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

[alsa-devel] [PATCH] ASoC: wm8997: Correct typo in ISRC mux routes
by Charles Keepax 12 Nov '13

12 Nov '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8997.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 6ec3de3..2e886ca 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -887,7 +887,7 @@ static const struct snd_soc_dapm_route wm8997_dapm_routes[] = { ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"), ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), - ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC2INT2"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"), ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), -- 1.7.2.5

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Re: [alsa-devel] ASoC: add Renesas R-Car module feature
by Dan Carpenter 12 Nov '13

12 Nov '13

Hello Kuninori Morimoto, This is a semi-automatic email about new static checker warnings. The patch cdaa3cdfb4a7: "ASoC: add Renesas R-Car module feature" from Jul 21, 2013, leads to the following Smatch complaint: sound/soc/sh/rcar/core.c:294 rsnd_dai_connect() warn: variable dereferenced before check 'mod' (see line 291) sound/soc/sh/rcar/core.c 290 { 291 struct rsnd_priv *priv = rsnd_mod_to_priv(mod); ^^^^^^^^^^^^^^^^^^^^^ Dereference. 292 struct device *dev = rsnd_priv_to_dev(priv); 293 294 if (!mod) { ^^^^ Check. 295 dev_err(dev, "NULL mod\n"); 296 return -EIO; regards, dan carpenter

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[alsa-devel] [RFC] ASoC: fsl-ssi: Missing DMA-Request workaround
by Markus Pargmann 12 Nov '13

12 Nov '13

There is a hardware issue when using capture and playback at the same time. In the tested scenario, the capture stream was started slightly before the playback stream. The first time the application starts after a powerup, there is a high probability that the SSI unit does not send a request to the SDMA unit. The playback does not work. I tested this on i.MX53 with SSI unit 2 kernel 3.10.10. Here is a register dump of the situation: SDMA 0x63fb0000 00000000: c0040000 00000000 0000001e 00000000 00000010: 00000001 ffffffff 00000000 e03efa41 00000020: 00000000 00000000 00000000 00000000 00000030: 00000064 00000000 00000003 00000000 00000040: 00000000 00000000 00000000 00006000 00000050: 00000000 00000000 00000001 00004050 00000060: 01208000 00000000 00000000 00000000 /* 0x60: EVT_MIRROR showing all received DMA requests */ ^ This should be a 3 instead. Bits 24 and 25 are the DMA requests for RX and TX. Bit 25 is not set, DMA request 25 is missing. SSI 0x50014000 00000000: 00000000 00000000 0000005f 00000000 00000010: 00000457 000050c1 01f80f00 0000028d /* 0x14: ISR Interrupt status register */ ^ At 0x14 we can clearly see the last bit is set (0x1) which is TFE0, Transmit FIFO 0 empty. So it should automatically request data from the DMA engine, but it doesn't. 00000020: 0000028d 0004e000 00040000 000010dd 00000030: 00001f1f 00000000 00000000 00000000 00000040: 00000000 00000000 00000000 00000000 00000050: 00000000 00000000 ........ STATS (/sys/bus/platform/devices/50014000.ssi/statistics) rfrc=0 tfrc=0 roe1=0 roe0=0 tue1=0 tue0=1094475 I could solve this issue by filling the transmit FIFO with data. After that, the SDMA unit receives DMA request 25 again and audio playback works. This patch adds TUE0 interrupt handler which fills FIFO0 with arbitrary data. Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> --- Hi, does anyone have an idea why the SSI unit does not request DMA by its own at the beginning? Or is it a hardware errata that is not listed yet? Regards, Markus Pargmann sound/soc/fsl/fsl_ssi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6b81d0c..8faf2b5 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -269,9 +269,14 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) } if (sisr & CCSR_SSI_SISR_TUE0) { + int i; + ssi_private->stats.tue0++; sisr2 |= CCSR_SSI_SISR_TUE0; ret = IRQ_HANDLED; + + for (i = 0; i != ssi_private->fifo_depth; ++i) + write_ssi(0x0, &ssi->stx0); } if (sisr & CCSR_SSI_SISR_TFS) { -- 1.8.4.2

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[alsa-devel] [PATCH] ALSA: hda - Fix Line Out automute on Realtek multifunction jacks
by David Henningsson 12 Nov '13

12 Nov '13

In case there is both a multifunction headset jack and a Line Out jack, automuting was not working properly from the Line Out jack. This patch fixes that issue. Cc: stable(a)vger.kernel.org (3.10+) BugLink: https://bugs.launchpad.net/bugs/1250377 Tested-by: Cyrus Lien <cyrus.lien(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 24d924d..04d1e6b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3344,8 +3344,10 @@ static void alc_update_headset_mode(struct hda_codec *codec) else new_headset_mode = ALC_HEADSET_MODE_HEADPHONE; - if (new_headset_mode == spec->current_headset_mode) + if (new_headset_mode == spec->current_headset_mode) { + snd_hda_gen_update_outputs(codec); return; + } switch (new_headset_mode) { case ALC_HEADSET_MODE_UNPLUGGED: -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - Check keep_eapd_on before inv_eapd
by Takashi Iwai 12 Nov '13

12 Nov '13

We don't change the EAPD bit in set_pin_eapd() if keep_eapd_on flag is set by the codec driver and enable is false. But, we also apply the flipping of enable value according to inv_eapd flag in the same function, and this confused the former check, handled as if it's turned ON. The inverted EAPD check must be applied after keep_eapd_on check, instead. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 276f6e759bac..3067ed4fe3b2 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -796,10 +796,10 @@ static void set_pin_eapd(struct hda_codec *codec, hda_nid_t pin, bool enable) if (spec->own_eapd_ctl || !(snd_hda_query_pin_caps(codec, pin) & AC_PINCAP_EAPD)) return; - if (codec->inv_eapd) - enable = !enable; if (spec->keep_eapd_on && !enable) return; + if (codec->inv_eapd) + enable = !enable; snd_hda_codec_update_cache(codec, pin, 0, AC_VERB_SET_EAPD_BTLENABLE, enable ? 0x02 : 0x00); -- 1.8.4.2

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Re: [alsa-devel] How to apply rate conversion in asound.conf
by Dolevo Jay 12 Nov '13

12 Nov '13

> Date: Wed, 6 Nov 2013 17:30:25 +0100 > From: clemens(a)ladisch.de > To: cmst(a)live.com; alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] How to apply rate conversion in asound.conf > > Dolevo Jay wrote: > > pcm.!default { > > type plug > > slave.pcm multi > > } > > > > pcm.multi { > > slaves.a.pcm "hw:0,1"; > > slaves.b.pcm "hw:1,1"; > > ... > > > > My purpose is to apply rate conversion to the "hw:1,1" and have 44100 rate > > out. > > All slaves must use the same rate. > > The easiest way is apply this to the plug plugin: > > pcm.!default { > type plug > slave { > pcm multi > rate 44100 > } > } > > > Regards, > Clemens > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel Hi, What you suggested worked perfect but I implemented the rate conversion in my software for some specific reasons. What I need to do now is to have a aosund.conf which has dmix into multi plugins together. So, here is my multi configuration: -------------------------------- pcm.!default { type plug slave.pcm multi } pcm.multi { type multi; slaves.a.pcm "hw:0,1"; slaves.a.channels 2; slaves.b.pcm "hw:1,1"; slaves.b.channels 2; bindings.0.slave a; bindings.0.channel 0; bindings.1.slave a; bindings.1.channel 1; bindings.2.slave b; bindings.2.channel 0; bindings.3.slave b; bindings.3.channel 1; } -------------------------------- The following dmix runs ok for 1,1 hardware: -------------------------------- pcm.dmixer { type dmix ipc_key 2048 slave { pcm "hw:1,1" } bindings { 0 0 1 1 } } pcm.!default { type plug slave.pcm dmixer } -------------------------------- Could you please tell me how I can implement the second configuration with dmixer into the first one which has multi. So, basically how can I call a plugin inside of another plugin? Thanks a lot.

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[alsa-devel] [snd_msnd_init] WARNING: CPU: 0 PID: 1 at fs/sysfs/dir.c:486 sysfs_warn_dup+0x7d/0xa0()
by Fengguang Wu 12 Nov '13

12 Nov '13

Greetings, I got this warning when booting a kernel with the attached kconfig. [ 5.073253] kobject: 'msnd-pinnacle.7' (cec33408): kobject_release, parent (null) (delayed) [ 5.074347] kobject: 'msnd-pinnacle' (cecd4980): kobject_release, parent cf3ad9b0 (delayed) [ 5.075576] ------------[ cut here ]------------ [ 5.076200] WARNING: CPU: 0 PID: 1 at fs/sysfs/dir.c:486 sysfs_warn_dup+0x7d/0xa0() [ 5.077434] sysfs: cannot create duplicate filename '/bus/isa/drivers/msnd-pinnacle' [ 5.078422] CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.12.0-wl-05537-gf1c1073 #242 [ 5.079529] Hardware name: Bochs Bochs, BIOS Bochs 01/01/2011 [ 5.080418] 00000000 00000000 cb857d74 c17a4989 cb857db4 cb857da4 c10944ba c1a23264 [ 5.081680] cb857dd0 00000001 c1a231f8 000001e6 c11c051d c11c051d c0025000 000001e6 [ 5.083104] c0025000 cb857dbc c109452e 00000009 cb857db4 c1a23264 cb857dd0 cb857de0 [ 5.084361] Call Trace: [ 5.084745] [<c17a4989>] dump_stack+0x4b/0x98 [ 5.085463] [<c10944ba>] warn_slowpath_common+0x7a/0xa0 [ 5.086182] [<c11c051d>] ? sysfs_warn_dup+0x7d/0xa0 [ 5.086785] [<c11c051d>] ? sysfs_warn_dup+0x7d/0xa0 [ 5.087397] [<c109452e>] warn_slowpath_fmt+0x2e/0x30 [ 5.087980] [<c11c051d>] sysfs_warn_dup+0x7d/0xa0 [ 5.088621] [<c11c0569>] sysfs_add_one+0x29/0x40 [ 5.089230] [<c11c0648>] create_dir+0x58/0xa0 [ 5.089797] [<c11c0a1d>] sysfs_create_dir_ns+0xfd/0x120 [ 5.090709] [<c12343c2>] kobject_add_internal+0xc2/0x240 [ 5.091505] [<c1179fbb>] ? kfree+0x7b/0x210 [ 5.092211] [<c1233fc2>] ? kobject_set_name_vargs+0x42/0x60 [ 5.092978] [<c123458d>] kobject_init_and_add+0x4d/0x60 [ 5.093706] [<c13e108a>] bus_add_driver+0x7a/0x200 [ 5.094351] [<c1c1c729>] ? snd_msnd_init+0x49/0x49 [ 5.095085] [<c13ebbb0>] ? isa_bus_remove+0x30/0x30 [ 5.095817] [<c1c1c729>] ? snd_msnd_init+0x49/0x49 [ 5.096521] [<c13e285f>] driver_register+0xaf/0xf0 [ 5.097136] [<c10c4492>] ? preempt_count_sub+0xf2/0x150 [ 5.097795] [<c13ebca6>] isa_register_driver+0x26/0x160 [ 5.098534] [<c1c1c6e0>] ? alsa_card_gusextreme_init+0x14/0x14 [ 5.099423] [<c17b3f20>] ? _raw_spin_unlock+0x50/0x60 [ 5.100093] [<c1c1c729>] ? snd_msnd_init+0x49/0x49 [ 5.100715] [<c1c1c73b>] snd_msnd_init+0x12/0x49 [ 5.101336] [<c1bd3bb3>] do_one_initcall+0x90/0x155 [ 5.102207] [<c10d00e0>] ? put_prev_task_fair+0x130/0x490 [ 5.102891] [<c123ecfd>] ? strlen+0xd/0x20 [ 5.103490] [<c1bd3500>] ? do_early_param+0x5e/0x7a [ 5.104164] [<c10b4a6c>] ? parse_args+0x1ec/0x330 [ 5.104746] [<c17b3fa8>] ? _raw_spin_unlock_irqrestore+0x78/0x90 [ 5.105526] [<c1bd3e18>] kernel_init_freeable+0x1a0/0x23f [ 5.106240] [<c1bd351c>] ? do_early_param+0x7a/0x7a [ 5.106961] [<c179bbcb>] kernel_init+0xb/0x150 [ 5.107566] [<c17b5937>] ret_from_kernel_thread+0x1b/0x28 [ 5.108356] [<c179bbc0>] ? rest_init+0xc0/0xc0 [ 5.109054] ---[ end trace f2f06bffaee363dd ]--- [ 5.109705] ------------[ cut here ]------------ Thanks, Fengguang

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[alsa-devel] [PATCHv1 0/8] ALSA: Add SAI driver and enable SGT15000 codec.
by Xiubo Li 12 Nov '13

12 Nov '13

This patch series is mostly Freescale's SAI SoC Digital Audio Interface driver implementation. And the implementation is only compatible with device tree definition. This patch series is based on linux-next and has been tested on Vybrid VF610 Tower board using device tree. Added in v1: - Add SAI SoC Digital Audio Interface driver. - Add Freescale SAI ALSA SoC Digital Audio Interface node for VF610. - Enables SAI ALSA SoC DAI device for Vybrid VF610 TOWER board. - Add device tree bindings for Freescale SAI. - Revise the bugs about the sgt15000 codec. - Add SGT15000 based audio machine driver. - Enable SGT15000 codec based audio driver node for VF610. - Add device tree bindings for Freescale VF610 sound.

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[alsa-devel] [COMPRESS] [PATCH] ALSA: compress_core: don't return -EBADFD from poll if paused
by Richard Fitzgerald 12 Nov '13

12 Nov '13

Pausing audio playback is not an illegal state so it doesn't seem sensible for poll() to return -EBADFD on a paused stream. There's also no reason to assume that we can't write more data to the DSP while playback is paused. Remove the -EBADFD so that a stream in paused state will still report the buffer availability from poll(). It is up to the user process to manage its state so that it knows whether it is paused or not. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index bea523a..30ca133 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -384,8 +384,7 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait) return -EFAULT; mutex_lock(&stream->device->lock); - if (stream->runtime->state == SNDRV_PCM_STATE_PAUSED || - stream->runtime->state == SNDRV_PCM_STATE_OPEN) { + if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) { retval = -EBADFD; goto out; } -- 1.7.2.5

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[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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