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

[alsa-devel] [PATCH v2] ASoC: Intel: Add Broadwell Machine support
by Jie Yang 16 Jul '14

16 Jul '14

From: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Add support for Broadwell based machines with SST DSP audio. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Signed-off-by: Jie Yang <yang.jie(a)linux.intel.com> --- sound/soc/intel/Kconfig | 12 +++ sound/soc/intel/Makefile | 2 + sound/soc/intel/broadwell.c | 251 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 265 insertions(+) create mode 100644 sound/soc/intel/broadwell.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c30fedb..0b305f9 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -58,3 +58,15 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH help This adds audio driver for Intel Baytrail platform based boards with the MAX98090 audio codec. + +config SND_SOC_INTEL_BROADWELL_MACH + tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + select SND_SOC_INTEL_HASWELL + select SND_COMPRESS_OFFLOAD + select SND_SOC_RT286 + help + This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell + Ultrabook platforms. + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 4bfca79..7acbfc4 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -24,7 +24,9 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-soc-sst-broadwell-objs := broadwell.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o +obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c new file mode 100644 index 0000000..0e550f1 --- /dev/null +++ b/sound/soc/intel/broadwell.c @@ -0,0 +1,251 @@ +/* + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * 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. + * + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include "sst-dsp.h" +#include "sst-haswell-ipc.h" + +#include "../codecs/rt286.h" + +static const struct snd_soc_dapm_widget broadwell_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC1", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_LINE("Line Jack", NULL), +}; + +static const struct snd_soc_dapm_route broadwell_rt286_map[] = { + + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphones", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + {"LINE1", NULL, "Line Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC1"}, + {"DMIC2 Pin", NULL, "DMIC2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int broadwell_rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops broadwell_rt286_ops = { + .hw_params = broadwell_rt286_hw_params, +}; + +static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); + struct sst_hsw *broadwell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings */ + ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_CLOCK_MASTER, 9); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set device config\n"); + return ret; + } + + /* always connected - check HP for jack detect */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Line Jack"); + snd_soc_dapm_enable_pin(dapm, "DMIC1"); + snd_soc_dapm_enable_pin(dapm, "DMIC2"); + + return 0; +} + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link broadwell_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback", + .cpu_dai_name = "System Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = broadwell_rtd_init, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload0", + .stream_name = "Offload0 Playback", + .cpu_dai_name = "Offload0 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload1", + .stream_name = "Offload1 Playback", + .cpu_dai_name = "Offload1 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Loopback PCM", + .stream_name = "Loopback", + .cpu_dai_name = "Loopback Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 0, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + { + .name = "Capture PCM", + .stream_name = "Capture", + .cpu_dai_name = "Capture Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .be_id = 0, + .cpu_dai_name = "snd-soc-dummy-dai", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt286-aif1", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broadwell_ssp0_fixup, + .ops = &broadwell_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +/* broadwell audio machine driver for WPT + RT286S */ +static struct snd_soc_card broadwell_rt286 = { + .name = "broadwell-rt286", + .owner = THIS_MODULE, + .dai_link = broadwell_rt286_dais, + .num_links = ARRAY_SIZE(broadwell_rt286_dais), + .dapm_widgets = broadwell_widgets, + .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), + .dapm_routes = broadwell_rt286_map, + .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), + .fully_routed = true, +}; + +static int broadwell_audio_probe(struct platform_device *pdev) +{ + broadwell_rt286.dev = &pdev->dev; + + return snd_soc_register_card(&broadwell_rt286); +} + +static int broadwell_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&broadwell_rt286); + return 0; +} + +static struct platform_driver broadwell_audio = { + .probe = broadwell_audio_probe, + .remove = broadwell_audio_remove, + .driver = { + .name = "broadwell-audio", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(broadwell_audio) + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:broadwell-audio"); -- 1.8.3.2

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[alsa-devel] [PATCH v2] ASoC: Intel: Update FW version readback
by Jie Yang 16 Jul '14

16 Jul '14

Update FW version readback. IPC_GLB_GET_FW_VERSION reads back the ABI version whilst the release version is in the mailbox. Update to use mailbox version for info logging. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 4342363..96373ab 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -183,7 +183,7 @@ struct sst_hsw_ipc_fw_ready { u32 inbox_size; u32 outbox_size; u32 fw_info_size; - u8 fw_info[1]; + u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)]; } __attribute__((packed)); struct ipc_message { @@ -569,6 +569,9 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) { struct sst_hsw_ipc_fw_ready fw_ready; u32 offset; + u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)]; + char *tmp[5], *pinfo; + int i = 0; offset = (header & 0x1FFFFFFF) << 3; @@ -589,6 +592,19 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) fw_ready.inbox_offset, fw_ready.inbox_size); dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n", fw_ready.outbox_offset, fw_ready.outbox_size); + if (fw_ready.fw_info_size < sizeof(fw_ready.fw_info)) { + fw_ready.fw_info[fw_ready.fw_info_size] = 0; + dev_dbg(hsw->dev, " Firmware info: %s \n", fw_ready.fw_info); + + /* log the FW version info got from the mailbox here. */ + memcpy(fw_info, fw_ready.fw_info, fw_ready.fw_info_size); + pinfo = &fw_info[0]; + for (i = 0; i < sizeof(tmp) / sizeof(char *); i++) + tmp[i] = strsep(&pinfo, " "); + dev_info(hsw->dev, "FW loaded, mailbox readback FW info: type %s, - " + "version: %s.%s, build %s, source commit id: %s\n", + tmp[0], tmp[1], tmp[2], tmp[3], tmp[4]); + } } static void hsw_notification_work(struct work_struct *work) @@ -1775,8 +1791,6 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) /* get the FW version */ sst_hsw_fw_get_version(hsw, &version); - dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n", - version.type, version.major, version.minor, version.build); /* get the globalmixer */ ret = sst_hsw_mixer_get_info(hsw); -- 1.8.3.2

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[alsa-devel] [PATCH v3] ASoC: core: add a helper for extended byte controls using TLV
by Vinod Koul 16 Jul '14

16 Jul '14

From: Omair Mohammed Abdullah <omair.m.abdullah(a)intel.com> ALSA supports arbitrary length TLVs for each kcontrol that can be used to pass metadata about the control (e.g. volumes, enum information). The same transport mechanism is now used for arbitrary length data by defining a new helper. Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Changes in v3: - rebased on tlv-opflag - fix return error for hanlders/missing ones include/sound/soc.h | 14 +++++++++++++- sound/soc/soc-core.c | 21 +++++++++++++++++++++ 2 files changed, 34 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7..688fa66 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -270,7 +270,14 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&(struct soc_bytes_ext) \ {.max = xcount} } - +#define SND_SOC_BYTES_TLV(xname, xcount, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ + .tlv.c = (snd_soc_bytes_tlv_callback), \ + .info = snd_soc_info_bytes_ext, \ + .private_value = (unsigned long)&(struct soc_bytes_ext) \ + {.max = xcount, .get = xhandler_get, .put = xhandler_put, } } #define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \ xmin, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -552,6 +559,8 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *ucontrol); +int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv); int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, @@ -1119,6 +1128,9 @@ struct soc_bytes { struct soc_bytes_ext { int max; + /* used for TLV byte control */ + int (*get)(unsigned int __user *bytes, unsigned int size); + int (*put)(const unsigned int __user *bytes, unsigned int size); }; /* multi register control */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b87d7d8..a9ce63f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3267,6 +3267,27 @@ int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_bytes_info_ext); +int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + unsigned int count = size < params->max ? size : params->max; + int ret = -ENXIO; + + switch (op_flag) { + case SNDRV_CTL_TLV_OP_READ: + if (params->get) + ret = params->get(tlv, count); + break; + case SNDRV_CTL_TLV_OP_WRITE: + if (params->put) + ret = params->put(tlv, count); + break; + } + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_tlv_callback); + /** * snd_soc_info_xr_sx - signed multi register info callback * @kcontrol: mreg control -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: mcasp: don't override bclk divider if it was provided by the machine
by Daniel Mack 16 Jul '14

16 Jul '14

If a machine driver provides an BCLK divider to the McASP driver, skip the automatic calculation. This fixes machines on which the physical sample transport always works in 32 bits, even though not all of them are actually used. snd_soc_params_to_bclk() will fail to address such cases properly. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/davinci/davinci-mcasp.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0518f27..dbeb4fd 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -64,6 +64,7 @@ struct davinci_mcasp { u8 num_serializer; u8 *serial_dir; u8 version; + u8 bclk_div; u16 bclk_lrclk_ratio; int streams; @@ -420,6 +421,7 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div ACLKXDIV(div - 1), ACLKXDIV_MASK); mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRDIV(div - 1), ACLKRDIV_MASK); + mcasp->bclk_div = div; break; case 2: /* BCLK/LRCLK ratio */ @@ -722,8 +724,11 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int period_size = params_period_size(params); int ret; - /* If mcasp is BCLK master we need to set BCLK divider */ - if (mcasp->bclk_master) { + /* + * If mcasp is BCLK master, and a BCLK divider was not provided by + * the machine driver, we need to calculate the ratio. + */ + if (mcasp->bclk_master && mcasp->bclk_div == 0) { unsigned int bclk_freq = snd_soc_params_to_bclk(params); if (mcasp->sysclk_freq % bclk_freq != 0) { dev_err(mcasp->dev, "Can't produce required BCLK\n"); -- 1.9.3

4 3

[alsa-devel] [PATCH 0/3] ASoC: davinci-mcasp/edma: Switch AM335x/43xx to edma-pcm
by Peter Ujfalusi 16 Jul '14

16 Jul '14

Hi, This series will switch the AM335x/AM43xx SoCs to use the edma-pcm as platform driver instead of the old davinic-pcm. Doing this we can benefit improvements within the dmaengine driver stack and are going to rely on standard interfaces towards the DMA. Regards, Peter --- Peter Ujfalusi (3): ASoC: edma-pcm: Remove PCM_INFO_BATCH and add PCM_INFO_NO_PERIOD_WAKEUP flag ASoC: edma-pcm: Add empty functions for !CONFIG_SND_EDMA_SOC builds ASoC: davinci-mcasp: Use dmaengine based platform driver for AM335x/447x sound/soc/davinci/Kconfig | 18 ++++++++++++++---- sound/soc/davinci/Makefile | 2 ++ sound/soc/davinci/davinci-mcasp.c | 11 +++++++++-- sound/soc/davinci/edma-pcm.c | 2 +- sound/soc/davinci/edma-pcm.h | 7 +++++++ 5 files changed, 33 insertions(+), 7 deletions(-) -- 2.0.0

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[alsa-devel] [PATCH] ALSA: pcm: Add tstamp_type and proto to sw_params compat layer
by Takashi Iwai 16 Jul '14

16 Jul '14

I forgot to add the new fields in sw_params to 32bit compat layer. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_compat.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index af49721ba0e3..1b61ca727750 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -101,7 +101,9 @@ struct snd_pcm_sw_params32 { u32 silence_threshold; u32 silence_size; u32 boundary; - unsigned char reserved[64]; + u32 tstamp_type; + u32 proto; + unsigned char reserved[56]; }; /* recalcuate the boundary within 32bit */ @@ -133,7 +135,9 @@ static int snd_pcm_ioctl_sw_params_compat(struct snd_pcm_substream *substream, get_user(params.start_threshold, &src->start_threshold) || get_user(params.stop_threshold, &src->stop_threshold) || get_user(params.silence_threshold, &src->silence_threshold) || - get_user(params.silence_size, &src->silence_size)) + get_user(params.silence_size, &src->silence_size) || + get_user(params.tstamp_type, &src->tstamp_type) || + get_user(params.proto, &src->proto)) return -EFAULT; /* * Check silent_size parameter. Since we have 64bit boundary, -- 2.0.1

1 0

[alsa-devel] [PATCH] alsa-lib: pcm: Fill sw_params proto field
by Takashi Iwai 16 Jul '14

16 Jul '14

Fill the new proto field introduced to sw_params with the current PCM protocol version. This makes tstamp_type evaluated properly in the kernel. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/asound.h | 4 ++-- src/pcm/pcm.c | 1 + src/pcm/pcm_params.c | 1 + 3 files changed, 4 insertions(+), 2 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 552f41bd1ca9..98708c60c320 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -386,8 +386,8 @@ struct snd_pcm_sw_params { snd_pcm_uframes_t silence_threshold; /* min distance from noise for silence filling */ snd_pcm_uframes_t silence_size; /* silence block size */ snd_pcm_uframes_t boundary; /* pointers wrap point */ - unsigned int tstamp_type; /* timestamp type */ - int pads; /* alignment, reserved */ + unsigned int tstamp_type; /* timestamp type (req. proto >= 2.0.12) */ + unsigned int proto; /* protocol version */ unsigned char reserved[56]; /* reserved for future */ }; diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 89844430bf35..1399a5b2e671 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -5610,6 +5610,7 @@ int snd_pcm_sw_params_current(snd_pcm_t *pcm, snd_pcm_sw_params_t *params) SNDMSG("PCM not set up"); return -EIO; } + params->proto = SNDRV_PCM_VERSION; params->tstamp_mode = pcm->tstamp_mode; params->tstamp_type = pcm->tstamp_type; params->period_step = pcm->period_step; diff --git a/src/pcm/pcm_params.c b/src/pcm/pcm_params.c index 4adbefae0530..6e57904e445b 100644 --- a/src/pcm/pcm_params.c +++ b/src/pcm/pcm_params.c @@ -2257,6 +2257,7 @@ static int snd_pcm_sw_params_default(snd_pcm_t *pcm, snd_pcm_sw_params_t *params { assert(pcm && params); assert(pcm->setup); + params->proto = SNDRV_PCM_VERSION; params->tstamp_mode = SND_PCM_TSTAMP_NONE; params->tstamp_type = pcm->tstamp_type; params->period_step = 1; -- 2.0.1

1 0

[alsa-devel] [PATCH 0/3] Revised patch(es) for CLOCK_MONOTONIC_RAW timestamp
by Takashi Iwai 16 Jul '14

16 Jul '14

Hi, here is a series of revised kernel patches for adding CLOCK_MONOTONIC_RAW timestamp type. In addition to Mark's original patch, the new field is added to sw_params. The corresponding alsa-lib patches will follow after these. Takashi

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[alsa-devel] [patch] ALSA: compress: fix an integer overflow check
by Dan Carpenter 16 Jul '14

16 Jul '14

I previously added an integer overflow check here but looking at it now, it's still buggy. The bug happens in snd_compr_allocate_buffer(). We multiply ".fragments" and ".fragment_size" and that doesn't overflow but then we save it in an unsigned int so it truncates the high bits away and we allocate a smaller than expected size. Fixes: b35cc8225845 ('ALSA: compress_core: integer overflow in snd_compr_allocate_buffer()') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- The other option would be to declare buffer_size as size_t instead of an unsigned int. This feels like a safer fix though. diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 7403f34..89028fa 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -491,7 +491,7 @@ static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ if (params->buffer.fragment_size == 0 || - params->buffer.fragments > SIZE_MAX / params->buffer.fragment_size) + params->buffer.fragments > INT_MAX / params->buffer.fragment_size) return -EINVAL; /* now codec parameters */

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[alsa-devel] [PATCH] ALSA: hda - Add NULL check to all PM ops in hda_intel.c
by Takashi Iwai 16 Jul '14

16 Jul '14

Since devptr can be NULL due to asynchronous probe, all PM ops should have NULL checks at the beginning. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 45 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 35 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8edcee5f034d..387fc73a5c89 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -772,10 +772,15 @@ static int azx_suspend(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct azx *chip; + struct hda_intel *hda; struct azx_pcm *p; + if (!card) + return 0; + + chip = card->private_data; + hda = container_of(chip, struct hda_intel, chip); if (chip->disabled || hda->init_failed) return 0; @@ -806,9 +811,14 @@ static int azx_resume(struct device *dev) { struct pci_dev *pci = to_pci_dev(dev); struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct azx *chip; + struct hda_intel *hda; + if (!card) + return 0; + + chip = card->private_data; + hda = container_of(chip, struct hda_intel, chip); if (chip->disabled || hda->init_failed) return 0; @@ -844,9 +854,14 @@ static int azx_resume(struct device *dev) static int azx_runtime_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct azx *chip; + struct hda_intel *hda; + if (!card) + return 0; + + chip = card->private_data; + hda = container_of(chip, struct hda_intel, chip); if (chip->disabled || hda->init_failed) return 0; @@ -869,12 +884,17 @@ static int azx_runtime_suspend(struct device *dev) static int azx_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct azx *chip; + struct hda_intel *hda; struct hda_bus *bus; struct hda_codec *codec; int status; + if (!card) + return 0; + + chip = card->private_data; + hda = container_of(chip, struct hda_intel, chip); if (chip->disabled || hda->init_failed) return 0; @@ -910,9 +930,14 @@ static int azx_runtime_resume(struct device *dev) static int azx_runtime_idle(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); - struct azx *chip = card->private_data; - struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct azx *chip; + struct hda_intel *hda; + + if (!card) + return 0; + chip = card->private_data; + hda = container_of(chip, struct hda_intel, chip); if (chip->disabled || hda->init_failed) return 0; -- 2.0.1

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