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

[alsa-devel] [PATCH 0/6] ASoC: Fix sparse warnings
by Lars-Peter Clausen 29 Jul '15

29 Jul '15

Hi, This series fixes a couple of sparse warnings that have appeared recently in the ASoC tree. Non of them are critical mostly just missing annotations or storage qualifiers. It still makes sense to fix those though since it reduces the noise that has to be filtered through when checking new code for new (potentially critical) warnings. - Lars Lars-Peter Clausen (6): ASoC: wm8804: Drop dublicate const ASoC: lpass-ipq806x: Staticise local symbols ASoC: rcar ctu: Staticise local symbols ASoC: uniperf: Add missing __iomem annotation ASoC: uniperf: Staticise local symbols ASoC: xtfpga-i2s: Add missing __rcu annotation sound/soc/codecs/wm8804.c | 2 +- sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/sh/rcar/ctu.c | 2 +- sound/soc/sti/uniperif.h | 2 +- sound/soc/sti/uniperif_player.c | 2 +- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) -- 2.1.4

7 18

[alsa-devel] [PATCH] ASoC: max98090: Simplify max98090_readable_register implementation
by Axel Lin 29 Jul '15

29 Jul '15

The readable registers are in consecutive ranges: 0x01 ~ 0x03, 0x0D ~ 0xD1, 0xFF So simplify the implementation by specifying a range of consecutive values in a single case label. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98090.c | 71 ++------------------------------------------- 1 file changed, 2 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 91c9f9f..315c71f 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -267,75 +267,8 @@ static bool max98090_volatile_register(struct device *dev, unsigned int reg) static bool max98090_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case M98090_REG_DEVICE_STATUS: - case M98090_REG_JACK_STATUS: - case M98090_REG_INTERRUPT_S: - case M98090_REG_RESERVED: - case M98090_REG_LINE_INPUT_CONFIG: - case M98090_REG_LINE_INPUT_LEVEL: - case M98090_REG_INPUT_MODE: - case M98090_REG_MIC1_INPUT_LEVEL: - case M98090_REG_MIC2_INPUT_LEVEL: - case M98090_REG_MIC_BIAS_VOLTAGE: - case M98090_REG_DIGITAL_MIC_ENABLE: - case M98090_REG_DIGITAL_MIC_CONFIG: - case M98090_REG_LEFT_ADC_MIXER: - case M98090_REG_RIGHT_ADC_MIXER: - case M98090_REG_LEFT_ADC_LEVEL: - case M98090_REG_RIGHT_ADC_LEVEL: - case M98090_REG_ADC_BIQUAD_LEVEL: - case M98090_REG_ADC_SIDETONE: - case M98090_REG_SYSTEM_CLOCK: - case M98090_REG_CLOCK_MODE: - case M98090_REG_CLOCK_RATIO_NI_MSB: - case M98090_REG_CLOCK_RATIO_NI_LSB: - case M98090_REG_CLOCK_RATIO_MI_MSB: - case M98090_REG_CLOCK_RATIO_MI_LSB: - case M98090_REG_MASTER_MODE: - case M98090_REG_INTERFACE_FORMAT: - case M98090_REG_TDM_CONTROL: - case M98090_REG_TDM_FORMAT: - case M98090_REG_IO_CONFIGURATION: - case M98090_REG_FILTER_CONFIG: - case M98090_REG_DAI_PLAYBACK_LEVEL: - case M98090_REG_DAI_PLAYBACK_LEVEL_EQ: - case M98090_REG_LEFT_HP_MIXER: - case M98090_REG_RIGHT_HP_MIXER: - case M98090_REG_HP_CONTROL: - case M98090_REG_LEFT_HP_VOLUME: - case M98090_REG_RIGHT_HP_VOLUME: - case M98090_REG_LEFT_SPK_MIXER: - case M98090_REG_RIGHT_SPK_MIXER: - case M98090_REG_SPK_CONTROL: - case M98090_REG_LEFT_SPK_VOLUME: - case M98090_REG_RIGHT_SPK_VOLUME: - case M98090_REG_DRC_TIMING: - case M98090_REG_DRC_COMPRESSOR: - case M98090_REG_DRC_EXPANDER: - case M98090_REG_DRC_GAIN: - case M98090_REG_RCV_LOUTL_MIXER: - case M98090_REG_RCV_LOUTL_CONTROL: - case M98090_REG_RCV_LOUTL_VOLUME: - case M98090_REG_LOUTR_MIXER: - case M98090_REG_LOUTR_CONTROL: - case M98090_REG_LOUTR_VOLUME: - case M98090_REG_JACK_DETECT: - case M98090_REG_INPUT_ENABLE: - case M98090_REG_OUTPUT_ENABLE: - case M98090_REG_LEVEL_CONTROL: - case M98090_REG_DSP_FILTER_ENABLE: - case M98090_REG_BIAS_CONTROL: - case M98090_REG_DAC_CONTROL: - case M98090_REG_ADC_CONTROL: - case M98090_REG_DEVICE_SHUTDOWN: - case M98090_REG_EQUALIZER_BASE ... M98090_REG_EQUALIZER_BASE + 0x68: - case M98090_REG_RECORD_BIQUAD_BASE ... M98090_REG_RECORD_BIQUAD_BASE + 0x0E: - case M98090_REG_DMIC3_VOLUME: - case M98090_REG_DMIC4_VOLUME: - case M98090_REG_DMIC34_BQ_PREATTEN: - case M98090_REG_RECORD_TDM_SLOT: - case M98090_REG_SAMPLE_RATE: - case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E: + case M98090_REG_DEVICE_STATUS ... M98090_REG_INTERRUPT_S: + case M98090_REG_LINE_INPUT_CONFIG ... 0xD1: case M98090_REG_REVISION_ID: return true; default: -- 2.1.0

3 3

[alsa-devel] [PATCH] sound/usb: add dB range mapping for some devices.
by Yao-Wen Mao 29 Jul '15

29 Jul '15

Add the correct dB ranges of Bose Companion 5 and Drangonfly DAC 1.2. Signed-off-by: Yao-Wen Mao <yaowen(a)google.com> --- sound/usb/mixer_maps.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index e5000da..f0c9e49 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -339,6 +339,19 @@ static struct usbmix_name_map gamecom780_map[] = { static const struct usbmix_name_map scms_usb3318_map[] = { { 10, NULL }, { 0 } + +/* Bose companion 5, the dB conversion factor is 16 instead of 256 */ +static struct usbmix_dB_map bose_companion5_dB = {-5006, -6}; +static struct usbmix_name_map bose_companion5_map[] = { + { 3, NULL, .dB = &bose_companion5_dB }, + { 0 } /* terminator */ +}; + +/* Dragonfly DAC 1.2, the dB conversion factor is 1 instead of 256 */ +static struct usbmix_dB_map dragonfly_1_2_dB = {0, 5000}; +static struct usbmix_name_map dragonfly_1_2_map[] = { + { 7, NULL, .dB = &dragonfly_1_2_dB }, + { 0 } /* terminator */ }; /* @@ -451,6 +464,16 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x25c4, 0x0003), .map = scms_usb3318_map, }, + { + /* Bose Companion 5 */ + .id = USB_ID(0x05a7, 0x1020), + .map = bose_companion5_map, + }, + { + /* Dragonfly DAC 1.2 */ + .id = USB_ID(0x21b4, 0x0081), + .map = dragonfly_1_2_map, + }, { 0 } /* terminator */ }; -- 2.5.0.rc2.392.g76e840b

2 2

Re: [alsa-devel] Bug on Dell Vostro 5480 (Intel Wildcat) kernel driver
by Paulo Roberto de Oliveira Castro 29 Jul '15

29 Jul '15

I'm not an expert on applying patches to the kernel (never did that before), therefore, I don't know if I did everything right. What I did was: downloaded 4.2-rc4 from kernel.org[1], added the line by hand (couldn't apply the patch using patch), copied the config file from /boot/ (the config fedora 22 uses for its kernel) to .config on the root of the kernel directory, runned "make oldconfig" (typing ENTER for every question, is that right?), "make", "make modules_install", "make install". Rebooted, and chose the new kernel I just installed. The sound worked like a charm, the speakers worked as expected, and the volume was as loud as on Windows. Is the patch going to be applied on master? Any chance that fix reaches my Fedora 22 (with backport to older kernel versions)? Thanks for the help, Paulo [1] https://www.kernel.org/pub/linux/kernel/v4.x/testing/linux-4.2-rc4.tar.xz 2015-07-28 10:49 GMT-03:00 Paulo Roberto de Oliveira Castro < p.oliveira.castro(a)gmail.com>: > > > ---------- Forwarded message --------- > From: Takashi Iwai <tiwai(a)suse.de> > Date: Tue, Jul 28, 2015, 03:35 > Subject: Re: [alsa-devel] Bug on Dell Vostro 5480 (Intel Wildcat) kernel > driver > To: Paulo Roberto de Oliveira Castro <p.oliveira.castro(a)gmail.com> > Cc: <alsa-devel(a)alsa-project.org> > > > On Tue, 28 Jul 2015 05:35:34 +0200, > Paulo Roberto de Oliveira Castro wrote: > > > > Hi everyone, > > > > After installing Fedora 22 on my Dell Vostro 5480, the speakers don't > work, > > and the sound only comes out of the subwoofer. Here is the output of > > alsa-info.sh. After talking to "debianuser" on IRC, we found this > > workaround: > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_PIN_WIDGET_CONTROL 0x40 # enable, > > set it to OUT > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_CONNECT_SEL 0 # switch to 0x0c > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0xa000 # unmute left > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0x9000 # unmute > right > > > > We suspect that it's a kernel driver bug. This makes the speakers work > > alongside the headphones (both at the same time) but I need help making > it > > working when headphones are not plugged in. > > Does the patch below work? Try with 4.2-rc kernel at best. > > > Takashi > > --- > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 4ae877c3b6a1..18ae17ebb356 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -5185,6 +5185,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] > = { > SND_PCI_QUIRK(0x1028, 0x064a, "Dell", > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x064b, "Dell", > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", > ALC288_FIXUP_DELL_XPS_13), > + SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", > ALC290_FIXUP_SUBWOOFER_HSJACK), > SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", > ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x06da, "Dell", > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), >

2 1

[alsa-devel] [PATCH] ASoC: dapm: fix module unload/reload failed issue by reset kcontrol's widget list
by Jie Yang 29 Jul '15

29 Jul '15

We need reset the related kcontrol's widget list during the freeing of a widget, otherwise, the widget list may be outdated and reference to it may introduce errors(they usually occurs during driver modules unloading/reloading). Here adding a func dapm_update_kcontrols_of_freeing_widget and call it at dapm_update_kcontrols_of_freeing_widget, to fix those issues. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/soc-dapm.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aa327c9..1ad8144 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2305,6 +2305,35 @@ static void dapm_free_path(struct snd_soc_dapm_path *path) kfree(path); } +/* update all dapm kcontrols that related to a widget which being freed*/ +static void dapm_update_kcontrols_of_freeing_widget(struct snd_soc_dapm_widget * w) +{ + struct dapm_kcontrol_data *data = NULL; + int i, j, n; + + if (w && w->kcontrols) { + for (i = 0; i < w->num_kcontrols; i++) { + if (w->kcontrols[i] == NULL) + continue; + data = snd_kcontrol_chip(w->kcontrols[i]); + if (data == NULL) + continue; + for (j = 0, n = 0; j < data->wlist->num_widgets; j++) { + if (data->wlist->widgets[j] == w) { + data->wlist->widgets[j] = NULL; + n++; + } + } + + if (n) { + data->wlist->num_widgets -= n; + data->widget = NULL; + } + } + } + +} + /* free all dapm widgets and resources */ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) { @@ -2326,6 +2355,8 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) list_for_each_entry_safe(p, next_p, &w->sinks, list_source) dapm_free_path(p); + dapm_update_kcontrols_of_freeing_widget(w); + kfree(w->kcontrols); kfree(w->name); kfree(w); -- 1.9.1

4 4

[alsa-devel] [PATCH 1/2] ALSA: hda - skip runtime PM if async probe is not finished
by U. Artie Eoff 28 Jul '15

28 Jul '15

Crash can occur if runtime PM is triggered before the async probe finishes via the azx_firmware_cb when CONFIG_SND_HDA_PATCH_LOADER is defined. Don't execute PM ops if the async probe has not completed yet. The probe is finished when chip->running is true. Signed-off-by: U. Artie Eoff <ullysses.a.eoff(a)intel.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735bdcb04ce8..b729b25a6ad6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1023,7 +1023,7 @@ static int azx_runtime_idle(struct device *dev) chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); - if (chip->disabled || hda->init_failed) + if (chip->disabled || hda->init_failed || !chip->running) return 0; if (!power_save_controller || !azx_has_pm_runtime(chip) || -- 1.9.3

4 9

[alsa-devel] [PATCH v13 0/3] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 28 Jul '15

28 Jul '15

This patch series asks for Russell King's patch http://article.gmane.org/gmane.linux.ports.arm.kernel/411574 drm/edid: add function to help find SADs to be applayed. v13: - rebase on 4.2.0-rc3 - remove Russell's patches v12: - use Russell King's DRM ELD helper (Mark Brown) v11: - reduce the patch series to adding the tda998x codec only v10: - add the generic dt-card - define the audio ports from a DT graph of ports (Russell King) - reuse HDMI constants (Andrew Jackson - Jyri Sarha) - alloc rate_constraints in codec (Jyri Sarha) - fix bad number of channels (Jyri Sarha) - correct codec generation from config (Russell King - Jyri Sarha) - no module init/exit (Russell King) v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (3): drm/i2c: tda998x: Add support of a DT graph of ports drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter .../devicetree/bindings/drm/i2c/tda998x.txt | 51 ++++++ drivers/gpu/drm/i2c/tda998x_drv.c | 178 +++++++++++++++++++-- include/sound/tda998x.h | 23 +++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 132 +++++++++++++++ 6 files changed, 381 insertions(+), 11 deletions(-) create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/tda998x.c -- 2.1.4

4 12

[alsa-devel] rockchip rk3288 registration
by Michael Trimarchi 28 Jul '15

28 Jul '15

Hi I have created a simple sound card to test registration on i2s but I have some problems that after a while in SND_SOC_DAIFMT_CBM_CFM the device stop to register and receive a input/output error. It means that I can not receive anymore from the device arecord -c 2 -r 96000 -f S32_LE -D hw:0 > record.pcm Recording WAVE 'stdin' : Signed 32 bit Little Endian, Rate 96000 Hz, Stereo arecord: pcm_read:2031: read error: Input/output error The result size is a bit random. I have no problem on playback until now. Any idea? Michael

1 1

[alsa-devel] Running PJSIP for voip calls in alsa
by Srinivasan S 28 Jul '15

28 Jul '15

Dear Clemens, Could you please let me know whether the below issue is in PJSIP or alsa or driver or in the ti sdk8 BSP's (rootfs) 1. We are running PJSIP for establishing VOIP calls on ti sdk 8 BSP's (linux kernel 3.14), we are seeing the below error messages in the PJSIP, Could you please let me know what could be issue ? 08:32:42.082 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.083 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.086 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.088 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.088 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.091 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.112 Master/sound Underflow, buf_cnt=0, will generate 1 frame In failure call pjsip reports Master/sound Underflow continuously, At this point the control stays at pjsip and application will wait for it and all functionality in idle state at application and after this, no other feature will work until restart the application. The above problem observed with new rootfs (ti sdk 8) but when we tried with old rootfs (ti sdk 7) we are able to make calls with no problem. Few calls are long duration calls(33 and 55 minutes). With ti sdk 8 kernel, I tried replacing even the alsa libraries from ti sdk 7 rootfs to ti sdk 8 rootfs, but still the issue persists, /usr/lib/libasound.la /usr/lib/libasound.so /usr/lib/alsa-lib/smixer/smixer-ac97.la /usr/lib/alsa-lib/smixer/smixer-ac97.so /usr/lib/alsa-lib/smixer/smixer-hda.la /usr/lib/alsa-lib/smixer/smixer-hda.so /usr/lib/alsa-lib/smixer/smixer-sbase.la /usr/lib/alsa-lib/smixer/smixer-sbase.so Firstly could you please let me know why this issue is seen in ti sdk 8 BSP's (ie., in rootfs), & what could be the problem in ti sdk 8 rootfs ?? Apart from the above is there any other ALSA related configuration files, needs to copied from the ti sdk 7 rootfs to ti sdk 8 rootfs, inorder to resolve this issue If possible, could you please validate this PJSIP version 2.3 on ti sdk 8 BSP's on your EVK kit & let me know whether this issue is reproducible Awaiting for your replies, Many Thanks in advance

3 11

[alsa-devel] [PATCH] sound/usb: add dB range mapping quirks.
by Yao-Wen Mao 28 Jul '15

28 Jul '15

Some usb audio devices don't follow the specification of the volume range. Add dB_mapping_quirks to fix some devices' dB ranges so that the function snd_mixer_selem_set_playback_dB in alsa-lib can set the correct value for those devices. Signed-off-by: Yao-Wen Mao <yaowen(a)google.com> --- sound/usb/mixer.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6b3acba..b6b27dc 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -933,6 +933,32 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, } } +/* dB mapping quirks */ +static void dB_mapping_quirks(struct usb_mixer_elem_info *cval, + struct snd_kcontrol *kctl) +{ + switch (cval->mixer->chip->usb_id) { + case USB_ID(0x05a7, 0x1020): /* Bose Companion 5 */ + if (strcmp(kctl->id.name, "PCM Playback Volume") == 0) { + cval->dBmin = (convert_signed_value(cval, cval->min) * 100) / 16; + cval->dBmax = (convert_signed_value(cval, cval->max) * 100) / 16; + } + break; + case USB_ID(0x21b4, 0x0081): /* Dragonfly DAC 1.2 */ + if (strcmp(kctl->id.name, "PCM Playback Volume") == 0) { + cval->dBmin = convert_signed_value(cval, cval->min) * 100; + cval->dBmax = convert_signed_value(cval, cval->max) * 100; + } + break; + case USB_ID(0x046d, 0x0a29): /* Logitech H800 */ + if (strcmp(kctl->id.name, "PCM Playback Volume") == 0) { + cval->dBmin = convert_signed_value(cval, cval->min); + cval->dBmax = convert_signed_value(cval, cval->max); + } + break; + } +} + /* * retrieve the minimum and maximum values for the specified control */ @@ -1027,6 +1053,10 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, */ cval->dBmin = (convert_signed_value(cval, cval->min) * 100) / 256; cval->dBmax = (convert_signed_value(cval, cval->max) * 100) / 256; + + if (kctl) + dB_mapping_quirks(cval, kctl); + if (cval->dBmin > cval->dBmax) { /* something is wrong; assume it's either from/to 0dB */ if (cval->dBmin < 0) -- 2.5.0.rc2.392.g76e840b

2 5