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

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Rename the pointer to davinci dma_params as dma_params
by Peter Ujfalusi 09 Mar '14

09 Mar '14

It is quite confusing to name the pointer to davinci_pcm_dma_params as dma_data. Rename the pointer to avoid confusion. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 44 +++++++++++++++++++-------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 35b9b5505f07..b0ae0677f023 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1025,7 +1025,7 @@ nodata: static int davinci_mcasp_probe(struct platform_device *pdev) { - struct davinci_pcm_dma_params *dma_data; + struct davinci_pcm_dma_params *dma_params; struct resource *mem, *ioarea, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; @@ -1094,41 +1094,41 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (dat) mcasp->dat_port = true; - dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; - dma_data->asp_chan_q = pdata->asp_chan_q; - dma_data->ram_chan_q = pdata->ram_chan_q; - dma_data->sram_pool = pdata->sram_pool; - dma_data->sram_size = pdata->sram_size_playback; + dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + dma_params->asp_chan_q = pdata->asp_chan_q; + dma_params->ram_chan_q = pdata->ram_chan_q; + dma_params->sram_pool = pdata->sram_pool; + dma_params->sram_size = pdata->sram_size_playback; if (dat) - dma_data->dma_addr = dat->start; + dma_params->dma_addr = dat->start; else - dma_data->dma_addr = mem->start + pdata->tx_dma_offset; + dma_params->dma_addr = mem->start + pdata->tx_dma_offset; /* Unconditional dmaengine stuff */ - mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = dma_data->dma_addr; + mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = dma_params->dma_addr; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (res) - dma_data->channel = res->start; + dma_params->channel = res->start; else - dma_data->channel = pdata->tx_dma_channel; + dma_params->channel = pdata->tx_dma_channel; - dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; - dma_data->asp_chan_q = pdata->asp_chan_q; - dma_data->ram_chan_q = pdata->ram_chan_q; - dma_data->sram_pool = pdata->sram_pool; - dma_data->sram_size = pdata->sram_size_capture; + dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; + dma_params->asp_chan_q = pdata->asp_chan_q; + dma_params->ram_chan_q = pdata->ram_chan_q; + dma_params->sram_pool = pdata->sram_pool; + dma_params->sram_size = pdata->sram_size_capture; if (dat) - dma_data->dma_addr = dat->start; + dma_params->dma_addr = dat->start; else - dma_data->dma_addr = mem->start + pdata->rx_dma_offset; + dma_params->dma_addr = mem->start + pdata->rx_dma_offset; /* Unconditional dmaengine stuff */ - mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = dma_data->dma_addr; + mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = dma_params->dma_addr; if (mcasp->version < MCASP_VERSION_3) { mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE; - /* dma_data->dma_addr is pointing to the data port address */ + /* dma_params->dma_addr is pointing to the data port address */ mcasp->dat_port = true; } else { mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE; @@ -1136,9 +1136,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (res) - dma_data->channel = res->start; + dma_params->channel = res->start; else - dma_data->channel = pdata->rx_dma_channel; + dma_params->channel = pdata->rx_dma_channel; /* Unconditional dmaengine stuff */ mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = "tx"; -- 1.9.0

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[alsa-devel] [PATCH] ASoC: tlv320aic23: add missing EXPORT_SYMBOLs
by Max Filippov 09 Mar '14

09 Mar '14

This fixes the following build errors when aic23 is configured as module: >> ERROR: "tlv320aic23_probe" >> [sound/soc/codecs/snd-soc-tlv320aic23-i2c.ko] undefined! >> ERROR: "tlv320aic23_regmap" >> [sound/soc/codecs/snd-soc-tlv320aic23-i2c.ko] undefined! Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- sound/soc/codecs/tlv320aic23.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 455250f..a934add 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -59,6 +59,7 @@ const struct regmap_config tlv320aic23_regmap = { .num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg), .cache_type = REGCACHE_RBTREE, }; +EXPORT_SYMBOL(tlv320aic23_regmap); static const char *rec_src_text[] = { "Line", "Mic" }; static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -634,6 +635,7 @@ int tlv320aic23_probe(struct device *dev, struct regmap *regmap) return snd_soc_register_codec(dev, &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1); } +EXPORT_SYMBOL(tlv320aic23_probe); MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS <arunks(a)mistralsolutions.com>"); -- 1.8.1.4

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[alsa-devel] [PATCH] ASoC: s6105-ipcam: Convert to table based DAPM setup
by Lars-Peter Clausen 08 Mar '14

08 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. While we are at it also remove the snd_soc_dapm_enable_pin() in the init callback, since pins are enabled by default. Also drop the snd_soc_dapm_sync() calls, since they are ignored by the core anyway until the card has been fully instantiated. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/s6000/s6105-ipcam.c | 28 ++++++++++------------------ 1 file changed, 10 insertions(+), 18 deletions(-) diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 945e8ab..b2fb37c 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -104,8 +104,8 @@ static int output_type_get(struct snd_kcontrol *kcontrol, static int output_type_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = kcontrol->private_data; - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_card *card = kcontrol->private_data; + struct snd_soc_dapm_context *dapm = &card->dapm; unsigned int val = (ucontrol->value.enumerated.item[0] != 0); char *differential = "Audio Out Differential"; char *stereo = "Audio Out Stereo"; @@ -137,13 +137,7 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Add s6105 specific widgets */ - snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, - ARRAY_SIZE(aic3x_dapm_widgets)); - - /* Set up s6105 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + struct snd_soc_card *card = rtd->card; /* not present */ snd_soc_dapm_nc_pin(dapm, "MONO_LOUT"); @@ -157,17 +151,10 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "RLOUT"); snd_soc_dapm_nc_pin(dapm, "HPRCOM"); - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Audio In"); - /* must correspond to audio_out_mux.private_value initializer */ - snd_soc_dapm_disable_pin(dapm, "Audio Out Differential"); - snd_soc_dapm_sync(dapm); - snd_soc_dapm_enable_pin(dapm, "Audio Out Stereo"); - - snd_soc_dapm_sync(dapm); + snd_soc_dapm_disable_pin(&card->dapm, "Audio Out Differential"); - snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&audio_out_mux, codec)); + snd_ctl_add(card->snd_card, snd_ctl_new1(&audio_out_mux, card)); return 0; } @@ -190,6 +177,11 @@ static struct snd_soc_card snd_soc_card_s6105 = { .owner = THIS_MODULE, .dai_link = &s6105_dai, .num_links = 1, + + .dapm_widgets = aic3x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct s6000_snd_platform_data s6105_snd_data __initdata = { -- 1.8.0

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[alsa-devel] UCM: How does its api relate to ALSA lib api?
by Kamil Wekalski 07 Mar '14

07 Mar '14

Where will be then best to ask these questions?

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[alsa-devel] compress: dpcm_path_get called in wrong place
by noman pouigt 07 Mar '14

07 Mar '14

Shouldn't dpcm_path_get be called earlier ? I think it should be called earlier as happens in soc-pcm file. static int soc_compr_open_fe(struct snd_compr_stream *cstream) { if (dpcm_path_get(fe, stream, &list) <= 0) { dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); In current code we open platform driver and then machine startup function gets called. After this we check for valid path which i think should happen earlier i.e. in the beginning of soc_compr_open_fe function. If you agree, then i will send a patch.

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Re: [alsa-devel] [GIT PULL v2] Enhancement of support for Firewire devices
by Takashi Iwai 07 Mar '14

07 Mar '14

At Wed, 05 Mar 2014 12:13:03 +0900, Takashi Sakamoto wrote: > > This pull-request is an update for my previous pull-request: > [alsa-devel] [GIT PULL][PATCH 00/39] Enhancement of support for Firewire > devices > http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073446.h… > > Updates since previous pull-request: > firewire-lib: > - use swap() instead of local macro > - check NULL at memory allocation for sort table > fireworks: > - use bitmap.h for card slots > - add 'firewire' subdirectory for card node in procfs > - add sanity check of module parameter with clamp() > bebob: > - use bitmap.h for card slots > - add 'firewire' subdirectory for card node in procfs > - simplify codes for metering > - use swap() instead of local macro > - change order of last two commits Clemens, could you review this new series? thanks, Takashi

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[alsa-devel] [TINYCOMPRESS][PATCH 1/1] compress: no need to set metadata before calling next_track
by Richard Fitzgerald 07 Mar '14

07 Mar '14

The metadata is mainly for MP3 gapless playback, since the MP3 audio stream does not contain enough information to enable gapless. Other audio formats do not necessarily require any additional metadata so we should allow next_track to be called without any metadata. Signed-off-by: Zhao Weijia <weijia.zhao(a)capelabs.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- compress.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/compress.c b/compress.c index 15dfdb7..0c9ecd2 100644 --- a/compress.c +++ b/compress.c @@ -534,8 +534,6 @@ int compress_next_track(struct compress *compress) if (!is_compress_running(compress)) return oops(compress, ENODEV, "device not ready"); - if (!compress->gapless_metadata) - return oops(compress, EPERM, "metadata not set"); if (ioctl(compress->fd, SNDRV_COMPRESS_NEXT_TRACK)) return oops(compress, errno, "cannot set next track\n"); compress->next_track = 1; -- 1.7.2.5

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[alsa-devel] [PATCH] ALSA: hda - Fix loud click noise with IdeaPad 410Y
by Takashi Iwai 07 Mar '14

07 Mar '14

Lenovo IdeaPad 410Y with ALC282 codec makes loud click noises at boot and shutdown. Also, it wrongly misdetects the acpi_thinkpad hook. This patch adds a device-specific fixup for disabling the shutup callback that is the cause of the click noise and also avoiding the thinpad_helper calls. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=71511 Reported-and-tested-by: Guilherme Amadio <guilherme.amadio(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 850296a1e0ff..8d0a84436674 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3616,6 +3616,19 @@ static void alc_fixup_auto_mute_via_amp(struct hda_codec *codec, } } +static void alc_no_shutup(struct hda_codec *codec) +{ +} + +static void alc_fixup_no_shutup(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + spec->shutup = alc_no_shutup; + } +} + static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3844,6 +3857,7 @@ enum { ALC269_FIXUP_HP_GPIO_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, + ALC269_FIXUP_NO_SHUTUP, ALC286_FIXUP_SONY_MIC_NO_PRESENCE, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -4020,6 +4034,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC269_FIXUP_NO_SHUTUP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_no_shutup, + }, [ALC269_FIXUP_LENOVO_DOCK] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -4405,6 +4423,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 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, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "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), -- 1.9.0

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Re: [alsa-devel] Call for testing: ALSA driver for Fireworks/BeBoB based devices
by Brand, Mark (Mr) (Summerstrand Campus South) 07 Mar '14

07 Mar '14

Hi Takashi, Thank you once again! Your comments have been very helpful. I now understand how to run ffado-dbus-server/ffado-mixer alongside your ALSA driver. I have pulled your revised Ozonic mixer from ffado-mixer, and when I now invoke ffado-mixer, it properly detects the Ozonic. I can also verify that the mixer control of my audio pass-through (which is plugged into inputs 3 & 4) works as expected. However, I have not yet been able to get audio playback (using "aplay -D plughw:2,0 Desktop/test.wav"). The .wav file is a short, 44.1/16 mono file saved from Audacity, if that's relevant. The terminal output of aplay seems to reflect successful playback, and I hear audible clicks as the file starts and ends its playback. The pass-through is also momentarily muted as aplay starts and stops. I have experimented with every conceivable combination of ffado-mixer settings, ultimately returning to the default settings that it was installed with. When aplay executes, the mixer app shows check marks next to both "Outgoing" and "Incoming". I have been testing the latest git version of the driver (updated this morning), and installing/uninstalling by DKMS. Please, is there anything you can suggest that I should try next? Is there anything obvious that I appear to have overlooked? I will try with a wider range of test materials tomorrow, as well as trying other ALSA apps, but I wanted to get this posted in case that's not the issue. I also recently saw some comments suggesting that pulseaudio be killed. Could that be the issue? I will investigate this tomorrow too. Thank you sincerely, Mark Brand. ________________________________________ From: Takashi Sakamoto [o-takashi(a)sakamocchi.jp] Sent: Sunday, February 23, 2014 9:55 AM To: Brand, Mark (Mr) (Summerstrand Campus South) Cc: alsa-devel(a)alsa-project.org; ffado-devel(a)lists.sf.net Subject: Re: Call for testing: ALSA driver for Fireworks/BeBoB based devices Hi Mark, Thanks for your testing. Ozonic is one of devices for which I test. > My testing is probably not particularly rigorous, and I would > sincerely appreciate your advice on what steps to follow > in order to verify your driver's functionality. Try playback/capture PCM samples/MIDI messages with ALSA applications. > In particular, you mention "(to control mixer channels please use > FFADO upstream)". My drivers don't give ways to control device's internal mixer. So you need the other stuff for this purpose. Please refer to this page. You can see some screenshots of ffado-mixer for 'M-Audio Ozonic'. I already commited these improvement to current FFADO trunk except for FW1814/ProjectMiX: MaudioBebob http://subversion.ffado.org/wiki/MaudioBebob > I have not yet set up FFADO on this machine, but I also see under > "Note": "- Don't use simultaneously 'ALSA PCM/MIDI playback/capture' > and 'jackd with Firewire (FFADO) backend'. Both of them try > connecting to the device when another is running." Would you please > explain? How should one rely on the "upstream" mixer functionality > while keeping jackd out of ALSA's way? Devices which my drivers support handle a pair of streams over IEEE1394 bus. PCM samples/MIDI messages are multiplexed into these streams. And snd-fireworks/snd-bebob uses 'duplex streams synchronization'. As a result: - ALSA PCM/MIDI functionality and jackd with firewire backend conflict for its streaming functionality. But: - ALSA PCM/MIDI fuctionality and ffado-dbus-server/ffado-mixer do not interrupt each other. > I open Audacity, which now lists "Ozonic: Ozonic PCM (hw:2,0)" as > ALSA input and output options. However, no audio input is shown when > I try to monitor the audio input I confirm that Audacity has such bugs to detect ALSA PCM character devices. In detail, its PortAudio v19 codes includes bugs. I have no will to investigate this bug because it's not within my work for these drivers. > $aplay -D hw:2,0 test.wav > (where test.wav has been saved from Audacity) > ... and this responds with: > aplay: main:722: audio open error: Input/output error According to aplay codes, snd_pcm_open() generate this error. I guess that another application still uses PCM character device for Ozonic. > $ aplay -D hw:2,0 Desktop/test.wav > Playing WAVE 'Desktop/test.wav' : Signed 16 bit Little Endian, Rate > 44100 Hz, Mono aplay: set_params:1239: Channels count not available You MUST add 'c' option to indicate the number of channels with 'hw' PCM device. For convinience, use 'plughw' PCM device instead, like: $ aplay -D plughw:2,0 Desktop/test.wav I note an item: - A few days ago, I fixed bug to fail to start stream for Ozonic. See: bebob: fix mappling for MIDI conformant data in stream channels https://github.com/takaswie/snd-firewire-improve/commit/545ab7068f1d53b2b17… Thanks Takashi Sakamoto o-takashi(a)sakamocchi.jp ________________________________ NOTICE: Please note that this eMail, and the contents thereof, is subject to the standard NMMU eMail disclaimer which may be found at: http://www.nmmu.ac.za/disclaimer/email.htm

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[alsa-devel] Controlling the cards order
by Olivier Langlois 07 Mar '14

07 Mar '14

Hi, I have added a second sound device in my system and it unexpectedly took the first card slot. I have found that it is easy to control the order as long as the devices use different drivers by modifying /etc/modprobe.d/alsa.conf ( http://alsa.opensrc.org/MultipleCards ) but in my situation I have 2 different devices using the same driver. I think that there is hope that I find an answer by digging the more general topic of ordering PCI devices drivers loading but if someone has few pointers to share, any help is welcome! Here are my 2 devices that I would like to change the order: lano1106@whippet2 /proc/asound $ cat cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xfbf20000 irq 84 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfbe60000 irq 88 thank you for your time, Olivier

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