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

[alsa-devel] [PATCH] ASoC: Intel: Fix check for pdata usage before dereference.
by Liam Girdwood 05 May '14

05 May '14

This patch fixes the following dereference check ordering. sound/soc/intel/sst-haswell-pcm.c:749 hsw_pcm_probe() warn: variable dereferenced before check 'pdata' (see line 746) git remote add asoc git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git git remote update asoc git checkout 0b708c87f66a15190fb43661c2320fd48c4dc6c8 vim +/pdata +749 sound/soc/intel/sst-haswell-pcm.c a4b12990 Mark Brown 2014-03-12 740 }; a4b12990 Mark Brown 2014-03-12 741 a4b12990 Mark Brown 2014-03-12 742 static int hsw_pcm_probe(struct snd_soc_platform *platform) a4b12990 Mark Brown 2014-03-12 743 { a4b12990 Mark Brown 2014-03-12 744 struct sst_pdata *pdata = dev_get_platdata(platform->dev); a4b12990 Mark Brown 2014-03-12 745 struct hsw_priv_data *priv_data; 0b708c87 Liam Girdwood 2014-05-02 @746 struct device *dma_dev = pdata->dma_dev; 0b708c87 Liam Girdwood 2014-05-02 747 int i, ret = 0; a4b12990 Mark Brown 2014-03-12 748 a4b12990 Mark Brown 2014-03-12 @749 if (!pdata) a4b12990 Mark Brown 2014-03-12 750 return -ENODEV; a4b12990 Mark Brown 2014-03-12 751 a4b12990 Mark Brown 2014-03-12 752 priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 8c6bd33..9d5f64a 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -772,12 +772,14 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) { struct sst_pdata *pdata = dev_get_platdata(platform->dev); struct hsw_priv_data *priv_data; - struct device *dma_dev = pdata->dma_dev; + struct device *dma_dev; int i, ret = 0; if (!pdata) return -ENODEV; + dma_dev = pdata->dma_dev; + priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); priv_data->hsw = pdata->dsp; snd_soc_platform_set_drvdata(platform, priv_data); -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: snd_soc_dapm_put_volsw: Make sure to always update the DAPM graph
by Lars-Peter Clausen 05 May '14

05 May '14

When using auto-muted controls it may happen that the register value will not change when changing a control from enabled to disabled (since the control might be physically disabled due to the auto-muting). We have to make sure to still update the DAPM graph and disconnect the mixer input. Fixes: commit 5729507 ("ASoC: dapm: Implement mixer input auto-disable") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 23 ++++++++++------------- 1 file changed, 10 insertions(+), 13 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1871dfc..5d54dfc 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2791,22 +2791,19 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = dapm_kcontrol_set_value(kcontrol, val); - - if (reg != SND_SOC_NOPM) { - mask = mask << shift; - val = val << shift; - - change = snd_soc_test_bits(codec, reg, mask, val); - } - if (change) { if (reg != SND_SOC_NOPM) { - update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; + mask = mask << shift; + val = val << shift; + + if (snd_soc_test_bits(codec, reg, mask, val)) { + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; + card->update = &update; + } - card->update = &update; } ret = soc_dapm_mixer_update_power(card, kcontrol, connect); -- 1.8.0

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[alsa-devel] [PATCH] ASoC: Make soc_find_matching_codec() static
by Lars-Peter Clausen 05 May '14

05 May '14

The function is only used locally, make it static. Fixes the following warning from sparse: sound/soc/soc-core.c:1644:22: warning: symbol 'soc_find_matching_codec' was not declared. Should it be static? Fixes: 3ca041ed ("ASoC: dt: Allow Aux Codecs to be specified using DT") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ff2b32f..e14bc6c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1641,7 +1641,8 @@ static void soc_unregister_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) } #endif -struct snd_soc_codec *soc_find_matching_codec(struct snd_soc_card *card, int num) +static struct snd_soc_codec *soc_find_matching_codec(struct snd_soc_card *card, + int num) { struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; struct snd_soc_codec *codec; -- 1.8.0

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[alsa-devel] [PATCH 0/5] ASoC: fsl_spdif: Add sysclk df support
by Nicolin Chen 05 May '14

05 May '14

This series of patches mainly add sysclk_df support to derive tx clock from sysclk, and meanwhile, accordingly do some additional improvements. Nicolin Chen (5): ASoC: fsl_spdif: Use clk_set_rate() for spdif root clock only ASoC: fsl_spdif: Rename all _div to _df ASoC: fsl_spdif: Add sysclk df support to derive txclk from sysclk ASoC: fsl_spdif: Print actual sample rate for debug ASoc: fsl_spdif: Add descriptions for fsl_spdif_priv sound/soc/fsl/fsl_spdif.c | 145 ++++++++++++++++++++++++++++++++-------------- sound/soc/fsl/fsl_spdif.h | 14 +++-- 2 files changed, 109 insertions(+), 50 deletions(-) -- 1.8.4

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[alsa-devel] [PATCH] ASoC: rt5640: correct 5640's device ID
by bardliao＠realtek.com 05 May '14

05 May '14

From: Bard Liao <bardliao(a)realtek.com> This patch correct rt5640's device ID Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- We will send another patch to fix the issue about the "Failed to add route" error. --- sound/soc/codecs/rt5640.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index ded2059..895ca14 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -195,7 +195,7 @@ /* SW Reset & Device ID (0x00) */ #define RT5640_ID_MASK (0x3 << 1) #define RT5640_ID_5639 (0x0 << 1) -#define RT5640_ID_5640 (0x1 << 1) +#define RT5640_ID_5640 (0x2 << 1) #define RT5640_ID_5642 (0x3 << 1) -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 1/2] ALSA: hda - hdmi: Set converter channel count even without sink
by Anssi Hannula 05 May '14

05 May '14

Since commit 1df5a06a ("ALSA: hda - hdmi: Fix programmed active channel count") channel count is no longer being set if monitor_present is 0. This is because setting the count was moved after the CA value is determined, which is only after the monitor_present check in hdmi_setup_audio_infoframe(). Unfortunately, in some cases, such as with a non-spec-compliant codec or with a problematic video driver, monitor_present is always 0. As a specific example, this seems to happen with gen1 ATV (SiI1390 codec), causing left-channel-only stereo playback (multi-channel playback has apparently never worked with this codec despite it reporting 8 channels, reason unknown). Simply setting converter channel count without setting the pin infoframe and channel mapping as well does not theoretically make much sense as this will just mean they are out-of-sync and multichannel playback will have a wrong channel mapping. However, adding back just setting the converter channel count even in no-monitor case is the safest change which at least fixes the stereo playback regression on SiI1390 codec. Do that. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Reported-by: Stephan Raue <stephan(a)openelec.tv> Tested-by: Stephan Raue <stephan(a)openelec.tv> Cc: <stable(a)vger.kernel.org> # 3.12+ --- sound/pci/hda/patch_hdmi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0cb5b89cd0c8..1edbb9c47c2d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1127,8 +1127,10 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, AMP_OUT_UNMUTE); eld = &per_pin->sink_eld; - if (!eld->monitor_present) + if (!eld->monitor_present) { + hdmi_set_channel_count(codec, per_pin->cvt_nid, channels); return; + } if (!non_pcm && per_pin->chmap_set) ca = hdmi_manual_channel_allocation(channels, per_pin->chmap); -- 1.8.4.5

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[alsa-devel] [RFC PATCH] ALSA: hda - hdmi: Disallow 3/4/5-channel playback to Statement D2
by Anssi Hannula 05 May '14

05 May '14

Anthem Statement D2 receiver plays only front channels when a 3/4/5-channel HDMI CA is selected. Only 2 and 6 channel maps work properly. Disallow 3/4/5-channel maps based on sink name, allowing userspace to retry with a 6-channel mode with additional silent channels. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Reported-by: Grant Warecki <gjwaudio1(a)hotmail.com> Tested-by: Grant Warecki <gjwaudio1(a)hotmail.com> --- Hi Takashi, I got a report about an issue concerning a fairly obscure high-end Anthem Statement D2 sink, and wrote this patch for it. However, I'm now having second thoughts about whether the kernel is the correct place for this, since (a) this is the only case I've encountered so far (and is obscure), and (b) these kind of issues might get fixed with fw upgrades, and (c) cross-platform media players will have to handle (possibly with manual configuration, though) these cases anyway on non-Linux... WDYT? sound/pci/hda/patch_hdmi.c | 74 ++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 72 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0cb5b89cd0c8..0a9c19cdb511 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -64,6 +64,9 @@ struct hdmi_spec_per_cvt { /* max. connections to a widget */ #define HDA_MAX_CONNECTIONS 32 +/* sink supports only CAs with 2/6 channels */ +#define SINK_QUIRK_2_6_CHANNELS 0x0001 + struct hdmi_spec_per_pin { hda_nid_t pin_nid; int num_mux_nids; @@ -83,6 +86,8 @@ struct hdmi_spec_per_pin { bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ char pcm_name[8]; /* filled in build_pcm callbacks */ + int sink_quirks; /* sink-specific quirks */ + #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; #endif @@ -342,6 +347,13 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { { .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } }, }; +static struct hdmi_sink_quirks { + int mfg_id; + char name[ELD_MAX_MNL + 1]; + int quirks; +} sink_quirks[] = { + { .mfg_id = 0xed40, .name = "Statement D2", .quirks = SINK_QUIRK_2_6_CHANNELS }, +}; /* * HDMI routines @@ -640,6 +652,39 @@ static int get_channel_allocation_order(int ca) return i; } +static unsigned int channels_2_6[] = { + 2, 6 +}; + +static struct snd_pcm_hw_constraint_list hw_constraints_2_6_channels = { + .count = ARRAY_SIZE(channels_2_6), + .list = channels_2_6, + .mask = 0, +}; + +static void set_constraints_from_sink_quirks(struct snd_pcm_runtime *runtime, + int sink_quirks) +{ + if (sink_quirks & SINK_QUIRK_2_6_CHANNELS) { + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &hw_constraints_2_6_channels); + } +} + +static bool ca_allowed_by_sink_quirks(int ca, int sink_quirks) +{ + if (sink_quirks & SINK_QUIRK_2_6_CHANNELS) { + int ordered_ca = get_channel_allocation_order(ca); + if (channel_allocations[ordered_ca].channels != 2 && + channel_allocations[ordered_ca].channels != 6) + return false; + } + + return true; +} + + /* * The transformation takes two steps: * @@ -1482,6 +1527,8 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_hda_spdif_ctls_unassign(codec, pin_idx); return -ENODEV; } + + set_constraints_from_sink_quirks(substream->runtime, per_pin->sink_quirks); } /* Store the updated parameters */ @@ -1518,6 +1565,24 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) return 0; } +static void hdmi_set_sink_quirks(struct hdmi_spec_per_pin *per_pin) +{ + int i; + + per_pin->sink_quirks = 0; + + if (!per_pin->sink_eld.eld_valid) + return; + + for (i = 0; i < ARRAY_SIZE(sink_quirks); i++) { + if (per_pin->sink_eld.info.manufacture_id == sink_quirks[i].mfg_id && + !strcmp(per_pin->sink_eld.info.monitor_name, sink_quirks[i].name)) { + per_pin->sink_quirks = sink_quirks[i].quirks; + break; + } + } +} + static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_jack_tbl *jack; @@ -1592,6 +1657,8 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) pin_eld->eld_size = eld->eld_size; pin_eld->info = eld->info; + hdmi_set_sink_quirks(per_pin); + /* * Re-setup pin and infoframe. This is needed e.g. when * - sink is first plugged-in (infoframe is not set up if !monitor_present) @@ -1916,6 +1983,8 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hda_codec *codec = info->private_data; struct hdmi_spec *spec = codec->spec; + int pin_idx = kcontrol->private_value; + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); unsigned int __user *dst; int chs, count = 0; @@ -1934,7 +2003,8 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, int type = spec->ops.chmap_cea_alloc_validate_get_type(cap, chs); unsigned int tlv_chmap[8]; - if (type < 0) + if (type < 0 || + !ca_allowed_by_sink_quirks(cap->ca_index, per_pin->sink_quirks)) continue; if (size < 8) return -ENOMEM; @@ -2007,7 +2077,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, if (!memcmp(chmap, per_pin->chmap, sizeof(chmap))) return 0; ca = hdmi_manual_channel_allocation(ARRAY_SIZE(chmap), chmap); - if (ca < 0) + if (ca < 0 || !ca_allowed_by_sink_quirks(ca, per_pin->sink_quirks)) return -EINVAL; if (spec->ops.chmap_validate) { err = spec->ops.chmap_validate(ca, ARRAY_SIZE(chmap), chmap); -- 1.8.4.5

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[alsa-devel] [Query] Play two audios simultaneously on two channels
by Prabhakar Lad 05 May '14

05 May '14

Hi folks, Is it possible to play two different audio on two separate channels ? For example a.wav on left channel and b.wav on the right channel. Any inputs on this would be helpful. Thanks, --Prabhakar

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[alsa-devel] arecord - bug with big file?
by korgman 05 May '14

05 May '14

Bugtracker is down -> https://bugtrack.alsa-project.org/alsa-bug from main page -> http://alsa-project.org So, excuse my reporting here. I can't record with arecord for long periods. time arecord -f dat -D hw:0,0 | aplay Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Playing WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo real 186m25.375s user 1m26.204s sys 0m20.772s It seems that this bug was also reported there. https://bugzilla.redhat.com/show_bug.cgi?id=213783 cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k3.12.9x40-122l. (custom kernel) arecord --version arecord: version 1.0.27.1 by Jaroslav Kysela <perex(a)perex.cz> I now trying with sox (rec test.wav) to check if sox is ok. Thank you for reading this!

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[alsa-devel] [Query]: Play only selected channel data
by Prabhakar Lad 05 May '14

05 May '14

Hi, I am working on application which captures data from a microphone, my requirement is to play only left channel data is it possible to get to know the recieved data is for left channel or right ? Any hints clues would be helpful Regards, --Prabhakar

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