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

[alsa-devel] [PATCH] ASoC: fsl: imx-ssi: omit ssi counter to avoid harm in unbalanced situation
by Oskar Schirmer 02 Dec '13

02 Dec '13

Unbalanced calls to imx_ssi_trigger() may result in endless SSI activity and thus provoke eternal sound. While on the first glance, the switch statement looks pretty symmetric, the SUSPEND/RESUME pair is not: the suspend case comes along snd_pcm_suspend_all(), which for fsl/imx-pcm-fiq is called only at snd_soc_suspend(), but the resume case originates straight from the SNDRV_PCM_IOCTL_RESUME. This way userland may provoke an unbalanced resume, which might cause the ssi->enabled counter to increase and never return to zero again, so eventually SSI_SCR_SSIEN is never disabled. Simply removing the ssi->enabled will solve the problem, as long as one never goes play and capture game simultaneously, but beware trying both at once, the early TRIGGER_STOP will cut off the other activity prematurely. So now playing and capturing is scrutinized separately, instead of by counting. This is essentially the same stuff as in sound/soc/fsl/imx-pcm-fiq.c which I send a patch for three days ago. Astonishing enough this highly fragile scheme is used twice in parallel to serve the very same control function, synchronously: Once out of sync you are lost until reboot. Note, that these fixes wont prevent state machine distortion on alsa level to cut sound or the like. It just makes sure we have a chance to synchronise again later on. Signed-off-by: Oskar Schirmer <oskar(a)scara.com> --- sound/soc/fsl/imx-ssi.c | 19 ++++++++++++------- sound/soc/fsl/imx-ssi.h | 3 ++- 2 files changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f5f248c..c68499b 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -298,27 +298,32 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { scr |= SSI_SCR_TE; - else + ssi->playing = 1; + } else { scr |= SSI_SCR_RE; + ssi->capturing = 1; + } sier |= sier_bits; - if (++ssi->enabled == 1) - scr |= SSI_SCR_SSIEN; + scr |= SSI_SCR_SSIEN; break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { scr &= ~SSI_SCR_TE; - else + ssi->playing = 0; + } else { scr &= ~SSI_SCR_RE; + ssi->capturing = 0; + } sier &= ~sier_bits; - if (--ssi->enabled == 0) + if (!ssi->playing && !ssi->capturing) scr &= ~SSI_SCR_SSIEN; break; diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index 560c40f..a3c90d5 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -213,7 +213,8 @@ struct imx_ssi { int fiq_init; int dma_init; - int enabled; + int playing; + int capturing; }; #endif /* _IMX_SSI_H */ -- 1.7.9.5

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[alsa-devel] [PATCH 1/1] ASoC: soc_compress: Add set_metadata
by Richard Fitzgerald 02 Dec '13

02 Dec '13

Pass the set_metadata() calls through to the codec driver. Signed-off-by: Zhao Weijia <weijia.zhao(a)capelabs.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 18 +++++++++++++++++- 1 files changed, 17 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 53c9ecd..186802b 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -318,6 +318,21 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, mutex_unlock(&rtd->pcm_mutex); return 0; } +static int soc_compr_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); + + if (platform->driver->compr_ops && platform->driver->compr_ops->set_metadata) + platform->driver->compr_ops->set_metadata(cstream, metadata); + + mutex_unlock(&rtd->pcm_mutex); + return 0; +} + static int soc_compr_copy(struct snd_compr_stream *cstream, char __user *buf, size_t count) @@ -372,7 +387,8 @@ static struct snd_compr_ops soc_compr_ops = { .pointer = soc_compr_pointer, .ack = soc_compr_ack, .get_caps = soc_compr_get_caps, - .get_codec_caps = soc_compr_get_codec_caps + .get_codec_caps = soc_compr_get_codec_caps, + .set_metadata = soc_compr_set_metadata, }; /* create a new compress */ -- 1.7.2.5

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[alsa-devel] Using devm_snd_soc_register_card() causes problem with imx-sgtl5000
by Shawn Guo 02 Dec '13

02 Dec '13

I'm trying to test Lars' devm_snd_dmaengine_pcm_register patches with imx-sgtl5000 machine driver, but running into the rmmod problem below on v3.13-rc1. Reverting commit 01984a4 (ASoC: imx-sgtl5000: Use devm_snd_soc_register_card) seems to fix the problem for me. Any hint on what's going wrong here? Shawn root@arm:~# lsmod Module Size Used by snd_soc_imx_sgtl5000 3506 0 snd_soc_sgtl5000 13677 2 snd_soc_imx_audmux 5324 1 snd_soc_imx_sgtl5000 snd_soc_fsl_ssi 8139 2 imx_pcm_dma 1380 1 snd_soc_fsl_ssi root@arm:~# rmmod snd_soc_imx_sgtl5000 Unable to handle kernel paging request at virtual address e594025c pgd = be134000 [e594025c] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: snd_soc_imx_sgtl5000(-) snd_soc_sgtl5000 snd_soc_imx_audmux snd_soc_fsl_ssi imx_pcm_dma CPU: 0 PID: 1793 Comm: rmmod Not tainted 3.13.0-rc1 #1570 task: bee28900 ti: bfbec000 task.ti: bfbec000 PC is at debugfs_remove_recursive+0x28/0x154 LR is at snd_soc_unregister_card+0xa0/0xcc pc : [<80252b38>] lr : [<80496ac4>] psr: a0000013 sp : bfbede00 ip : bfbede28 fp : bfbede24 r10: 803281d4 r9 : bfbec000 r8 : 803271ac r7 : bef54440 r6 : 00000004 r5 : bf9a4010 r4 : bf9a4010 r3 : e5940224 r2 : 00000000 r1 : bef54450 r0 : 803271ac Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c53c7d Table: 4e13404a DAC: 00000015 Process rmmod (pid: 1793, stack limit = 0xbfbec240) Stack: (0xbfbede00 to 0xbfbee000) de00: 00000000 bf9a4010 bf9a4010 00000004 bef54440 bec89000 bfbede44 bfbede28 de20: 80496ac4 80252b1c 804a4b60 bfbede60 bf9a4010 00000004 bfbede54 bfbede48 de40: 804a4b74 80496a30 bfbede94 bfbede58 80328728 804a4b6c bfbede94 a0000013 de60: bf1b5800 bef54440 00000002 bf9a4010 7f0169f8 bf9a4044 00000081 8000e9c4 de80: bfbec000 00000000 bfbedeac bfbede98 80328cb0 80328618 7f016000 bf9a4010 dea0: bfbedec4 bfbedeb0 8032561c 80328c84 bf9a4010 7f0169f8 bfbedee4 bfbedec8 dec0: 80325e84 803255a8 bee28900 7f0169f8 00000000 78208d30 bfbedefc bfbedee8 dee0: 80325410 80325dd4 beca8100 7f0169f8 bfbedf14 bfbedf00 803264f8 803253c8 df00: 7f01635c 7f016a3c bfbedf24 bfbedf18 80327098 803264d4 bfbedf34 bfbedf28 df20: 7f016370 80327090 bfbedfa4 bfbedf38 80085ef0 7f016368 bfbedf54 5f646e73 df40: 5f636f73 5f786d69 6c746773 30303035 00000000 78208008 bfbedf84 bfbedf68 df60: 800613b0 80061194 fffffffe 78208d00 7efc2f07 00000081 7f016a3c 00000800 df80: bfbedf84 00000000 00000000 fffffffe 78208d00 7efc2f07 00000000 bfbedfa8 dfa0: 8000e800 80085dcc fffffffe 78208d00 78208d30 00000800 a8c82400 a8c82400 dfc0: fffffffe 78208d00 7efc2f07 00000081 00000002 00000000 78208008 00000800 dfe0: 7efc2e1c 7efc2ba8 76f5ca47 76edec7c 80000010 78208d30 00000000 00000000 Backtrace: [<80252b10>] (debugfs_remove_recursive+0x0/0x154) from [<80496ac4>] (snd_soc_unregister_card+0xa0/0xcc) r8:bec89000 r7:bef54440 r6:00000004 r5:bf9a4010 r4:bf9a4010 r3:00000000 [<80496a24>] (snd_soc_unregister_card+0x0/0xcc) from [<804a4b74>] (devm_card_release+0x14/0x18) r6:00000004 r5:bf9a4010 r4:bfbede60 r3:804a4b60 [<804a4b60>] (devm_card_release+0x0/0x18) from [<80328728>] (release_nodes+0x11c/0x1dc) [<8032860c>] (release_nodes+0x0/0x1dc) from [<80328cb0>] (devres_release_all+0x38/0x54) [<80328c78>] (devres_release_all+0x0/0x54) from [<8032561c>] (__device_release_driver+0x80/0xd4) r4:bf9a4010 r3:7f016000 [<8032559c>] (__device_release_driver+0x0/0xd4) from [<80325e84>] (driver_detach+0xbc/0xc0) r5:7f0169f8 r4:bf9a4010 [<80325dc8>] (driver_detach+0x0/0xc0) from [<80325410>] (bus_remove_driver+0x54/0x98) r6:78208d30 r5:00000000 r4:7f0169f8 r3:bee28900 [<803253bc>] (bus_remove_driver+0x0/0x98) from [<803264f8>] (driver_unregister+0x30/0x50) r4:7f0169f8 r3:beca8100 [<803264c8>] (driver_unregister+0x0/0x50) from [<80327098>] (platform_driver_unregister+0x14/0x18) r4:7f016a3c r3:7f01635c [<80327084>] (platform_driver_unregister+0x0/0x18) from [<7f016370>] (imx_sgtl5000_driver_exit+0x14/0x1c [snd_soc_imx_sgtl5000]) [<7f01635c>] (imx_sgtl5000_driver_exit+0x0/0x1c [snd_soc_imx_sgtl5000]) from [<80085ef0>] (SyS_delete_module+0x130/0x18c) [<80085dc0>] (SyS_delete_module+0x0/0x18c) from [<8000e800>] (ret_fast_syscall+0x0/0x48) r6:7efc2f07 r5:78208d00 r4:fffffffe Code: 889da9f8 e5983020 e3530000 089da9f8 (e5933038) ---[ end trace 825e7e125251a225 ]---

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[alsa-devel] [PATCH 1/6] ASoC: Add resource managed snd_dmaengine_pcm_register()
by Lars-Peter Clausen 02 Dec '13

02 Dec '13

For many drivers using the generic dmaengine PCM driver one of the few (or the only) things left to do in the drivers remove function is to unregister the PCM device. This patch adds a resource managed version of snd_dmaengine_pcm_register() which makes it possible to simplify the remove function as well as the error path in the probe function for those drivers. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Moved snd_dmaengine_pcm_register() to soc-devres.c --- include/sound/dmaengine_pcm.h | 4 ++++ sound/soc/soc-devres.c | 41 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 45 insertions(+) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 1501731..4ef986c 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -140,6 +140,10 @@ int snd_dmaengine_pcm_register(struct device *dev, unsigned int flags); void snd_dmaengine_pcm_unregister(struct device *dev); +int devm_snd_dmaengine_pcm_register(struct device *dev, + const struct snd_dmaengine_pcm_config *config, + unsigned int flags); + int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config); diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index b1d7322..9998619 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> static void devm_component_release(struct device *dev, void *res) { @@ -84,3 +85,43 @@ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) return ret; } EXPORT_SYMBOL_GPL(devm_snd_soc_register_card); + +#ifdef CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM + +static void devm_dmaengine_pcm_release(struct device *dev, void *res) +{ + snd_dmaengine_pcm_unregister(*(struct device **)res); +} + +/** + * devm_snd_dmaengine_pcm_register - resource managed dmaengine PCM registration + * @dev: The parent device for the PCM device + * @config: Platform specific PCM configuration + * @flags: Platform specific quirks + * + * Register a dmaengine based PCM device with automatic unregistration when the + * device is unregistered. + */ +int devm_snd_dmaengine_pcm_register(struct device *dev, + const struct snd_dmaengine_pcm_config *config, unsigned int flags) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_dmaengine_pcm_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_dmaengine_pcm_register(dev, config, flags); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_dmaengine_pcm_register); + +#endif -- 1.8.0

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[alsa-devel] [PATCH] ASoC: generic-dmaengine-pcm: Set BATCH flag when residue reporting is not supported
by Lars-Peter Clausen 02 Dec '13

02 Dec '13

For dmaengine drivers which do not support transfer residue reporting we update the PCM pointer with period granularity. Set the SNDRV_PCM_INFO_BATCH flag in this case to let userspace know about this. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-generic-dmaengine-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index cbc9c96..87e8635 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -137,6 +137,9 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea hw.buffer_bytes_max = SIZE_MAX; hw.fifo_size = dma_data->fifo_size; + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) + hw.info |= SNDRV_PCM_INFO_BATCH; + ret = dma_get_slave_caps(chan, &dma_caps); if (ret == 0) { if (dma_caps.cmd_pause) -- 1.8.0

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[alsa-devel] [PATCH] ASoC: Set SNDRV_PCM_INFO_JOINT_DUPLEX for PCMs with symmetry constraints
by Lars-Peter Clausen 02 Dec '13

02 Dec '13

If there are symmetry constraints between the playback and the capture channel set the SNDRV_PCM_INFO_JOINT_DUPLEX flag to let userspace know about this. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-pcm.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 22af038..eb340a8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -180,6 +180,21 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, return 0; } +static bool soc_pcm_has_symmetry(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_driver *cpu_driver = rtd->cpu_dai->driver; + struct snd_soc_dai_driver *codec_driver = rtd->codec_dai->driver; + struct snd_soc_dai_link *link = rtd->dai_link; + + return cpu_driver->symmetric_rates || codec_driver->symmetric_rates || + link->symmetric_rates || cpu_driver->symmetric_channels || + codec_driver->symmetric_channels || link->symmetric_channels || + cpu_driver->symmetric_samplebits || + codec_driver->symmetric_samplebits || + link->symmetric_samplebits; +} + /* * List of sample sizes that might go over the bus for parameter * application. There ought to be a wildcard sample size for things @@ -316,6 +331,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) &cpu_dai_drv->capture); } + if (soc_pcm_has_symmetry(substream)) + runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; + ret = -EINVAL; if (!runtime->hw.rates) { printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n", -- 1.8.0

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[alsa-devel] [PATCH] ASoC: imx-spdif: Remove error message upon devm_kzalloc() failure
by Fabio Estevam 02 Dec '13

02 Dec '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> No need to have a specific OOM message, since there is generic MM out of memory message in place. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-spdif.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 8499d52..980dd1f 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -35,7 +35,6 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); if (!data) { - dev_err(&pdev->dev, "failed to allocate memory\n"); ret = -ENOMEM; goto end; } -- 1.8.1.2

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[alsa-devel] MAXIM 98089 codec support in Linux
by VS 02 Dec '13

02 Dec '13

Hi, We are trying to integrate MAXIM 98089 (Audio codec) into our Linux product build. This has 2 DAI's available with it - DAI1 with both Playback and Capture facility and DAI2 with Playback capability alone. When we use this codec in our build, we find that the dai1_hw_params and dai1_set_fmt functions never gets called but it is always dai2_hw_params and dai2_set_fmt function that gets called because of which apparently, we are able to hear the voice that is incoming but not able to capture the voice and send it out externally. Here are our definitions: static struct snd_soc_dai_driver max98089_dai[] = { { .name = "max98089.0", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = MAX98089_RATES, .formats = MAX98089_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = MAX98089_RATES, .formats = MAX98089_FORMATS, }, .ops = &max98089_dai1_ops, }, { .name = "max98089.1", .playback = { .stream_name = "Aux Playback", .channels_min = 1, .channels_max = 2, .rates = MAX98089_RATES, .formats = MAX98089_FORMATS, }, .ops = &max98089_dai2_ops, .symmetric_rates = 1, } }; The codec specific registers only when we put in dai2_ops functions, we are able to see the effects but there is no effect if we put it in dai1_ops functions. Is there any specific way of referring to DAI when we have multiple DAI support in the codec that we are using? Thanks in advance for your inputs VS

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[alsa-devel] [PATCH] ALSA: hda - Fix headset mic input after muted internal mic (Dell/Realtek)
by David Henningsson 02 Dec '13

02 Dec '13

By trial and error, I found this patch could work around an issue where the headset mic would stop working if you switch between the internal mic and the headset mic, and the internal mic was muted. It still takes a second or two before the headset mic actually starts working, but still better than nothing. I'd like to add stable to this patch, but I'd like Kailang's ack first because I don't know what the verbs really do. BugLink: https://bugs.launchpad.net/bugs/1256840 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 537991c..85d1c8b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3338,6 +3338,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) alc_write_coef_idx(codec, 0x18, 0x7388); break; case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); alc_write_coef_idx(codec, 0x15, 0x0d60); alc_write_coef_idx(codec, 0xc3, 0x0000); break; @@ -3366,6 +3367,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) alc_write_coef_idx(codec, 0x18, 0x7388); break; case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); alc_write_coef_idx(codec, 0x15, 0x0d50); alc_write_coef_idx(codec, 0xc3, 0x0000); break; -- 1.7.9.5

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[alsa-devel] Remove fixup list for HP
by Kailang 02 Dec '13

02 Dec '13

Hi Takashi, Chrome OS was loaded driver by models. The patch was remove the alc269 fixup entry for HP chrome book. Thanks. Kialang

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