- Patch - mailman.alsa-project.org

[PATCH] ALSA: usb-audio: ignore broken processing/extension unit
by tom.ty89＠gmail.com 16 Aug '20

16 Aug '20

From: Tom Yan <tom.ty89(a)gmail.com> Some devices have broken extension unit where getting current value doesn't work. Attempt that once when creating mixer control for it. If it fails, just ignore it, so that it won't cripple the device entirely (and/or make the error floods). --- sound/usb/mixer.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index eab0fd4fd7c3..e0b7174c1043 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2367,7 +2367,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, int num_ins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; - int i, err, nameid, type, len; + int i, err, nameid, type, len, val; const struct procunit_info *info; const struct procunit_value_info *valinfo; const struct usbmix_name_map *map; @@ -2470,6 +2470,12 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, break; } + err = get_cur_ctl_value(cval, cval->control << 8, &val); + if (err < 0) { + usb_mixer_elem_info_free(cval); + return -EINVAL; + } + kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); if (!kctl) { usb_mixer_elem_info_free(cval); -- 2.28.0

1 0

Re: [ALSA patch] [PATCH] pcm: dmix: fix access to sum-buffer in non-interleaved mixing mode
by Takashi Iwai 13 Aug '20

13 Aug '20

On Wed, 12 Aug 2020 08:18:25 +0200, Vijay Palaniswamy (RBEI/ECF3) wrote: > > Hi All, > > When dmix uses non-interleaved mixing mode the offset and step width to > sum_buffer was calculated by using the dmix channels instead of the slave > channels. This leads to audio distortions due to frame corruption. > > Issue happens while doing channel wise mixing of src and destination buffers. > > Attached the patch to fix the offset and step width calculation based on slave > channels for sum buffer. So mixed data will be saved in respective channels of > sum buffer. > > Steps to reproduce the issue: > > With below configuratio, Do aplay on both device in parallel for audio > distortion > > pcm.dmix_2_channels { > > type dmix > > ipc_key 5678293 > > ipc_perm 0660 > > ipc_gid audio > > bindings [0 1] > > slave { > > pcm "hardware" > > channels 2 > > periods 4 > > period_time 40000 > > } > > } > > pcm.dmix_1_channels { > > type dmix > > ipc_key 5678293 > > ipc_perm 0660 > > ipc_gid audio > > bindings [0] > > slave { > > pcm "hardware" > > channels 1 > > periods 4 > > period_time 40000 > > } > > } > > pcm.hardware { > > type hw > > card 0 > > channels 2 > > rate 48000 > > format S16_LE > > } > > Best regards > > Palaniswamy Vijay Applied now. Thanks. Takashi

1 0

Re: [ALSA patch] [PATCH 1/1] ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Xtreme
by Takashi Iwai 18 May '20

18 May '20

On Mon, 18 May 2020 07:38:44 +0200, Christian Lachner wrote: > > The Gigabyte X570 Aorus Xtreme motherboard with ALC1220 codec > requires a similar workaround for Clevo laptops to enforce the > DAC/mixer connection path. Set up a quirk entry for that. > > BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205275 > Signed-off-by: Christian Lachner <gladiac(a)gmail.com> Applied now. thanks, Takashi

1 0

Re: [ALSA patch] [PATCH alsa-lib] conf: USB-Audio: Fix S/PDIF output of ASUS Xonar AE
by Jaroslav Kysela 17 Apr '20

17 Apr '20

Dne 26. 03. 20 v 20:46 Ivan Mironov napsal(a): > ASUS Xonar AE is a PCI-Express card containing USB controller: > > USB controller [0c03]: ASMedia Technology Inc. ASM1042A USB 3.0 Host Controller [1b21:1142] > > and the actual USB sound card: > > ID 0b05:180f ASUSTek Computer, Inc. XONAR SOUND CARD > > As other Xonar USB sound cards, it uses second device for digital > output. > > Signed-off-by: Ivan Mironov <mironov.ivan(a)gmail.com> > > diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf > index 89811086db1d..3f4c421b7e2c 100644 > --- a/src/conf/cards/USB-Audio.conf > +++ b/src/conf/cards/USB-Audio.conf > @@ -40,6 +40,7 @@ USB-Audio.pcm.iec958_device { > "USB Sound Blaster HD" 1 > "Xonar U7" 1 > "ASUS XONAR U5" 1 > + "XONAR SOUND CARD" 1 > > # The below don't have digital in/out, so prevent them from being opened. > "Andrea PureAudio USB-SA Headset" 999 > Merged. Thank you. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

Re: [ALSA patch] [alsa-devel] [PATCH] do not set close-on-exec flag on descriptor if it was already set
by Jaroslav Kysela 11 Feb '20

11 Feb '20

Dne 11. 02. 20 v 11:22 Rolf Eike Beer napsal(a): > There is no need to set this again if O_CLOEXEC is supported. Thanks. Applied. Jroslav > > Signed-off-by: Rolf Eike Beer <eb(a)emlix.com> > --- > include/local.h | 2 ++ > 1 file changed, 2 insertions(+) > > diff --git a/include/local.h b/include/local.h > index ea0ec32d..ed6ba936 100644 > --- a/include/local.h > +++ b/include/local.h > @@ -320,8 +320,10 @@ static inline int snd_open_device(const char *filename, int fmode) > fd = rsm_open_device(filename, fmode); > } > #endif > +#ifndef O_CLOEXEC > if (fd >= 0) > fcntl(fd, F_SETFD, FD_CLOEXEC); > +#endif > return fd; > } > > -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

Re: [ALSA patch] [alsa-devel] [PATCH - 1/1] pcm: direct: Round down of slave_hw_ptr when buffer size is two period size
by Takashi Iwai 06 Feb '20

06 Feb '20

On Thu, 06 Feb 2020 09:14:42 +0100, Channaiah Vanitha (RBEI/ECF3) wrote: > > Hello Takashi-san, > > This patch is regarding the fix for rounding down/up of slave pointers. > For buffer_size >= 2*period_size, round down of slave pointers and > for buffer_size < 2*period_size, round up of slave pointers will avoid xruns. > which otherwise causes snd_pcm_wait() to block for more than expected snd_pcm_period_elapsed() which leads to xruns. > > We had similar discussion for same issue in below link: > https://mailman.alsa-project.org/pipermail/alsa-devel/2019-June/151169.html Yes, and I still don't get why this must be required. Doesn't this imply that you drop the samples instead? Takashi > > Regards, > Vanitha > > > -----Original Message----- > From: Channaiah Vanitha (RBEI/ECF3) <Vanitha.Channaiah(a)in.bosch.com> > Sent: Thursday, February 6, 2020 1:43 PM > To: Channaiah Vanitha (RBEI/ECF3) <Vanitha.Channaiah(a)in.bosch.com>; patch(a)alsa-project.org > Cc: alsa-devel(a)alsa-project.org; Wischer Timo (ADITG/ESS) <twischer(a)de.adit-jv.com> > Subject: [PATCH - 1/1] pcm: direct: Round down of slave_hw_ptr when buffer size is two period size > > From: Vanitha Channaiah <vanitha.channaiah(a)in.bosch.com> > > For buffer size equal to two period size, the start position of slave_hw_ptr is rounded down in order to avoid xruns > > For e.g.: > Considering below parameters and its values: > Period size = 96 (0x60) > Buffer size = 192 (0xC0) > Timer ticks = 1 > avail_min = 0x60 > slave_hw_ptr = unaligned value during dmix_start() > > Issue: > - Initial, app_ptr = hw_ptr = 0 > - Application fills buffer size of data. so app_ptr = 0xC0, hw_ptr = 0 > - During dmix_start(), current slave_hw_ptr is not rounded down. current slave_hw_ptr would be 0x66 > - slave_hw_ptr is keep on updating at the hardware 0x67, 0x68, 0x69 > - The diff calculation between old_slave_hw_ptr(0x66) and new_slave_hw_ptr(0x69) > results in avail = 0x6 > - After 1 snd_pcm_period_elapsed(), slave_hw_ptr = 0xC0 > - The result of avail = 0x5A which is less than avail_min(0x60) > - Here, xruns will be observed > > Fix: > - Initial, app_ptr = hw_ptr = 0 > - Application fills buffer size of data. so app_ptr = 0xC0, hw_ptr = 0 > - Round down of slave_hw_ptr during dmix_start() leads to below calculation: > - During dmix_start(), slave_hw_ptr rounded to 0x60 (old slave_hw_ptr) > - The diff calculation between old_slave_hw_ptr(0x60) and new_slave_hw_ptr(0x69) > results in avail = 0x9 > - After 1 snd_pcm_period_elapsed(), slave_hw_ptr = 0xC0 > - The result of avail = 0x60 which is avail_min(0x60) > - Here, xruns can be avoided > > Signed-off-by: Vanitha Channaiah <vanitha.channaiah(a)in.bosch.com> > > diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 54d9900..a201fa3 100644 > --- a/src/pcm/pcm_direct.c > +++ b/src/pcm/pcm_direct.c > @@ -2043,10 +2043,14 @@ int snd_pcm_direct_parse_open_conf(snd_config_t *root, snd_config_t *conf, > > void snd_pcm_direct_reset_slave_ptr(snd_pcm_t *pcm, snd_pcm_direct_t *dmix) { > + /* > + * For buffer size equal to two period size, slave_hw_ptr is rounded down > + * to avoid xruns > + */ > > if (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_ROUNDUP || > (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_AUTO && > - pcm->buffer_size <= pcm->period_size * 2)) > + pcm->buffer_size < pcm->period_size * 2)) > dmix->slave_appl_ptr = > ((dmix->slave_appl_ptr + dmix->slave_period_size - 1) / > dmix->slave_period_size) * dmix->slave_period_size; > -- > 2.7.4 > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > https://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

1 1

Re: [ALSA patch] [alsa-devel] [PATCH] ASoC: rt274: Disable jack report IRQ with disabling jack
by Harlozinski, Pawel 13 Nov '19

13 Nov '19

On 11/12/2019 6:10 PM, Mark Brown wrote: > On Tue, Nov 12, 2019 at 02:02:37PM +0100, Pawel Harlozinski wrote: > >> /* Disable jack detection */ >> regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, >> RT274_IRQ_EN, RT274_IRQ_DIS); >> - >> + disable_irq(rt274->i2c->irq); >> return 0; > Shouldn't the register update above be suppressing interrupts? For rt274 disable_irq is also needed, otherwise we're getting flood of irq's in case of not loaded machine board. > disable_irq() is a bit of a hammer and interferes with things like > possible share use. This irq should be handled in codec code anyway - control of jack detect events from non-codec code is done with set_jack. Similar solutions for jack report irq enable/disable flow are implemented in rt5640 and rt5651. > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > https://mailman.alsa-project.org/mailman/listinfo/alsa-devel

1 0

[PATCH] ASoC: Jack: Fix NULL pointer dereference in snd_soc_jack_report
by Pawel Harlozinski 12 Nov '19

12 Nov '19

Check for existance of jack before tracing. NULL pointer dereference has been reported by KASAN while unloading machine driver (snd_soc_cnl_rt274). Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/soc/soc-jack.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index a71d2340eb05..b5748dcd490f 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -82,10 +82,9 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) unsigned int sync = 0; int enable; - trace_snd_soc_jack_report(jack, mask, status); - if (!jack) return; + trace_snd_soc_jack_report(jack, mask, status); dapm = &jack->card->dapm; -- 2.17.1

1 1

[PATCH v3] ASoC: Intel: Skylake: Recover BXT FW on DSP boot timeout error
by Pawel Harlozinski 12 Sep '19

12 Sep '19

When DSP boots with timeout error try to reload firmware to keep best user expeirience. Change-Id: Ic7377da687e7441a487c443f9de1ebae5967afa2 Signed-off-by: Szymon Mielczarek <szymonx.mielczarek(a)intel.com> Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/soc/intel/skylake/bxt-sst.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 92a82e6b5fe6..fdf4c198445e 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -446,14 +446,19 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) /* If core 1 was turned on for booting core 0, turn it off */ skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1)); + /* FIXME this FW recovery applies only for BXT */ if (ret == 0) { - dev_err(ctx->dev, "%s: DSP boot timeout\n", __func__); - dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + dev_warn(ctx->dev, + "DSP boot timeout: Error code=0x%x: FW status=0x%x. Reloading FW\n", sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); - dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); - ret = -EIO; - goto err; + + ret = bxt_sst_init_fw(skl->dev, skl); + if (ret < 0) { + dev_err(ctx->dev, "Reload fw failed: %d. Failed to set core0 to D0 state\n", + ret); + goto err; + } } } @@ -478,7 +483,6 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) if (core_id == SKL_DSP_CORE0_ID) core_mask |= SKL_DSP_CORE_MASK(1); skl_dsp_disable_core(ctx, core_mask); - return ret; } -- 2.17.1

1 0

[RFC PATCH 1/3] ALSA:hda: Simplify and clear calculating SDxFMT
by Pawel Harlozinski 06 Sep '19

06 Sep '19

Set SDxFMT based only on given format, due to maxbps not always being set. Split cases for formats 20,24,32 bits. For format SNDRV_PCM_FORMAT_FLOAT_LE width is equal 32 so it will end up with same mask. Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/hda/hdac_device.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b26cc93e7e10..add758e11b85 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -759,14 +759,13 @@ unsigned int snd_hdac_calc_stream_format(unsigned int rate, val |= AC_FMT_BITS_16; break; case 20: + val |= AC_FMT_BITS_20; + break; case 24: + val |= AC_FMT_BITS_24; + break; case 32: - if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) - val |= AC_FMT_BITS_32; - else if (maxbps >= 24) - val |= AC_FMT_BITS_24; - else - val |= AC_FMT_BITS_20; + val |= AC_FMT_BITS_32; break; default: return 0; -- 2.17.1

3 7