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

[alsa-devel] [PATCH 1/2] ASoC: rt5677: Correct the routing paths of that after IF1/2 DACx Mux
by Oder Chiou 26 Feb '15

26 Feb '15

The patch corrects the routing paths of that after IF1/2 DACx Mux Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index ab62777..f87c929 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3297,8 +3297,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IB45 Bypass Mux", "Bypass", "IB45 Mux" }, { "IB45 Bypass Mux", "Pass SRC", "IB45 Mux" }, - { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6" }, - { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6" }, + { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6 Mux" }, + { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6 Mux" }, { "IB6 Mux", "SLB DAC 6", "SLB DAC6" }, { "IB6 Mux", "STO4 ADC MIX L", "Stereo4 ADC MIXL" }, { "IB6 Mux", "IF4 DAC L", "IF4 DAC L" }, @@ -3306,8 +3306,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IB6 Mux", "STO2 ADC MIX L", "Stereo2 ADC MIXL" }, { "IB6 Mux", "STO3 ADC MIX L", "Stereo3 ADC MIXL" }, - { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7" }, - { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7" }, + { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7 Mux" }, + { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7 Mux" }, { "IB7 Mux", "SLB DAC 7", "SLB DAC7" }, { "IB7 Mux", "STO4 ADC MIX R", "Stereo4 ADC MIXR" }, { "IB7 Mux", "IF4 DAC R", "IF4 DAC R" }, @@ -3648,15 +3648,15 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC1 FS", NULL, "DAC1 MIXL" }, { "DAC1 FS", NULL, "DAC1 MIXR" }, - { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2" }, - { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2" }, + { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2 Mux" }, + { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2 Mux" }, { "DAC2 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC2 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC2 L Mux", "SLB DAC 2", "SLB DAC2" }, { "DAC2 L Mux", "OB 2", "OutBound2" }, - { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3" }, - { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3" }, + { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3 Mux" }, + { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3 Mux" }, { "DAC2 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC2 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC2 R Mux", "SLB DAC 3", "SLB DAC3" }, @@ -3664,29 +3664,29 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC2 R Mux", "Haptic Generator", "Haptic Generator" }, { "DAC2 R Mux", "VAD ADC", "VAD ADC Mux" }, - { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4" }, - { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4" }, + { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4 Mux" }, + { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4 Mux" }, { "DAC3 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC3 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC3 L Mux", "SLB DAC 4", "SLB DAC4" }, { "DAC3 L Mux", "OB 4", "OutBound4" }, - { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC4" }, - { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC4" }, + { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC5 Mux" }, + { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC5 Mux" }, { "DAC3 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC3 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC3 R Mux", "SLB DAC 5", "SLB DAC5" }, { "DAC3 R Mux", "OB 5", "OutBound5" }, - { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6" }, - { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6" }, + { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6 Mux" }, + { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6 Mux" }, { "DAC4 L Mux", "IF3 DAC L", "IF3 DAC L" }, { "DAC4 L Mux", "IF4 DAC L", "IF4 DAC L" }, { "DAC4 L Mux", "SLB DAC 6", "SLB DAC6" }, { "DAC4 L Mux", "OB 6", "OutBound6" }, - { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7" }, - { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7" }, + { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7 Mux" }, + { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7 Mux" }, { "DAC4 R Mux", "IF3 DAC R", "IF3 DAC R" }, { "DAC4 R Mux", "IF4 DAC R", "IF4 DAC R" }, { "DAC4 R Mux", "SLB DAC 7", "SLB DAC7" }, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: Intel: wrap runtime_pm usage count under CONFIG_PM
by Vinod Koul 26 Feb '15

26 Feb '15

The struct dev_pm_ops defines usage_count only when CONFIG_PM is defined. So we should use this variable only in cases where this falg is true. So we define a local variable and read the value under this flag. In non PM cases, we set this to 1. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/sst/sst_drv_interface.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 549af7d7f6d0..f0e4b99b3aeb 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -139,17 +139,23 @@ static int sst_power_control(struct device *dev, bool state) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); int ret = 0; + int usage_count = 0; + +#ifdef CONFIG_PM + usage_count = atomic_read(&dev->power.usage_count); +#else + usage_count = 1; +#endif if (state == true) { ret = pm_runtime_get_sync(dev); - dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", - atomic_read(&dev->power.usage_count)); + + dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); if (ret < 0) { dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); return ret; } - if ((ctx->sst_state == SST_RESET) && - (atomic_read(&dev->power.usage_count) == 1)) { + if ((ctx->sst_state == SST_RESET) && (usage_count == 1)) { ret = sst_load_fw(ctx); if (ret) { dev_err(dev, "FW download fail %d\n", ret); @@ -158,8 +164,7 @@ static int sst_power_control(struct device *dev, bool state) } } } else { - dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", - atomic_read(&dev->power.usage_count)); + dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", usage_count); return sst_pm_runtime_put(ctx); } return ret; -- 1.7.9.5

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[alsa-devel] audio mcbsp1 broken for beagleboard-xm -OMAP
by noman pouigt 25 Feb '15

25 Feb '15

Hello, I am trying to integrate max98090 codec with $subject board. I am running the mcBSP in slave mode as i have not changed the hw_parms for omap-twl4030.c file. With below changes i see wait_for_avail is getting stuck i.e. mcbsp is not getting any interrupt and nothing is playing. Kernel version: VERSION = 3 PATCHLEVEL = 19 SUBLEVEL = 0 EXTRAVERSION = NAME = Diseased Newt Setup information: Using beagle board -xm and connected max98090 codec to mcbsp1 pins. Connected only 4 pins. Changed the omap-twl4030.c machine file as below to disable the twl4030: static struct snd_soc_dai_link omap_twl4030_dai_links[] = { { + .name = "max98090", + .stream_name = "max98090", .cpu_dai_name = "omap-mcbsp.1", + .codec_dai_name = "HiFi", .platform_name = "omap-mcbsp.1", + .codec_name = "max98090.1-0010", .ops = &omap_twl4030_ops, }, DTSI file change:arch/arm/boot/dts/omap3-beagle-xm.dts &i2c2 { clock-frequency = <400000>; + + max98090: max98090@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + }; }; omap/omap-twl4030.c added in hw_params below function to set internal oscillator for 98090 for generating PLL for master clock. snd_soc_dai_set_sysclk(codec_dai, 26000000, 0, SND_SOC_CLOCK_IN); I am able to see that the cpu_dai is mapped with codec_dai and configuration is ok. kernel logs at https://ideone.com/oezNSj I think i don't need to configure the mcbsp1 pins right as it uses default. Please guide me.I have tried IRC channel also but not getting the right folks. https://github.com/CircuitCo/BeagleBoard-xM-RevC/blob/master/BeagleBoard-xM… Thanks,

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[alsa-devel] [patch] ALSA: cmipci: remove a stray space character
by Dan Carpenter 25 Feb '15

25 Feb '15

Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 1d0f2ca..6cf464d 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2062,7 +2062,7 @@ static int snd_cmipci_get_volume(struct snd_kcontrol *kcontrol, val = (snd_cmipci_mixer_read(cm, reg.right_reg) >> reg.right_shift) & reg.mask; if (reg.invert) val = reg.mask - val; - ucontrol->value.integer.value[1] = val; + ucontrol->value.integer.value[1] = val; } spin_unlock_irq(&cm->reg_lock); return 0;

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[alsa-devel] [patch] ALSA: azt3328: some indenting cleanups
by Dan Carpenter 25 Feb '15

25 Feb '15

A few minor tweaks to make things line up correctly. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index a40a2b4..33b2a0a 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1385,8 +1385,8 @@ snd_azf3328_ctrl_codec_activity(struct snd_azf3328 *chip, .running) && (!chip->codecs[peer_codecs[codec_type].other2] .running)); - } - if (call_function) + } + if (call_function) snd_azf3328_ctrl_enable_codecs(chip, enable); /* ...and adjust clock, too @@ -2126,7 +2126,8 @@ static struct snd_pcm_ops snd_azf3328_i2s_out_ops = { static int snd_azf3328_pcm(struct snd_azf3328 *chip) { -enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS }; /* pcm devices */ + /* pcm devices */ + enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS }; struct snd_pcm *pcm; int err;

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[alsa-devel] [patch] sound/sb_midi: a couple indenting fixes
by Dan Carpenter 25 Feb '15

25 Feb '15

Let's make things line up a little bit better. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/sb_midi.c b/sound/oss/sb_midi.c index f139028..551ee75 100644 --- a/sound/oss/sb_midi.c +++ b/sound/oss/sb_midi.c @@ -179,14 +179,14 @@ void sb_dsp_midi_init(sb_devc * devc, struct module *owner) { printk(KERN_WARNING "Sound Blaster: failed to allocate MIDI memory.\n"); sound_unload_mididev(dev); - return; + return; } memcpy((char *) midi_devs[dev], (char *) &sb_midi_operations, sizeof(struct midi_operations)); if (owner) - midi_devs[dev]->owner = owner; - + midi_devs[dev]->owner = owner; + midi_devs[dev]->devc = devc;

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[alsa-devel] [patch] sound/sb_ess: white space cleanups
by Dan Carpenter 25 Feb '15

25 Feb '15

These weren't aligned on the same lines as the surrounding code and the printk was quite messy. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index b47a690..57f7d25 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -604,7 +604,7 @@ static void ess_audio_output_block_audio2 ess_chgmixer (devc, 0x78, 0x03, 0x03); /* Go */ devc->irq_mode_16 = IMODE_OUTPUT; - devc->intr_active_16 = 1; + devc->intr_active_16 = 1; } static void ess_audio_output_block @@ -1183,17 +1183,12 @@ FKS_test (devc); chip = "ES1688"; } - printk ( KERN_INFO "ESS chip %s %s%s\n" - , chip - , ( devc->sbmo.esstype == ESSTYPE_DETECT || devc->sbmo.esstype == ESSTYPE_LIKE20 - ? "detected" - : "specified" - ) - , ( devc->sbmo.esstype == ESSTYPE_LIKE20 - ? " (kernel 2.0 compatible)" - : "" - ) - ); + printk(KERN_INFO "ESS chip %s %s%s\n", chip, + (devc->sbmo.esstype == ESSTYPE_DETECT || + devc->sbmo.esstype == ESSTYPE_LIKE20) ? + "detected" : "specified", + devc->sbmo.esstype == ESSTYPE_LIKE20 ? + " (kernel 2.0 compatible)" : ""); sprintf(name,"ESS %s AudioDrive (rev %d)", chip, ess_minor & 0x0f); } else {

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[alsa-devel] [patch] sound/oss/opl3: remove some stray whitespace
by Dan Carpenter 25 Feb '15

25 Feb '15

Removed an extra tab and a extra space character. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index 607cee4..b6d19ad 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -666,7 +666,7 @@ static int opl3_start_note (int dev, int voice, int note, int volume) opl3_command(map->ioaddr, FNUM_LOW + map->voice_num, data); data = 0x20 | ((block & 0x7) << 2) | ((fnum >> 8) & 0x3); - devc->voc[voice].keyon_byte = data; + devc->voc[voice].keyon_byte = data; opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num, data); if (voice_mode == 4) opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num + 3, data); @@ -717,7 +717,7 @@ static void freq_to_fnum (int freq, int *block, int *fnum) static void opl3_command (int io_addr, unsigned int addr, unsigned int val) { - int i; + int i; /* * The original 2-OP synth requires a quite long delay after writing to a

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[alsa-devel] [patch] sound: sys_timer: indent poll_def_tmr() correctly
by Dan Carpenter 25 Feb '15

25 Feb '15

The indenting here was really whacky and not consistent from one line to the next. I also reverse the "if (opened)" and "if (tmr_running)" tests so that I could remove two indent levels. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/sys_timer.c b/sound/oss/sys_timer.c index 9f03983..2226dda 100644 --- a/sound/oss/sys_timer.c +++ b/sound/oss/sys_timer.c @@ -50,29 +50,24 @@ tmr2ticks(int tmr_value) static void poll_def_tmr(unsigned long dummy) { + if (!opened) + return; + def_tmr.expires = (1) + jiffies; + add_timer(&def_tmr); - if (opened) - { + if (!tmr_running) + return; - { - def_tmr.expires = (1) + jiffies; - add_timer(&def_tmr); - } + spin_lock(&lock); + tmr_ctr++; + curr_ticks = ticks_offs + tmr2ticks(tmr_ctr); - if (tmr_running) - { - spin_lock(&lock); - tmr_ctr++; - curr_ticks = ticks_offs + tmr2ticks(tmr_ctr); - - if (curr_ticks >= next_event_time) - { - next_event_time = (unsigned long) -1; - sequencer_timer(0); - } - spin_unlock(&lock); - } - } + if (curr_ticks >= next_event_time) { + next_event_time = (unsigned long) -1; + sequencer_timer(0); + } + + spin_unlock(&lock); } static void

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[alsa-devel] No sound from speakers with Realtek ALC3226
by John Frankish 25 Feb '15

25 Feb '15

I posted this on the alsa-user list, but did not get a reply. Although it occasionally works, I am usually unable to get any sound out of the speakers on a dell latitude e7240 using alsa-1.0.28 and linux-3.16.6 - see details at the end of the post. 1. HDMI is the default sound device - why, the analogue device would make more sense? 2. On the analogue sound device, the headphones are the default instead of the speakers. On the few occasions I get sound out of the speakers, the headphones need to be muted first - even though no headphones are connected. 3. Shutting windows 7 64bit down with "shutdown.exe -s" seems to help with getting sound from the speakers with linux on the same machine. Most of the time, I see the error: snd_hda_intel 0000:00:1b.0: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. I've tried various module options: options snd_hda_intel probe_only=0,1 options snd-hda-intel enable_msi=1 bdl_pos_adj=1,32 [and 48 and 64] options snd-hda-intel id=PCH index=1 ..all to no avail. Any additional troubleshooting suggestions (or a patch) would be much appreciated. John ---------- $ lspci | grep -i audio 00:03.0 Audio device: Intel Corporation Haswell-ULT HD Audio Controller (rev0b) 00:1b.0 Audio device: Intel Corporation 8 Series HD Audio Controller (rev04) $ cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA Intel HDMI HDA Intel HDMI at 0xf7e34000 irq 65 1 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xf7e30000 irq 64 $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 0: ALC3226 Analog [ALC3226 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 $ cat /proc/asound/card*/codec* | grep Codec Codec: Intel Haswell HDMI Codec: Realtek ALC3226

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