Alsa-devel September 2013

alsa-devel@alsa-project.org

119 participants
213 discussions

[alsa-devel] [PATCH] ASoC: kirkwood: fix compilation warning in kirkwood_dma_open
by Vladimir Murzin 01 Oct '13

01 Oct '13

writel() supposes the first argument of type unsigned int. This fix the warning: sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_open': sound/soc/kirkwood/kirkwood-dma.c:164:3: warning: large integer implicitly truncated to unsigned type [-Woverflow] Signed-off-by: Vladimir Murzin <murzin.v(a)gmail.com> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index b238434..1135c30 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -161,7 +161,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) * Enable Error interrupts. We're only ack'ing them but * it's useful for diagnostics */ - writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK); + writel((unsigned int)-1, priv->io + KIRKWOOD_ERR_MASK); } dram = mv_mbus_dram_info(); -- 1.8.1.5

2 1

[alsa-devel] [PATCH] snd_tlv_convert_from_dB: fix decreasing gain across entries
by Stephen Warren 01 Oct '13

01 Oct '13

From: Stephen Warren <swarren(a)nvidia.com> Currently, for a TLV consisting of TLV_DB_SCALE_ITEMs, if e.g. alsamixer calls snd_mixer_selem_set_playback_dB() with a value that is in-between two TLV_DB_SCALE_ITEMs, and xdir is negative, the selected raw hardware value is the minimum in the first range above that value, rather than the maximum in the last range below that value. The user-visible symptom is that in alsamixer, pressing the down key to reduce the value sticks at certain points, and cannot be incrementally reduced any further, although directly selecting a much lower value (e.g. by pressing 0..9) works as expected. This is triggered e.g. by sound/soc/codec/max98090.c's max98090_hp_tlv[]. Fix this by checking whether xdir is positive or not, rather than checking whether it has a non-zero value. The code to select the previous range's max value is already present. This matches how xdir is used in other parts of the code. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- src/control/tlv.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/control/tlv.c b/src/control/tlv.c index 6b0b9f4..b08d887 100644 --- a/src/control/tlv.c +++ b/src/control/tlv.c @@ -312,7 +312,7 @@ int snd_tlv_convert_from_dB(unsigned int *tlv, long rangemin, long rangemax, submin, submax, db_gain, value, xdir); else if (db_gain < dbmin) { - *value = xdir || pos == 2 ? submin : prev_submax; + *value = xdir > 0 || pos == 2 ? submin : prev_submax; return 0; } prev_submax = submax; -- 1.8.1.5

2 1

[alsa-devel] [PATCH] amixer: actually print all TLVs in a container TLV
by Anssi Hannula 01 Oct '13

01 Oct '13

Due to a wrong indexing of tlv[] decode_tlv() always skips to the end of the container TLV since the used tlv[1] contains the container TLV size instead of the intended element size. This causes, for example, only the first HDMI channel map TLV to be shown. Fix the indexing to actually use the element size. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- amixer/amixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index 7727943..b83b0c3 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -468,7 +468,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ return; } decode_tlv(spaces + 2, tlv + idx, tlv[idx+1] + 8); - idx += 2 + (tlv[1] + sizeof(unsigned int) - 1) / sizeof(unsigned int); + idx += 2 + (tlv[idx+1] + sizeof(unsigned int) - 1) / sizeof(unsigned int); } break; case SND_CTL_TLVT_DB_SCALE: -- 1.8.1.5

2 1

[alsa-devel] [PATCH v2] ASoC: imx-sgtl5000: do not use devres on a foreign device
by Philipp Zabel 01 Oct '13

01 Oct '13

Calling devm_clk_get with any device pointer other than our own confuses devres. Use clk_get instead. This avoids hitting the following warning in the imx-sgtl5000 error path: imx-sgtl5000 sound.12: snd_soc_register_card failed (-517) platform sound.12: Driver imx-sgtl5000 requests probe deferral ------------[ cut here ]------------ WARNING: CPU: 0 PID: 75 at drivers/base/dd.c:272 driver_probe_device+0x194/0x218() Modules linked in: snd_soc_sgtl5000(+) snd_soc_imx_sgtl5000 coda snd_soc_imx_audmux imx_sdma snd_soc_fsl_spdif snd_soc_fsl_ssi CPU: 0 PID: 75 Comm: udevd Not tainted 3.11.0-rc6+ #4682 Backtrace: [<80010bc4>] (dump_backtrace+0x0/0x10c) from [<80010d60>] (show_stack+0x18/0x1c) r6:00000110 r5:00000009 r4:00000000 r3:00000000 [<80010d48>] (show_stack+0x0/0x1c) from [<804f0764>] (dump_stack+0x20/0x28) [<804f0744>] (dump_stack+0x0/0x28) from [<8001a4a4>] (warn_slowpath_common+0x6c/0x8c) [<8001a438>] (warn_slowpath_common+0x0/0x8c) from [<8001a4e8>] (warn_slowpath_null+0x24/0x2c) r8:7f032000 r7:7f02f93c r6:cf8eaa54 r5:cf8eaa20 r4:80728a0c [<8001a4c4>] (warn_slowpath_null+0x0/0x2c) from [<80286bdc>] (driver_probe_device+0x194/0x218) [<80286a48>] (driver_probe_device+0x0/0x218) from [<80286cf4>] (__driver_attach+0x94/0x98) r7:00000000 r6:cf8eaa54 r5:7f02f93c r4:cf8eaa20 [<80286c60>] (__driver_attach+0x0/0x98) from [<802851c8>] (bus_for_each_dev+0x5c/0x90) r6:80286c60 r5:7f02f93c r4:00000000 r3:cf8ef03c [<8028516c>] (bus_for_each_dev+0x0/0x90) from [<80286654>] (driver_attach+0x24/0x28) r6:806d0424 r5:cf16a580 r4:7f02f93c [<80286630>] (driver_attach+0x0/0x28) from [<802861e4>] (bus_add_driver+0xdc/0x234) [<80286108>] (bus_add_driver+0x0/0x234) from [<802871d4>] (driver_register+0x80/0x154) r8:7f032000 r7:00000001 r6:7f02fa68 r5:7f02fa74 r4:7f02f93c [<80287154>] (driver_register+0x0/0x154) from [<8033c278>] (i2c_register_driver+0x34/0xbc) [<8033c244>] (i2c_register_driver+0x0/0xbc) from [<7f032018>] (sgtl5000_i2c_driver_init+0x18/0x24 [snd_soc_sgtl5000]) r5:7f02fa74 r4:cfb7ff48 [<7f032000>] (sgtl5000_i2c_driver_init+0x0/0x24 [snd_soc_sgtl5000]) from [<80008738>] (do_one_initcall+0xf4/0x150) [<80008644>] (do_one_initcall+0x0/0x150) from [<80053f64>] (load_module+0x174c/0x1db4) [<80052818>] (load_module+0x0/0x1db4) from [<800546ac>] (SyS_init_module+0xe0/0xf4) [<800545cc>] (SyS_init_module+0x0/0xf4) from [<8000e540>] (ret_fast_syscall+0x0/0x30) r6:00005b22 r5:00afed68 r4:00000000 ---[ end trace b24c5c3bb145dbdd ]--- Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> Reviewed-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Rebased onto asoc/for-next, adding back imx_sgtl5000_remove. --- sound/soc/fsl/imx-sgtl5000.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 78f86d8..c49e4cb 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -128,7 +128,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) goto fail; } - data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + data->codec_clk = clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); goto fail; @@ -172,6 +172,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; fail: + if (!IS_ERR(data->codec_clk)) + clk_put(data->codec_clk); if (ssi_np) of_node_put(ssi_np); if (codec_np) @@ -180,6 +182,15 @@ fail: return ret; } +static int imx_sgtl5000_remove(struct platform_device *pdev) +{ + struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); + + clk_put(data->codec_clk); + + return 0; +} + static const struct of_device_id imx_sgtl5000_dt_ids[] = { { .compatible = "fsl,imx-audio-sgtl5000", }, { /* sentinel */ } -- 1.8.4.rc3

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[alsa-devel] [PATCH 2/2] [RESEND] snd-ac97-codec: Use complete ID when the codec is not known
by Ondrej Zary 30 Sep '13

30 Sep '13

Currently, if the codec ID is not known, the last byte of the ID is used as name. It's not helpful at all, use full codec ID instead. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/ac97/ac97_codec.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index bf578ba2..30a40a8 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1830,7 +1830,7 @@ void snd_ac97_get_name(struct snd_ac97 *ac97, unsigned int id, char *name, int m pid->patch(ac97); } } else - sprintf(name + strlen(name), " id %x", id & 0xff); + sprintf(name + strlen(name), " id %x", id); } /** -- Ondrej Zary

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[alsa-devel] [PATCH] ASoC: generic: Add generic DT based simple codec
by Jean-Francois Moine 30 Sep '13

30 Sep '13

This patch adds a simple sound codec which is described by the DT. This codec may be used when no specific codec action is needed. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- .../devicetree/bindings/sound/simple-codec.txt | 103 +++++++++++++ sound/soc/generic/Kconfig | 6 + sound/soc/generic/Makefile | 2 + sound/soc/generic/simple-codec.c | 197 +++++++++++++++++++++++++ 4 files changed, 3088 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-codec.txt b/Documentation/devicetree/bindings/sound/simple-codec.txt new file mode 100644 index 0000000..75be747 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-codec.txt @@ -0,0 +1,103 @@ +Device-Tree bindings for the simple codec + +Required properties: +- compatible: should be "linux,simple-codec". +- dai-name: name of the codec + +Optional properties: +- capture: information about capture +- playback: information about playback +At least one of the 'capture' or 'playback' nodes must be present. + +Child 'capture' and 'playback' required properties: +- stream-name: name of the stream +- formats: list of the supported formats (see below) +- rates: list of the supported rates (see below) +- #channels: minimum and maximum numbers of channels + +Formats: + "s8" + "u8" + "s16_le" + "s16_be" + "u16_le" + "u16_be" + "s24_le" + "s24_be" + "u24_le" + "u24_be" + "s32_le" + "s32_be" + "u32_le" + "u32_be" + "float_le" + "float_be" + "float64_le" + "float64_be" + "iec958_subframe_le" + "iec958_subframe_be" + "mu_law" + "a_law" + "ima_adpcm" + "mpeg" + "gsm" + "special" + "s24_3l" + "s24_3be" + "u24_3le" + "u24_3be" + "s20_3le" + "s20_3be" + "u20_3le" + "u20_3be" + "s18_3le" + "s18_3b" + "u18_3le" + "u18_3be" + "g723_24" + "g723_24_1b" + "g723_40" + "g723_40_1b" + "dsd_u8" + "dsd_u16_le" + +Rates: + 5512 + 8000 + 11025 + 16000 + 22050 + 32000 + 44100 + 48000 + 64000 + 88200 + 96000 + 176400 + 192000 + +Examples: + + spdifo: spdif-transmitter { + compatible = "linux,simple-codec"; + dai-name = "dit-hifi"; + playback { + stream-name = "S/PDIF Playback"; + formats = "s16_le"; + rates = <8000 11025 16000 22050 32000 44100 + 48000 64000 88200 96000>; + #channels = <1 384>; + }; + }; + + hdmio: hdmi-transmitter { + compatible = "linux,simple-codec"; + dai-name = "hdmi-hifi"; + playback { + stream-name = "HDMI Playback"; + formats = "s16_le", "s24_le"; + rates = <32000 44100 48000 88200 96000 + 176400 192000>; + #channels = <2 8>; + }; + }; diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index 610f612..d9221539 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -2,3 +2,9 @@ config SND_SIMPLE_CARD tristate "ASoC Simple sound card support" help This option enables generic simple sound card support + +config SND_SIMPLE_CODEC + tristate "ASoC Simple sound codec support" + depends on OF + help + This option enables generic simple sound codec support diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9c3b246..f94620e 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -1,3 +1,5 @@ snd-soc-simple-card-objs := simple-card.o +snd-soc-simple-codec-objs := simple-codec.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o +obj-$(CONFIG_SND_SIMPLE_CODEC) += snd-soc-simple-codec.o diff --git a/sound/soc/generic/simple-codec.c b/sound/soc/generic/simple-codec.c new file mode 100644 index 0000000..217cd0e --- /dev/null +++ b/sound/soc/generic/simple-codec.c @@ -0,0 +1,197 @@ +/* + * ALSA SoC simple codec driver + * + * This driver is used by controllers which do not need any codec + * (s/pdif, hdmi..). + * Its parameters are built from a DT description. + * + * Copyright: (C) 2013 Jean-François Moine <moinejf(a)free.fr> + * + * 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. + */ + +#include <linux/module.h> +#include <sound/soc.h> +#include <linux/of.h> + +#define DRV_NAME "simple-codec" + +static struct snd_soc_codec_driver soc_codec_simple_codec; + +static char *format_tb[] = { + "s8", "u8", "s16_le", "s16_be", + "u16_le", "u16_be", "s24_le", "s24_be", + "u24_le", "u24_be", "s32_le", "s32_be", + "u32_le", "u32_be", "float_le", "float_be", + + "float64_le", "float64_be", "iec958_subframe_le", "iec958_subframe_be", + "mu_law", "a_law", "ima_adpcm", "mpeg", + "gsm", "", "", "", + "", "", "", "special", + + "s24_3l", "s24_3be", "u24_3le", "u24_3be", + "s20_3le", "s20_3be", "u20_3le", "u20_3be", + "s18_3le", "s18_3b", "u18_3le", "u18_3be", + "g723_24", "g723_24_1b", "g723_40", "g723_40_1b", + + "dsd_u8", "dsd_u16_le", +}; + +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 +#error "Change this table" +#endif +static u32 rate_tb[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, + 48000, 64000, 88200, 96000, 176400, 192000 }; + +static int get_format(u64 *p_formats, char *p) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(format_tb); i++) { + if (strcmp(format_tb[i], p) == 0) { + *p_formats |= 1 << i; + return 0; + } + } + return -EINVAL; +} + +static int get_rate(u32 *p_rates, u32 val) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rate_tb); i++) { + if (rate_tb[i] == val) { + *p_rates |= 1 << i; + return 0; + } + } + return -EINVAL; +} + +static int stream_populate(struct device *dev, + struct snd_soc_pcm_stream *stream, + struct device_node *np) +{ + u64 formats; + u32 rates, val; + char *p; + int i, ret; + + ret = of_property_read_string(np, "stream-name", &stream->stream_name); + if (ret) { + dev_err(dev, "Failed to parse stream-name string\n"); + return ret; + } + + i = 0; + formats = 0; + for (;;) { + ret = of_property_read_string_index(np, "formats", i, + (const char **) &p); + if (ret) + break; + ret = get_format(&formats, p); + if (ret) { + dev_err(dev, "Bad codec format\n"); + return ret; + } + i++; + } + stream->formats = formats; + + i = 0; + rates = 0; + for (;;) { + ret = of_property_read_u32_index(np, "rates", i, &val); + if (ret) + break; + ret = get_rate(&rates, val); + if (ret) { + dev_err(dev, "Bad rate\n"); + return ret; + } + i++; + } + stream->rates = rates; + + ret = of_property_read_u32_index(np, "#channels", 0, &val); + if (ret) { + dev_err(dev, "Bad min channel\n"); + return ret; + } + stream->channels_min = val; + ret = of_property_read_u32_index(np, "#channels", 1, &val); + if (ret) { + dev_err(dev, "Bad max channel\n"); + return ret; + } + stream->channels_max = val; + + return 0; +} + +static int simple_codec_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct device_node *np = dev->of_node; + struct device_node *np2; + struct snd_soc_dai_driver *dai; + int ret; + + dai = devm_kzalloc(dev, sizeof *dai, GFP_KERNEL); + if (!dai) + return -ENOMEM; + + ret = of_property_read_string(np, "dai-name", &dai->name); + if (ret) { + dev_err(dev, "Failed to parse dai-name\n"); + return ret; + } + + np2 = of_find_node_by_name(np, "capture"); + if (np2) { + ret = stream_populate(dev, &dai->capture, np2); + if (ret) + return ret; + } + np2 = of_find_node_by_name(np, "playback"); + if (np2) { + ret = stream_populate(dev, &dai->playback, np2); + if (ret) + return ret; + } + + return snd_soc_register_codec(dev, &soc_codec_simple_codec, + dai, 1); +} + +static int simple_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static const struct of_device_id simple_codec_of_match[] = { + { .compatible = "linux,simple-codec", }, + { } +}; +MODULE_DEVICE_TABLE(of, simple_codec_of_match); + +static struct platform_driver simple_codec_driver = { + .probe = simple_codec_probe, + .remove = simple_codec_remove, + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = simple_codec_of_match, + }, +}; + +module_platform_driver(simple_codec_driver); + +MODULE_AUTHOR("Jean-Francois Moine <moinejf(a)free.fr>"); +MODULE_DESCRIPTION("simple codec driver"); +MODULE_LICENSE("GPL"); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH] ASoC: pcm: Remove extra spaces from dev_ prints
by Jarkko Nikula 30 Sep '13

30 Sep '13

dev_ prints are already prefixed by ": " before format string so there is no need for extra spaces. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/soc-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 330c9a6..d449872 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -721,7 +721,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); - dev_dbg(fe->dev, " connected new DPCM %s path %s %s %s\n", + dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, stream ? "<-" : "->", be->dai_link->name); @@ -749,7 +749,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, if (dpcm->fe == fe) continue; - dev_dbg(fe->dev, " reparent %s path %s %s %s\n", + dev_dbg(fe->dev, "reparent %s path %s %s %s\n", stream ? "capture" : "playback", dpcm->fe->dai_link->name, stream ? "<-" : "->", dpcm->be->dai_link->name); @@ -773,7 +773,7 @@ static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) if (dpcm->state != SND_SOC_DPCM_LINK_STATE_FREE) continue; - dev_dbg(fe->dev, " freed DSP %s path %s %s %s\n", + dev_dbg(fe->dev, "freed DSP %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, stream ? "<-" : "->", dpcm->be->dai_link->name); @@ -2116,7 +2116,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) pcm->private_free = platform->driver->pcm_free; out: - dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name, + dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; } -- 1.8.4.rc3

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[alsa-devel] [RFC] ALSA: Reduce delay for a blocking pcm_drain.
by Dylan Reid 30 Sep '13

30 Sep '13

This patch addresses two issues related to calling snd_pcm_drain. If no_period_wakeup is set, then snd_pcm_drain would wait MAX_SCHEDULE_TIMEOUT jiffies, without a wakeup pending this will leave the calling task blocked indefinitely. Instead block for double the buffer size played back at a low sample rate (4k chosen arbitrarily). Also if the stream is running with period wakeups but with a long period, the delay could be seconds. If only a small part of the buffer is being used, this is unnecessary. Instead wait for the remaining samples to play out, plus one millisecond. This allows systems that fill the buffer only part way to still do a blocking drain before closing to avoid losing samples. Change-Id: I84ee1a52d9bdf80ea6065fb9533e565d3e6c8021 Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- This can be worked around in user space as well by using a nonblocking drain and sleeping based on how many samples are queued. Would that be the right thing to do or is this (or something similar) worth fixing here. To test this I opened a pcm and set no period wakeups, played a few samples and then called drain, which hung indefinitely. Using a USB headset with a max period size of 128k but only 15ms of the buffer filled, having the drain time shorter without losing samples makes a blocking call feasible. --- sound/core/pcm_native.c | 17 +++++++---------- 1 file changed, 7 insertions(+), 10 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f4aaf5a..a5f0695 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1481,7 +1481,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, } for (;;) { - long tout; + long tout, tout_ms; struct snd_pcm_runtime *to_check; if (signal_pending(current)) { result = -ERESTARTSYS; @@ -1505,16 +1505,13 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, snd_pcm_stream_unlock_irq(substream); up_read(&snd_pcm_link_rwsem); snd_power_unlock(card); - if (runtime->no_period_wakeup) - tout = MAX_SCHEDULE_TIMEOUT; - else { - tout = 10; - if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; - tout = max(t, tout); - } - tout = msecs_to_jiffies(tout * 1000); + tout_ms = runtime->buffer_size * 2 * 1000 / 4000; + if (runtime->rate) { + snd_pcm_sframes_t hw_avail; + hw_avail = snd_pcm_playback_hw_avail(runtime); + tout_ms = 1 + hw_avail * 1000 / runtime->rate; } + tout = msecs_to_jiffies(tout_ms); tout = schedule_timeout_interruptible(tout); snd_power_lock(card); down_read(&snd_pcm_link_rwsem); -- 1.8.1.3.605.g02339dd

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[alsa-devel] [PATCH] Enable internal mic on a Thinkpad machine
by David Henningsson 30 Sep '13

30 Sep '13

From: Kailang <kailang(a)realtek.com> More thorough testing showed that these verbs were necessary to improve quality of the internal mic. Patch originally from Realtek. Cc: stable(a)kernel.org (3.11) BugLink: https://bugs.launchpad.net/bugs/1231931 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bc07d36..c4d54a9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3531,6 +3531,7 @@ enum { ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, ALC282_FIXUP_ASUS_TX300, + ALC283_FIXUP_INT_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -3790,6 +3791,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc282_fixup_asus_tx300, }, + [ALC283_FIXUP_INT_MIC] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x20, AC_VERB_SET_COEF_INDEX, 0x1a}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0011}, + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3874,7 +3885,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), -- 1.7.9.5

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Re: [alsa-devel] pcm_rate API question
by Rob Sykes 30 Sep '13

30 Sep '13

> From: Rob Sykes <aquegg(a)yahoo.co.uk> >To: "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org> >Sent: Friday, 27 September 2013, 16:28 >Subject: Re: [alsa-devel] pcm_rate API question > >> From: Clemens Ladisch <clemens(a)ladisch.de> >> To: Rob Sykes <aquegg(a)yahoo.co.uk>; "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org> >> Cc: >> Sent: Friday, 27 September 2013, 13:38 >> Subject: Re: [alsa-devel] pcm_rate API question >> >> Rob Sykes wrote: >>> void (*convert_s16)(void *obj, int16_t *dst, unsigned int >> dst_frames, >>> const int16_t *src, unsigned int src_frames); >>> >>> is the plugin obliged always to consume src_frames and generate dst_frames >> samples? >> >> Yes. src_frames and dst_frames are the period sizes of the respective >> devices; their values will never change, and they define the actual >> conversion ratio. > >Thanks, Clemens. Typically there is some buffering/delay within a resampler -- how does the pcm_rate API allow for this? E.g. from the libsamplerate FAQ: > >Q6 : I'm use the SRC_SINC_* converters and up-sampling by a ratio of 2. I reset the converter and put in 1000 samples and I expect to get 2000 samples out, but I'm getting less than that. Why? >The short answer is that there is a transport delay inside the converter itself. Long answer follows. By way of example, the first time you call src_process() you might only get 1900 samples out. > However, after that first call all subsequent calls will probably get you about 2000 samples out for every 1000 samples you put in. Also, just noticed that when performing a rate conversion from 44100 -> 48000, the buffer sizes used by alsa are 5512 and 6000, however the actual ratio for the required conversion is 5512.5 to 6000. If, as you say above, the buffer sizes define the conversion ratio, then this seems to imply that alsa is unable to perform accurate rate conversion for one of the most common uses! — am I missing something here? A pointer to the API documentation, or perhaps some relevant threads, would be very useful. Cheers, Rob

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Alsa-devel September 2013