* + * snd_compr_next_track: set the next track for stream + * + * return 0 on success, negative on error + * + * @compr: compress stream to be transistioned to next track + */ +int snd_compr_next_track(struct snd_compr *compr); + +/* + * snd_compr_partial_drain: drain will return after the last frame is decoded + * by DSP and will play the , All the data written into compressed + * ring buffer is decoded + * + * return 0 on success, negative on error + * + * @compr: compress stream to be drain + */ +int snd_compr_partial_drain(struct snd_compr *compr); + +/* + * snd_compr_set_gapless_metadata: set gapless metadata of a compress strem + * + * return 0 on success, negative on error + * + * @compr: compress stream for which metadata has to set + * @mdata: metadata encoder delay and padding + */ + +int snd_compr_set_gapless_metadata(struct snd_compr *compr, + struct snd_compr_gapless_mdata *mdata); + +/* + * snd_compr_is_codec_supported: check if the given codec is supported + * returns true when supported, false if not + * + * @card: sound card number + * @device: device number + * @flags: stream flags + * @codec: codec type and parameters to be checked + */ +bool snd_compr_is_codec_supported(unsigned int card, unsigned int device, + unsigned int flags, struct snd_codec *codec); + +/* + * snd_compr_set_max_poll_wait: set the maximum time tinycompress + * will wait for driver to signal a poll(). Interval is in + * milliseconds. + * Pass interval of -1 to disable timeout and make poll() wait + * until driver signals. + * If this function is not used the timeout defaults to 20 seconds. + */ +void snd_compr_set_max_poll_wait(struct snd_compr *compr, int milliseconds); + +/* Enable or disable non-blocking mode for write and read */ +void snd_compr_nonblock(struct snd_compr *compr, int nonblock); + +/* Wait for ring buffer to ready for next read or write */ +int snd_compr_wait(struct snd_compr *compr, int timeout_ms); + +int snd_compr_is_running(struct snd_compr *compr); + +int snd_compr_is_ready(struct snd_compr *compr); + +/* Returns a human readable reason for the last error */ +const char *snd_compr_get_error(struct snd_compr *compr); + +#endif diff --git a/include/sound/compress_offload.h b/include/sound/compress_offload.h new file mode 100644 index 000000000000..22ed8cb7800b --- /dev/null +++ b/include/sound/compress_offload.h @@ -0,0 +1,191 @@ +/* + * compress_offload.h - compress offload header definations + * + * Copyright (C) 2011 Intel Corporation + * Authors: Vinod Koul <vinod.koul(a)linux.intel.com> + * Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.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; version 2 of the License. + * + * 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. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ +#ifndef __COMPRESS_OFFLOAD_H +#define __COMPRESS_OFFLOAD_H + +#include <linux/types.h> +#include <sound/asound.h> +#include <sound/compress_params.h> + + +#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 2) +/** + * struct snd_compressed_buffer - compressed buffer + * @fragment_size: size of buffer fragment in bytes + * @fragments: number of such fragments + */ +struct snd_compressed_buffer { + __u32 fragment_size; + __u32 fragments; +} __attribute__((packed, aligned(4))); + +/** + * struct snd_compr_params - compressed stream params + * @buffer: buffer description + * @codec: codec parameters + * @no_wake_mode: dont wake on fragment elapsed + */ +struct snd_compr_params { + struct snd_compressed_buffer buffer; + struct snd_codec codec; + __u8 no_wake_mode; +} __attribute__((packed, aligned(4))); + +/** + * struct snd_compr_tstamp - timestamp descriptor + * @byte_offset: Byte offset in ring buffer to DSP + * @copied_total: Total number of bytes copied from/to ring buffer to/by DSP + * @pcm_frames: Frames decoded or encoded by DSP. This field will evolve by + * large steps and should only be used to monitor encoding/decoding + * progress. It shall not be used for timing estimates. + * @pcm_io_frames: Frames rendered or received by DSP into a mixer or an audio + * output/input. This field should be used for A/V sync or time estimates. + * @sampling_rate: sampling rate of audio + */ +struct snd_compr_tstamp { + __u32 byte_offset; + __u32 copied_total; + __u32 pcm_frames; + __u32 pcm_io_frames; + __u32 sampling_rate; +} __attribute__((packed, aligned(4))); + +/** + * struct snd_compr_avail - avail descriptor + * @avail: Number of bytes available in ring buffer for writing/reading + * @tstamp: timestamp infomation + */ +struct snd_compr_avail { + __u64 avail; + struct snd_compr_tstamp tstamp; +} __attribute__((packed, aligned(4))); + +enum snd_compr_direction { + SND_COMPRESS_PLAYBACK = 0, + SND_COMPRESS_CAPTURE +}; + +/** + * struct snd_compr_caps - caps descriptor + * @codecs: pointer to array of codecs + * @direction: direction supported. Of type snd_compr_direction + * @min_fragment_size: minimum fragment supported by DSP + * @max_fragment_size: maximum fragment supported by DSP + * @min_fragments: min fragments supported by DSP + * @max_fragments: max fragments supported by DSP + * @num_codecs: number of codecs supported + * @reserved: reserved field + */ +struct snd_compr_caps { + __u32 num_codecs; + __u32 direction; + __u32 min_fragment_size; + __u32 max_fragment_size; + __u32 min_fragments; + __u32 max_fragments; + __u32 codecs[MAX_NUM_CODECS]; + __u32 reserved[11]; +} __attribute__((packed, aligned(4))); + +/** + * struct snd_compr_codec_caps - query capability of codec + * @codec: codec for which capability is queried + * @num_descriptors: number of codec descriptors + * @descriptor: array of codec capability descriptor + */ +struct snd_compr_codec_caps { + __u32 codec; + __u32 num_descriptors; + struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS]; +} __attribute__((packed, aligned(4))); + +/** + * enum sndrv_compress_encoder + * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the + * end of the track + * @SNDRV_COMPRESS_ENCODER_DELAY: no of samples inserted by the encoder at the + * beginning of the track + */ +enum sndrv_compress_encoder { + SNDRV_COMPRESS_ENCODER_PADDING = 1, + SNDRV_COMPRESS_ENCODER_DELAY = 2, +}; + +/** + * struct snd_compr_metadata - compressed stream metadata + * @key: key id + * @value: key value + */ +struct snd_compr_metadata { + __u32 key; + __u32 value[8]; +} __attribute__((packed, aligned(4))); + +/** + * compress path ioctl definitions + * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP + * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec + * SNDRV_COMPRESS_SET_PARAMS: Set codec and stream parameters + * Note: only codec params can be changed runtime and stream params cant be + * SNDRV_COMPRESS_GET_PARAMS: Query codec params + * SNDRV_COMPRESS_TSTAMP: get the current timestamp value + * SNDRV_COMPRESS_AVAIL: get the current buffer avail value. + * This also queries the tstamp properties + * SNDRV_COMPRESS_PAUSE: Pause the running stream + * SNDRV_COMPRESS_RESUME: resume a paused stream + * SNDRV_COMPRESS_START: Start a stream + * SNDRV_COMPRESS_STOP: stop a running stream, discarding ring buffer content + * and the buffers currently with DSP + * SNDRV_COMPRESS_DRAIN: Play till end of buffers and stop after that + * SNDRV_COMPRESS_IOCTL_VERSION: Query the API version + */ +#define SNDRV_COMPRESS_IOCTL_VERSION _IOR('C', 0x00, int) +#define SNDRV_COMPRESS_GET_CAPS _IOWR('C', 0x10, struct snd_compr_caps) +#define SNDRV_COMPRESS_GET_CODEC_CAPS _IOWR('C', 0x11,\ + struct snd_compr_codec_caps) +#define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params) +#define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec) +#define SNDRV_COMPRESS_SET_METADATA _IOW('C', 0x14,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_GET_METADATA _IOWR('C', 0x15,\ + struct snd_compr_metadata) +#define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) +#define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) +#define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) +#define SNDRV_COMPRESS_RESUME _IO('C', 0x31) +#define SNDRV_COMPRESS_START _IO('C', 0x32) +#define SNDRV_COMPRESS_STOP _IO('C', 0x33) +#define SNDRV_COMPRESS_DRAIN _IO('C', 0x34) +#define SNDRV_COMPRESS_NEXT_TRACK _IO('C', 0x35) +#define SNDRV_COMPRESS_PARTIAL_DRAIN _IO('C', 0x36) +/* + * TODO + * 1. add mmap support + * + */ +#define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */ +#define SND_COMPR_TRIGGER_NEXT_TRACK 8 +#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9 +#endif diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h new file mode 100644 index 000000000000..d9bd9ca0d5b0 --- /dev/null +++ b/include/sound/compress_params.h @@ -0,0 +1,404 @@ +/* + * compress_params.h - codec types and parameters for compressed data + * streaming interface + * + * Copyright (C) 2011 Intel Corporation + * Authors: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> + * Vinod Koul <vinod.koul(a)linux.intel.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; version 2 of the License. + * + * 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. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * The definitions in this file are derived from the OpenMAX AL version 1.1 + * and OpenMAX IL v 1.1.2 header files which contain the copyright notice below. + * + * Copyright (c) 2007-2010 The Khronos Group Inc. + * + * Permission is hereby granted, free of charge, to any person obtaining + * a copy of this software and/or associated documentation files (the + * "Materials "), to deal in the Materials without restriction, including + * without limitation the rights to use, copy, modify, merge, publish, + * distribute, sublicense, and/or sell copies of the Materials, and to + * permit persons to whom the Materials are furnished to do so, subject to + * the following conditions: + * + * The above copyright notice and this permission notice shall be included + * in all copies or substantial portions of the Materials. + * + * THE MATERIALS ARE PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS + * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY + * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, + * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE + * MATERIALS OR THE USE OR OTHER DEALINGS IN THE MATERIALS. + * + */ +#ifndef __SND_COMPRESS_PARAMS_H +#define __SND_COMPRESS_PARAMS_H + +#include <linux/types.h> + +/* AUDIO CODECS SUPPORTED */ +#define MAX_NUM_CODECS 32 +#define MAX_NUM_CODEC_DESCRIPTORS 32 +#define MAX_NUM_BITRATES 32 +#define MAX_NUM_SAMPLE_RATES 32 + +/* Codecs are listed linearly to allow for extensibility */ +#define SND_AUDIOCODEC_PCM ((__u32) 0x00000001) +#define SND_AUDIOCODEC_MP3 ((__u32) 0x00000002) +#define SND_AUDIOCODEC_AMR ((__u32) 0x00000003) +#define SND_AUDIOCODEC_AMRWB ((__u32) 0x00000004) +#define SND_AUDIOCODEC_AMRWBPLUS ((__u32) 0x00000005) +#define SND_AUDIOCODEC_AAC ((__u32) 0x00000006) +#define SND_AUDIOCODEC_WMA ((__u32) 0x00000007) +#define SND_AUDIOCODEC_REAL ((__u32) 0x00000008) +#define SND_AUDIOCODEC_VORBIS ((__u32) 0x00000009) +#define SND_AUDIOCODEC_FLAC ((__u32) 0x0000000A) +#define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B) +#define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) +#define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) +#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_G729 + +/* + * Profile and modes are listed with bit masks. This allows for a + * more compact representation of fields that will not evolve + * (in contrast to the list of codecs) + */ + +#define SND_AUDIOPROFILE_PCM ((__u32) 0x00000001) + +/* MP3 modes are only useful for encoders */ +#define SND_AUDIOCHANMODE_MP3_MONO ((__u32) 0x00000001) +#define SND_AUDIOCHANMODE_MP3_STEREO ((__u32) 0x00000002) +#define SND_AUDIOCHANMODE_MP3_JOINTSTEREO ((__u32) 0x00000004) +#define SND_AUDIOCHANMODE_MP3_DUAL ((__u32) 0x00000008) + +#define SND_AUDIOPROFILE_AMR ((__u32) 0x00000001) + +/* AMR modes are only useful for encoders */ +#define SND_AUDIOMODE_AMR_DTX_OFF ((__u32) 0x00000001) +#define SND_AUDIOMODE_AMR_VAD1 ((__u32) 0x00000002) +#define SND_AUDIOMODE_AMR_VAD2 ((__u32) 0x00000004) + +#define SND_AUDIOSTREAMFORMAT_UNDEFINED ((__u32) 0x00000000) +#define SND_AUDIOSTREAMFORMAT_CONFORMANCE ((__u32) 0x00000001) +#define SND_AUDIOSTREAMFORMAT_IF1 ((__u32) 0x00000002) +#define SND_AUDIOSTREAMFORMAT_IF2 ((__u32) 0x00000004) +#define SND_AUDIOSTREAMFORMAT_FSF ((__u32) 0x00000008) +#define SND_AUDIOSTREAMFORMAT_RTPPAYLOAD ((__u32) 0x00000010) +#define SND_AUDIOSTREAMFORMAT_ITU ((__u32) 0x00000020) + +#define SND_AUDIOPROFILE_AMRWB ((__u32) 0x00000001) + +/* AMRWB modes are only useful for encoders */ +#define SND_AUDIOMODE_AMRWB_DTX_OFF ((__u32) 0x00000001) +#define SND_AUDIOMODE_AMRWB_VAD1 ((__u32) 0x00000002) +#define SND_AUDIOMODE_AMRWB_VAD2 ((__u32) 0x00000004) + +#define SND_AUDIOPROFILE_AMRWBPLUS ((__u32) 0x00000001) + +#define SND_AUDIOPROFILE_AAC ((__u32) 0x00000001) + +/* AAC modes are required for encoders and decoders */ +#define SND_AUDIOMODE_AAC_MAIN ((__u32) 0x00000001) +#define SND_AUDIOMODE_AAC_LC ((__u32) 0x00000002) +#define SND_AUDIOMODE_AAC_SSR ((__u32) 0x00000004) +#define SND_AUDIOMODE_AAC_LTP ((__u32) 0x00000008) +#define SND_AUDIOMODE_AAC_HE ((__u32) 0x00000010) +#define SND_AUDIOMODE_AAC_SCALABLE ((__u32) 0x00000020) +#define SND_AUDIOMODE_AAC_ERLC ((__u32) 0x00000040) +#define SND_AUDIOMODE_AAC_LD ((__u32) 0x00000080) +#define SND_AUDIOMODE_AAC_HE_PS ((__u32) 0x00000100) +#define SND_AUDIOMODE_AAC_HE_MPS ((__u32) 0x00000200) + +/* AAC formats are required for encoders and decoders */ +#define SND_AUDIOSTREAMFORMAT_MP2ADTS ((__u32) 0x00000001) +#define SND_AUDIOSTREAMFORMAT_MP4ADTS ((__u32) 0x00000002) +#define SND_AUDIOSTREAMFORMAT_MP4LOAS ((__u32) 0x00000004) +#define SND_AUDIOSTREAMFORMAT_MP4LATM ((__u32) 0x00000008) +#define SND_AUDIOSTREAMFORMAT_ADIF ((__u32) 0x00000010) +#define SND_AUDIOSTREAMFORMAT_MP4FF ((__u32) 0x00000020) +#define SND_AUDIOSTREAMFORMAT_RAW ((__u32) 0x00000040) + +#define SND_AUDIOPROFILE_WMA7 ((__u32) 0x00000001) +#define SND_AUDIOPROFILE_WMA8 ((__u32) 0x00000002) +#define SND_AUDIOPROFILE_WMA9 ((__u32) 0x00000004) +#define SND_AUDIOPROFILE_WMA10 ((__u32) 0x00000008) + +#define SND_AUDIOMODE_WMA_LEVEL1 ((__u32) 0x00000001) +#define SND_AUDIOMODE_WMA_LEVEL2 ((__u32) 0x00000002) +#define SND_AUDIOMODE_WMA_LEVEL3 ((__u32) 0x00000004) +#define SND_AUDIOMODE_WMA_LEVEL4 ((__u32) 0x00000008) +#define SND_AUDIOMODE_WMAPRO_LEVELM0 ((__u32) 0x00000010) +#define SND_AUDIOMODE_WMAPRO_LEVELM1 ((__u32) 0x00000020) +#define SND_AUDIOMODE_WMAPRO_LEVELM2 ((__u32) 0x00000040) +#define SND_AUDIOMODE_WMAPRO_LEVELM3 ((__u32) 0x00000080) + +#define SND_AUDIOSTREAMFORMAT_WMA_ASF ((__u32) 0x00000001) +/* + * Some implementations strip the ASF header and only send ASF packets + * to the DSP + */ +#define SND_AUDIOSTREAMFORMAT_WMA_NOASF_HDR ((__u32) 0x00000002) + +#define SND_AUDIOPROFILE_REALAUDIO ((__u32) 0x00000001) + +#define SND_AUDIOMODE_REALAUDIO_G2 ((__u32) 0x00000001) +#define SND_AUDIOMODE_REALAUDIO_8 ((__u32) 0x00000002) +#define SND_AUDIOMODE_REALAUDIO_10 ((__u32) 0x00000004) +#define SND_AUDIOMODE_REALAUDIO_SURROUND ((__u32) 0x00000008) + +#define SND_AUDIOPROFILE_VORBIS ((__u32) 0x00000001) + +#define SND_AUDIOMODE_VORBIS ((__u32) 0x00000001) + +#define SND_AUDIOPROFILE_FLAC ((__u32) 0x00000001) + +/* + * Define quality levels for FLAC encoders, from LEVEL0 (fast) + * to LEVEL8 (best) + */ +#define SND_AUDIOMODE_FLAC_LEVEL0 ((__u32) 0x00000001) +#define SND_AUDIOMODE_FLAC_LEVEL1 ((__u32) 0x00000002) +#define SND_AUDIOMODE_FLAC_LEVEL2 ((__u32) 0x00000004) +#define SND_AUDIOMODE_FLAC_LEVEL3 ((__u32) 0x00000008) +#define SND_AUDIOMODE_FLAC_LEVEL4 ((__u32) 0x00000010) +#define SND_AUDIOMODE_FLAC_LEVEL5 ((__u32) 0x00000020) +#define SND_AUDIOMODE_FLAC_LEVEL6 ((__u32) 0x00000040) +#define SND_AUDIOMODE_FLAC_LEVEL7 ((__u32) 0x00000080) +#define SND_AUDIOMODE_FLAC_LEVEL8 ((__u32) 0x00000100) + +#define SND_AUDIOSTREAMFORMAT_FLAC ((__u32) 0x00000001) +#define SND_AUDIOSTREAMFORMAT_FLAC_OGG ((__u32) 0x00000002) + +/* IEC61937 payloads without CUVP and preambles */ +#define SND_AUDIOPROFILE_IEC61937 ((__u32) 0x00000001) +/* IEC61937 with S/PDIF preambles+CUVP bits in 32-bit containers */ +#define SND_AUDIOPROFILE_IEC61937_SPDIF ((__u32) 0x00000002) + +/* + * IEC modes are mandatory for decoders. Format autodetection + * will only happen on the DSP side with mode 0. The PCM mode should + * not be used, the PCM codec should be used instead. + */ +#define SND_AUDIOMODE_IEC_REF_STREAM_HEADER ((__u32) 0x00000000) +#define SND_AUDIOMODE_IEC_LPCM ((__u32) 0x00000001) +#define SND_AUDIOMODE_IEC_AC3 ((__u32) 0x00000002) +#define SND_AUDIOMODE_IEC_MPEG1 ((__u32) 0x00000004) +#define SND_AUDIOMODE_IEC_MP3 ((__u32) 0x00000008) +#define SND_AUDIOMODE_IEC_MPEG2 ((__u32) 0x00000010) +#define SND_AUDIOMODE_IEC_AACLC ((__u32) 0x00000020) +#define SND_AUDIOMODE_IEC_DTS ((__u32) 0x00000040) +#define SND_AUDIOMODE_IEC_ATRAC ((__u32) 0x00000080) +#define SND_AUDIOMODE_IEC_SACD ((__u32) 0x00000100) +#define SND_AUDIOMODE_IEC_EAC3 ((__u32) 0x00000200) +#define SND_AUDIOMODE_IEC_DTS_HD ((__u32) 0x00000400) +#define SND_AUDIOMODE_IEC_MLP ((__u32) 0x00000800) +#define SND_AUDIOMODE_IEC_DST ((__u32) 0x00001000) +#define SND_AUDIOMODE_IEC_WMAPRO ((__u32) 0x00002000) +#define SND_AUDIOMODE_IEC_REF_CXT ((__u32) 0x00004000) +#define SND_AUDIOMODE_IEC_HE_AAC ((__u32) 0x00008000) +#define SND_AUDIOMODE_IEC_HE_AAC2 ((__u32) 0x00010000) +#define SND_AUDIOMODE_IEC_MPEG_SURROUND ((__u32) 0x00020000) + +#define SND_AUDIOPROFILE_G723_1 ((__u32) 0x00000001) + +#define SND_AUDIOMODE_G723_1_ANNEX_A ((__u32) 0x00000001) +#define SND_AUDIOMODE_G723_1_ANNEX_B ((__u32) 0x00000002) +#define SND_AUDIOMODE_G723_1_ANNEX_C ((__u32) 0x00000004) + +#define SND_AUDIOPROFILE_G729 ((__u32) 0x00000001) + +#define SND_AUDIOMODE_G729_ANNEX_A ((__u32) 0x00000001) +#define SND_AUDIOMODE_G729_ANNEX_B ((__u32) 0x00000002) + +/* <FIXME: multichannel encoders aren't supported for now. Would need + an additional definition of channel arrangement> */ + +/* VBR/CBR definitions */ +#define SND_RATECONTROLMODE_CONSTANTBITRATE ((__u32) 0x00000001) +#define SND_RATECONTROLMODE_VARIABLEBITRATE ((__u32) 0x00000002) + +/* Encoder options */ + +struct snd_enc_wma { + __u32 super_block_align; /* WMA Type-specific data */ +}; + + +/** + * struct snd_enc_vorbis + * @quality: Sets encoding quality to n, between -1 (low) and 10 (high). + * In the default mode of operation, the quality level is 3. + * Normal quality range is 0 - 10. + * @managed: Boolean. Set bitrate management mode. This turns off the + * normal VBR encoding, but allows hard or soft bitrate constraints to be + * enforced by the encoder. This mode can be slower, and may also be + * lower quality. It is primarily useful for streaming. + * @max_bit_rate: Enabled only if managed is TRUE + * @min_bit_rate: Enabled only if managed is TRUE + * @downmix: Boolean. Downmix input from stereo to mono (has no effect on + * non-stereo streams). Useful for lower-bitrate encoding. + * + * These options were extracted from the OpenMAX IL spec and Gstreamer vorbisenc + * properties + * + * For best quality users should specify VBR mode and set quality levels. + */ + +struct snd_enc_vorbis { + __s32 quality; + __u32 managed; + __u32 max_bit_rate; + __u32 min_bit_rate; + __u32 downmix; +} __attribute__((packed, aligned(4))); + + +/** + * struct snd_enc_real + * @quant_bits: number of coupling quantization bits in the stream + * @start_region: coupling start region in the stream + * @num_regions: number of regions value + * + * These options were extracted from the OpenMAX IL spec + */ + +struct snd_enc_real { + __u32 quant_bits; + __u32 start_region; + __u32 num_regions; +} __attribute__((packed, aligned(4))); + +/** + * struct snd_enc_flac + * @num: serial number, valid only for OGG formats + * needs to be set by application + * @gain: Add replay gain tags + * + * These options were extracted from the FLAC online documentation + * at http://flac.sourceforge.net/documentation_tools_flac.html + * + * To make the API simpler, it is assumed that the user will select quality + * profiles. Additional options that affect encoding quality and speed can + * be added at a later stage if needed. + * + * By default the Subset format is used by encoders. + * + * TAGS such as pictures, etc, cannot be handled by an offloaded encoder and are + * not supported in this API. + */ + +struct snd_enc_flac { + __u32 num; + __u32 gain; +} __attribute__((packed, aligned(4))); + +struct snd_enc_generic { + __u32 bw; /* encoder bandwidth */ + __s32 reserved[15]; +} __attribute__((packed, aligned(4))); + +union snd_codec_options { + struct snd_enc_wma wma; + struct snd_enc_vorbis vorbis; + struct snd_enc_real real; + struct snd_enc_flac flac; + struct snd_enc_generic generic; +} __attribute__((packed, aligned(4))); + +/** struct snd_codec_desc - description of codec capabilities + * @max_ch: Maximum number of audio channels + * @sample_rates: Sampling rates in Hz, use values like 48000 for this + * @num_sample_rates: Number of valid values in sample_rates array + * @bit_rate: Indexed array containing supported bit rates + * @num_bitrates: Number of valid values in bit_rate array + * @rate_control: value is specified by SND_RATECONTROLMODE defines. + * @profiles: Supported profiles. See SND_AUDIOPROFILE defines. + * @modes: Supported modes. See SND_AUDIOMODE defines + * @formats: Supported formats. See SND_AUDIOSTREAMFORMAT defines + * @min_buffer: Minimum buffer size handled by codec implementation + * @reserved: reserved for future use + * + * This structure provides a scalar value for profiles, modes and stream + * format fields. + * If an implementation supports multiple combinations, they will be listed as + * codecs with different descriptors, for example there would be 2 descriptors + * for AAC-RAW and AAC-ADTS. + * This entails some redundancy but makes it easier to avoid invalid + * configurations. + * + */ + +struct snd_codec_desc { + __u32 max_ch; + __u32 sample_rates[MAX_NUM_SAMPLE_RATES]; + __u32 num_sample_rates; + __u32 bit_rate[MAX_NUM_BITRATES]; + __u32 num_bitrates; + __u32 rate_control; + __u32 profiles; + __u32 modes; + __u32 formats; + __u32 min_buffer; + __u32 reserved[15]; +} __attribute__((packed, aligned(4))); + +/** struct snd_codec + * @id: Identifies the supported audio encoder/decoder. + * See SND_AUDIOCODEC macros. + * @ch_in: Number of input audio channels + * @ch_out: Number of output channels. In case of contradiction between + * this field and the channelMode field, the channelMode field + * overrides. + * @sample_rate: Audio sample rate of input data in Hz, use values like 48000 + * for this. + * @bit_rate: Bitrate of encoded data. May be ignored by decoders + * @rate_control: Encoding rate control. See SND_RATECONTROLMODE defines. + * Encoders may rely on profiles for quality levels. + * May be ignored by decoders. + * @profile: Mandatory for encoders, can be mandatory for specific + * decoders as well. See SND_AUDIOPROFILE defines. + * @level: Supported level (Only used by WMA at the moment) + * @ch_mode: Channel mode for encoder. See SND_AUDIOCHANMODE defines + * @format: Format of encoded bistream. Mandatory when defined. + * See SND_AUDIOSTREAMFORMAT defines. + * @align: Block alignment in bytes of an audio sample. + * Only required for PCM or IEC formats. + * @options: encoder-specific settings + * @reserved: reserved for future use + */ + +struct snd_codec { + __u32 id; + __u32 ch_in; + __u32 ch_out; + __u32 sample_rate; + __u32 bit_rate; + __u32 rate_control; + __u32 profile; + __u32 level; + __u32 ch_mode; + __u32 format; + __u32 align; + union snd_codec_options options; + __u32 reserved[3]; +} __attribute__((packed, aligned(4))); + +#endif diff --git a/src/Makefile.am b/src/Makefile.am index fa255ff43ee0..3930986946cf 100644 --- a/src/Makefile.am +++ b/src/Makefile.am @@ -26,6 +26,10 @@ if BUILD_PCM SUBDIRS += pcm timer libasound_la_LIBADD += pcm/libpcm.la timer/libtimer.la endif +if BUILD_COMPRESS +SUBDIRS += compress +libasound_la_LIBADD += compress/libcompress.la +endif if BUILD_RAWMIDI SUBDIRS += rawmidi libasound_la_LIBADD += rawmidi/librawmidi.la @@ -66,6 +70,9 @@ pcm/libpcm.la: ordinary_pcm/libordinarypcm.la: $(MAKE) -C ordinary_pcm libordinarypcm.la +pcm/libcompress.la: + $(MAKE) -C compress libcompress.la + rawmidi/librawmidi.la: $(MAKE) -C rawmidi librawmidi.la diff --git a/src/compress/Makefile.am b/src/compress/Makefile.am new file mode 100644 index 000000000000..893871ab00a5 --- /dev/null +++ b/src/compress/Makefile.am @@ -0,0 +1,8 @@ +EXTRA_LTLIBRARIES=libcompress.la + +libcompress_la_SOURCES = compress.c + +all: libcompress.la + + +AM_CPPFLAGS=-I$(top_srcdir)/include diff --git a/src/compress/compress.c b/src/compress/compress.c new file mode 100644 index 000000000000..e3fe828f2b1b --- /dev/null +++ b/src/compress/compress.c @@ -0,0 +1,599 @@ +/* + * tinycompress library for compress audio offload in alsa + * Copyright (c) 2011-2012, Intel Corporation. + * + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU Lesser General Public License, + * version 2.1, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public + * License for more details. + * + * You should have received a copy of the GNU Lesser General Public License + * along with this program; if not, write to + * the Free Software Foundation, Inc., + * 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include <stdio.h> +#include <stdlib.h> +#include <fcntl.h> +#include <stdarg.h> +#include <string.h> +#include <errno.h> +#include <unistd.h> +#include <poll.h> +#include <stdbool.h> +#include <sys/ioctl.h> +#include <sys/mman.h> +#include <sys/time.h> +#include <limits.h> + +#include <linux/types.h> +#include <linux/ioctl.h> +#define __force +#define __bitwise +#define __user +#include <sound/asound.h> +#include "sound/compress_params.h" +#include "sound/compress_offload.h" + +#include "compress.h" + +#define COMPR_ERR_MAX 128 + +/* Default maximum time we will wait in a poll() - 20 seconds */ +#define DEFAULT_MAX_POLL_WAIT_MS 20000 + +struct snd_compr { + int fd; + unsigned int flags; + char error[COMPR_ERR_MAX]; + struct snd_compr_config *config; + int running; + int max_poll_wait_ms; + int nonblocking; + unsigned int gapless_metadata; + unsigned int next_track; +}; + +static int oops(struct snd_compr *compr, int e, const char *fmt, ...) +{ + va_list ap; + int sz; + + va_start(ap, fmt); + vsnprintf(compr->error, COMPR_ERR_MAX, fmt, ap); + va_end(ap); + sz = strlen(compr->error); + + snprintf(compr->error + sz, COMPR_ERR_MAX - sz, + ": %s", strerror(e)); + errno = e; + + return -1; +} + +const char *snd_compr_get_error(struct snd_compr *compr) +{ + return compr->error; +} +static struct snd_compr bad_compress = { + .fd = -1, +}; + +int snd_compr_is_running(struct snd_compr *compr) +{ + return ((compr->fd > 0) && compr->running) ? 1 : 0; +} + +int snd_compr_is_ready(struct snd_compr *compr) +{ + return (compr->fd > 0) ? 1 : 0; +} + +static int snd_compr_get_version(struct snd_compr *compr) +{ + int version = 0; + + if (ioctl(compr->fd, SNDRV_COMPRESS_IOCTL_VERSION, &version)) { + oops(compr, errno, "cant read version"); + return -1; + } + return version; +} + +static bool _snd_compr_snd_compr_is_codec_supported(struct snd_compr *compr, + struct snd_compr_config *config, const struct snd_compr_caps *caps) +{ + bool codec = false; + unsigned int i; + + for (i = 0; i < caps->num_codecs; i++) { + if (caps->codecs[i] == config->codec->id) { + /* found the codec */ + codec = true; + break; + } + } + if (codec == false) { + oops(compr, ENXIO, "this codec is not supported"); + return false; + } + + if (config->fragment_size < caps->min_fragment_size) { + oops(compr, EINVAL, "requested fragment size %d is below min supported %d", + config->fragment_size, caps->min_fragment_size); + return false; + } + if (config->fragment_size > caps->max_fragment_size) { + oops(compr, EINVAL, "requested fragment size %d is above max supported %d", + config->fragment_size, caps->max_fragment_size); + return false; + } + if (config->fragments < caps->min_fragments) { + oops(compr, EINVAL, "requested fragments %d are below min supported %d", + config->fragments, caps->min_fragments); + return false; + } + if (config->fragments > caps->max_fragments) { + oops(compr, EINVAL, "requested fragments %d are above max supported %d", + config->fragments, caps->max_fragments); + return false; + } + + /* TODO: match the codec properties */ + return true; +} + +static bool _snd_compr_is_codec_type_supported(int fd, struct snd_codec *codec) +{ + struct snd_compr_caps caps; + bool found = false; + unsigned int i; + + if (ioctl(fd, SNDRV_COMPRESS_GET_CAPS, &caps)) { + oops(&bad_compress, errno, "cannot get device caps"); + return false; + } + + for (i = 0; i < caps.num_codecs; i++) { + if (caps.codecs[i] == codec->id) { + /* found the codec */ + found = true; + break; + } + } + /* TODO: match the codec properties */ + return found; +} + +static inline void +snd_compr_fill_params(struct snd_compr_config *config, struct snd_compr_params *params) +{ + params->buffer.fragment_size = config->fragment_size; + params->buffer.fragments = config->fragments; + memcpy(&params->codec, config->codec, sizeof(params->codec)); +} + +struct snd_compr *snd_compr_open(unsigned int card, unsigned int device, + unsigned int flags, struct snd_compr_config *config) +{ + struct snd_compr *compr; + struct snd_compr_params params; + struct snd_compr_caps caps; + char fn[256]; + + if (!config) { + oops(&bad_compress, EINVAL, "passed bad config"); + return &bad_compress; + } + + compr = calloc(1, sizeof(struct snd_compr)); + if (!compr) { + oops(&bad_compress, errno, "cannot allocate compress object"); + return &bad_compress; + } + + compr->next_track = 0; + compr->gapless_metadata = 0; + compr->config = calloc(1, sizeof(*config)); + if (!compr->config) + goto input_fail; + + snprintf(fn, sizeof(fn), "/dev/snd/comprC%uD%u", card, device); + + compr->max_poll_wait_ms = DEFAULT_MAX_POLL_WAIT_MS; + + compr->flags = flags; + if (!((flags & COMPRESS_OUT) || (flags & COMPRESS_IN))) { + oops(&bad_compress, EINVAL, "can't deduce device direction from given flags"); + goto config_fail; + } + + if (flags & COMPRESS_OUT) { + compr->fd = open(fn, O_RDONLY); + } else { + compr->fd = open(fn, O_WRONLY); + } + if (compr->fd < 0) { + oops(&bad_compress, errno, "cannot open device '%s'", fn); + goto config_fail; + } + + if (ioctl(compr->fd, SNDRV_COMPRESS_GET_CAPS, &caps)) { + oops(compr, errno, "cannot get device caps"); + goto codec_fail; + } + + /* If caller passed "don't care" fill in default values */ + if ((config->fragment_size == 0) || (config->fragments == 0)) { + config->fragment_size = caps.min_fragment_size; + config->fragments = caps.max_fragments; + } + +#if 0 + /* FIXME need to turn this On when DSP supports + * and treat in no support case + */ + if (_snd_compr_snd_compr_is_codec_supported(compr, config, &caps) == false) { + oops(compr, errno, "codec not supported\n"); + goto codec_fail; + } +#endif + + memcpy(compr->config, config, sizeof(*compr->config)); + snd_compr_fill_params(config, &params); + + if (ioctl(compr->fd, SNDRV_COMPRESS_SET_PARAMS, &params)) { + oops(&bad_compress, errno, "cannot set device"); + goto codec_fail; + } + + return compr; + +codec_fail: + close(compr->fd); + compr->fd = -1; +config_fail: + free(compr->config); +input_fail: + free(compr); + return &bad_compress; +} + +void snd_compr_close(struct snd_compr *compr) +{ + if (compr == &bad_compress) + return; + + if (compr->fd >= 0) + close(compr->fd); + compr->running = 0; + compr->fd = -1; + free(compr->config); + free(compr); +} + +int snd_compr_get_hpointer(struct snd_compr *compr, + unsigned int *avail, struct timespec *tstamp) +{ + struct snd_compr_avail kavail; + __u64 time; + + if (!snd_compr_is_ready(compr)) + return oops(compr, ENODEV, "device not ready"); + + if (ioctl(compr->fd, SNDRV_COMPRESS_AVAIL, &kavail)) + return oops(compr, errno, "cannot get avail"); + if (0 == kavail.tstamp.sampling_rate) + return oops(compr, ENODATA, "sample rate unknown"); + *avail = (unsigned int)kavail.avail; + time = kavail.tstamp.pcm_io_frames / kavail.tstamp.sampling_rate; + tstamp->tv_sec = time; + time = kavail.tstamp.pcm_io_frames % kavail.tstamp.sampling_rate; + tstamp->tv_nsec = time * 1000000000 / kavail.tstamp.sampling_rate; + return 0; +} + +int snd_compr_get_tstamp(struct snd_compr *compr, + unsigned int *samples, unsigned int *sampling_rate) +{ + struct snd_compr_tstamp ktstamp; + + if (!snd_compr_is_ready(compr)) + return oops(compr, ENODEV, "device not ready"); + + if (ioctl(compr->fd, SNDRV_COMPRESS_TSTAMP, &ktstamp)) + return oops(compr, errno, "cannot get tstamp"); + + *samples = ktstamp.pcm_io_frames; + *sampling_rate = ktstamp.sampling_rate; + return 0; +} + +int snd_compr_write(struct snd_compr *compr, const void *buf, unsigned int size) +{ + struct snd_compr_avail avail; + struct pollfd fds; + int to_write = 0; /* zero indicates we haven't written yet */ + int written, total = 0, ret; + const char* cbuf = buf; + const unsigned int frag_size = compr->config->fragment_size; + + if (!(compr->flags & COMPRESS_IN)) + return oops(compr, EINVAL, "Invalid flag set"); + if (!snd_compr_is_ready(compr)) + return oops(compr, ENODEV, "device not ready"); + fds.fd = compr->fd; + fds.events = POLLOUT; + + /*TODO: treat auto start here first */ + while (size) { + if (ioctl(compr->fd, SNDRV_COMPRESS_AVAIL, &avail)) + return oops(compr, errno, "cannot get avail"); + + /* We can write if we have at least one fragment available + * or there is enough space for all remaining data + */ + if ((avail.avail < frag_size) && (avail.avail < size)) { + + if (compr->nonblocking) + return total; + + ret = poll(&fds, 1, compr->max_poll_wait_ms); + if (fds.revents & POLLERR) { + return oops(compr, EIO, "poll returned error!"); + } + /* A pause will cause -EBADFD or zero. + * This is not an error, just stop writing */ + if ((ret == 0) || (ret == -EBADFD)) + break; + if (ret < 0) + return oops(compr, errno, "poll error"); + if (fds.revents & POLLOUT) { + continue; + } + } + /* write avail bytes */ + if (size > avail.avail) + to_write = avail.avail; + else + to_write = size; + written = write(compr->fd, cbuf, to_write); + /* If play was paused the write returns -EBADFD */ + if (written == -EBADFD) + break; + if (written < 0) + return oops(compr, errno, "write failed!"); + + size -= written; + cbuf += written; + total += written; + } + return total; +} + +int snd_compr_read(struct snd_compr *compr, void *buf, unsigned int size) +{ + struct snd_compr_avail avail; + struct pollfd fds; + int to_read = 0; + int num_read, total = 0, ret; + char* cbuf = buf; + const unsigned int frag_size = compr->config->fragment_size; + + if (!(compr->flags & COMPRESS_OUT)) + return oops(compr, EINVAL, "Invalid flag set"); + if (!snd_compr_is_ready(compr)) + return oops(compr, ENODEV, "device not ready"); + fds.fd = compr->fd; + fds.events = POLLIN; + + while (size) { + if (ioctl(compr->fd, SNDRV_COMPRESS_AVAIL, &avail)) + return oops(compr, errno, "cannot get avail"); + + if ( (avail.avail < frag_size) && (avail.avail < size) ) { + /* Less than one fragment available and not at the + * end of the read, so poll + */ + if (compr->nonblocking) + return total; + + ret = poll(&fds, 1, compr->max_poll_wait_ms); + if (fds.revents & POLLERR) { + return oops(compr, EIO, "poll returned error!"); + } + /* A pause will cause -EBADFD or zero. + * This is not an error, just stop reading */ + if ((ret == 0) || (ret == -EBADFD)) + break; + if (ret < 0) + return oops(compr, errno, "poll error"); + if (fds.revents & POLLIN) { + continue; + } + } + /* read avail bytes */ + if (size > avail.avail) + to_read = avail.avail; + else + to_read = size; + num_read = read(compr->fd, cbuf, to_read); + /* If play was paused the read returns -EBADFD */ + if (num_read == -EBADFD) + break; + if (num_read < 0) + return oops(compr, errno, "read failed!"); + + size -= num_read; + cbuf += num_read; + total += num_read; + } + + return total; +} + +int snd_compr_start(struct snd_compr *compr) +{ + if (!snd_compr_is_ready(compr)) + return oops(compr, ENODEV, "device not ready"); + if (ioctl(compr->fd, SNDRV_COMPRESS_START)) + return oops(compr, errno, "cannot start the stream"); + compr->running = 1; + return 0; + +} + +int snd_compr_stop(struct snd_compr *compr) +{ + if (!snd_compr_is_running(compr)) + return oops(compr, ENODEV, "device not ready"); + if (ioctl(compr->fd, SNDRV_COMPRESS_STOP)) + return oops(compr, errno, "cannot stop the stream"); + return 0; +} + +int snd_compr_pause(struct snd_compr *compr) +{ + if (!snd_compr_is_running(compr)) + return oops(compr, ENODEV, "device not ready"); + if (ioctl(compr->fd, SNDRV_COMPRESS_PAUSE)) + return oops(compr, errno, "cannot pause the stream"); + return 0; +} + +int snd_compr_resume(struct snd_compr *compr) +{ + if (ioctl(compr->fd, SNDRV_COMPRESS_RESUME)) + return oops(compr, errno, "cannot resume the stream"); + return 0; +} + +int snd_compr_drain(struct snd_compr *compr) +{ + if (!snd_compr_is_running(compr)) + return oops(compr, ENODEV, "device not ready"); + if (ioctl(compr->fd, SNDRV_COMPRESS_DRAIN)) + return oops(compr, errno, "cannot drain the stream"); + return 0; +} + +int snd_compr_partial_drain(struct snd_compr *compr) +{ + if (!snd_compr_is_running(compr)) + return oops(compr, ENODEV, "device not ready"); + + if (!compr->next_track) + return oops(compr, EPERM, "next track not signalled"); + if (ioctl(compr->fd, SNDRV_COMPRESS_PARTIAL_DRAIN)) + return oops(compr, errno, "cannot drain the stream\n"); + compr->next_track = 0; + return 0; +} + +int snd_compr_next_track(struct snd_compr *compr) +{ + if (!snd_compr_is_running(compr)) + return oops(compr, ENODEV, "device not ready"); + + if (!compr->gapless_metadata) + return oops(compr, EPERM, "metadata not set"); + if (ioctl(compr->fd, SNDRV_COMPRESS_NEXT_TRACK)) + return oops(compr, errno, "cannot set next track\n"); + compr->next_track = 1; + compr->gapless_metadata = 0; + return 0; +} + +int snd_compr_set_gapless_metadata(struct snd_compr *compr, + struct snd_compr_gapless_mdata *mdata) +{ + struct snd_compr_metadata metadata; + int version; + + if (!snd_compr_is_ready(compr)) + return oops(compr, ENODEV, "device not ready"); + + version = snd_compr_get_version(compr); + if (version <= 0) + return -1; + + if (version < SNDRV_PROTOCOL_VERSION(0, 1, 1)) + return oops(compr, ENXIO, "gapless apis not supported in kernel"); + + metadata.key = SNDRV_COMPRESS_ENCODER_PADDING; + metadata.value[0] = mdata->encoder_padding; + if (ioctl(compr->fd, SNDRV_COMPRESS_SET_METADATA, &metadata)) + return oops(compr, errno, "can't set metadata for stream\n"); + + metadata.key = SNDRV_COMPRESS_ENCODER_DELAY; + metadata.value[0] = mdata->encoder_delay; + if (ioctl(compr->fd, SNDRV_COMPRESS_SET_METADATA, &metadata)) + return oops(compr, errno, "can't set metadata for stream\n"); + compr->gapless_metadata = 1; + return 0; +} + +bool snd_compr_is_codec_supported(unsigned int card, unsigned int device, + unsigned int flags, struct snd_codec *codec) +{ + unsigned int dev_flag; + bool ret; + int fd; + char fn[256]; + + snprintf(fn, sizeof(fn), "/dev/snd/comprC%uD%u", card, device); + + if (flags & COMPRESS_OUT) + dev_flag = O_RDONLY; + else + dev_flag = O_WRONLY; + + fd = open(fn, dev_flag); + if (fd < 0) + return oops(&bad_compress, errno, "cannot open device '%s'", fn); + + ret = _snd_compr_is_codec_type_supported(fd, codec); + + close(fd); + return ret; +} + +void snd_compr_set_max_poll_wait(struct snd_compr *compr, int milliseconds) +{ + compr->max_poll_wait_ms = milliseconds; +} + +void snd_compr_nonblock(struct snd_compr *compr, int nonblock) +{ + compr->nonblocking = !!nonblock; +} + +int snd_compr_wait(struct snd_compr *compr, int timeout_ms) +{ + struct pollfd fds; + int ret; + + fds.fd = compr->fd; + fds.events = POLLOUT | POLLIN; + + ret = poll(&fds, 1, timeout_ms); + if (ret > 0) { + if (fds.revents & POLLERR) + return oops(compr, EIO, "poll returned error!"); + if (fds.revents & (POLLOUT | POLLIN)) + return 0; + } + if (ret == 0) + return oops(compr, ETIME, "poll timed out"); + if (ret < 0) + return oops(compr, errno, "poll error"); + + return oops(compr, EIO, "poll signalled unhandled event"); +} + -- 2.1.0

3 4

[alsa-devel] Facing issues in a2dp sink profile support on linux
by jags gediya 05 Mar '15

05 Mar '15

Below is the detail of my software SDK: Kernel version : 3.10.17 Build tool : Yocto Processor : i.MX6 Sololite Pulseaudio version : 5.0 Bluez version : 4.101 I want to play music from my phone to the linux box running the above described setup. When i play song from the mobile, I am seeing the ACL data through hcidump. but it is not playing on the headphone. In the pulseaudio logs, i am seeing the errors. Below are my changes, for pulseaudio and bluetooth. /etc/pulse/daemon.conf resample-method = trivial enable-remixing = no enable-lfe-remixing = no default-sample-format = s16le default-sample-rate = 48000 alternate-sample-rate = 24000 default-sample-channels = 2 /etc/pulse/system.pa .ifexists module-bluetooth-policy.so load-module module-bluetooth-policy .endif .ifexists module-bluetooth-discover.so load-module module-bluetooth-discover .endif Below are my steps to connect to mobile. pulseaudio --system --realtime -vvvvv --log-time=1 --log-target=file:pulse.log & ADAPTER=`dbus-send --print-reply --system --dest=org.bluez / org.bluez.Manager.DefaultAdapter | cut -d'"' -f 2 | grep org` DEVICE_PATH=`dbus-send --system --dest=org.bluez --print-reply $ADAPTER org.bluez.Adapter.ListDevices | grep path | cut -d'"' -f 2` dbus-send --print-reply --system --dest=org.bluez $DEVICE_PATH org.bluez.AudioSource.Connect in, "pactl modules list short" I am seeing, module-bluetooth-device address="00:73:8D:35:2A:B8" path="/org/bluez/351/hci0/dev_00_73_8D_35_2A_B8" module-loopback source="bluez_source.00_73_8D_35_2A_B8" source_dont_move="true" sink_input_properties="media.role=music" pactl list sources short bluez_source.00_73_8D_35_2A_B8 module-bluetooth-device.c s16le 2ch 44100Hz SUSPENDED this is the logs, i am getting. D ( 1.163| 0.002) [pulseaudio] alsa-sink.c: Read hardware volume: 0: 75% 1: 75% D ( 1.163| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Thread starting up D ( 1.164| 0.000) [alsa-sink-HiFi sgtl5000-0] core-util.c: SCHED_RR|SCHED_RESET_ON_FORK worked. I ( 1.164| 0.000) [alsa-sink-HiFi sgtl5000-0] core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5. I ( 1.165| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Starting playback. D ( 1.165| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 1.165| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 1.166| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 1.166| 0.000) [pulseaudio] alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... I ( 1.167| 0.000) [pulseaudio] conf.c: Unknown parameters 0 I ( 1.167| 0.000) [pulseaudio] pcm.c: Unknown PCM front:0 I ( 1.167| 0.000) [pulseaudio] alsa-util.c: Error opening PCM device front:0: Invalid argument D ( 1.167| 0.000) [pulseaudio] alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... D ( 1.169| 0.001) [pulseaudio] alsa-util.c: Managed to open hw:0 I ( 1.169| 0.000) [pulseaudio] alsa-util.c: cannot disable ALSA period wakeups D ( 1.169| 0.000) [pulseaudio] alsa-util.c: Maximum hw buffer size is 371 ms D ( 1.174| 0.005) [pulseaudio] alsa-util.c: Set buffer size first (to 88200 samples), period size second (to 88200 samples). I ( 1.175| 0.000) [pulseaudio] alsa-util.c: ALSA period wakeups were not disabled I ( 1.175| 0.000) [pulseaudio] alsa-source.c: Successfully opened device hw:0. I ( 1.175| 0.000) [pulseaudio] alsa-source.c: Selected mapping 'Analog Stereo' (analog-stereo). I ( 1.175| 0.000) [pulseaudio] alsa-source.c: Successfully enabled mmap() mode. I ( 1.175| 0.000) [pulseaudio] alsa-source.c: Successfully enabled timer-based scheduling mode. I ( 1.177| 0.001) [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:0' D ( 1.179| 0.002) [pulseaudio] alsa-mixer.c: Added 1 ports I ( 1.180| 0.000) [pulseaudio] source.c: Created source 1 "alsa_input.platform-sound.25.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right . . . . I ( 1.194| 0.000) [pulseaudio] module.c: Loaded "module-alsa-card" (index: #0; argument: "device_id="0" name="platform-sound.25" card_name="alsa_card.platform-sound.25" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes card_properties="module-udev-detect.discovered=1""). I ( 1.194| 0.000) [pulseaudio] module-udev-detect.c: Card /devices/soc0/sound.25/sound/card0 (alsa_card.platform-sound.25) module loaded. I ( 1.194| 0.000) [pulseaudio] module-udev-detect.c: Found 1 cards. I ( 1.194| 0.000) [pulseaudio] module.c: Loaded "module-udev-detect" (index: #1; argument: ""). D ( 1.195| 0.000) [pulseaudio] cli-command.c: Checking for existence of '/usr/lib/pulse-4.0/modules/module-esound-protocol-unix.so': failure W ( 1.212| 0.016) [pulseaudio] authkey.c: Failed to open cookie file '/var/run/pulse/.config/pulse/cookie': No such file or directory W ( 1.212| 0.000) [pulseaudio] authkey.c: Failed to load authorization key '/var/run/pulse/.config/pulse/cookie': No such file or directory W ( 1.212| 0.000) [pulseaudio] authkey.c: Failed to open cookie file '/var/run/pulse/.pulse-cookie': No such file or directory W ( 1.212| 0.000) [pulseaudio] authkey.c: Failed to load authorization key '/var/run/pulse/.pulse-cookie': No such file or directory D ( 1.213| 0.000) [pulseaudio] authkey.c: Got 0 bytes from cookie file '/var/run/pulse/.config/pulse/cookie', expected 256 I ( 1.214| 0.001) [pulseaudio] module.c: Loaded "module-native-protocol-unix" (index: #2; argument: "auth-anonymous=1"). I ( 1.228| 0.013) [pulseaudio] module-stream-restore.c: Successfully opened database file '/var/lib/pulse/79e7ab36632556972e086ba654e49e31-stream-volumes'. . . . . D ( 107.002| 34.839) [pulseaudio] bluetooth-util.c: dbus: interface=org.bluez.AudioSource, path=/org/bluez/351/hci0/dev_00_73_8D_35_2A_B8, member=PropertyChanged D ( 107.003| 0.000) [pulseaudio] bluetooth-util.c: Device /org/bluez/351/hci0/dev_00_73_8D_35_2A_B8 interface org.bluez.AudioSource property 'State' changed to value 'connecting' D ( 107.363| 0.360) [pulseaudio] bluetooth-util.c: dbus: interface=org.bluez.Device, path=/org/bluez/351/hci0/dev_00_73_8D_35_2A_B8, member=PropertyChanged D ( 107.496| 0.132) [pulseaudio] bluetooth-util.c: dbus: interface=org.bluez.MediaEndpoint, path=/MediaEndpoint/A2DPSink, member=SelectConfiguration D ( 107.496| 0.000) [pulseaudio] bluetooth-util.c: dbus: interface=org.bluez.MediaEndpoint, path=/MediaEndpoint/A2DPSink, member=SelectConfiguration D ( 107.510| 0.013) [pulseaudio] bluetooth-util.c: dbus: interface=org.bluez.MediaEndpoint, path=/MediaEndpoint/A2DPSink, member=SetConfiguration D ( 107.510| 0.000) [pulseaudio] bluetooth-util.c: dbus: interface=org.bluez.MediaEndpoint, path=/MediaEndpoint/A2DPSink, member=SetConfiguration D ( 107.510| 0.000) [pulseaudio] bluetooth-util.c: Transport /org/bluez/351/hci0/dev_00_73_8D_35_2A_B8/fd0 profile 1 available D ( 107.567| 0.056) [pulseaudio] bluetooth-util.c: dbus: interface=org.bluez.AudioSource, path=/org/bluez/351/hci0/dev_00_73_8D_35_2A_B8, member=PropertyChanged D ( 107.567| 0.000) [pulseaudio] bluetooth-util.c: Device /org/bluez/351/hci0/dev_00_73_8D_35_2A_B8 interface org.bluez.AudioSource property 'State' changed to value 'connected' D ( 107.567| 0.000) [pulseaudio] bluetooth-util.c: Transport /org/bluez/351/hci0/dev_00_73_8D_35_2A_B8/fd0 (profile a2dp_source) changed state from disconnected to idle. D ( 107.567| 0.000) [pulseaudio] module-bluetooth-discover.c: Loading module-bluetooth-device address="00:73:8D:35:2A:B8" path="/org/bluez/351/hci0/dev_00_73_8D_35_2A_B8" I ( 107.594| 0.027) [pulseaudio] card.c: Created 1 "bluez_card.00_73_8D_35_2A_B8" W ( 107.595| 0.000) [pulseaudio] module-bluetooth-device.c: Default profile not connected, selecting off profile I ( 107.595| 0.000) [pulseaudio] module.c: Loaded "module-bluetooth-device" (index: #13; argument: "address="00:73:8D:35:2A:B8" path="/org/bluez/351/hci0/dev_00_73_8D_35_2A_B8""). . . . . D ( 195.725| 0.003) [pulseaudio] module-bluetooth-device.c: Acquiring transport /org/bluez/351/hci0/dev_00_73_8D_35_2A_B8/fd0 I ( 195.725| 0.000) [pulseaudio] bluetooth-util.c: Failed optional acquire of transport /org/bluez/351/hci0/dev_00_73_8D_35_2A_B8/fd0 I ( 195.725| 0.000) [pulseaudio] module-bluetooth-device.c: Failed optional acquire of transport /org/bluez/351/hci0/dev_00_73_8D_35_2A_B8/fd0 I ( 195.725| 0.000) [pulseaudio] module-bluetooth-device.c: SBC parameters: I ( 195.725| 0.000) [pulseaudio] module-bluetooth-device.c: allocation=0 I ( 195.725| 0.000) [pulseaudio] module-bluetooth-device.c: subbands=1 I ( 195.725| 0.000) [pulseaudio] module-bluetooth-device.c: blocks=3 I ( 195.725| 0.000) [pulseaudio] module-bluetooth-device.c: bitpool=19 D ( 195.726| 0.000) [pulseaudio] module-device-restore.c: Database contains invalid data for key: source:bluez_source.00_73_8D_35_2A_B8:null I ( 195.726| 0.000) [pulseaudio] source.c: Created source 2 "bluez_source.00_73_8D_35_2A_B8" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I ( 195.726| 0.000) [pulseaudio] source.c: bluetooth.protocol = "a2dp_source" I ( 195.726| 0.000) [pulseaudio] source.c: device.description = "jags gediya" I ( 195.726| 0.000) [pulseaudio] source.c: device.string = "00:73:8D:35:2A:B8" I ( 195.726| 0.000) [pulseaudio] source.c: device.api = "bluez" I ( 195.726| 0.000) [pulseaudio] source.c: device.class = "sound" I ( 195.726| 0.000) [pulseaudio] source.c: device.bus = "bluetooth" I ( 195.726| 0.000) [pulseaudio] source.c: device.form_factor = "phone" I ( 195.726| 0.000) [pulseaudio] source.c: bluez.path = "/org/bluez/351/hci0/dev_00_73_8D_35_2A_B8" I ( 195.726| 0.000) [pulseaudio] source.c: bluez.class = "0x5a020c" I ( 195.726| 0.000) [pulseaudio] source.c: bluez.name = "jags gediya" I ( 195.726| 0.000) [pulseaudio] source.c: device.icon_name = "audio-card-bluetooth" D ( 195.728| 0.001) [bluetooth] module-bluetooth-device.c: IO Thread starting up D ( 195.728| 0.000) [bluetooth] core-util.c: SCHED_RR|SCHED_RESET_ON_FORK worked. I ( 195.728| 0.000) [bluetooth] core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5. D ( 195.729| 0.000) [pulseaudio] module-suspend-on-idle.c: Source bluez_source.00_73_8D_35_2A_B8 becomes idle, timeout in 5 seconds. D ( 195.747| 0.018) [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.platform-sound.25.analog-stereo becomes busy, resuming. D ( 195.747| 0.000) [pulseaudio] sink.c: Suspend cause of sink alsa_output.platform-sound.25.analog-stereo is 0x0000, resuming I ( 195.747| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Trying resume... I ( 195.749| 0.001) [alsa-sink-HiFi sgtl5000-0] alsa-util.c: cannot disable ALSA period wakeups D ( 195.749| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-util.c: Maximum hw buffer size is 371 ms D ( 195.757| 0.007) [alsa-sink-HiFi sgtl5000-0] alsa-util.c: Set buffer size first (to 16384 samples), period size second (to 8192 samples). I ( 195.758| 0.001) [alsa-sink-HiFi sgtl5000-0] alsa-util.c: ALSA period wakeups were not disabled D ( 195.758| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: hwbuf_unused=0 D ( 195.758| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: setting avail_min=15502 I ( 195.759| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Time scheduling watermark is 20.00ms I ( 195.759| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Resumed successfully.... . . . I ( 195.773| 0.000) [pulseaudio] protocol-native.c: Connection died. D ( 195.945| 0.172) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! I ( 195.945| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Underrun! I ( 195.945| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing wakeup watermark to 30.00 ms D ( 196.131| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! I ( 196.131| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Underrun! I ( 196.131| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing wakeup watermark to 40.00 ms D ( 196.316| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! I ( 196.317| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing wakeup watermark to 50.00 ms I ( 196.502| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing wakeup watermark to 56.64 ms I ( 196.688| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing minimal latency to 1.00 ms D ( 196.688| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Latency set to 66.67ms D ( 196.688| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: hwbuf_unused=53780 D ( 196.688| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: setting avail_min=13888 D ( 196.689| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Latency set to 66.67ms D ( 196.689| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: hwbuf_unused=53780 D ( 196.689| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: setting avail_min=13888 I ( 196.874| 0.184) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing minimal latency to 2.00 ms D ( 196.874| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Latency set to 66.67ms D ( 196.874| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: hwbuf_unused=53780 D ( 196.874| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: setting avail_min=13888 . . D ( 304.751| 0.000) [alsa-sink-HiFi sgtl5000-0] ratelimit.c: 47 events suppressed D ( 304.751| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.751| 0.000) [pulseaudio] module-suspend-on-idle.c: Source bluez_source.00_73_8D_35_2A_B8 becomes busy, resuming. D ( 304.751| 0.000) [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D ( 304.752| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.752| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.753| 0.001) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.753| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.754| 0.000) [pulseaudio] module-loopback.c: Loopback overall latency is 63.82 ms + 0.00 ms + 74.89 ms = 138.70 ms D ( 304.754| 0.000) [pulseaudio] module-loopback.c: Should buffer 23512 bytes, buffered at minimum 0 bytes I ( 304.754| 0.000) [pulseaudio] module-loopback.c: New rate of 43513 Hz not within 2‰ of 44100 Hz, forcing smaller adjustment D ( 304.754| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.759| 0.005) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.760| 0.000) [pulseaudio] module-loopback.c: [alsa_output.platform-sound.25.analog-stereo] Updated sampling rate to 44011 Hz. D ( 304.760| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.760| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.760| 0.000) [pulseaudio] module-loopback.c: Loopback overall latency is 56.99 ms + 0.00 ms + 81.77 ms = 138.76 ms D ( 304.761| 0.000) [pulseaudio] module-loopback.c: Should buffer 23472 bytes, buffered at minimum 0 bytes I ( 304.761| 0.000) [pulseaudio] module-loopback.c: New rate of 43514 Hz not within 2‰ of 44011 Hz, forcing smaller adjustment D ( 304.761| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.764| 0.002) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Cutting sleep time for the initial iterations by half. D ( 304.764| 0.000) [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D ( 304.764| 0.000) [pulseaudio] module-loopback.c: [alsa_output.platform-sound.25.analog-stereo] Updated sampling rate to 43922 Hz. D ( 304.915| 0.150) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Requesting rewind due to end of underrun. D ( 304.922| 0.007) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Requesting rewind due to end of underrun. D ( 304.925| 0.002) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Requesting rewind due to end of underrun. I ( 304.936| 0.011) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing wakeup watermark to 30.00 ms I ( 304.955| 0.018) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Could not peek into queue D ( 304.955| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.966| 0.011) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.966| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.971| 0.004) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.973| 0.002) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.976| 0.002) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.976| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.979| 0.003) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.980| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 304.998| 0.018) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.017| 0.018) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.021| 0.004) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.021| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.026| 0.004) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.026| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.031| 0.004) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.031| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.036| 0.004) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.036| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.055| 0.018) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.074| 0.018) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.080| 0.006) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.081| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.086| 0.005) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.086| 0.000) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Requesting rewind due to end of underrun. D ( 305.086| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.092| 0.005) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.092| 0.000) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Requesting rewind due to end of underrun. D ( 305.092| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.095| 0.003) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.095| 0.000) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Requesting rewind due to end of underrun. D ( 305.095| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) D ( 305.114| 0.018) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1920 bytes ago (9836 bytes ahead in playback buffer) I ( 305.122| 0.007) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Increasing wakeup watermark to 40.00 ms I ( 305.129| 0.007) [alsa-sink-HiFi sgtl5000-0] module-loopback.c: Could not peek into queue D ( 305.129| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.140| 0.010) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.140| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.144| 0.003) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.144| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.148| 0.003) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.148| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.152| 0.003) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.152| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.179| 0.026) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.196| 0.017) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.197| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.202| 0.004) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.202| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.207| 0.005) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.207| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.212| 0.004) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.212| 0.000) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.239| 0.026) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) D ( 305.254| 0.015) [alsa-sink-HiFi sgtl5000-0] sink.c: Found underrun 1924 bytes ago (9832 bytes ahead in playback buffer) . . I ( 312.115| 0.000) [pulseaudio] protocol-native.c: Connection died. D ( 314.782| 2.666) [alsa-sink-HiFi sgtl5000-0] ratelimit.c: 27 events suppressed D ( 314.782| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 314.967| 0.184) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 315.153| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 315.339| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 315.525| 0.186) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 315.711| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 315.896| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 316.082| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 316.268| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 316.454| 0.186) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 316.641| 0.187) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 319.797| 3.155) [alsa-sink-HiFi sgtl5000-0] ratelimit.c: 16 events suppressed D ( 319.797| 0.000) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 319.983| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 320.172| 0.188) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 320.354| 0.182) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 320.540| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 320.726| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 320.912| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 321.097| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 321.284| 0.187) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 321.469| 0.184) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! D ( 321.655| 0.185) [alsa-sink-HiFi sgtl5000-0] alsa-sink.c: Wakeup from ALSA! . . . D ( 381.721| 0.066) [pulseaudio] module-loopback.c: Loopback overall latency is 0.00 ms + 1412.45 ms + 62.23 ms = 1474.68 ms D ( 381.721| 0.000) [pulseaudio] module-loopback.c: Should buffer 64888 bytes, buffered at minimum 210984 bytes I ( 381.721| 0.000) [pulseaudio] module-loopback.c: New rate of 47752 Hz not within 2‰ of 43608 Hz, forcing smaller adjustment D ( 381.722| 0.000) [pulseaudio] module-loopback.c: [alsa_output.platform-sound.25.analog-stereo] Updated sampling rate to 43695 Hz. >From the logs, what can be the issue?

1 0

[alsa-devel] [PATCH] amixer: Don't set only the first item in sset_enum()
by Peter Meerwald 05 Mar '15

05 Mar '15

From: Peter Meerwald <p.meerwald(a)bct-electronic.com> Signed-off-by: Peter Meerwald <pmeerw(a)pmeerw.net> --- amixer/amixer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index ed60e7c..36c92eb 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -1278,7 +1278,7 @@ static int get_enum_item_index(snd_mixer_elem_t *elem, char **ptrp) static int sset_enum(snd_mixer_elem_t *elem, unsigned int argc, char **argv) { - unsigned int idx, chn = 0; + unsigned int idx, item = 0; int check_flag = ignore_error ? 0 : -1; for (idx = 1; idx < argc; idx++) { @@ -1287,7 +1287,7 @@ static int sset_enum(snd_mixer_elem_t *elem, unsigned int argc, char **argv) int ival = get_enum_item_index(elem, &ptr); if (ival < 0) return check_flag; - if (snd_mixer_selem_set_enum_item(elem, chn, ival) >= 0) + if (snd_mixer_selem_set_enum_item(elem, item++, ival) >= 0) check_flag = 1; /* skip separators */ while (*ptr == ',' || isspace(*ptr)) -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 1/2] ucm: document some standard values
by han.lu＠intel.com 05 Mar '15

05 Mar '15

From: "Lu, Han" <han.lu(a)intel.com> add standard value JackDev, JackControl and JackHWMute for speaker mute control on jack insertion. These values will be applied on pulseaudio. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/include/use-case.h b/include/use-case.h index 697377a..3728906 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -284,6 +284,10 @@ int snd_use_case_get_list(snd_use_case_mgr_t *uc_mgr, * PlaybackMixerID - mixer playback ID * CaptureMixer - name of capture mixer * CaptureMixerID - mixer capture ID + * JackDev - jack device name + * JackControl - jack control name + * JackHWMute - indicate if the HW mutes a device on jack insertion + * or not. */ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, const char *identifier, -- 2.1.0

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[alsa-devel] [PATCH 00/30] ASoC: Move jack registration to the card level
by Lars-Peter Clausen 05 Mar '15

05 Mar '15

Audio jacks are typically card level elements and when initially introduced the API to register a jack took a card as its parameter. Back at that time there was only one CODEC per card and DAPM support was only available at the CODEC level and all DAPM elements, including the card level elements, were registered with the CODEC's DAPM context. So naturally for all its DAPM operations the jack API used the CODEC's DAPM context. With the multi-component patchset support for multiple CODECs per card was introduced and since it was no longer clear which DAPM context to use for the jack operations the jack API was updated to take a CODEC instead of a card to make it unambiguous again. Shortly after DAPM support was decoupled from the CODECs and became available at the card level as well. Since then card drivers have been updated to register their card level DAPM elements with the card's DAPM context, but the jack API is still CODEC centric and uses the CODEC's DAPM context. This leaves us with some issues. The DAPM pins that are managed by the jack detection logic are part of the card DAPM context but are accessed through a CODEC DAPM context. Currently this works fine and is more of an inelegance, but might break in the future if we take a more hierarchical approach to DAPM contexts. Furthermore with componentization progressing systems that do not register a snd_soc_codec might appear, while these system may still want to able to register a jack. This patchset introduces a new function called snd_soc_card_jack_register() which allows to register the jacks at the card level again and updates all the existing card drivers to use the new API. - Lars Lars-Peter Clausen (30): ASoC: sn95031: Pass CODEC to sn95031_jack_detection() ASoC: Allow to register jacks at the card level ASoC: simple-card: Register jacks at the card level ASoC: imx-es8328: Register jacks at the card level ASoC: wm1133-ev: Register jacks at the card level ASoC: broadwell: Register jacks at the card level ASoC: byt-max98090: Register jacks at the card level ASoC: cht_bsw_rt5645: Register jacks at the card level ASoC: mfld_machine: Register jacks at the card level ASoC: ams-deltea: Register jacks at the card level ASoC: omap-abe-twl6040: Register jacks at the card level ASoC: omap-twl4030: Register jacks at the card level ASoC: rx51: Register jacks at the card level ASoC: hx4700: Register jacks at the card level ASoC: palm27x: Register jacks at the card level ASoC: ttc-dkb: Register jacks at the card level ASoC: z2: Register jacks at the card level ASoC: h1980_uda1380: Register jacks at the card level ASoC: littlemill: Register jacks at the card level ASoC: lowland: Register jacks at the card level ASoC: rx1950_uda1380: Register jacks at the card level ASoC: smartq: Register jacks at the card level ASoC: speyside: Register jacks at the card level ASoC: tobermory: Register jacks at the card level ASoC: tegra_alc5632: Register jacks at the card level ASoC: tegra_max98090: Register jacks at the card level ASoC: tegra_rt5640: Register jacks at the card level ASoC: tegra_rt5677: Register jacks at the card level ASoC: tegra_wm8903: Register jacks at the card level ASoC: Remove snd_soc_jack_new() include/sound/soc.h | 8 +++++--- sound/soc/codecs/sn95031.c | 14 +++++++------ sound/soc/codecs/sn95031.h | 3 ++- sound/soc/fsl/imx-es8328.c | 6 +++--- sound/soc/fsl/wm1133-ev1.c | 12 +++++------ sound/soc/generic/simple-card.c | 20 +++++++++--------- sound/soc/intel/broadwell.c | 12 +++-------- sound/soc/intel/byt-max98090.c | 11 +++------- sound/soc/intel/cht_bsw_rt5645.c | 12 +++++------ sound/soc/intel/mfld_machine.c | 24 +++++++++------------- sound/soc/omap/ams-delta.c | 4 ++-- sound/soc/omap/omap-abe-twl6040.c | 10 ++++----- sound/soc/omap/omap-twl4030.c | 12 ++++------- sound/soc/omap/rx51.c | 6 +++--- sound/soc/pxa/hx4700.c | 11 +++------- sound/soc/pxa/palm27x.c | 11 +++------- sound/soc/pxa/ttc-dkb.c | 15 ++++++-------- sound/soc/pxa/z2.c | 10 +++------ sound/soc/samsung/h1940_uda1380.c | 9 ++------ sound/soc/samsung/littlemill.c | 12 +++++------ sound/soc/samsung/lowland.c | 14 ++++--------- sound/soc/samsung/rx1950_uda1380.c | 9 ++------ sound/soc/samsung/smartq_wm8987.c | 11 ++++------ sound/soc/samsung/speyside.c | 14 ++++--------- sound/soc/samsung/tobermory.c | 13 ++++-------- sound/soc/soc-jack.c | 42 +++++++++++++++++++++++--------------- sound/soc/tegra/tegra_alc5632.c | 9 ++++---- sound/soc/tegra/tegra_max98090.c | 22 +++++++++----------- sound/soc/tegra/tegra_rt5640.c | 10 +++------ sound/soc/tegra/tegra_rt5677.c | 14 ++++++------- sound/soc/tegra/tegra_wm8903.c | 18 ++++++++-------- 31 files changed, 167 insertions(+), 231 deletions(-) -- 1.8.0

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[alsa-devel] [PATCH][ASoC]fsl_ssi: fix of_property_read_u32_array return value check
by Maciej S. Szmigiero 05 Mar '15

05 Mar '15

of_property_read_u32_array returns 0 on success, so the return value shouldn't be inverted twice, first on assignment then in condition expression. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1227,7 +1227,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0; ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0; - ret = !of_property_read_u32_array(np, "dmas", dmas, 4); + ret = of_property_read_u32_array(np, "dmas", dmas, 4); if (ssi_private->use_dma && !ret && dmas[2] == IMX_DMATYPE_SSI_DUAL) { ssi_private->use_dual_fifo = true; /* When using dual fifo mode, we need to keep watermark

2 1

[alsa-devel] [PATCH 1/3] ASoC: ak4671: Fix control-less DAPM routes
by Lars-Peter Clausen 05 Mar '15

05 Mar '15

Routes without a control must use NULL for the control name. The ak4671 driver uses "NULL" instead in a few places. Previous to commit 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") the DAPM core silently ignored non-NULL controls on non-mixer and non-mux routes. But starting with that commit it will complain and not add the route breaking the ak4671 driver in the process. This patch replaces the incorrect "NULL" control name with NULL to fix the issue. Fixes: 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ak4671.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 632e89f..2a58b1d 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -343,25 +343,25 @@ static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = { }; static const struct snd_soc_dapm_route ak4671_intercon[] = { - {"DAC Left", "NULL", "PMPLL"}, - {"DAC Right", "NULL", "PMPLL"}, - {"ADC Left", "NULL", "PMPLL"}, - {"ADC Right", "NULL", "PMPLL"}, + {"DAC Left", NULL, "PMPLL"}, + {"DAC Right", NULL, "PMPLL"}, + {"ADC Left", NULL, "PMPLL"}, + {"ADC Right", NULL, "PMPLL"}, /* Outputs */ - {"LOUT1", "NULL", "LOUT1 Mixer"}, - {"ROUT1", "NULL", "ROUT1 Mixer"}, - {"LOUT2", "NULL", "LOUT2 Mix Amp"}, - {"ROUT2", "NULL", "ROUT2 Mix Amp"}, - {"LOUT3", "NULL", "LOUT3 Mixer"}, - {"ROUT3", "NULL", "ROUT3 Mixer"}, + {"LOUT1", NULL, "LOUT1 Mixer"}, + {"ROUT1", NULL, "ROUT1 Mixer"}, + {"LOUT2", NULL, "LOUT2 Mix Amp"}, + {"ROUT2", NULL, "ROUT2 Mix Amp"}, + {"LOUT3", NULL, "LOUT3 Mixer"}, + {"ROUT3", NULL, "ROUT3 Mixer"}, {"LOUT1 Mixer", "DACL", "DAC Left"}, {"ROUT1 Mixer", "DACR", "DAC Right"}, {"LOUT2 Mixer", "DACHL", "DAC Left"}, {"ROUT2 Mixer", "DACHR", "DAC Right"}, - {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"}, - {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"}, + {"LOUT2 Mix Amp", NULL, "LOUT2 Mixer"}, + {"ROUT2 Mix Amp", NULL, "ROUT2 Mixer"}, {"LOUT3 Mixer", "DACSL", "DAC Left"}, {"ROUT3 Mixer", "DACSR", "DAC Right"}, @@ -381,18 +381,18 @@ static const struct snd_soc_dapm_route ak4671_intercon[] = { {"LIN2", NULL, "Mic Bias"}, {"RIN2", NULL, "Mic Bias"}, - {"ADC Left", "NULL", "LIN MUX"}, - {"ADC Right", "NULL", "RIN MUX"}, + {"ADC Left", NULL, "LIN MUX"}, + {"ADC Right", NULL, "RIN MUX"}, /* Analog Loops */ - {"LIN1 Mixing Circuit", "NULL", "LIN1"}, - {"RIN1 Mixing Circuit", "NULL", "RIN1"}, - {"LIN2 Mixing Circuit", "NULL", "LIN2"}, - {"RIN2 Mixing Circuit", "NULL", "RIN2"}, - {"LIN3 Mixing Circuit", "NULL", "LIN3"}, - {"RIN3 Mixing Circuit", "NULL", "RIN3"}, - {"LIN4 Mixing Circuit", "NULL", "LIN4"}, - {"RIN4 Mixing Circuit", "NULL", "RIN4"}, + {"LIN1 Mixing Circuit", NULL, "LIN1"}, + {"RIN1 Mixing Circuit", NULL, "RIN1"}, + {"LIN2 Mixing Circuit", NULL, "LIN2"}, + {"RIN2 Mixing Circuit", NULL, "RIN2"}, + {"LIN3 Mixing Circuit", NULL, "LIN3"}, + {"RIN3 Mixing Circuit", NULL, "RIN3"}, + {"LIN4 Mixing Circuit", NULL, "LIN4"}, + {"RIN4 Mixing Circuit", NULL, "RIN4"}, {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"}, {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"}, -- 1.8.0

4 5