April 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 1/2] sound: usb: add UAC2 clock sources as mixer controls
by Daniel Mack 09 Apr '16

09 Apr '16

UAC2 specifies clock sources that optionally have validity controls. This patch exposes them as mixer controls, so they can be read (and at least in theory even be written) by userspace applications in order to make clock selection policy decisions. This implementation does nothing if the device is not UAC2 compliant, or if the clock source does not define said validity control bits. Tested with a miniDSP USBStreamer (0x2752/0x0016). Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/usb/mixer.c | 69 ++++++++++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4f85757..cab6f52 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -45,6 +45,7 @@ #include <linux/bitops.h> #include <linux/init.h> #include <linux/list.h> +#include <linux/log2.h> #include <linux/slab.h> #include <linux/string.h> #include <linux/usb.h> @@ -1378,6 +1379,71 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_add_control(&cval->head, kctl); } +static int parse_clock_source_unit(struct mixer_build *state, int unitid, + void *_ftr) +{ + struct uac_clock_source_descriptor *hdr = _ftr; + struct usb_mixer_elem_info *cval; + struct snd_kcontrol *kctl; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + int ret; + + if (state->mixer->protocol != UAC_VERSION_2) + return -EINVAL; + + if (hdr->bLength != sizeof(*hdr)) { + usb_audio_dbg(state->chip, + "Bogus clock source descriptor length of %d, ignoring.\n", + hdr->bLength); + return 0; + } + + /* + * The only property of this unit we are interested in is the + * clock source validity. If that isn't readable, just bail out. + */ + if (!uac2_control_is_readable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + return 0; + + cval = kzalloc(sizeof(*cval), GFP_KERNEL); + if (!cval) + return -ENOMEM; + + snd_usb_mixer_elem_init_std(&cval->head, state->mixer, hdr->bClockID); + + cval->min = 0; + cval->max = 1; + cval->channels = 1; + cval->val_type = USB_MIXER_BOOLEAN; + cval->control = UAC2_CS_CONTROL_CLOCK_VALID; + + if (uac2_control_is_writeable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); + else { + cval->master_readonly = 1; + kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval); + } + + if (!kctl) { + kfree(cval); + return -ENOMEM; + } + + kctl->private_free = snd_usb_mixer_elem_free; + ret = snd_usb_copy_string_desc(state, hdr->iClockSource, + name, sizeof(name)); + if (ret > 0) + snprintf(kctl->id.name, sizeof(kctl->id.name), + "%s Validity", name); + else + snprintf(kctl->id.name, sizeof(kctl->id.name), + "Clock Source %d Validity", hdr->bClockID); + + return snd_usb_mixer_add_control(&cval->head, kctl); +} + /* * parse a feature unit * @@ -2126,10 +2192,11 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) switch (p1[2]) { case UAC_INPUT_TERMINAL: - case UAC2_CLOCK_SOURCE: return 0; /* NOP */ case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); + case UAC2_CLOCK_SOURCE: + return parse_clock_source_unit(state, unitid, p1); case UAC_SELECTOR_UNIT: case UAC2_CLOCK_SELECTOR: return parse_audio_selector_unit(state, unitid, p1); -- 2.4.3

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[alsa-devel] [PATCH] ALSA: constify ct_timer_ops structures
by Julia Lawall 09 Apr '16

09 Apr '16

The ct_timer_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/pci/ctxfi/cttimer.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index a5d4604..8f94534 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -49,7 +49,7 @@ struct ct_timer { spinlock_t lock; /* global timer lock (for xfitimer) */ spinlock_t list_lock; /* lock for instance list */ struct ct_atc *atc; - struct ct_timer_ops *ops; + const struct ct_timer_ops *ops; struct list_head instance_head; struct list_head running_head; unsigned int wc; /* current wallclock */ @@ -128,7 +128,7 @@ static void ct_systimer_prepare(struct ct_timer_instance *ti) #define ct_systimer_free ct_systimer_prepare -static struct ct_timer_ops ct_systimer_ops = { +static const struct ct_timer_ops ct_systimer_ops = { .init = ct_systimer_init, .free_instance = ct_systimer_free, .prepare = ct_systimer_prepare, @@ -322,7 +322,7 @@ static void ct_xfitimer_free_global(struct ct_timer *atimer) ct_xfitimer_irq_stop(atimer); } -static struct ct_timer_ops ct_xfitimer_ops = { +static const struct ct_timer_ops ct_xfitimer_ops = { .prepare = ct_xfitimer_prepare, .start = ct_xfitimer_start, .stop = ct_xfitimer_stop,

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[alsa-devel] [PATCH] ASoC: samsung: Remove unused Odroid x2/u3 machine driver
by Sylwester Nawrocki 08 Apr '16

08 Apr '16

We have been using "simple-audio-card" for Odroid X2/U3 boards, as can be seen from sound node in arch/arm/boot/dts/ exynos4412-odroid-common.dtsi. A dedicated machine driver is not needed and it is removed in this patch. There is no dts files using "samsung,odroidx2-audio" or "samsung,odroidu3-audio" compatible strings. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- .../bindings/sound/samsung,odroidx2-max98090.txt | 35 ---- sound/soc/samsung/Kconfig | 8 - sound/soc/samsung/Makefile | 2 - sound/soc/samsung/odroidx2_max98090.c | 185 --------------------- 4 files changed, 230 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt delete mode 100644 sound/soc/samsung/odroidx2_max98090.c diff --git a/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt deleted file mode 100644 index 9148f72..0000000 --- a/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt +++ /dev/null @@ -1,35 +0,0 @@ -Samsung Exynos Odroid X2/U3 audio complex with MAX98090 codec - -Required properties: - - compatible : "samsung,odroidx2-audio" - for Odroid X2 board, - "samsung,odroidu3-audio" - for Odroid U3 board - - samsung,model : the user-visible name of this sound complex - - samsung,i2s-controller : the phandle of the I2S controller - - samsung,audio-codec : the phandle of the MAX98090 audio codec - - samsung,audio-routing : a list of the connections between audio - components; each entry is a pair of strings, the first being the - connection's sink, the second being the connection's source; - valid names for sources and sinks are the MAX98090's pins (as - documented in its binding), and the jacks on the board - For Odroid X2: - * Headphone Jack - * Mic Jack - * DMIC - - For Odroid U3: - * Headphone Jack - * Speakers - -Example: - -sound { - compatible = "samsung,odroidu3-audio"; - samsung,i2s-controller = <&i2s0>; - samsung,audio-codec = <&max98090>; - samsung,model = "Odroid-X2"; - samsung,audio-routing = - "Headphone Jack", "HPL", - "Headphone Jack", "HPR", - "IN1", "Mic Jack", - "Mic Jack", "MICBIAS"; -}; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 78baa26..7b722b0 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -224,14 +224,6 @@ config SND_SOC_SNOW Say Y if you want to add audio support for various Snow boards based on Exynos5 series of SoCs. -config SND_SOC_ODROIDX2 - tristate "Audio support for Odroid-X2 and Odroid-U3" - depends on SND_SOC_SAMSUNG && I2C - select SND_SOC_MAX98090 - select SND_SAMSUNG_I2S - help - Say Y here to enable audio support for the Odroid-X2/U3. - config SND_SOC_ARNDALE_RT5631_ALC5631 tristate "Audio support for RT5631(ALC5631) on Arndale Board" depends on SND_SOC_SAMSUNG && I2C diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 052fe71..5d03f5c 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -43,7 +43,6 @@ snd-soc-tobermory-objs := tobermory.o snd-soc-lowland-objs := lowland.o snd-soc-littlemill-objs := littlemill.o snd-soc-bells-objs := bells.o -snd-soc-odroidx2-max98090-objs := odroidx2_max98090.o snd-soc-arndale-rt5631-objs := arndale_rt5631.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o @@ -69,5 +68,4 @@ obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o -obj-$(CONFIG_SND_SOC_ODROIDX2) += snd-soc-odroidx2-max98090.o obj-$(CONFIG_SND_SOC_ARNDALE_RT5631_ALC5631) += snd-soc-arndale-rt5631.o diff --git a/sound/soc/samsung/odroidx2_max98090.c b/sound/soc/samsung/odroidx2_max98090.c deleted file mode 100644 index 0421727..0000000 --- a/sound/soc/samsung/odroidx2_max98090.c +++ /dev/null @@ -1,185 +0,0 @@ -/* - * Copyright (C) 2014 Samsung Electronics Co., Ltd. - * - * 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; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/of.h> -#include <linux/module.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> -#include "i2s.h" - -struct odroidx2_drv_data { - const struct snd_soc_dapm_widget *dapm_widgets; - unsigned int num_dapm_widgets; -}; - -/* The I2S CDCLK output clock frequency for the MAX98090 codec */ -#define MAX98090_MCLK 19200000 - -static struct snd_soc_dai_link odroidx2_dai[]; - -static int odroidx2_late_probe(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - cpu_dai = rtd->cpu_dai; - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, MAX98090_MCLK, - SND_SOC_CLOCK_IN); - - if (ret < 0 || of_find_property(odroidx2_dai[0].codec_of_node, - "clocks", NULL)) - return ret; - - /* Set the cpu DAI configuration in order to use CDCLK */ - return snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK, - 0, SND_SOC_CLOCK_OUT); -} - -static const struct snd_soc_dapm_widget odroidx2_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), - SND_SOC_DAPM_MIC("DMIC", NULL), -}; - -static const struct snd_soc_dapm_widget odroidu3_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speakers", NULL), -}; - -static struct snd_soc_dai_link odroidx2_dai[] = { - { - .name = "MAX98090", - .stream_name = "MAX98090 PCM", - .codec_dai_name = "HiFi", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - } -}; - -static struct snd_soc_card odroidx2 = { - .owner = THIS_MODULE, - .dai_link = odroidx2_dai, - .num_links = ARRAY_SIZE(odroidx2_dai), - .fully_routed = true, - .late_probe = odroidx2_late_probe, -}; - -static const struct odroidx2_drv_data odroidx2_drvdata = { - .dapm_widgets = odroidx2_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(odroidx2_dapm_widgets), -}; - -static const struct odroidx2_drv_data odroidu3_drvdata = { - .dapm_widgets = odroidu3_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(odroidu3_dapm_widgets), -}; - -static const struct of_device_id odroidx2_audio_of_match[] = { - { - .compatible = "samsung,odroidx2-audio", - .data = &odroidx2_drvdata, - }, { - .compatible = "samsung,odroidu3-audio", - .data = &odroidu3_drvdata, - }, - { }, -}; -MODULE_DEVICE_TABLE(of, odroidx2_audio_of_match); - -static int odroidx2_audio_probe(struct platform_device *pdev) -{ - struct device_node *snd_node = pdev->dev.of_node; - struct snd_soc_card *card = &odroidx2; - struct device_node *i2s_node, *codec_node; - struct odroidx2_drv_data *dd; - const struct of_device_id *of_id; - int ret; - - of_id = of_match_node(odroidx2_audio_of_match, snd_node); - dd = (struct odroidx2_drv_data *)of_id->data; - - card->num_dapm_widgets = dd->num_dapm_widgets; - card->dapm_widgets = dd->dapm_widgets; - - card->dev = &pdev->dev; - - ret = snd_soc_of_parse_card_name(card, "samsung,model"); - if (ret < 0) - return ret; - - ret = snd_soc_of_parse_audio_routing(card, "samsung,audio-routing"); - if (ret < 0) - return ret; - - codec_node = of_parse_phandle(snd_node, "samsung,audio-codec", 0); - if (!codec_node) { - dev_err(&pdev->dev, - "Failed parsing samsung,i2s-codec property\n"); - return -EINVAL; - } - - i2s_node = of_parse_phandle(snd_node, "samsung,i2s-controller", 0); - if (!i2s_node) { - dev_err(&pdev->dev, - "Failed parsing samsung,i2s-controller property\n"); - ret = -EINVAL; - goto err_put_codec_n; - } - - odroidx2_dai[0].codec_of_node = codec_node; - odroidx2_dai[0].cpu_of_node = i2s_node; - odroidx2_dai[0].platform_of_node = i2s_node; - - ret = snd_soc_register_card(card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - goto err_put_i2s_n; - } - return 0; - -err_put_i2s_n: - of_node_put(i2s_node); -err_put_codec_n: - of_node_put(codec_node); - return ret; -} - -static int odroidx2_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - of_node_put(odroidx2_dai[0].cpu_of_node); - of_node_put(odroidx2_dai[0].codec_of_node); - - return 0; -} - -static struct platform_driver odroidx2_audio_driver = { - .driver = { - .name = "odroidx2-audio", - .of_match_table = odroidx2_audio_of_match, - .pm = &snd_soc_pm_ops, - }, - .probe = odroidx2_audio_probe, - .remove = odroidx2_audio_remove, -}; -module_platform_driver(odroidx2_audio_driver); - -MODULE_AUTHOR("Chen Zhen <zhen1.chen(a)samsung.com>"); -MODULE_AUTHOR("Sylwester Nawrocki <s.nawrocki(a)samsung.com>"); -MODULE_DESCRIPTION("ALSA SoC Odroid X2/U3 Audio Support"); -MODULE_LICENSE("GPL v2"); -- 1.9.1

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[alsa-devel] [PATCH 1/3] ASoC: samsung: Remove unused "samsung-i2sv4" platform_device_id entry
by Sylwester Nawrocki 08 Apr '16

08 Apr '16

"samsung-i2sv4" identifier was previously used for the I2S device of the S5PV210 SoCs, it can be removed now when s5pv210 is a dt-only platform. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 70a2559..ededac9 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1492,9 +1492,6 @@ static const struct platform_device_id samsung_i2s_driver_ids[] = { }, { .name = "samsung-i2s-sec", .driver_data = (kernel_ulong_t)&samsung_dai_type_sec, - }, { - .name = "samsung-i2sv4", - .driver_data = (kernel_ulong_t)&i2sv5_dai_type, }, {}, }; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: arizona: Do not create OUT4R widget for CS47L24/WM1831
by Richard Fitzgerald 08 Apr '16

08 Apr '16

The CS47L24 and WM1831 codecs only use the OUT4L widget so we can skip creation of the OUT4R widget. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 92d22a0..d8a6823 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -221,6 +221,8 @@ int arizona_init_spk(struct snd_soc_codec *codec) switch (arizona->type) { case WM8997: + case CS47L24: + case WM1831: break; default: ret = snd_soc_dapm_new_controls(dapm, &arizona_spkr, 1); -- 1.9.1

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[alsa-devel] [PATCH v2 0/6] Propagate errors out from compressed streams
by Charles Keepax 08 Apr '16

08 Apr '16

If the DSP suffers an unrecoverable error, the driver likely knows about this, however the framework may not get informed because errors returned from pointer requests are ignored within the framework. Things work out fine if user-space is doing a read as reads return error status back to user-space so the user can find out that things have gone bad. However, if user-space is doing an avail request there is no path for the error to come back up to user-space. The pointer request returns zero available data, so a read never happens and we basically just end up sitting waiting for data on a stream that we know full well has died. This patch set attempts to address this and ensure that errors are fully propagated to user-space and we don't ever end up wait for data that will never come. Changes since v1: - Based on a recommendation from Vinod, rather than handling errors from pointer requests, add a new call that allows us to inform the compress core that the stream has gone bad. Then return errors from the avail and poll callbacks if the stream is bad. Thanks, Charles Charles Keepax (6): ASoC: compress: Pass error out of soc_compr_pointer ALSA: compress: Add function to indicate the stream has gone bad ALSA: compress: Replace complex if statement with switch ASoC: wm_adsp: Factor out fetching of stream errors from the DSP ASoC: wm_adsp: Improve DSP error handling ASoC: wm_adsp: Use new snd_compr_stop_xrun to signal stream failure include/sound/compress_driver.h | 3 ++ sound/core/compress_offload.c | 67 ++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 43 +++++++++++++++++++------- sound/soc/soc-compress.c | 5 +-- 4 files changed, 101 insertions(+), 17 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH] [linux-next]:ALSA: Fix a typo in timestamping.txt
by Masanari Iida 08 Apr '16

08 Apr '16

This patch fix a spelling typo found in Documentation/sound/alsa/timestamping.txt Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> --- Documentation/sound/alsa/timestamping.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt index 0b191a23f534..1b6473f393a8 100644 --- a/Documentation/sound/alsa/timestamping.txt +++ b/Documentation/sound/alsa/timestamping.txt @@ -129,7 +129,7 @@ will be required to issue multiple queries and perform an interpolation of the results In some hardware-specific configuration, the system timestamp is -latched by a low-level audio subsytem, and the information provided +latched by a low-level audio subsystem, and the information provided back to the driver. Due to potential delays in the communication with the hardware, there is a risk of misalignment with the avail and delay information. To make sure applications are not confused, a -- 2.8.0

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[alsa-devel] [PATCH v3 0/7] topology: Add support for vendor tuples
by mengdong.lin＠linux.intel.com 07 Apr '16

07 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series addes support for vendor tuples to topology, to avoid importing binary data blob from other files. Backward compatibility of ABI is not impacted. A kernel patch is also submitted "ASoC: topology: ABI - Define types for vendor tuples". The 1st patch is small code cleanup. The 2nd patch is a preparation, since tuples will need the type-specific free handler. History: v2: add check on string length, use strtol() to get hex value, and fix memory leak. v3: Fix overflow on uuid string copy. Use strtoul() to get unsigned vendor values and check range, tested on 32 & 64 bit systems. Mengdong Lin (7): topology: Use the generic pointer to free an element's object topology: Define a free handler for the element topology: Add doc for vendor tuples topology: ABI - Define types for vendor tuples topology: Add support for vendor tokens topology: Add support for parsing vendor tuples topology: Build data objects with tuples include/sound/asoc.h | 42 +++- include/topology.h | 79 +++++++- src/topology/data.c | 503 +++++++++++++++++++++++++++++++++++++++++++++- src/topology/elem.c | 15 +- src/topology/parser.c | 24 +++ src/topology/tplg_local.h | 47 +++++ 6 files changed, 702 insertions(+), 8 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH] pcm_plugin: fix appl pointer not correct when mmap_commit() return error
by Shengjiu Wang 07 Apr '16

07 Apr '16

When snd_pcm_mmap_commit() return error, the appl pointer is also updated. which cause the avail_update()'s result wrong. This patch move the snd_pcm_mmap_appl_forward() to the place when snd_pcm_mmap_commit() is successfully returned. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- src/pcm/pcm_plugin.c | 48 ++++++++++++++++++++++++++++++++---------------- 1 file changed, 32 insertions(+), 16 deletions(-) diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index d007e8c..940491d 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -279,18 +279,22 @@ static snd_pcm_sframes_t snd_pcm_plugin_write_areas(snd_pcm_t *pcm, return -EPIPE; } snd_atomic_write_begin(&plugin->watom); - snd_pcm_mmap_appl_forward(pcm, frames); result = snd_pcm_mmap_commit(slave, slave_offset, slave_frames); if (result > 0 && (snd_pcm_uframes_t)result != slave_frames) { snd_pcm_sframes_t res; res = plugin->undo_write(pcm, slave_areas, slave_offset + result, slave_frames, slave_frames - result); - if (res < 0) + if (res < 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : res; + } frames -= res; } - snd_atomic_write_end(&plugin->watom); - if (result <= 0) + if (result <= 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : result; + } + snd_pcm_mmap_appl_forward(pcm, frames); + snd_atomic_write_end(&plugin->watom); offset += frames; xfer += frames; size -= frames; @@ -325,19 +329,23 @@ static snd_pcm_sframes_t snd_pcm_plugin_read_areas(snd_pcm_t *pcm, return -EPIPE; } snd_atomic_write_begin(&plugin->watom); - snd_pcm_mmap_appl_forward(pcm, frames); result = snd_pcm_mmap_commit(slave, slave_offset, slave_frames); if (result > 0 && (snd_pcm_uframes_t)result != slave_frames) { snd_pcm_sframes_t res; res = plugin->undo_read(slave, areas, offset, frames, slave_frames - result); - if (res < 0) + if (res < 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : res; + } frames -= res; } - snd_atomic_write_end(&plugin->watom); - if (result <= 0) + if (result <= 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : result; + } + snd_pcm_mmap_appl_forward(pcm, frames); + snd_atomic_write_end(&plugin->watom); offset += frames; xfer += frames; size -= frames; @@ -423,19 +431,23 @@ snd_pcm_plugin_mmap_commit(snd_pcm_t *pcm, frames = plugin->write(pcm, areas, appl_offset, frames, slave_areas, slave_offset, &slave_frames); snd_atomic_write_begin(&plugin->watom); - snd_pcm_mmap_appl_forward(pcm, frames); result = snd_pcm_mmap_commit(slave, slave_offset, slave_frames); - snd_atomic_write_end(&plugin->watom); if (result > 0 && (snd_pcm_uframes_t)result != slave_frames) { snd_pcm_sframes_t res; res = plugin->undo_write(pcm, slave_areas, slave_offset + result, slave_frames, slave_frames - result); - if (res < 0) + if (res < 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? xfer : res; + } frames -= res; } - if (result <= 0) + if (result <= 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? xfer : result; + } + snd_pcm_mmap_appl_forward(pcm, frames); + snd_atomic_write_end(&plugin->watom); if (frames == cont) appl_offset = 0; else @@ -490,19 +502,23 @@ static snd_pcm_sframes_t snd_pcm_plugin_avail_update(snd_pcm_t *pcm) frames = (plugin->read)(pcm, areas, hw_offset, frames, slave_areas, slave_offset, &slave_frames); snd_atomic_write_begin(&plugin->watom); - snd_pcm_mmap_hw_forward(pcm, frames); result = snd_pcm_mmap_commit(slave, slave_offset, slave_frames); - snd_atomic_write_end(&plugin->watom); if (result > 0 && (snd_pcm_uframes_t)result != slave_frames) { snd_pcm_sframes_t res; res = plugin->undo_read(slave, areas, hw_offset, frames, slave_frames - result); - if (res < 0) + if (res < 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : res; + } frames -= res; } - if (result <= 0) + if (result <= 0) { + snd_atomic_write_end(&plugin->watom); return xfer > 0 ? (snd_pcm_sframes_t)xfer : result; + } + snd_pcm_mmap_hw_forward(pcm, frames); + snd_atomic_write_end(&plugin->watom); if (frames == cont) hw_offset = 0; else -- 1.9.1

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[alsa-devel] Synchonize video with audio playback
by Li, Steve 07 Apr '16

07 Apr '16

Hi, I am new to ALSA. I am trying to implement video and audio playback in my app. I am able to find a lot of sample code for audio playback, but I am wondering what is the best way to synchronize video playback with audio. What kind of timing information can I query through the ALSA interface in terms of the audio playback? Your help is appreciated. Thanks Steve ______________________________________________________________________ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com ______________________________________________________________________

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