June 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda - Don't actually write registers for caps overwrites
by Takashi Iwai 10 Jun '15

10 Jun '15

Along with the transition to regmap for managing the cached parameter reads, the caps overwrite was also moved to regmap cache. The cache change itself works, but it still tries to write the non-existing verb (the HDA parameter is read-only) wrongly. It's harmless in most cases, but some chips are picky and may result in the codec communication stall. This patch avoids it just by adding the missing flag check in reg_write ops. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_regmap.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 7371e0c3926f..c4f5e61d4404 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -265,6 +265,9 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) unsigned int verb; int i, bytes, err; + if (codec->caps_overwriting) + return 0; + reg &= ~0x00080000U; /* drop GET bit */ reg |= (codec->addr << 28); verb = get_verb(reg); -- 2.4.3

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[alsa-devel] OPL3 on cmipci doesn't survive suspend/resume
by Adam Goode 10 Jun '15

10 Jun '15

Hi, I have this card: 05:00.0 Multimedia audio controller: C-Media Electronics Inc CMI8738/CMI8768 PCI Audio (rev 10) Subsystem: C-Media Electronics Inc CMI8738/C3DX PCI Audio Device Flags: bus master, stepping, medium devsel, latency 32, IRQ 19 I/O ports at d000 [size=256] Capabilities: [c0] Power Management version 2 Kernel driver in use: snd_cmipci Kernel modules: snd_cmipci When I suspend/resume, the sound from the OPL3 is permanently distorted. It sounds like not all of the operators are active. I can fix this by doing a full PCI remove/rescan cycle: the driver correctly initializes the card at this time. Any ideas before I peek into the code? Thanks, Adam

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Re: [alsa-devel] [pulseaudio-discuss] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed (-5)
by Raymond Yau 10 Jun '15

10 Jun '15

>> >> Some error messages can be reproduced by set default-sample-format to s24le >> >> But cannot reproduce SNDRV_PCM_IOCTL_HW_PARAMS >> >> Any error meesage in system log , it seem snd_usb_pcm_hw_params return error with your usb sound car >> > The only thing related to sound card found in log, last 3 lines are multiple times repeated afterwards: > > [ 1306.010420] usb 1-6: new full-speed USB device number 23 using xhci_hcd > [ 1306.177219] usb 1-6: New USB device found, idVendor=041e, idProduct=322c > [ 1306.177222] usb 1-6: New USB device strings: Mfr=1, Product=2, SerialNumber=3 > [ 1306.177223] usb 1-6: Product: SB Omni Surround 5.1 > [ 1306.177225] usb 1-6: Manufacturer: Creative Technology Ltd > [ 1306.177226] usb 1-6: SerialNumber: 000000Q6 > [ 1306.177381] usb 1-6: ep 0x83 - rounding interval to 64 microframes, ep desc says 80 microframes > [ 1306.208636] input: Creative Technology Ltd SB Omni Surround 5.1 as /devices/pci0000:00/0000:00:14.0/usb1/1-6/1-6:1.4/0003:041E:322C.0013/input/input42 > [ 1306.259885] hid-generic 0003:041E:322C.0013: input,hidraw0: USB HID v1.11 Keyboard [Creative Technology Ltd SB Omni Surround 5.1] on usb-0000:00:14.0-6/input4 > [ 1306.292463] usb 1-6: Not enough bandwidth for new device state. > [ 1306.292470] usb 1-6: Not enough bandwidth for altsetting 2 > [ 1306.292472] usb 1-6: 3:2: usb_set_interface failed (-28) > > I have no idea why is that the case, device is USB 2.0, even disabling all other external USB devices and pluggin it into USB 3.0 doesn't help. > Nothing more in log. > Can "not enough bandwidth" or "usb_set_interface failed (-28)" message repoduced by using aplay or arecord with s24le/s16le at different rate instead of using pulseaudio ?

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[alsa-devel] wm8731 - redundant register initialization, stream name vs DAPM widgets, snd_soc_io to regmap_io conversion
by Sergey Kiselev 09 Jun '15

09 Jun '15

Hi, I have a few questions regarding wm8731. Some might be applicable to other codecs as well. 1. According to the datasheet bit #8 of all the registers below is set to '0' by the hardware reset. Is there a good reason to set them anyway? (Perhaps an errata in previous chip version?) /* Latch the update bits */ snd_soc_update_bits(codec, WM8731_LOUT1V, 0x100, 0); snd_soc_update_bits(codec, WM8731_ROUT1V, 0x100, 0); snd_soc_update_bits(codec, WM8731_LINVOL, 0x100, 0); snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0); 2. In wm8731's snd_soc_dai_driver structure the steam names are 'Playback' and 'Capture', while DAPM widgets have 'HiFi' prepended to the names: SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8731_PWR, 3, 1), SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8731_PWR, 2, 1), Should the name match? DAPM.txt says so, but the DAPM code actually looks for a substring match (DAI name as a substring in widget name). 3. Do I understand correctly the direction is to move from snd_soc_${io} to more generic regmap_${io}? If so, I'll submit a patch. Thanks -- Sergey

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[alsa-devel] [PATCH v5 0/2] ASoC: qcom: add support to apq8016 sbc machine driver
by Srinivas Kandagatla 09 Jun '15

09 Jun '15

This patchset adds sound card support to APQ8016 SBC board aka DB410c. APQ8016 has 4 MI2S( Primary, Secondary, Tertiary, Quaternary) which can be routed to internal wcd codec or external codecs. This routing and various board specifics are controlled by 2 mux registers. All these patches are tested for HDMI audio via adv7533 bridge and Analog audio on APQ8016-SBC, msm8916-mtp boards. Changes since v4 (https://lkml.org/lkml/2015/5/22/628) - moved all the dt parsing to single function, spotted by Mark. - removed unnecessary setting of card->dev = NULL spotted by Mark. - renamed the dai links to the correct codec names. - removed the special case for EPROBE_DEFER in probe, spotted by Mark. - Dropped all the Changes since comments as most of the patches in last series are merged to topic/qcom branch. --srini Srinivas Kandagatla (2): ASoC: qcom: document apq8016 sbc machine driver bindings ASoC: qcom: add apq8016 sound card support .../devicetree/bindings/sound/qcom,apq8016-sbc.txt | 61 ++++++ sound/soc/qcom/Kconfig | 9 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/apq8016_sbc.c | 208 +++++++++++++++++++++ 4 files changed, 280 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt create mode 100644 sound/soc/qcom/apq8016_sbc.c -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: davinci-mcasp: Choose PCM driver based on configured DMA controller
by Jyri Sarha 09 Jun '15

09 Jun '15

Find the configured DMA controller by asking for a DMA channel in the probe phase and releasing it right after. The controller device can be found via the dma_chan struct and the controller is recognized from the compatible property of its device node. The patch assumes EDMA if there is no device node. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Changes since v2: - Print only one error message if DMA "probing" fails and none in case of -EPROBE_DEFER Changes since first patch version: - Change enum names from MCASP_?DMA to PCM_?DMA - Return PCM_EDMA if !mcasp->dev->of_node - Do not ignore possible error code returned by davinci_mcasp_get_dma_type() sound/soc/davinci/davinci-mcasp.c | 70 +++++++++++++++++++++++++++++++++------ 1 file changed, 60 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d793494..a73cd57 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1565,6 +1565,49 @@ static int davinci_mcasp_init_ch_constraints(struct davinci_mcasp *mcasp) return ret; } +enum { + PCM_EDMA, + PCM_SDMA, +}; +static const char *sdma_prefix = "ti,omap"; + +static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) +{ + struct dma_chan *chan; + const char *tmp; + int ret = PCM_EDMA; + + if (!mcasp->dev->of_node) + return PCM_EDMA; + + tmp = mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data; + chan = dma_request_slave_channel_reason(mcasp->dev, tmp); + if (IS_ERR(chan)) { + if (PTR_ERR(chan) != -EPROBE_DEFER) + dev_err(mcasp->dev, + "Can't verify DMA configuration (%ld)\n", + PTR_ERR(chan)); + return PTR_ERR(chan); + } + BUG_ON(!chan->device || !chan->device->dev); + + if (chan->device->dev->of_node) + ret = of_property_read_string(chan->device->dev->of_node, + "compatible", &tmp); + else + dev_dbg(mcasp->dev, "DMA controller has no of-node\n"); + + dma_release_channel(chan); + if (ret) + return ret; + + dev_dbg(mcasp->dev, "DMA controller compatible = \"%s\"\n", tmp); + if (!strncmp(tmp, sdma_prefix, strlen(sdma_prefix))) + return PCM_SDMA; + + return PCM_EDMA; +} + static int davinci_mcasp_probe(struct platform_device *pdev) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -1763,27 +1806,34 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (ret != 0) goto err; - switch (mcasp->version) { + ret = davinci_mcasp_get_dma_type(mcasp); + switch (ret) { + case PCM_EDMA: #if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ IS_MODULE(CONFIG_SND_EDMA_SOC)) - case MCASP_VERSION_1: - case MCASP_VERSION_2: - case MCASP_VERSION_3: ret = edma_pcm_platform_register(&pdev->dev); - break; +#else + dev_err(&pdev->dev, "Missing SND_EDMA_SOC\n"); + ret = -EINVAL; + goto err; #endif + break; + case PCM_SDMA: #if IS_BUILTIN(CONFIG_SND_OMAP_SOC) || \ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ IS_MODULE(CONFIG_SND_OMAP_SOC)) - case MCASP_VERSION_4: ret = omap_pcm_platform_register(&pdev->dev); - break; +#else + dev_err(&pdev->dev, "Missing SND_SDMA_SOC\n"); + ret = -EINVAL; + goto err; #endif + break; default: - dev_err(&pdev->dev, "Invalid McASP version: %d\n", - mcasp->version); - ret = -EINVAL; + dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); + case -EPROBE_DEFER: + goto err; break; } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rx51: Automatically disconnect non-connected pins
by Lars-Peter Clausen 09 Jun '15

09 Jun '15

According to the schematics the external speaker is differential, so the HPLCOM and HPRCOM pins of the CODEC should be connected to it. Otherwise the routing looks complete, so add the missing routes and set the fully_routed flag of the card instead of manually marking the unused inputs and outputs as not connected. This makes the code a bit shorter and cleaner. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/omap/rx51.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index fded993..3bebfb1 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -245,6 +245,8 @@ static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "HPLOUT"}, {"Ext Spk", NULL, "HPROUT"}, + {"Ext Spk", NULL, "HPLCOM"}, + {"Ext Spk", NULL, "HPRCOM"}, {"Headphone Jack", NULL, "LLOUT"}, {"Headphone Jack", NULL, "RLOUT"}, {"FM Transmitter", NULL, "LLOUT"}, @@ -288,15 +290,8 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = rtd->card; struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card); - - struct snd_soc_dapm_context *dapm = &codec->dapm; int err; - /* Set up NC codec pins */ - snd_soc_dapm_nc_pin(dapm, "MIC3L"); - snd_soc_dapm_nc_pin(dapm, "MIC3R"); - snd_soc_dapm_nc_pin(dapm, "LINE1R"); - err = tpa6130a2_add_controls(codec); if (err < 0) { dev_err(card->dev, "Failed to add TPA6130A2 controls\n"); @@ -383,6 +378,7 @@ static struct snd_soc_card rx51_sound_card = { .num_aux_devs = ARRAY_SIZE(rx51_aux_dev), .codec_conf = rx51_codec_conf, .num_configs = ARRAY_SIZE(rx51_codec_conf), + .fully_routed = true, .controls = aic34_rx51_controls, .num_controls = ARRAY_SIZE(aic34_rx51_controls), -- 2.1.4

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[alsa-devel] [PATCH v2 1/2] ASoC: wm_adsp: Add code_probe and codec_remove stubs
by Richard Fitzgerald 09 Jun '15

09 Jun '15

Currently the only init function in wm_adsp is called by the codec driver early in its probe before the codec has been registered with SOC. This patch adds stubs for the codec_probe and codec_remove stages and calls them from WM5102 and WM5110 codec drivers. This allows us to hang anything that needs setup during the codec probe stage off these functions without further modification of the codec drivers. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 6 ++++++ sound/soc/codecs/wm5110.c | 12 +++++++++++- sound/soc/codecs/wm_adsp.c | 12 ++++++++++++ sound/soc/codecs/wm_adsp.h | 2 ++ 4 files changed, 31 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 11eba0e..341d96e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1875,6 +1875,10 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; + ret = wm_adsp2_codec_probe(&priv->core.adsp[0], codec); + if (ret) + return ret; + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 2); if (ret != 0) return ret; @@ -1893,6 +1897,8 @@ static int wm5102_codec_remove(struct snd_soc_codec *codec) { struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); + wm_adsp2_codec_remove(&priv->core.adsp[0], codec); + priv->core.arizona->dapm = NULL; return 0; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index d65364e..6e15d9c 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1599,7 +1599,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { static int wm5110_codec_probe(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); - int ret; + int i, ret; priv->core.arizona->dapm = &codec->dapm; @@ -1607,6 +1607,12 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_gpio(codec); arizona_init_mono(codec); + for (i = 0; i < WM5110_NUM_ADSP; ++i) { + ret = wm_adsp2_codec_probe(&priv->core.adsp[i], codec); + if (ret) + return ret; + } + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8); if (ret != 0) return ret; @@ -1621,6 +1627,10 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) static int wm5110_codec_remove(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < WM5110_NUM_ADSP; ++i) + wm_adsp2_codec_remove(&priv->core.adsp[i], codec); priv->core.arizona->dapm = NULL; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6e358b5..490edb3 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1906,6 +1906,18 @@ err: } EXPORT_SYMBOL_GPL(wm_adsp2_event); +int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec) +{ + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_codec_probe); + +int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec) +{ + return 0; +} +EXPORT_SYMBOL_GPL(wm_adsp2_codec_remove); + int wm_adsp2_init(struct wm_adsp *dsp) { int ret; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 0e5f07c..5584e34 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -79,6 +79,8 @@ extern const struct snd_kcontrol_new wm_adsp2_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); +int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec); +int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, -- 1.7.2.5

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[alsa-devel] [PATCH 1/2] ASoC: qcom: storm: allocate snd_soc_card struct dynamically.
by Srinivas Kandagatla 09 Jun '15

09 Jun '15

This patch moves static allocation of snd_soc_card to dynamic allocation, the reason to do this is to avoid holding up any dangling pointers in this static structures. And I see no use for having this struct as static given that the card->name is also populated dynamically from dt. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/storm.c | 17 +++++------------ 1 file changed, 5 insertions(+), 12 deletions(-) diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c index b8bd296..ee06613 100644 --- a/sound/soc/qcom/storm.c +++ b/sound/soc/qcom/storm.c @@ -69,11 +69,6 @@ static struct snd_soc_dai_link storm_dai_link = { .ops = &storm_soc_ops, }; -static struct snd_soc_card storm_soc_card = { - .name = "ipq806x-storm", - .dev = NULL, -}; - static int storm_parse_of(struct snd_soc_card *card) { struct snd_soc_dai_link *dai_link = card->dai_link; @@ -99,14 +94,13 @@ static int storm_parse_of(struct snd_soc_card *card) static int storm_platform_probe(struct platform_device *pdev) { - struct snd_soc_card *card = &storm_soc_card; + struct snd_soc_card *card; int ret; - if (card->dev) { - dev_err(&pdev->dev, "%s() error, existing soundcard\n", - __func__); - return -ENODEV; - } + card = devm_kzalloc(&pdev->dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); @@ -129,7 +123,6 @@ static int storm_platform_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret == -EPROBE_DEFER) { - card->dev = NULL; return ret; } else if (ret) { dev_err(&pdev->dev, "%s() error registering soundcard: %d\n", -- 1.9.1

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[alsa-devel] Softvol only after device has opened
by Ricard Wanderlof 09 Jun '15

09 Jun '15

I'm working on implementing support for a MEMS microphone with an I2S connection. Since the microphone has no controls but simply outputs I2S data from the word go, this means that any gain must be applied later. (Microphones of this type are generally specified at outputting 0 dB FS at some rediculously high SPL, 120 dB SPL being typical). Since we use 16 bit audio in our (SOC) system, I'm trying to apply gain using the softvol ALSA plugin, before converting to 16 bit audio (the microphone outputs 24 bit audio). One thing though, the 'softvol' plugin very correctly creates a mixer control and it all works fine, but only after the device has been opened. Before that the mixer control doesn't seem to exist, which means that we cannot adjust the gain until the device is opened by some application. Is this intentional, a bug, or something that is misconfigured on my part? /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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