April 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ALSA: ASoC davinci-pcm, davinci-mcasp Clean up active_serializers
by Michal Bachraty 23 Apr '13

23 Apr '13

As pointed of by Vaibhav, commit message: "ASoC: davinci-mcasp: Add support for multichannel playback" number of active serializers can be hidden into fifo_level variable, which is set in davimci-mcasp. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 8 +++++--- sound/soc/davinci/davinci-pcm.c | 11 +++++------ sound/soc/davinci/davinci-pcm.h | 1 - 3 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 8a9c0ee..63633e9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -834,17 +834,20 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int word_length; u8 fifo_level; u8 slots = dev->tdm_slots; + u8 active_serializers; int channels; struct snd_interval *pcm_channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); channels = pcm_channels->min; + active_serializers = (channels + slots - 1) / slots; + if (davinci_hw_common_param(dev, substream->stream, channels) == -EINVAL) return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - fifo_level = dev->txnumevt; + fifo_level = dev->txnumevt * active_serializers; else - fifo_level = dev->rxnumevt; + fifo_level = dev->rxnumevt * active_serializers; if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); @@ -889,7 +892,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; - dma_params->active_serializers = (channels + slots - 1) / slots; davinci_config_channel_size(dev, word_length); return 0; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 0161579..8460edc 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,7 +182,6 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; - unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -196,14 +195,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) data_type = prtd->params->data_type; count = period_size / data_type; if (fifo_level) - count /= fifo_level * serializers; + count /= fifo_level; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; dst_bidx = 4; - src_cidx = data_type * fifo_level * serializers; + src_cidx = data_type * fifo_level; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -211,7 +210,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level * serializers; + dst_cidx = data_type * fifo_level; } acnt = prtd->params->acnt; @@ -226,8 +225,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) ASYNC); else edma_set_transfer_params(prtd->asp_link[0], acnt, - fifo_level * serializers, - count, fifo_level * serializers, + fifo_level, + count, fifo_level, ABSYNC); } diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 0d84d32..fbb710c 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,7 +27,6 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; - unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ALSA: ASoc: spdif_transciever: Add DT support
by Michal Bachraty 23 Apr '13

23 Apr '13

Add devicetree support for this dummy audio soc driver. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/codecs/spdif_transciever.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index 112a49d..4c8c295 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -20,6 +20,7 @@ #include <sound/soc.h> #include <sound/pcm.h> #include <sound/initval.h> +#include <linux/of.h> #define DRV_NAME "spdif-dit" @@ -52,12 +53,21 @@ static int spdif_dit_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id spdif_dit_dt_ids[] = { + { .compatible = DRV_NAME, }, + { } +}; +MODULE_DEVICE_TABLE(of, spdif_dits_dt_ids); +#endif + static struct platform_driver spdif_dit_driver = { .probe = spdif_dit_probe, .remove = spdif_dit_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(spdif_dit_dt_ids), }, }; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: generic-dmaengine-pcm: call dma_request_slave_channel()
by Shawn Guo 23 Apr '13

23 Apr '13

dma_request_slave_channel() is a more appropriate API for dmaengine clients that adopt generic DMA bindings to call. Let's use it instead of of_dma_request_slave_channel() to save <linux/of_dma.h> include. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/soc-generic-dmaengine-pcm.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 98e1310..e29ec3c 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -21,7 +21,6 @@ #include <sound/soc.h> #include <linux/dma-mapping.h> #include <linux/of.h> -#include <linux/of_dma.h> #include <sound/dmaengine_pcm.h> @@ -219,19 +218,19 @@ static const char * const dmaengine_pcm_dma_channel_names[] = { }; static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, - struct device_node *of_node) + struct device *dev) { unsigned int i; - if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !of_node) + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node) return; if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { - pcm->chan[0] = of_dma_request_slave_channel(of_node, "rx-tx"); + pcm->chan[0] = dma_request_slave_channel(dev, "rx-tx"); pcm->chan[1] = pcm->chan[0]; } else { for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - pcm->chan[i] = of_dma_request_slave_channel(of_node, + pcm->chan[i] = dma_request_slave_channel(dev, dmaengine_pcm_dma_channel_names[i]); } } @@ -255,7 +254,7 @@ int snd_dmaengine_pcm_register(struct device *dev, pcm->config = config; pcm->flags = flags; - dmaengine_pcm_request_chan_of(pcm, dev->of_node); + dmaengine_pcm_request_chan_of(pcm, dev); if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) return snd_soc_add_platform(dev, &pcm->platform, -- 1.7.9.5

3 2

[alsa-devel] [PATCH] ASoC: generic-dmaengine-pcm: use a more common dma name
by Shawn Guo 23 Apr '13

23 Apr '13

The examples in Documentation/devicetree/bindings/dma/dma.txt recommends the name for dma channel doing both RX and TX to be "rx-tx". This becomes a common pattern that has been adopted by platforms that converts to generic DMA bindings. Let's follow this common pattern in generic-dmaengine-pcm. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 5fd5ed4..98e1310 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -227,7 +227,7 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, return; if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { - pcm->chan[0] = of_dma_request_slave_channel(of_node, "tx_rx"); + pcm->chan[0] = of_dma_request_slave_channel(of_node, "rx-tx"); pcm->chan[1] = pcm->chan[0]; } else { for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { -- 1.7.9.5

3 2

[alsa-devel] [PATCH, for-next] ALSA: usb-audio: use FMTBITs in parse_audio_format_i
by Eldad Zack 23 Apr '13

23 Apr '13

Replace the usage of SNDRV_PCM_FORMAT_* macros with the equivalent SNDRV_PCM_FMTBIT_* macros, and the result be can assigned directly to the formats field. Modify the variable name to reflect the change. The following sparse messages are silenced: sound/usb/format.c:377:44: warning: incorrect type in assignment (different base types) sound/usb/format.c:377:44: expected int [signed] pcm_format sound/usb/format.c:377:44: got restricted snd_pcm_format_t [usertype] <noident> sound/usb/format.c:379:44: warning: incorrect type in assignment (different base types) sound/usb/format.c:379:44: expected int [signed] pcm_format sound/usb/format.c:379:44: got restricted snd_pcm_format_t [usertype] <noident> sound/usb/format.c:382:36: warning: incorrect type in assignment (different base types) sound/usb/format.c:382:36: expected int [signed] pcm_format sound/usb/format.c:382:36: got restricted snd_pcm_format_t [usertype] <noident> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/format.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 020ede0..e025e28 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -365,7 +365,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, { struct usb_interface_descriptor *altsd = get_iface_desc(iface); int protocol = altsd->bInterfaceProtocol; - int pcm_format, ret; + int pcm_formats, ret; if (fmt->bFormatType == UAC_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx @@ -377,14 +377,14 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ if (chip->setup == 0x00 && fp->altsetting == 6) - pcm_format = SNDRV_PCM_FORMAT_S16_BE; + pcm_formats = SNDRV_PCM_FMTBIT_S16_BE; else - pcm_format = SNDRV_PCM_FORMAT_S16_LE; + pcm_formats = SNDRV_PCM_FMTBIT_S16_LE; break; default: - pcm_format = SNDRV_PCM_FORMAT_S16_LE; + pcm_formats = SNDRV_PCM_FMTBIT_S16_LE; } - fp->formats = 1uLL << pcm_format; + fp->formats = pcm_formats; } else { fp->formats = parse_audio_format_i_type(chip, fp, format, fmt, protocol); -- 1.8.1.5

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[alsa-devel] [RFC] ASoC: Current state of Dynamic PCM API
by Sebastian Hesselbarth 22 Apr '13

22 Apr '13

Hi, I am rewriting Marvell Kirkwood ASoC driver to also support Dove and reflect the audio controller's capabilities. The audio controller can have one DMA stream for playback and capture respectively. While capture source can be either I2S or SPDIF, playback can be on both interfaces simultaneously. The setup I have in mind for a generic Kirkwood/Dove soundcard is: (DMA)-->(CPU_DAI)--+<-->(I2S Codec) | +--->(SPDIF TX Codec) | +<---(SPDIF RX Codec) I sent a mail on how to handle multiple codecs on a single cpu_dai earlier, and Mark pointed me to Dynamic PCM API. Unfortunately, current mainline ASoC has no driver using DPCM as an example and integration of DPCM patches look kind-of incomplete to me. Can any of you point me to patches, repositories, or documentation about DPCM where I can start with? Regards, Sebastian

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Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Bard 22 Apr '13

22 Apr '13

Dear Mark, Please see my reply below. Thanks. -----Original Message----- From: Mark Brown [mailto:broonie@opensource.wolfsonmicro.com] Sent: Tuesday, April 16, 2013 10:38 PM To: Bard Cc: Flove; Oder Chiou; swarren(a)nvidia.com; bard Subject: Re: [PATCH] ASoC: add RT5640 CODEC driver On Tue, Apr 16, 2013 at 10:10:37PM +0800, bardliao(a)realtek.com wrote: Please remember to post things to the list... just a quick scan through here: > +static int rt5640_reg_init(struct snd_soc_codec *codec) { > + int i; > + for (i = 0; i < RT5640_INIT_REG_LEN; i++) > + snd_soc_write(codec, init_list[i].reg, init_list[i].val); > + return 0; > +} This should be converted over to regmap. There's support for paging in the core, look at regmap_range. Bard: I will use regmap_register_patch to do that. > +/* IN1/IN2 Input Type */ > +static const char * const rt5640_input_mode[] = { > + "Single ended", "Differential"}; > + > +static const SOC_ENUM_SINGLE_DECL( > + rt5640_in1_mode_enum, RT5640_IN1_IN2, > + RT5640_IN_SFT1, rt5640_input_mode); > + > +static const SOC_ENUM_SINGLE_DECL( > + rt5640_in2_mode_enum, RT5640_IN3_IN4, > + RT5640_IN_SFT2, rt5640_input_mode); Platform data... Bard: Can I export a function that machine can configure it by this function? > +static int rt5640_vol_rescale_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) { > + struct soc_mixer_control *mc = > + (struct soc_mixer_control *)kcontrol->private_value; > + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); > + unsigned int val, val2; > + > + val = VOL_RESCALE_MAX_VOL - ucontrol->value.integer.value[0]; > + val2 = VOL_RESCALE_MAX_VOL - ucontrol->value.integer.value[1]; > + return snd_soc_update_bits_locked(codec, mc->reg, RT5640_L_VOL_MASK | > + RT5640_R_VOL_MASK, val << mc->shift | val2); } This looks like a variant on the _RANGE controls? Bard: Yes, we want to limit the max value of volume. If it is not good, I will change it. > +static int hp_event(struct snd_soc_dapm_widget *w, > + struct snd_kcontrol *kcontrol, int event) { > + switch (event) { > + case SND_SOC_DAPM_POST_PMU: > + break; > + > + case SND_SOC_DAPM_PRE_PMD: > + break; > + > + default: > + return 0; > + } > + return 0; > +} Remove this, it doesn't actually do anything any more. Bard: I will remove it. ------Please consider the environment before printing this e-mail.

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[alsa-devel] I connected my digital piano and it just worked :-)
by David Henningsson 22 Apr '13

22 Apr '13

On a slightly different note; I needed to connect my digital piano to my laptop the other day, but the laptop had no MIDI interface. So I tried connecting it through USB, because the piano had a USB port too. I knew it required a 3rd party driver on Windows, so I didn't expect much. It just worked. :-) From someone who knows that fixing hardware often takes a lot of effort and hard work, I just wanted to shout a "thank you" and a "keep up the good work" to all contributors! :-) -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

1 0

[alsa-devel] [PATCH] ALSA: hda - Limit internal mic boost for a few Asus machines
by David Henningsson 22 Apr '13

22 Apr '13

These are being reported as being so noisy at high mic boost levels, so they are unusable in practice. Therefore artificially limit the boosts. BugLink: https://bugs.launchpad.net/bugs/1089795 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 42 ++++++++++++++++++++++++++++++++++++++++- 1 file changed, 41 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 28b3f5a..6e5d383 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3156,6 +3156,38 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, } } +static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + int i; + + /* The mic boosts on level 2 and 3 are too noisy + on the internal mic input. + Therefore limit the boost to 0 or 1. */ + + if (action != HDA_FIXUP_ACT_PROBE) + return; + + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; + unsigned int defcfg; + if (cfg->inputs[i].type != AUTO_PIN_MIC) + continue; + defcfg = snd_hda_codec_get_pincfg(codec, nid); + if (snd_hda_get_input_pin_attr(defcfg) != INPUT_PIN_ATTR_INT) + continue; + + snd_hda_override_amp_caps(codec, nid, HDA_INPUT, + (0x00 << AC_AMPCAP_OFFSET_SHIFT) | + (0x01 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x2f << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); + } +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3190,6 +3222,7 @@ enum { ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, + ALC269_FIXUP_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -3428,6 +3461,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC271_FIXUP_DMIC, }, + [ALC269_FIXUP_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3458,11 +3495,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), + SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), - SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/7] Add capture support to compress API
by Richard Fitzgerald 22 Apr '13

22 Apr '13

This chain of patches does some clearing up and add capture support to the compressed API. Missing is support for capture streams that don't use the copy callback. Also I have included two patches at the end of the chain that remove the buffer pointers and just use the cumulative values however as our setup uses the copy callback these are untested past build so should be taken with caution. Thanks, Charles Thess patches apply to: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git branch "for-next" based on top of commit 126825e ALSA: snd-usb: add quirks handler for DSD streams - RichF Charles Keepax (7): ALSA: compress_core: Update calc_avail to use cumulative values ALSA: compress_core: Calculate avail correctly for capture streams ALSA: compress_core: Deconstify copy callback buffer ALSA: compress_core: Add support for capture streams ASoC: soc-compress: Deduce stream direction ALSA: compress_core: Remove unused hw_pointer ALSA: compress_core: Rework writes to use cumulative values include/sound/compress_driver.h | 4 +- sound/core/compress_offload.c | 117 +++++++++++++++++++++++++------------- sound/soc/soc-compress.c | 11 +++- 3 files changed, 87 insertions(+), 45 deletions(-) -- 1.7.2.5

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[alsa-devel] [PATCH v3] ALSA: snd-usb-audio: set the timeout for usb control set messages to 5000 ms
by Daniel Schürmann 22 Apr '13

22 Apr '13

This is an other version of the patch send via git send-email. I still distinguish between set and get but as long both are 5000 ms GCC will remove it anyway. IMHO this is more easy read and there is no need to explain why we use a get timeout for set messages. Commit message: Set the timeout for USB control set messages according to the USB 2 spec, using the macros from include/linux/usb.h. The get timout becomes 5000 ms even though it is 500 ms in the spec. This patch is required to run the Hercules RMX2 which needs a timeout of 1240 ms. Signed-off-by: Daniel Schürmann <daschuer(a)mixxx.org> --- sound/usb/helper.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/usb/helper.c b/sound/usb/helper.c index c1db28f..6209024 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -86,14 +86,22 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, { int err; void *buf = NULL; + int timeout; if (size > 0) { buf = kmemdup(data, size, GFP_KERNEL); if (!buf) return -ENOMEM; } + + if (requesttype & USB_DIR_IN) + timeout = USB_CTRL_GET_TIMEOUT; + else + timeout = USB_CTRL_SET_TIMEOUT; + err = usb_control_msg(dev, pipe, request, requesttype, - value, index, buf, size, 1000); + value, index, buf, size, timeout); + if (size > 0) { memcpy(data, buf, size); kfree(buf); -- 1.7.9.5

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[alsa-devel] [patch] ALSA: compress: info leak in snd_compr_get_caps()
by Dan Carpenter 22 Apr '13

22 Apr '13

If the ->get_caps() function doesn't clear the buffer then there would stack information leaked to userspace. For example, soc_compr_get_caps() can return success without clearing the buffer. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Perhaps the soc_compr_get_caps() function should return an error code if the platform->driver->compr_ops is NULL. I'm not sure about that, and it's a separate issue anyway. diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index c84abc8..8d3190a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -375,6 +375,7 @@ snd_compr_get_caps(struct snd_compr_stream *stream, unsigned long arg) if (!stream->ops->get_caps) return -ENXIO; + memset(&caps, 0, sizeof(caps)); retval = stream->ops->get_caps(stream, &caps); if (retval) goto out;

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Re: [alsa-devel] [PATCH v2] Add M2Tech hiFace USB-SPDIF driver
by Pavel Hofman 22 Apr '13

22 Apr '13

On 22.4.2013 10:31, Pietro Cipriano wrote: > Il 22/04/2013 09:40, Pavel Hofman ha scritto: >> >> >> On 20.4.2013 22:15, Daniel Mack wrote: >>> On 13.02.2013 18:11, Antonio Ospite wrote: >>>> Add driver for M2Tech hiFace USB-SPDIF interface and compatible >>>> devices. >>>> >>>> M2Tech hiFace and compatible devices offer a Hi-End S/PDIF Output >>>> Interface, see http://www.m2tech.biz/hiface.html >>> >>> Just curious: is there a new version of this patch set coming? >> >> I have read users report large latencies (unable to watch movies) of >> hiFace1 in OSX while reportedly they have not identified any major >> latency in windows. Perhaps the fixed PCM_PACKET_SIZE > The issue is related to a no controllable buffer dimension, which > depends by Audio Core thus present only on OSx. A easy workaround is of > to set up a higher output sampling frequency, for instance at 192KHz the > delay is no more perceptible, obviously the DAC must be compliant with > this setting. > > Pietro Thanks for very fast reponse. So is it possible to reduce the period size in the linux driver? Right now if I calculate correctly it is 4096 bytes / 32bits / 2 channels / 48000 Hz = 10 ms. Thanks, Pavel.

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Re: [alsa-devel] [PATCH 26/28] proc: Supply a function to remove a proc entry by PDE [RFC]
by Rafael J. Wysocki 22 Apr '13

22 Apr '13

On Tuesday, April 16, 2013 07:27:34 PM David Howells wrote: > Supply a function (proc_remove()) to remove a proc entry (and any subtree > rooted there) by proc_dir_entry pointer rather than by name and (optionally) > root dir entry pointer. This allows us to eliminate all remaining pde->name > accesses outside of procfs. > > Signed-off-by: David Howells <dhowells(a)redhat.com> > cc: linux-acpi(a)vger.kernel.org > cc: openipmi-developer(a)lists.sourceforge.net > cc: devicetree-discuss(a)lists.ozlabs.org > cc: linux-pci(a)vger.kernel.org > cc: netdev(a)vger.kernel.org > cc: netfilter-devel(a)vger.kernel.org > cc: alsa-devel(a)alsa-project.org The ACPI changes look good to me. Acked-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> > --- > > drivers/acpi/sbs.c | 21 ++++----------------- > drivers/char/ipmi/ipmi_msghandler.c | 2 +- > drivers/misc/sgi-gru/gruprocfs.c | 2 +- > drivers/of/base.c | 11 +---------- > drivers/pci/proc.c | 12 +++--------- > fs/proc/generic.c | 7 +++++++ > fs/proc/vmcore.c | 2 +- > include/linux/proc_fs.h | 2 ++ > kernel/irq/proc.c | 6 +----- > net/8021q/vlanproc.c | 9 ++------- > net/core/pktgen.c | 6 ++---- > net/ipv4/netfilter/ipt_CLUSTERIP.c | 4 ++-- > net/ipv6/proc.c | 3 +-- > net/netfilter/xt_hashlimit.c | 4 ++-- > sound/core/info.c | 19 +++++-------------- > 15 files changed, 35 insertions(+), 75 deletions(-) > > diff --git a/drivers/acpi/sbs.c b/drivers/acpi/sbs.c > index a296e08..b6241ee 100644 > --- a/drivers/acpi/sbs.c > +++ b/drivers/acpi/sbs.c > @@ -521,19 +521,6 @@ acpi_sbs_add_fs(struct proc_dir_entry **dir, > return 0; > } > > -static void > -acpi_sbs_remove_fs(struct proc_dir_entry **dir, > - struct proc_dir_entry *parent_dir) > -{ > - if (*dir) { > - remove_proc_entry(ACPI_SBS_FILE_INFO, *dir); > - remove_proc_entry(ACPI_SBS_FILE_STATE, *dir); > - remove_proc_entry(ACPI_SBS_FILE_ALARM, *dir); > - remove_proc_entry((*dir)->name, parent_dir); > - *dir = NULL; > - } > -} > - > /* Smart Battery Interface */ > static struct proc_dir_entry *acpi_battery_dir = NULL; > > @@ -836,8 +823,8 @@ static void acpi_battery_remove(struct acpi_sbs *sbs, int id) > power_supply_unregister(&battery->bat); > } > #ifdef CONFIG_ACPI_PROCFS_POWER > - if (battery->proc_entry) > - acpi_sbs_remove_fs(&battery->proc_entry, acpi_battery_dir); > + proc_remove(battery->proc_entry); > + battery->proc_entry = NULL; > #endif > } > > @@ -873,8 +860,8 @@ static void acpi_charger_remove(struct acpi_sbs *sbs) > if (sbs->charger.dev) > power_supply_unregister(&sbs->charger); > #ifdef CONFIG_ACPI_PROCFS_POWER > - if (sbs->charger_entry) > - acpi_sbs_remove_fs(&sbs->charger_entry, acpi_ac_dir); > + proc_remove(sbs->charger_entry); > + sbs->charger_entry = NULL; > #endif > } > -- I speak only for myself. Rafael J. Wysocki, Intel Open Source Technology Center.

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[alsa-devel] [PATCH v2] ALSA: snd-usb-audio: set the timeout for usb control set messages to 5000 ms
by Daniel Schürmann 20 Apr '13

20 Apr '13

Thank you Daniel for review. Here is an updated version of the patch against https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/ 581cbef46ae60073252208fc1f26dd1044f6e215 Set the timeout for USB control set messages according to the USB 2 spec §9.2.6.4 to 5000 ms. To avoid new issues, the get timeout is unchanged at 1000 ms, though it is 500 ms in the spec. This patch is required to run the Hercules RMX2 which needs a timeout > 1240 ms Signed-off-by: Daniel Schürmann <daschuer(a)mixxx.org> --- diff --git a/sound/usb/helper.c b/sound/usb/helper.c index c1db28f..c1caae4 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -23,6 +23,11 @@ #include "helper.h" #include "quirks.h" +/* Value from 9.2.6.4 USB 2 spec */ +#define USB_MSG_SET_TIMEOUT 5000 +/* Value from spec is 500 but we pick 1000 for legacy reasons */ +#define USB_MSG_GET_TIMEOUT 1000 + /* * combine bytes and get an integer value */ @@ -86,14 +91,19 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, { int err; void *buf = NULL; + int timeout; if (size > 0) { buf = kmemdup(data, size, GFP_KERNEL); if (!buf) return -ENOMEM; } + if (requesttype & USB_DIR_IN) + timeout = USB_MSG_GET_TIMEOUT; + else + timeout = USB_MSG_SET_TIMEOUT; err = usb_control_msg(dev, pipe, request, requesttype, - value, index, buf, size, 1000); + value, index, buf, size, timeout); if (size > 0) { memcpy(data, buf, size); kfree(buf);

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[alsa-devel] DAPM widget power off
by yuvaraj r 20 Apr '13

20 Apr '13

Hi All, I have two different dais 1. Hifi Multimedia 2. Voice call dai Before opening voice call dai. I closed Hifi stream. After 5 sec Voice call stream dapm widget got powered off automatically. I am assuming that stream close event of Multimedia causing problem. What will issue in my cause. Thanks Yuvaraj

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[alsa-devel] snd_pcm_avail() and snd_pcm_avail_update() always returns buffer_size
by Nate Case 20 Apr '13

20 Apr '13

Hello, I'm debugging an issue where both snd_pcm_avail_update() and snd_pcm_avail() are always returning buffer_size, even after successfully calling snd_pcm_writei() repeatedly. This is causing problems when I first start playback. My goal was to fill up the playback buffer until it's almost full when first starting, and then enter my regular playback loop where I write period_size at a time. However, I'm not able to tell when the playback buffer is full from snd_pcm_avail() or snd_pcm_avail_update() as expected. In case it's useful, the full snd_pcm_dump() output of the playback device is available here: http://pastebin.com/ai3e0cer To summarize the configuration, I'm using a chain of PCM devices that looks like this: Soft volume -> Rate conversion -> Route conversion -> HW PCM (USB audio) (notes: buffer_size for soft volume PCM is 640 [80ms], period size is 160 [20ms]. buffer_size for HW PCM is 3846 and period_size is 960. I'm opening the playback device in blocking mode, although the same thing appears to happen in non-blocking mode) Here is the sequence of events I'm seeing: 1) Call snd_pcm_writei() with 160 frames successfully. Function returns immediately in less than 1ms because the buffer is empty. 2) snd_pcm_avail() returns 640 (expected 640-160 = 480) snd_pcm_dump() shows the following for the HW PCM device: appl_ptr : 960 hw_ptr : 0 3) Call snd_pcm_writei() again with 160 frames successfully. Function returns immediately again. 4) snd_pcm_avail() returns 640 (expected 480-160 = 320) snd_pcm_dump() shows the following for the HW PCM device: appl_ptr : 1920 hw_ptr : 0 5) Cycle repeats two more times, with these values for apply_ptr/hw_ptr reported after each subsequent snd_pcm_writei() call: appl_ptr : 2880 hw_ptr : 288 appl_ptr : 3840 hw_ptr : 288 All along, snd_pcm_avail() returns 640. 6) Call snd_pcm_writei() with 160 frames again. This time, the buffer is actually full, so it blocks for 20 ms as expected. 7) snd_pcm_avail() returns 640 snd_pcm_dump() shows the following for the HW PCM device: appl_ptr : 4800 hw_ptr : 960 8) Continue this loop, with each subsequent snd_pcm_writei() call blocking for 20 ms, while snd_pcm_avail() always returns 640. Any ideas on why this could be happening? Or am I misunderstanding something here? Thanks, Nate Case <ncase(a)xes-inc.com>

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Re: [alsa-devel] PCM buffering ussues ( or probably the lack of understanding )
by Lars-Peter Clausen 20 Apr '13

20 Apr '13

On 04/20/2013 12:59 PM, marcelg wrote: > Lars Peter, > > Thanks for your reply. > > Some things are more clear now, however one issue I still wonder. > > > Quote > > " >> Suppose I want to have some buffering inside my pcm driver, how will I be >> able to get some samples in advance before starting playback ? Can I just >> call pcm_period_elapsed() a few times in a row to get the buffer filled >> before starting the actual playcback, or does this cause side effects ? >> > Why'd you want to do that? > " > > > If Alsa is not having an internal buffer as you pointed out. An my driver > for example advertises > min periods = 1 and max_periods = 10. Then it means my drivers is capable > of buffering up to 10 periods. > > However what I notice is that the Alsa layer pcm_copy() always supplies > "period_bytes" after each call to period_elapsed() > > This would imply that even my driver is able to buffer 10 periods, it never > receives more data as one period, because I call period_elapsed() > as soon as the data is consumed. The ALSA core keeps track of how many bytes are available in your buffer. It will supply your driver with as many bytes as there is space in the buffer. So if you start playback you should see it supplying you more than one period worth of bytes. Once the buffer is full you'll naturally only get one new period of each period that has elapsed. But as I said before if you don't do anything other then copying data to your buffer in the copy callback, you should probably get rid of the copy callback altogether and let ALSA take care of writing the data to your buffer. This will also allow you to use for example mmap. - Lars

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[alsa-devel] Invitation to connect on LinkedIn
by Barry Song via LinkedIn 20 Apr '13

20 Apr '13

LinkedIn ------------ Barry Song requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, I'd like to add you to my professional network on LinkedIn. - Barry Accept invitation from Barry Song http://www.linkedin.com/e/jpxkk3-hfqntbbx-71/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of Barry Song http://www.linkedin.com/e/jpxkk3-hfqntbbx-71/rso/196461153/a2mu/name/138036… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-hfqntbbx-71/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

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[alsa-devel] PCM buffering ussues ( or probably the lack of understanding )
by marcelg 20 Apr '13

20 Apr '13

I am working on a PCM driver, have some issues and thing which are unclear to me, hopefully somebody here can clarify something for me. My PCM driver is straigh forward: - Copy callback copies the frames received in the callback into a small ringbuffer and starts a DMA transfer to the hardware. - DMA interrupt each times take the next chunk from the ringbuffer. - In the DMA interrupt the number of frames is counted, if te number >= period_size reported in runtime_struct pcm_period_elapsed() is called. - the pointer which is read by the pcm_pointer() callback is updated in each DMA interrupt. When using mplayer this works without problems, however with other applications like mpg123 its causes a lot of problems, for some reaosns I do not understand yet. It has something to do with buffer handling. basically in the cm driver I can influence a few parameters: periods_min/periods_max period_bytes_min, period_bytes_max buffer_size ( should be sufficient to hold periods_max with period_bytes_max I assume ) As far as I understand periods_min/periods_max is the size (number of buffer entries) of the internal ringbuffer Alsa uses. - Why is there both a min and a max ? How is chosen what will be used ? - Currently my period_min is set to 1 and my period_max set to 4, If I increase the period_max value it does not start properly, it seems that the samples are not deliverered in time. Is there any explanation for this, does the Alsa layer not any pre-buffering before the first copy() is called ? - Is there any advice / guidline for period min/max values ? I checked sourecs of severl drivers, some use 1 and some use values like 32 or more for period_max. The same applies to buffer_bytes min/max, currently my min is set to 128bytes , max to 2kB, larger values of max seem to cause more problems. From the documentation I understand that it does not matter if pcm_period_elapsed() is not always called, How dows the ALSA layer then know how many samples are played. the pointer value read the pcm_pointer() only counts from [0..period_max] so what if 2 or more calls to period_elapsed() are skipped ? Suppose I want to have some buffering inside my pcm driver, how will I be able to get some samples in advance before starting playback ? Can I just call pcm_period_elapsed() a few times in a row to get the buffer filled before starting the actual playcback, or does this cause side effects ? I noticed that the number of frames passed to pcm_copy is often less than the periods_size, is this determined by the application ? Lots of questions, but I try to understand the odd behaviour with certain applications ( and the best settings for a pcm driver which will work with most applications ) Any help would be appreciated. Marcel

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Re: [alsa-devel] [PATCH 4/7] ALSA: compress_core: Add support for capture streams
by Vinod Koul 20 Apr '13

20 Apr '13

On Thu, Apr 18, 2013 at 11:02:08AM +0100, Charles Keepax wrote: > Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> > Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> > --- > sound/core/compress_offload.c | 43 +++++++++++++++++++++++++++++++++++++++- > 1 files changed, 41 insertions(+), 2 deletions(-) > > diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c > index 52ca4cc..52a2765 100644 > --- a/sound/core/compress_offload.c > +++ b/sound/core/compress_offload.c > @@ -296,7 +296,41 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, > static ssize_t snd_compr_read(struct file *f, char __user *buf, > size_t count, loff_t *offset) > { > - return -ENXIO; > + struct snd_compr_file *data = f->private_data; > + struct snd_compr_stream *stream; > + size_t avail; > + int retval; > + > + if (snd_BUG_ON(!data)) > + return -EFAULT; > + > + stream = &data->stream; > + mutex_lock(&stream->device->lock); > + > + /* read is allowed when stream is running */ > + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { > + retval = -EBADFD; > + goto out; minor one but shouldnt we allow read even when we have stoped or paused the stream. Stop is important as you might want to get all buffers after you have stopped. > + } > + > + avail = snd_compr_get_avail(stream); > + pr_debug("avail returned %ld\n", (unsigned long)avail); > + /* calculate how much we can read from buffer */ > + if (avail > count) > + avail = count; > + > + if (stream->ops->copy) { > + retval = stream->ops->copy(stream, buf, avail); > + } else { > + retval = -ENXIO; > + goto out; > + } > + if (retval > 0) > + stream->runtime->total_bytes_transferred += retval; > + > +out: > + mutex_unlock(&stream->device->lock); > + return retval; > } > > static int snd_compr_mmap(struct file *f, struct vm_area_struct *vma) > @@ -481,9 +515,14 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) > retval = stream->ops->set_params(stream, params); > if (retval) > goto out; > - stream->runtime->state = SNDRV_PCM_STATE_SETUP; > + > stream->metadata_set = false; > stream->next_track = false; > + > + if (stream->direction == SND_COMPRESS_PLAYBACK) > + stream->runtime->state = SNDRV_PCM_STATE_SETUP; > + else > + stream->runtime->state = SNDRV_PCM_STATE_PREPARED; > } else { > return -EPERM; > } > -- > 1.7.2.5 >

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[alsa-devel] Solved Hercules RMX2
by Daniel Schürmann 20 Apr '13

20 Apr '13

Hallo Alsa Developers, I have fixed Support for Hercules RMX2. You can find the Bug and the patch here: https://bugs.launchpad.net/mixxx/+bug/1096687 The issue was caused by the first time setting the rate. I have changed the usb_msg_timeout to 1500 ms to make it work. The measures time is a ~1240 ms. All following rate settings are finished in not time < 4 ms Sound is now working fine but there are still IO errors just after connecting the device errno 5 - input/output error Here the extended Log: Apr 14 23:50:37 KellerPc kernel: [19213.746929] snd_usb_autoresume() -19 ID:6f8b113 chip->shutdown 0 chip->probing 1. Apr 14 23:50:37 KellerPc kernel: [19213.746934] -5 1 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0x0, type = 4 Apr 14 23:50:37 KellerPc kernel: [19213.747053] snd_usb_autoresume() -19 ID:6f8b113 chip->shutdown 0 chip->probing 1. Apr 14 23:50:37 KellerPc kernel: [19213.747058] -5 1 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0x0, type = 4 Apr 14 23:50:37 KellerPc kernel: [19213.747430] snd_usb_autoresume() -19 ID:6f8b113 chip->shutdown 0 chip->probing 1. Apr 14 23:50:37 KellerPc kernel: [19213.747435] -5 1 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0x0, type = 4 Apr 14 23:50:37 KellerPc kernel: [19213.747554] snd_usb_autoresume() -19 ID:6f8b113 chip->shutdown 0 chip->probing 1. Apr 14 23:50:37 KellerPc kernel: [19213.747559] -5 1 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0x0, type = 4 Apr 14 23:50:37 KellerPc kernel: [19213.752599] usbcore: registered new interface driver snd-usb-audio Apr 14 23:50:37 KellerPc kernel: [19213.775204] snd_usb_ctl_msg() dev:ed5eb000 pipe:80000d80 request:02, requesttype:a1, value:0201, index:0a00, data_size:8. Apr 14 23:50:37 KellerPc kernel: [19213.776264] snd_usb_ctl_msg() dev:ed5eb000 pipe:80000d80 request:02, requesttype:a1, value:0201, index:0a00, data_size:8. Apr 14 23:50:37 KellerPc kernel: [19213.776436] snd_usb_ctl_msg() dev:ed5eb000 pipe:80000d80 request:02, requesttype:a1, value:0201, index:0a00, data_size:8. The current solution is fine for me, but maybe there is an other place for tweaking the startup timing. Any ideas? Is it possible to include the fix in trunk? Thank you. Kind regards, Daniel Schürmann

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[alsa-devel] [PATCH 2/2] ALSA: ASoC: spdif_receiver: Add DT support
by Michal Bachraty 19 Apr '13

19 Apr '13

Add devicetree support for this dummy audio soc driver. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/codecs/spdif_receiver.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c index dd8d856..5d469c7 100644 --- a/sound/soc/codecs/spdif_receiver.c +++ b/sound/soc/codecs/spdif_receiver.c @@ -21,6 +21,9 @@ #include <sound/soc.h> #include <sound/pcm.h> #include <sound/initval.h> +#include <linux/of.h> + +#define DRV_NAME "spdif-dir" #define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -51,12 +54,21 @@ static int spdif_dir_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id spdif_dit_dt_ids[] = { + { .compatible = DRV_NAME, }, + { } +}; +MODULE_DEVICE_TABLE(of, spdif_dits_dt_ids); +#endif + static struct platform_driver spdif_dir_driver = { .probe = spdif_dir_probe, .remove = spdif_dir_remove, .driver = { .name = "spdif-dir", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(spdif_dit_dt_ids), }, }; -- 1.7.9.5

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Re: [alsa-devel] [PATCH 5/7] ASoC: soc-compress: Deduce stream direction
by Vinod Koul 19 Apr '13

19 Apr '13

On Thu, Apr 18, 2013 at 11:02:38AM +0100, Charles Keepax wrote: > Previously we just hard coded all streams as playback streams, this > patch checks the DAI to see if it is a capture or playback stream. It is > worth noting that at this time only unidirectional streams are > supported. well the intention was the compress device would be either playback or record. you should have two devices if you support both. > > Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> > Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> > --- > sound/soc/soc-compress.c | 9 ++++++++- > 1 files changed, 8 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c > index da83e56..3853f7e 100644 > --- a/sound/soc/soc-compress.c > +++ b/sound/soc/soc-compress.c > @@ -384,7 +384,14 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) > /* check client and interface hw capabilities */ > snprintf(new_name, sizeof(new_name), "%s %s-%d", > rtd->dai_link->stream_name, codec_dai->name, num); > - direction = SND_COMPRESS_PLAYBACK; > + > + if (codec_dai->driver->playback.channels_min) > + direction = SND_COMPRESS_PLAYBACK; > + else if (codec_dai->driver->capture.channels_min) > + direction = SND_COMPRESS_CAPTURE; > + else > + return -EINVAL; > + > compr = kzalloc(sizeof(*compr), GFP_KERNEL); > if (compr == NULL) { > snd_printk(KERN_ERR "Cannot allocate compr\n"); > -- > 1.7.2.5 >

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[alsa-devel] ALSA framework problem: FE disconnected, BE still connected!
by lkasam＠codeaurora.org 19 Apr '13

19 Apr '13

Hi Liam, We have a problem encountered in ALSA framework wrt FE/BE disconnection and fix to it is mentioned in below code change. please check it and provide your valuable comments. ---------------------- From: Laxminath Kasam <lkasam(a)codeaurora.org> Date: Wed, 3 Apr 2013 14:03:17 +0530 Subject: [PATCH] ASoC: audio: Fix BE not disconnected even codec teardown Scenario: When FE1(LowLatency) comes in which opens BE1 and puts it in state SND_SOC_DPCM_STATE_START. Similarly FE2(LPA) comes in with connecting to same BE. Now FE2 is in SND_SOC_DPCM_STATE_PREPARE state and waits for TRIGGER_START. Meanwhile FE1 is closed which doesn't affect BE1 and FE2 which are connected. Also, for FE2 pcm_close is called before TRIGGER_START itself. In such a case, BE1 is not getting disconnected while Codec is teared down resulting in failures. Fix: Whenever BE1 is found to be in SND_SOC_DPCM_STATE_START state and FE1 being connected is not started yet, and close sequence happens for this FE, then allow hw_free to happen for BE1 if this is the only FE connected to this BE1. This allows BE to be disconnected which calls afe_close then followed by Codec teardown not resulting in any failures. Signed-off-by: Laxminath Kasam <lkasam(a)codeaurora.org> --- sound/soc/soc-pcm.c | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 915c3c2..ed0ea86 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1765,7 +1765,12 @@ static int soc_dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + !((be->dpcm[stream].state == SND_SOC_DPCM_STATE_START)&& + ((fe->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (fe->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && + (fe->dpcm[stream].state != + SND_SOC_DPCM_STATE_SUSPEND)))) continue; dev_dbg(be->dev, "dpcm: hw_free BE %s\n", -- ----------------------------------------- Please let me know if you are looking for more details. Thanks Kasam

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