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

[alsa-devel] [TINYCOMPRESS][PATCH] compress: compress_wait() must return error if timed out
by Richard Fitzgerald 12 Nov '13

12 Nov '13

The caller must be certain that a return of 0 really means that compress is ready for more data, so when poll() returns 0 for a timeout we must report that as an error. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- compress.c | 18 ++++++++++-------- 1 files changed, 10 insertions(+), 8 deletions(-) diff --git a/compress.c b/compress.c index 5cd0966..d1a2283 100644 --- a/compress.c +++ b/compress.c @@ -616,15 +616,17 @@ int compress_wait(struct compress *compress, int timeout_ms) fds.events = POLLOUT | POLLIN; ret = poll(&fds, 1, timeout_ms); - if (fds.revents & POLLERR) { - return oops(compress, EIO, "poll returned error!"); + if (ret > 0) { + if (fds.revents & POLLERR) + return oops(compress, EIO, "poll returned error!"); + if (fds.revents & (POLLOUT | POLLIN)) + return 0; } - /* A pause will cause -EBADFD or zero. */ - if ((ret < 0) && (ret != -EBADFD)) + if (ret == 0) + return oops(compress, ETIME, "poll timed out"); + if (ret < 0) return oops(compress, errno, "poll error"); - if (fds.revents & (POLLOUT | POLLIN)) { - return 0; - } - return ret; + + return oops(compress, EIO, "poll signalled unhandled event"); } -- 1.7.2.5

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Re: [alsa-devel] no reset_resume for driver snd-usb-audio for logitech headset H600
by Alan Stern 10 Nov '13

10 Nov '13

On Tue, 8 Oct 2013 baumber(a)hotmail.com wrote: > Hello, > > please help me with the bug for my Logitech Headset; > > https://bugzilla.kernel.org/show_bug.cgi?id=62691 > > Bug ID: 62691 > Summary: no reset_resume for driver snd-usb-audio for logitech headset H600 > > Thank you > > Best regards, Bernhard Actually this should be brought to the attention of the ALSA developers (CC'ed). They are the people in charge of the snd-usb-audio driver. Alan Stern

4 8

[alsa-devel] [RFC/RFT v2 0/4] ALSA: hda - hdmi: ATI/AMD multi-channel and HBR support
by Anssi Hannula 09 Nov '13

09 Nov '13

Hi all! Here is a second revision of the ATI/AMD multichannel patch, now a patchset. Since the last revision from a bit over the week ago, AMD has released more documentation ( http://www.x.org/docs/AMD/AMD_HDA_verbs_v2.pdf ), so the patchset contains new additions: - HBR bitstreaming support (HD passthrough, for DTS-HD and Dolby TrueHD) o This completes the set, now we have HBR on NVIDIA, Intel and AMD :) - More complete ELD emulation on revision 3+ (0x100300) codecs o Untested, though from what I can see the newly added fields may not be filled in yet by the radeon driver. - Ramp up/down configuration according to non-pcm bit on rev3+ o I guess this means gradual fade-in/out? Untested. Other changes since the previous post: - ELD emulation moved to hda_eld.c and cleaned up a bit - Some fixes to hdmi_chmap_ctl_tlv() changes (and maybe something else I forgot - if so, sorry) Also, the pairwise channel mapping code has been tested to work with custom channel maps. The TLV contents are shown as follows by amixer: | chmap-paired=FL,FR | chmap-paired=FL,FR,NA,LFE | chmap-paired=FL,FR,FC,NA | chmap-paired=FL,FR,RL,RR | chmap-paired=FL,FR,FC,LFE | chmap-paired=FL,FR,NA,LFE,RL,RR | chmap-paired=FL,FR,FC,NA,RL,RR [etc] Rafał, do you know about the missing (AFAICS) support for lipsync (F7B) and sink information (F81) in radeon driver? This patchset requires at least more testing before it should be applied. A few other somewhat open things: - I named everything ati/ATI except for is_amdhdmi_rev3() and has_amd_full_remap_support() that are AMD-only, though AMD-only verbs I still named ATI to have similar names to other verbs. Does that seem sensible or how should this be done? - Currently setup_audio_infoframe() writes the channel mapping every time for ATI/AMD, should we try to avoid that? If so, any suggestions? (argh, just noticed this is broken for manual channel maps on non-ATI; if just the chmap is changed, channel mapping is not updated; so this requires fixes in the generic side as well) I'm especially interested in testers with: - Older codecs other than 0x1002aa01. My best guess still is that the new code works on them as well. o On these I'd like to know if multichannel and the new formats work, i.e. e.g. speaker-test -D hdmi:CARD=Generic,DEV=0 -c8 -r192000 -F S32_LE - Codec 0x1002aa01 revisions 0x100300 (Rev ID 3) or later (see /proc/asound/cardX/codec#Y). These are HD7750+ I think. Stuff to be tested on these: o The ramp up/down stuff, i.e. patch 4. Is there any difference seen with these, in the beginning/end (i.e. fade-out/in): speaker-test -D hdmi:CARD=Generic,DEV=0,AES0=0x04 -c2 -r48000 speaker-test -D hdmi:CARD=Generic,DEV=0,AES0=0x06 -c2 -r48000 Also, is there a difference in the beginning of these (maybe garbage sound and/or slightly slower startup?): aplay -Dhdmi:CARD=Generic,DEV=0,AES0=4 -r44100 -f s16_le -c2 testi.dts.cut.spdif aplay -Dhdmi:CARD=Generic,DEV=0,AES0=6 -r44100 -f s16_le -c2 testi.dts.cut.spdif o Contents of /proc/asound/cardX/eld#0. I'd like to see the contents both with radeon and with the proprietary fglrx driver in use Olivier, can you test again on your rev3 hw? :) The patchset can be found in combined form (for e.g. testing purposes) at: http://onse.fi/files/atihdmi5.patch The test file referenced above can be found at: http://onse.fi/files/testi.dts.cut.spdif.gz (just regular DTS) Anssi Hannula (4): ALSA: hda - hdmi: Add ATI/AMD multi-channel audio support ALSA: hda - hdmi: Add ELD emulation for ATI/AMD codecs ALSA: hda - hdmi: Add HBR bitstreaming support for ATI/AMD HDMI codecs ALSA: hda - hdmi: Disable ramp-up/down for non-PCM on AMD codecs sound/pci/hda/hda_eld.c | 157 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_local.h | 5 +++ sound/pci/hda/patch_hdmi.c | 424 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++------------------ 3 files changed, 547 insertions(+), 39 deletions(-) -- Anssi Hannula

4 29

[alsa-devel] async between dmaengine_pcm_dma_complete and snd_pcm_release
by Qiao Zhou 05 Nov '13

05 Nov '13

Hi Mark, Liam, Jaroslav, Takashi I met an issue in which kernel panic appears in dmaengine_pcm_dma_complete function on a quad-core system. The dmaengine_pcm_dma_complete is running core0, while snd_pcm_release has already been executed on core1, due to in low memory stress oom killer kills the audio thread to release some memory. snd_pcm_release frees the runtime parameters, and runtime is used in dmaengine_pcm_dma_complete, which is a callback from tasklet in dmaengine. In current audio driver, we can't promise that dmaengine_pcm_dma_complete is not executed after snd_pcm_release on multi cores. Maybe we should add some protection. Do you have any suggestion? I have tried to apply below workaround, which can fix the panic, but I'm not confident it's proper. Need your comment and better suggestion. From d568a88e8f66ee21d44324bdfb48d2a3106cf0d1 Mon Sep 17 00:00:00 2001 From: Qiao Zhou <zhouqiao(a)marvell.com> Date: Wed, 9 Oct 2013 15:24:29 +0800 Subject: [PATCH] ASoC: soc-dmaengine: add mutex to protect runtime param under SMP arch, the dmaengine_pcm_dma_complete callback has the chance to run on one cpu while at the same time, the substream is released on another cpu. thus it may access param which is already freed. we need to add mutes to protect such access, and check PCM availability before using it. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/soc-dmaengine-pcm.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 111b7d9..5917029 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -125,13 +125,22 @@ EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config); static void dmaengine_pcm_dma_complete(void *arg) { struct snd_pcm_substream *substream = arg; - struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct dmaengine_pcm_runtime_data *prtd; + + mutex_lock(&substream->pcm->open_mutex); + if (!substream || !substream->runtime) { + mutex_unlock(&substream->pcm->open_mutex); + return; + } + + prtd = substream_to_prtd(substream); prtd->pos += snd_pcm_lib_period_bytes(substream); if (prtd->pos >= snd_pcm_lib_buffer_bytes(substream)) prtd->pos = 0; snd_pcm_period_elapsed(substream); + mutex_unlock(&substream->pcm->open_mutex); } static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) -- 1.7.0.4 -- Best Regards Qiao

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[alsa-devel] NVIDIA HDMI: Channel remapping does not work for stereo?
by Anssi Hannula 04 Nov '13

04 Nov '13

Hi! I just noticed while testing the ALSA HDMI channel remapping code that my NVIDIA codec 0x10de0015 does not seem to follow the Audio Sample Packet (ASP) channel mapping (set by verb F34h as per HDA spec 7.3.3.41) when playing back stereo audio. This can be reproduced by e.g.: $ speaker-test -c2 -Dhdmi:CARD=NVidia,DEV=0 -m FR,FL The channels are reversed compared to what the printout is (i.e. the codec plays the streams straight-through instead of swapping channels as instructed). E.g. this works expectedly, however: $ speaker-test -c4 -Dhdmi:CARD=NVidia,DEV=1 -m FR,FL,RR,RL (note that recent sound git master or for-next is needed for these kind of remappings to work properly at all) On Intel HDMI codec 0x80862806 both work. While stereo right-left swapping is admittedly a rather corner-case situation, I guess we should probably refuse such chmaps when they are not supported. Aaron, do you have any information on whether this affects all NVIDIA HDMI codecs? -- Anssi Hannula

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[alsa-devel] [PATCH v2 5/6] ASoC: sirf-inner: add mach driver for SiRFSoC internal codec
by Barry Song 04 Nov '13

04 Nov '13

From: Rongjun Ying <Rongjun.Ying(a)csr.com> This connects DMA, CPU DAI and Codec DAI together and works as a mach driver. Signed-off-by: Rongjun Ying <Rongjun.Ying(a)csr.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- -v2: use extcon for both Linux and Android support; use DAPM to replace legacy kcontrol sound/soc/sirf/Kconfig | 5 + sound/soc/sirf/Makefile | 2 + sound/soc/sirf/sirf-inner.c | 237 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 244 insertions(+) create mode 100644 sound/soc/sirf/sirf-inner.c diff --git a/sound/soc/sirf/Kconfig b/sound/soc/sirf/Kconfig index afa3952..564b0ec 100644 --- a/sound/soc/sirf/Kconfig +++ b/sound/soc/sirf/Kconfig @@ -11,3 +11,8 @@ config SND_SIRF_SOC_INNER config SND_SOC_SIRF_USP tristate + +config SND_SIRF_INNER + tristate "SoC Audio support for SiRF inner codec of SiRF EVB" + depends on SND_SIRF_SOC + select SND_SIRF_SOC_INNER diff --git a/sound/soc/sirf/Makefile b/sound/soc/sirf/Makefile index 8517c67..80abdf6 100644 --- a/sound/soc/sirf/Makefile +++ b/sound/soc/sirf/Makefile @@ -1,9 +1,11 @@ snd-soc-sirf-objs := sirf-pcm.o +snd-soc-sirf-inner-objs := sirf-inner.o snd-soc-sirf-soc-inner-objs := sirf-soc-inner.o snd-soc-sirf-i2s-objs := sirf-i2s.o snd-soc-sirf-usp-objs := sirf-usp.o obj-$(CONFIG_SND_SIRF_SOC) += snd-soc-sirf.o +obj-$(CONFIG_SND_SIRF_INNER) += snd-soc-sirf-inner.o obj-$(CONFIG_SND_SIRF_SOC_INNER) += snd-soc-sirf-soc-inner.o obj-$(CONFIG_SND_SOC_SIRF_I2S) += snd-soc-sirf-i2s.o obj-$(CONFIG_SND_SOC_SIRF_USP) += snd-soc-sirf-usp.o diff --git a/sound/soc/sirf/sirf-inner.c b/sound/soc/sirf/sirf-inner.c new file mode 100644 index 0000000..e36ac5e --- /dev/null +++ b/sound/soc/sirf/sirf-inner.c @@ -0,0 +1,237 @@ +/* + * SiRF inner audio device driver + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#include <linux/platform_device.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> + +#include <linux/extcon.h> +#include <linux/extcon/extcon-gpio.h> + +#define SIRF_JACK_GPIO_DEBOUNCE_TIME 200 /* in ms */ +struct sirf_inner_extcon_info { + struct platform_device pdev; + struct gpio_extcon_platform_data extcon_data; + int last_state; + int state_changed; +}; + +struct sirf_inner_card { + unsigned int gpio_hp_pa; + unsigned int gpio_spk_pa; + struct platform_device *sirf_inner_device; + struct sirf_inner_extcon_info extcon_info; +}; + +static int sirf_inner_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *ctrl, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct sirf_inner_card *sinner_card = snd_soc_card_get_drvdata(card); + int on = !SND_SOC_DAPM_EVENT_OFF(event); + if (gpio_is_valid(sinner_card->gpio_hp_pa)) + gpio_direction_output(sinner_card->gpio_hp_pa, on); + return 0; +} + +static int sirf_inner_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *ctrl, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct sirf_inner_card *sinner_card = snd_soc_card_get_drvdata(card); + int on = !SND_SOC_DAPM_EVENT_OFF(event); + + if (sinner_card->extcon_info.state_changed) { + sinner_card->extcon_info.state_changed = 0; + if (!sinner_card->extcon_info.last_state) + gpio_direction_output(sinner_card->gpio_spk_pa, 0); + else + gpio_direction_output(sinner_card->gpio_spk_pa, 1); + } else { + if (gpio_is_valid(sinner_card->gpio_spk_pa)) + gpio_direction_output(sinner_card->gpio_spk_pa, on); + } + + return 0; +} +static const struct snd_soc_dapm_widget sirf_inner_dapm_widgets[] = { + SND_SOC_DAPM_HP("Hp", sirf_inner_hp_event), + SND_SOC_DAPM_SPK("Ext Spk", sirf_inner_spk_event), + SND_SOC_DAPM_MIC("Ext Mic", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"Hp", NULL, "HPOUTL"}, + {"Hp", NULL, "HPOUTR"}, + {"Ext Spk", NULL, "SPKOUT"}, + {"MICIN1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Ext Mic"}, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sirf_inner_dai_links[] = { + { + .name = "SiRF inner", + .stream_name = "SiRF inner", + .codec_dai_name = "sirf-soc-inner", + .platform_name = "sirf-pcm-audio.1", + }, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_sirf_inner_card = { + .name = "SiRF inner", + .owner = THIS_MODULE, + .dai_link = sirf_inner_dai_links, + .num_links = ARRAY_SIZE(sirf_inner_dai_links), + .dapm_widgets = sirf_inner_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sirf_inner_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), +}; + +static int sirf_inner_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_sirf_inner_card; + struct sirf_inner_card *sinner_card; + int ret; + + sinner_card = devm_kzalloc(&pdev->dev, sizeof(struct sirf_inner_card), + GFP_KERNEL); + if (sinner_card == NULL) + return -ENOMEM; + + sinner_card->sirf_inner_device = platform_device_alloc("soc-audio", -1); + if (!sinner_card->sirf_inner_device) + return -ENOMEM; + + sirf_inner_dai_links[0].cpu_of_node = + of_parse_phandle(pdev->dev.of_node, "sirf,inner-platform", 0); + sirf_inner_dai_links[0].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "sirf,inner-codec", 0); + sinner_card->gpio_spk_pa = of_get_named_gpio(pdev->dev.of_node, + "spk-pa-gpios", 0); + sinner_card->gpio_hp_pa = of_get_named_gpio(pdev->dev.of_node, + "hp-pa-gpios", 0); + if (gpio_is_valid(sinner_card->gpio_spk_pa)) + gpio_request(sinner_card->gpio_spk_pa, "SPA_PA_SD"); + if (gpio_is_valid(sinner_card->gpio_hp_pa)) + gpio_request(sinner_card->gpio_hp_pa, "HP_PA_SD"); + + card->dev = &pdev->dev; + snd_soc_card_set_drvdata(card, sinner_card); + platform_set_drvdata(pdev, card); + if (gpio_is_valid(sinner_card->gpio_hp_pa)) + gpio_direction_output(sinner_card->gpio_hp_pa, 0); + if (gpio_is_valid(sinner_card->gpio_spk_pa)) + gpio_direction_output(sinner_card->gpio_spk_pa, 0); + + platform_set_drvdata(sinner_card->sirf_inner_device, + &snd_soc_sirf_inner_card); + + ret = platform_device_add(sinner_card->sirf_inner_device); + if (ret) { + platform_device_put(sinner_card->sirf_inner_device); + return ret; + } + + sinner_card->extcon_info.extcon_data.name = "h2w"; + sinner_card->extcon_info.extcon_data.debounce = + SIRF_JACK_GPIO_DEBOUNCE_TIME; + sinner_card->extcon_info.extcon_data.irq_flags = + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_SHARED; + sinner_card->extcon_info.extcon_data.state_on = "0"; + sinner_card->extcon_info.extcon_data.state_off = "1"; + sinner_card->extcon_info.extcon_data.gpio = + of_get_named_gpio(pdev->dev.of_node, + "hp-switch-gpios", 0); + + sinner_card->extcon_info.pdev.name = "extcon-gpio"; + sinner_card->extcon_info.pdev.id = pdev->id; + sinner_card->extcon_info.pdev.dev.platform_data = + &sinner_card->extcon_info.extcon_data; + + return platform_device_register(&sinner_card->extcon_info.pdev); +} + +static int sirf_inner_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct sirf_inner_card *sinner_card = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(sinner_card->gpio_hp_pa)) + gpio_free(sinner_card->gpio_hp_pa); + if (gpio_is_valid(sinner_card->gpio_spk_pa)) + gpio_free(sinner_card->gpio_spk_pa); + + platform_device_put(sinner_card->sirf_inner_device); + return 0; +} + +#ifdef CONFIG_PM_SLEEP +static int sirf_inner_resume(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + struct sirf_inner_card *sinner_card = snd_soc_card_get_drvdata(card); + struct extcon_dev *edev; + int state; + + edev = extcon_get_extcon_dev(sinner_card->extcon_info.extcon_data.name); + state = gpio_get_value(sinner_card->extcon_info.extcon_data.gpio); + if (state != sinner_card->extcon_info.last_state) { + sinner_card->extcon_info.state_changed = 1; + sinner_card->extcon_info.last_state = state; + extcon_set_state(edev, state); + } + return 0; +} + +static int sirf_inner_suspend(struct device *dev) +{ + struct snd_soc_card *card = dev_get_drvdata(dev); + struct sirf_inner_card *sinner_card = snd_soc_card_get_drvdata(card); + sinner_card->extcon_info.last_state = gpio_get_value(sinner_card->extcon_info.extcon_data.gpio); + if (gpio_is_valid(sinner_card->gpio_spk_pa)) + gpio_direction_output(sinner_card->gpio_spk_pa, 0); + return 0; +} +#endif + +static const struct of_device_id sirf_inner_of_match[] = { + {.compatible = "sirf,sirf-inner", }, + { }, +}; +MODULE_DEVICE_TABLE(of, sirf_inner_of_match); + +static const struct dev_pm_ops sirf_inner_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(sirf_inner_suspend, sirf_inner_resume) +}; + +static struct platform_driver sirf_inner_driver = { + .driver = { + .name = "sirf-inner", + .owner = THIS_MODULE, + .pm = &sirf_inner_pm_ops, + .of_match_table = sirf_inner_of_match, + }, + .probe = sirf_inner_probe, + .remove = sirf_inner_remove, +}; +module_platform_driver(sirf_inner_driver); + +MODULE_AUTHOR("RongJun Ying <RongJun.Ying(a)csr.com>"); +MODULE_DESCRIPTION("ALSA SoC SIRF inner AUDIO driver"); +MODULE_LICENSE("GPL v2"); -- 1.8.2.3

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[alsa-devel] include/uapi/firewire.h for other firewire drivers
by Takashi Sakamoto 03 Nov '13

03 Nov '13

Hi Clemens, About include/uapi/firewire.h, I'd like you to give me an advice. I'm sorry but I have little knowledgement about Dice chipset. So here I want to know what I should do for snd-fireworks/snd-bebob. The header file includes an interface for users to know its already streaming or not, and lock it. But for Fireworks/BeBoB, I think checking CMP connections do the same thing. Current FFADO can be failed when CMP connections on the device are established, and ALSA can do the same. (but there are some problems. both drivers set sampling rate before checking CMP connections. For this I have a plan to post a patch for both FFADO and ALSA.) Should I implement the same lock for Fireworks/BeBoB drivers? Regards Takashi Sakamoto

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[alsa-devel] [PATCH] ASoC: dpcm: improve robustness
by Russell King - ARM Linux 01 Nov '13

01 Nov '13

Mark, Liam, You may also wish to consider adding this patch as well - it avoids oopsing due to a NULL be_substream when writing be_substream->runtime if the frontend is marked as having a playback/capture stream but there is no backend. I found this with Liam's "Explicitly set capture/playback support in DAI links" patch applied, and I'd configured the FE to have both capture and playback, but I'd forgotten to configure the BE appropriately. It seems nicer to gracefully handle this than oops the kernel. 8<===== From: Russell King <rmk+kernel(a)arm.linux.org.uk> ASoC: dpcm: improve robustness Avoid oopsing if there is no backend stream associated with a front end stream. Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> --- sound/soc/soc-pcm.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7798072c45ea..419af79fa30e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1036,6 +1036,12 @@ static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); + if (!be_substream) { + dev_err(be->dev, "ASoC: no backend %s stream\n", + stream ? "capture" : "playback"); + continue; + } + /* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue; @@ -1053,7 +1059,8 @@ static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) (be->dpcm[stream].state != SND_SOC_DPCM_STATE_CLOSE)) continue; - dev_dbg(be->dev, "ASoC: open BE %s\n", be->dai_link->name); + dev_dbg(be->dev, "ASoC: open %s BE %s\n", + stream ? "capture" : "playback", be->dai_link->name); be_substream->runtime = be->dpcm[stream].runtime; err = soc_pcm_open(be_substream); -- 1.7.4.4

2 1

[alsa-devel] Audinst HUD-mx2 covertly muting itself
by Tobias Klausmann 01 Nov '13

01 Nov '13

Hey, I have an Audinst HUD-mx2 external USB audio device: Bus 007 Device 006: ID 040d:340f VIA Technologies, Inc. $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HUDmx2 [Audinst HUD-mx2], device 0: USB Audio [USB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: HUDmx2 [Audinst HUD-mx2], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 While the device works right in principle, every time an app closes the device, it mutes itself (or crash, this may be a Hw problem). There is no indication of the broken state in the mixer displays of alsamixer or alsamixer-gui, and amixer output does not indicate any change, either (see below). I am a bit stumped as to what to do next. The vendor does not officially support anything but Windows, so getting help from them is very unlikely. amixer output in working/not working/working again condition: ~ $ amixer -c0 contents # working numid=1,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=2,iface=MIXER,name='PCM Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=3,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=45,step=0 : values=45,45 | dBminmax-min=42949627.96dB,max=0.00dB ~ $ amixer -c0 contents # not working numid=1,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=2,iface=MIXER,name='PCM Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=3,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=45,step=0 : values=45,45 | dBminmax-min=42949627.96dB,max=0.00dB ~ $ amixer -c0 contents # working again numid=1,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=2,iface=MIXER,name='PCM Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=3,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=45,step=0 : values=45,45 | dBminmax-min=42949627.96dB,max=0.00dB Kernel version is 3.2.5 (Ubuntu). $ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.24. $ cat /proc/asound/cards 0 [HUDmx2 ]: USB-Audio - Audinst HUD-mx2 VIA Technologies Inc. Audinst HUD-mx2 at usb-0000:00:1d.1-2, full speed $ cat /proc/asound/modules 0 snd_usb_audio $ cat /proc/asound/devices 1: : sequencer 2: [ 0- 1]: digital audio playback 3: [ 0- 0]: digital audio playback 4: [ 0] : control 33: : timer $ lsusb -v [...] Bus 007 Device 006: ID 040d:340f VIA Technologies, Inc. Couldn't open device, some information will be missing Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x040d VIA Technologies, Inc. idProduct 0x340f bcdDevice 1.01 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 255 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xe0 Self Powered Remote Wakeup MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 59 bInCollection 1 baInterfaceNr( 0) 1 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 12 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bSourceID 11 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 1 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x00 bmaControls( 2) 0x00 iFeature 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 5 bSourceID 1 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 8193 IEC1937_AC-3 AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 3 (FORMAT_TYPE_III) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 1 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 tSamFreq[ 2] 88200 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0180 1x 384 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 1 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 4 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 tSamFreq[ 2] 88200 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0240 1x 576 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 1 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 61 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x8f EP 15 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 16 [...] Any help is appreciated. Regards, Tobias -- Aussen Stop-Bits, innen gepackt!

2 5

[alsa-devel] [PATCH 1/3] ASoC: wm_adsp: Print error when regmap reads/writes fail
by Dimitris Papastamos 01 Nov '13

01 Nov '13

Change-Id: I782f7684d9ef38adcbee6c6ed6809534dc7ff9b9 Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index e09ac94..cd007cc 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -339,8 +339,8 @@ static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, ret = regmap_raw_write(adsp->regmap, reg, scratch, ctl->len); if (ret) { - adsp_err(adsp, "Failed to write %zu bytes to %x\n", - ctl->len, reg); + adsp_err(adsp, "Failed to write %zu bytes to %x: %d\n", + ctl->len, reg, ret); kfree(scratch); return ret; } @@ -393,8 +393,8 @@ static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, ret = regmap_raw_read(adsp->regmap, reg, scratch, ctl->len); if (ret) { - adsp_err(adsp, "Failed to read %zu bytes from %x\n", - ctl->len, reg); + adsp_err(adsp, "Failed to read %zu bytes from %x: %d\n", + ctl->len, reg, ret); kfree(scratch); return ret; } @@ -1227,8 +1227,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) le32_to_cpu(blk->len)); if (ret != 0) { adsp_err(dsp, - "%s.%d: Failed to write to %x in %s\n", - file, blocks, reg, region_name); + "%s.%d: Failed to write to %x in %s: %d\n", + file, blocks, reg, region_name, ret); } kfree(buf); -- 1.8.4.2

2 3

[alsa-devel] [PATCH] ALSA: hda - not choose assigned converters for unused pins of Valleyview
by mengdong.lin＠intel.com 01 Nov '13

01 Nov '13

From: Mengdong Lin <mengdong.lin(a)intel.com> For Valleyview display codec, if an unused pin chooses an assgined converter selected by a used pin, playback on the unused pin can also give sound to the output device of the used pin. It's because data flows from the same convertor to the display port of the used pin. This issue is same as Haswell. So this patch avoids using assinged convertors for unused pins. The related function haswell_config_cvts() is renamed for code reuse. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e332db8..bcf764f 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -45,6 +45,7 @@ module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell(codec) ((codec)->vendor_id == 0x80862807) +#define is_valleyview(codec) ((codec)->vendor_id == 0x80862882) struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; @@ -1148,7 +1149,7 @@ static int hdmi_choose_cvt(struct hda_codec *codec, return 0; } -static void haswell_config_cvts(struct hda_codec *codec, +static void not_share_unassigned_cvt(struct hda_codec *codec, hda_nid_t pin_nid, int mux_idx) { struct hdmi_spec *spec = codec->spec; @@ -1226,8 +1227,8 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (is_haswell(codec)) - haswell_config_cvts(codec, per_pin->pin_nid, mux_idx); + if (is_haswell(codec) || is_valleyview(codec)) + not_share_unassigned_cvt(codec, per_pin->pin_nid, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); -- 1.8.1.2

3 2

[alsa-devel] [PATCH v4 0/4] Add dual-fifo mode support of i.MX ssi
by Nicolin Chen 01 Nov '13

01 Nov '13

Changelog v4: * PATCH-3: Drop useless register configuration. * * Nothing changes for the other three patches v3: * PATCH-1: Add comments to indicate the end of v1 and v2 array. * PATCH-3: Use better way to keep watermark as even number. * * Nothing changes for PATCH-2 and PATCH-4 v2: * Instead of adding rogue scripts to current SDMA driver based on firmware * V1, we define the new SDMA firmware as version 2 and bisect the PATCH-1 * to two patches: The first is to add version check code to the SDMA driver; * And the second is to add SSI dual FIFO DMATYPE. * * Nothing changes for the last two patches. v1: * SSI can reduce hardware overrun/underrun possibility when using dual * fifo mode. To support this mode, we need to first update sdma sciprt * list, and then enable dual fifo BIT in SSI driver, and last update DT * bindings of i.MX series. * * ! This series of patches has a direct dependency between them. When * ! applying them, we need to apply to one single branch. Otherwise, * ! it would break currect branches. Nicolin Chen (4): dma: imx-sdma: Add sdma firmware version 2 support dma: imx-sdma: Add new dma type for ssi dual fifo script ASoC: fsl_ssi: Add dual fifo mode support ARM: dts: imx: use dual-fifo sdma script for ssi .../devicetree/bindings/dma/fsl-imx-sdma.txt | 1 + arch/arm/boot/dts/imx51.dtsi | 4 ++-- arch/arm/boot/dts/imx53.dtsi | 4 ++-- arch/arm/boot/dts/imx6qdl.dtsi | 12 ++++++------ arch/arm/boot/dts/imx6sl.dtsi | 12 ++++++------ drivers/dma/imx-sdma.c | 19 ++++++++++++++++++- include/linux/platform_data/dma-imx-sdma.h | 5 +++++ include/linux/platform_data/dma-imx.h | 1 + sound/soc/fsl/fsl_ssi.c | 18 +++++++++++++++++- 9 files changed, 58 insertions(+), 18 deletions(-) -- 1.8.4

3 7

[alsa-devel] [PATCHv7][ 1/5] ASoC: eukrea-tlv320: Use dev_err instead of pr_err.
by Denis Carikli 01 Nov '13

01 Nov '13

It also contains a minor style cleanup. Cc: Rob Herring <rob.herring(a)calxeda.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Sascha Hauer <kernel(a)pengutronix.de> Cc: linux-arm-kernel(a)lists.infradead.org Cc: Eric Bénard <eric(a)eukrea.com> Signed-off-by: Denis Carikli <denis(a)eukrea.com> --- ChangeLog v6->v7: - eukrea-tlv320 cleanups are now moved into that patch. --- sound/soc/fsl/eukrea-tlv320.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 9a4a0ca..5983740 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -42,7 +42,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { - pr_err("%s: failed set cpu dai format\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the cpu dai format.\n"); return ret; } @@ -50,14 +51,16 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { - pr_err("%s: failed set codec dai format\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the codec format.\n"); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_OUT); if (ret) { - pr_err("%s: failed setting codec sysclk\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the codec sysclk.\n"); return ret; } snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); @@ -65,7 +68,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, SND_SOC_CLOCK_IN); if (ret) { - pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); + dev_err(cpu_dai->dev, + "Can't set the IMX_SSP_SYS_CLK CPU system clock.\n"); return ret; } @@ -155,7 +159,8 @@ static struct platform_driver eukrea_tlv320_driver = { .owner = THIS_MODULE, }, .probe = eukrea_tlv320_probe, - .remove = eukrea_tlv320_remove,}; + .remove = eukrea_tlv320_remove, +}; module_platform_driver(eukrea_tlv320_driver); -- 1.7.9.5

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[alsa-devel] [RFC PATCH] ASoC: sgtl5000: interconnect ADC with Mic Bias
by Hector Palacios 31 Oct '13

31 Oct '13

The interconnection of the Mic Bias to the ADC allows to control the Mic Bias Output Impedance. Without it, recording from a PC microphone results in silence. Signed-off-by: Hector Palacios <hector.palacios(a)digi.com> --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index ea47938..60d5012 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -231,6 +231,7 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ {"ADC", NULL, "VAG_POWER"}, + {"ADC", NULL, "Mic Bias"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */

1 0

[alsa-devel] [PATCH] ALSA: fix oops in snd_pcm_info() caused by ASoC DPCM
by Russell King - ARM Linux 31 Oct '13

31 Oct '13

Liam, Mark, As discussed privately, here's the fix for the ASoC oops I've been seeing with DPCM in ALSA. This avoids unnecessarily exposing the ASoC internal PCM devices to userspace, which is complimentary to your patch fixing the deref of the PCM ops. 8<======== From: Russell King <rmk+kernel(a)arm.linux.org.uk> Subject: [PATCH] ALSA: fix oops in snd_pcm_info() caused by ASoC DPCM Unable to handle kernel NULL pointer dereference at virtual address 00000008 pgd = d5300000 [00000008] *pgd=0d265831, *pte=00000000, *ppte=00000000 Internal error: Oops: 17 [#1] PREEMPT ARM CPU: 0 PID: 2295 Comm: vlc Not tainted 3.11.0+ #755 task: dee74800 ti: e213c000 task.ti: e213c000 PC is at snd_pcm_info+0xc8/0xd8 LR is at 0x30232065 pc : [<c031b52c>] lr : [<30232065>] psr: a0070013 sp : e213dea8 ip : d81cb0d0 fp : c05f7678 r10: c05f7770 r9 : fffffdfd r8 : 00000000 r7 : d8a968a8 r6 : d8a96800 r5 : d8a96200 r4 : d81cb000 r3 : 00000000 r2 : d81cb000 r1 : 00000001 r0 : d8a96200 Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c5387d Table: 15300019 DAC: 00000015 Process vlc (pid: 2295, stack limit = 0xe213c248) [<c031b52c>] (snd_pcm_info) from [<c031b570>] (snd_pcm_info_user+0x34/0x9c) [<c031b570>] (snd_pcm_info_user) from [<c03164a4>] (snd_pcm_control_ioctl+0x274/0x280) [<c03164a4>] (snd_pcm_control_ioctl) from [<c0311458>] (snd_ctl_ioctl+0xc0/0x55c) [<c0311458>] (snd_ctl_ioctl) from [<c00eca84>] (do_vfs_ioctl+0x80/0x31c) [<c00eca84>] (do_vfs_ioctl) from [<c00ecd5c>] (SyS_ioctl+0x3c/0x60) [<c00ecd5c>] (SyS_ioctl) from [<c000e500>] (ret_fast_syscall+0x0/0x48) Code: e1a00005 e59530dc e3a01001 e1a02004 (e5933008) ---[ end trace cb3d9bdb8dfefb3c ]--- This is provoked when the ASoC front end is open along with its backend, (which causes the backend to have a runtime assigned to it) and then the SNDRV_CTL_IOCTL_PCM_INFO is requested for the (visible) backend device. Resolve this by ensuring that ASoC internal backend devices are not visible to userspace, just as the commentry for snd_pcm_new_internal() says it should be. Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> --- sound/core/pcm.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 17f45e8aa89c..e1e9e0c999fe 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -49,6 +49,8 @@ static struct snd_pcm *snd_pcm_get(struct snd_card *card, int device) struct snd_pcm *pcm; list_for_each_entry(pcm, &snd_pcm_devices, list) { + if (pcm->internal) + continue; if (pcm->card == card && pcm->device == device) return pcm; } @@ -60,6 +62,8 @@ static int snd_pcm_next(struct snd_card *card, int device) struct snd_pcm *pcm; list_for_each_entry(pcm, &snd_pcm_devices, list) { + if (pcm->internal) + continue; if (pcm->card == card && pcm->device > device) return pcm->device; else if (pcm->card->number > card->number) -- 1.7.4.4

3 2

[alsa-devel] Digidesign Eleven Rack support
by Aidan Schurr 31 Oct '13

31 Oct '13

Hi guys, am wondering if support for the Digidesign Eleven Rack is on the cards for the future at all? Alsa mixer currently recognises it but lists it as having no available controls, while aplay -l and arecord -l miss it altogether. I found one thread from around 18 months ago that mentions it uses a non standard implementation of USB audio; does this pretty much preclude it from getting up and finding running? Happy to be a beta tester of that could help in any way.

2 1

[alsa-devel] [PATCH 0/3] dma: edma: Add cyclic DMA support
by Joel Fernandes 31 Oct '13

31 Oct '13

The following series adds Cyclic DMA support to TI EDMA DMA Engine driver. First we split out the calculations for the Slave DMA case into a separate function so that we may reuse it for the calculations of Cyclic DMA parameters. Next patch then adds the actual support for Cyclic DMA, enables interrupts correctly and uses's the callbacks in virt-dma during interrupts thus signalling back to the ALSA layer that a period was transmitted. Some background on motivation for this series: Currently, only user of Cyclic DMA in EDMA is davinci-pcm driver. As of today, this driver directly calls into the EDMA private API (arch/arm/common/edma.c) without going through the DMAEngine. davinci-pcm in future will be modified to use DMA Engine framework for Cyclic DMA instead of directly using the Private API. However that's a much larger effort, involving dealing with ping-pong from SRAM on user's of the Davinci McASP, etc. As a first step, we add Cyclic DMA support to the EDMA driver so that this may be used when the actual conversion of davinci-pcm happens. Tested series along with couple of hacks to davinci-pcm to work with DMA Engine: git@github.com:joelagnel/linux-kernel.git (branch dma/cyclic) Joel Fernandes (3): dma: edma: Split out PaRAM set calculations into its own function dma: edma: Add support for Cyclic DMA dma: edma: Increase maximum SG limit to 20 drivers/dma/edma.c | 350 +++++++++++++++++++++++++++++++++++++++++------------ 1 file changed, 273 insertions(+), 77 deletions(-) -- 1.8.1.2

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[alsa-devel] [PATCH v3 0/4] Add dual-fifo mode support of i.MX ssi
by Nicolin Chen 31 Oct '13

31 Oct '13

Changelog v3: * PATCH-1: Add comments to indicate the end of v1 and v2 array. * PATCH-3: Use better way to keep watermark as even number. * * Nothing changes for PATCH-2 and PATCH-4 v2: * Instead of adding rogue scripts to current SDMA driver based on firmware * V1, we define the new SDMA firmware as version 2 and bisect the PATCH-1 * to two patches: The first is to add version check code to the SDMA driver; * And the second is to add SSI dual FIFO DMATYPE. * * Nothing changes for the last two patches. v1: * SSI can reduce hardware overrun/underrun possibility when using dual * fifo mode. To support this mode, we need to first update sdma sciprt * list, and then enable dual fifo BIT in SSI driver, and last update DT * bindings of i.MX series. * * ! This series of patches has a direct dependency between them. When * ! applying them, we need to apply to one single branch. Otherwise, * ! it would break currect branches. Nicolin Chen (4): dma: imx-sdma: Add sdma firmware version 2 support dma: imx-sdma: Add new dma type for ssi dual fifo script ASoC: fsl_ssi: Add dual fifo mode support ARM: dts: imx: use dual-fifo sdma script for ssi .../devicetree/bindings/dma/fsl-imx-sdma.txt | 1 + arch/arm/boot/dts/imx51.dtsi | 4 ++-- arch/arm/boot/dts/imx53.dtsi | 4 ++-- arch/arm/boot/dts/imx6qdl.dtsi | 12 ++++++------ arch/arm/boot/dts/imx6sl.dtsi | 12 ++++++------ drivers/dma/imx-sdma.c | 19 ++++++++++++++++++- include/linux/platform_data/dma-imx-sdma.h | 5 +++++ include/linux/platform_data/dma-imx.h | 1 + sound/soc/fsl/fsl_ssi.c | 22 +++++++++++++++++++++- 9 files changed, 62 insertions(+), 18 deletions(-) -- 1.8.4

3 6

[alsa-devel] [PATCH] asoc-tools: Render png image instead of jpeg in vizdapm.sh
by Jarkko Nikula 31 Oct '13

31 Oct '13

Lossless png image suits better to technical drawings than jpeg which may even have artifacts around sharp objects. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- For git://opensource.wolfsonmicro.com/asoc-tools.git --- bin/vizdapm.sh | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/bin/vizdapm.sh b/bin/vizdapm.sh index 11d1361..8d61dd2 100755 --- a/bin/vizdapm.sh +++ b/bin/vizdapm.sh @@ -16,7 +16,7 @@ usage() { cat << EOF -usage: $0 <dapm-debugfs-directory> <graph-dot-file> <graph-jpg-file> +usage: $0 <dapm-debugfs-directory> <graph-dot-file> <graph-png-file> EOF exit 1 } @@ -29,7 +29,7 @@ active= curpath=`pwd` dapm_debugfs="$1" graph_file=${curpath}"/$2" -graph_file_jpg=${curpath}"/$3" +graph_file_png=${curpath}"/$3" trap "{ [ -f "$graph_file" ] && rm -f "$graph_file"; exit; }" SIGINT SIGTERM @@ -62,4 +62,4 @@ done popd >/dev/null echo "}" >> "$graph_file" -dot -Kfdp -Tjpg "$graph_file" -o "$graph_file_jpg" +dot -Kfdp -Tpng "$graph_file" -o "$graph_file_png" -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH] ALSA: hda - Add extra chmap for 2.1 outputs on ASUS laptops
by Takashi Iwai 31 Oct '13

31 Oct '13

ASUS N56VZ and N76VZ laptops have a bass speaker but its output comes only from the right channel. This patch adds the extra chmap specific to these models. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=846531 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 30 ++++++++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 13ad49ca9357..2d7c86d81196 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4426,6 +4426,25 @@ static void alc272_fixup_mario(struct hda_codec *codec, "hda_codec: failed to override amp caps for NID 0x2\n"); } +static const struct snd_pcm_chmap_elem asus_pcm_2_1_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_NA, SNDRV_CHMAP_LFE } }, /* LFE only on right */ + { } +}; + +/* override the 2.1 chmap */ +static void alc662_fixup_bass_chmap(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_BUILD) { + struct alc_spec *spec = codec->spec; + spec->gen.pcm_rec[0].stream[0].chmap = asus_pcm_2_1_chmaps; + } +} + enum { ALC662_FIXUP_ASPIRE, ALC662_FIXUP_IDEAPAD, @@ -4446,6 +4465,7 @@ enum { ALC662_FIXUP_INV_DMIC, ALC668_FIXUP_DELL_MIC_NO_PRESENCE, ALC668_FIXUP_HEADSET_MODE, + ALC662_FIXUP_BASS_CHMAP, }; static const struct hda_fixup alc662_fixups[] = { @@ -4620,6 +4640,12 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_alc668, }, + [ALC662_FIXUP_BASS_CHMAP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc662_fixup_bass_chmap, + .chained = true, + .chain_id = ALC662_FIXUP_ASUS_MODE4 + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -4633,8 +4659,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), - SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_ASUS_MODE4), - SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_ASUS_MODE4), + SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), + SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), -- 1.8.4.1

2 4

[alsa-devel] Realtek HD-A codec ALC262: How to unmute mixer node 0x24 Amp-In for Mic Pin 0x18
by Lin, Mengdong 31 Oct '13

31 Oct '13

Hi, We found when recording on ALC262, the audio mixer node 0x24 mutes the input from Mic Pin node 0x18, so we get silent data. Would someone please give me some advice to solve this? Node 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 9 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x0b I cannot find user-space amixer controls to unmute node 0x18. The recording data flows from Mic Pin 0x18 -> Mixer 0x24 -> audio input 0x07. The Amp-In value on pin 0x18 (Rear Mic Boost) and input 0x07 (Capture Volume) are okay. Thanks Mengdong

3 9

[alsa-devel] [PATCH v2 0/4] Add dual-fifo mode support of i.MX ssi
by Nicolin Chen 31 Oct '13

31 Oct '13

Changelog v2: * Instead of adding rogue scripts to current SDMA driver based on firmware * V1, we define the new SDMA firmware as version 2 and bisect the PATCH-1 * to two patches: The first is to add version check code to the SDMA driver; * And the second is to add SSI dual FIFO DMATYPE. * * Nothing changes for the last two patches. v1: * SSI can reduce hardware overrun/underrun possibility when using dual * fifo mode. To support this mode, we need to first update sdma sciprt * list, and then enable dual fifo BIT in SSI driver, and last update DT * bindings of i.MX series. * * ! This series of patches has a direct dependency between them. When * ! applying them, we need to apply in one single branch. Otherwise, * ! it would break currect branches. Nicolin Chen (4): dma: imx-sdma: Add sdma firmware version 2 support dma: imx-sdma: Add new dma type for ssi dual fifo script ASoC: fsl_ssi: Add dual fifo mode support ARM: dts: imx: use dual-fifo sdma script for ssi .../devicetree/bindings/dma/fsl-imx-sdma.txt | 1 + arch/arm/boot/dts/imx51.dtsi | 4 ++-- arch/arm/boot/dts/imx53.dtsi | 4 ++-- arch/arm/boot/dts/imx6qdl.dtsi | 12 +++++------ arch/arm/boot/dts/imx6sl.dtsi | 12 +++++------ drivers/dma/imx-sdma.c | 19 ++++++++++++++++- include/linux/platform_data/dma-imx-sdma.h | 3 +++ include/linux/platform_data/dma-imx.h | 1 + sound/soc/fsl/fsl_ssi.c | 24 +++++++++++++++++++++- 9 files changed, 62 insertions(+), 18 deletions(-) -- 1.8.4

3 7

[alsa-devel] [PATCH 0/9] ASoC: Yet another small fixes
by Takashi Iwai 31 Oct '13

31 Oct '13

Hi, this is yet another series of small fixes for ASoC codec bugs spotted by coverity. All minor issues, obviously. Takashi

2 25

[alsa-devel] [PATCH] ALSA: compress: fix drain calls blocking other compress functions
by Vinod Koul 31 Oct '13

31 Oct '13

The drain and drain_notify callback were blocked by low level driver untill the draining was complete. Due to this being invoked with big fat mutex held, others ops like reading timestamp, calling pause, drop were blocked. So to fix this we add a new snd_compr_drain_notify() API. This would be required to be invoked by low level driver when drain or partial drain has been completed by the DSP. Thus we make the drain and partial_drain callback as non blocking and driver returns immediately after notifying DSP. The waiting is done while relasing the lock so that other ops can go ahead. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> CC: stable(a)vger.kernel.org --- include/sound/compress_driver.h | 12 ++++++++++ sound/core/compress_offload.c | 43 +++++++++++++++++++++++++++++++++++--- 2 files changed, 51 insertions(+), 4 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 9031a26..eff5f84 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -48,6 +48,8 @@ struct snd_compr_ops; * the ring buffer * @total_bytes_transferred: cumulative bytes transferred by offload DSP * @sleep: poll sleep + * @wait: drain wait queuq + * @draining: draining wait condition */ struct snd_compr_runtime { snd_pcm_state_t state; @@ -59,6 +61,8 @@ struct snd_compr_runtime { u64 total_bytes_available; u64 total_bytes_transferred; wait_queue_head_t sleep; + wait_queue_head_t wait; + unsigned int draining; void *private_data; }; @@ -171,4 +175,12 @@ static inline void snd_compr_fragment_elapsed(struct snd_compr_stream *stream) wake_up(&stream->runtime->sleep); } +static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) +{ + snd_BUG_ON(!stream); + + stream->runtime->draining = 1; + wake_up(&stream->runtime->wait); +} + #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index bea523a..3582f9f 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -123,6 +123,7 @@ static int snd_compr_open(struct inode *inode, struct file *f) } runtime->state = SNDRV_PCM_STATE_OPEN; init_waitqueue_head(&runtime->sleep); + init_waitqueue_head(&runtime->wait); data->stream.runtime = runtime; f->private_data = (void *)data; mutex_lock(&compr->lock); @@ -682,12 +683,36 @@ static int snd_compr_stop(struct snd_compr_stream *stream) if (!retval) { stream->runtime->state = SNDRV_PCM_STATE_SETUP; wake_up(&stream->runtime->sleep); + stream->runtime->draining = 1; + wake_up(&stream->runtime->wait); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; } return retval; } +static int snd_compress_wait_for_drain(struct snd_compr_stream *stream) +{ + int retval = 0; + + /* + * we are called with lock held. So drop the lock while we wait for + * drain complete notfication from the driver + */ + stream->runtime->state = SNDRV_PCM_STATE_DRAINING; + mutex_unlock(&stream->device->lock); + + retval = wait_event_interruptible(stream->runtime->wait, stream->runtime->draining); + if (retval) + pr_err("Wait for drain failed %d\n", retval); + + wake_up(&stream->runtime->sleep); + mutex_lock(&stream->device->lock); + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + + return retval; +} + static int snd_compr_drain(struct snd_compr_stream *stream) { int retval; @@ -695,12 +720,16 @@ static int snd_compr_drain(struct snd_compr_stream *stream) if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || stream->runtime->state == SNDRV_PCM_STATE_SETUP) return -EPERM; + + stream->runtime->draining = 0; retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN); - if (!retval) { - stream->runtime->state = SNDRV_PCM_STATE_DRAINING; + if (retval) { + pr_err("SND_COMPR_TRIGGER_DRAIN failed %d\n", retval); wake_up(&stream->runtime->sleep); + return retval; } - return retval; + + return snd_compress_wait_for_drain(stream); } static int snd_compr_next_track(struct snd_compr_stream *stream) @@ -735,10 +764,16 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream) if (stream->next_track == false) return -EPERM; + stream->runtime->draining = 0; retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + if (retval) { + pr_err("Partial drain returned failure\n"); + wake_up(&stream->runtime->sleep); + return retval; + } stream->next_track = false; - return retval; + return snd_compress_wait_for_drain(stream); } static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) -- 1.7.0.4

2 18

[alsa-devel] [PATCH] ASoC: wm_adsp: Release firmware on memory allocation failure
by Dimitris Papastamos 30 Oct '13

30 Oct '13

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b3fbe4a..a2144b0 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -630,7 +630,8 @@ static int wm_adsp_load(struct wm_adsp *dsp) GFP_KERNEL | GFP_DMA); if (!buf) { adsp_err(dsp, "Out of memory\n"); - return -ENOMEM; + ret = -ENOMEM; + goto out_fw; } ret = regmap_raw_write(regmap, reg, buf, -- 1.8.4.2

2 1