September 2012 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH] ALSA: fix possible memory leak in snd_mixer_oss_build_input()
by Takashi Iwai 03 Sep '12

03 Sep '12

At Sun, 2 Sep 2012 22:10:27 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > uinfo has been allocated in this function and should be > freed before leaving from the error handling cases. > > spatch with a semantic match is used to found this problem. > (http://coccinelle.lip6.fr/) > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Applied now. Thanks. Takashi > --- > sound/core/oss/mixer_oss.c | 2 ++ > 1 file changed, 2 insertions(+) > > diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c > index 18297f7..29f6ded 100644 > --- a/sound/core/oss/mixer_oss.c > +++ b/sound/core/oss/mixer_oss.c > @@ -1046,6 +1046,7 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, struct snd_mix > > if (kctl->info(kctl, uinfo)) { > up_read(&mixer->card->controls_rwsem); > + kfree(uinfo); > return 0; > } > strcpy(str, ptr->name); > @@ -1061,6 +1062,7 @@ static int snd_mixer_oss_build_input(struct snd_mixer_oss *mixer, struct snd_mix > uinfo->value.enumerated.item = slot.capture_item; > if (kctl->info(kctl, uinfo)) { > up_read(&mixer->card->controls_rwsem); > + kfree(uinfo); > return 0; > } > if (!strcmp(uinfo->value.enumerated.name, str)) { > >

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[alsa-devel] [PATCH] ALSA: hda - Fix runtime PM accounting
by Takashi Iwai 02 Sep '12

02 Sep '12

The flag codec->d3_clk_stop_ok isn't always set when the codec is powered off. If it's not set, the controller driver doesn't change the runtime PM count. This causes an unblanced accounting when multiple codecs are present. Usually, the first codec is powered down with CLK_STOP_OK=0, then the next codec with CLK_STOP_OK=1. At this moment, the first codec is changed also to CLK_STOP_OK=1, but this isn't informed to the controller. Thus it still thinks as if one codec is active. For fixing this issue, move the check of CLK_STOP_OK bit at the runtime suspend time, and simple up/down by the codec power save in azx_power_notifier(). For CLK_STOP_OK bit checks of all codecs, a new helper function is introduced. With this change, d3_clk_stop_ok field is no longer referred, thus removed. Also, d3_clk_stop bit field is moved to a place where more efficiently packed. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- I faced this issue while I'm checking the pm issue. Mengdong, could you verify whether this change is OK? sound/pci/hda/hda_codec.c | 30 ++++++++++++++++++++---------- sound/pci/hda/hda_codec.h | 6 +++--- sound/pci/hda/hda_intel.c | 4 ++-- 3 files changed, 25 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4a2f35c..b7ac1ff 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3546,10 +3546,6 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; -#ifdef CONFIG_SND_HDA_POWER_SAVE - codec->d3_stop_clk_ok = 0; -#endif - if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; @@ -3572,12 +3568,6 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, if (!(state & AC_PWRST_ERROR)) break; } - -#ifdef CONFIG_SND_HDA_POWER_SAVE - if ((power_state == AC_PWRST_D3) - && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) - codec->d3_stop_clk_ok = 1; -#endif } #ifdef CONFIG_SND_HDA_HWDEP @@ -4532,6 +4522,26 @@ void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait) } EXPORT_SYMBOL_HDA(snd_hda_power_save); +/* check whether all codecs are ready to stop clock */ +bool snd_hda_bus_ready_for_d3(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + unsigned int state; + int fg = codec->afg ? codec->afg : codec->mfg; + + if (!codec->d3_stop_clk) + return false; + state = snd_hda_codec_read(codec, fg, 0, + AC_VERB_GET_POWER_STATE, 0); + if (!(state & AC_PWRST_CLK_STOP_OK)) + return false; + } + return true; +} +EXPORT_SYMBOL_HDA(snd_hda_bus_ready_for_d3); + /** * snd_hda_check_amp_list_power - Check the amp list and update the power * @codec: HD-audio codec diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 13c834f..abd85e5 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -868,6 +868,7 @@ struct hda_codec { unsigned int pcm_format_first:1; /* PCM format must be set first */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ + unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ @@ -875,9 +876,6 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; - - unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ - unsigned int d3_stop_clk_ok:1; /* BCLK can stop */ #endif /* codec-specific additional proc output */ @@ -1068,9 +1066,11 @@ const char *snd_hda_get_jack_location(u32 cfg); #ifdef CONFIG_SND_HDA_POWER_SAVE void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait); void snd_hda_update_power_acct(struct hda_codec *codec); +bool snd_hda_bus_ready_for_d3(struct hda_bus *bus); #else static inline void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait) {} +static inline bool snd_hda_bus_ready_for_d3(struct hda_bus *bus) { return false; } #endif /** diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1c9c779..9b02174 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2410,7 +2410,7 @@ static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec) { struct azx *chip = bus->private_data; - if (bus->power_keep_link_on || !codec->d3_stop_clk_ok) + if (bus->power_keep_link_on) return; if (codec->power_on) @@ -2529,7 +2529,7 @@ static int azx_runtime_suspend(struct device *dev) struct azx *chip = card->private_data; #ifdef CONFIG_SND_HDA_POWER_SAVE - if (!power_save_controller) + if (!power_save_controller || !snd_hda_bus_ready_for_d3(chip->bus)) return -EAGAIN; #endif -- 1.7.11.4

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[alsa-devel] [PATCH] ASoC: dapm: Ensure bypass paths are suspended and resumed
by Mark Brown 01 Sep '12

01 Sep '12

Since bypass paths aren't part of DAPM streams and we may not have any DAPM streams there may not be anything that triggers a DAPM sync for them. Mark all input and output widgets as dirty and then sync to do so at the end of suspend and resume. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 1 + sound/soc/soc-core.c | 8 ++++++++ sound/soc/soc-dapm.c | 22 ++++++++++++++++++++++ 3 files changed, 31 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index abe373d..5b584cb 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -412,6 +412,7 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec); /* Mostly internal - should not normally be used */ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason); +void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 01ec2d6..3be2e40 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -609,6 +609,10 @@ int snd_soc_suspend(struct device *dev) SND_SOC_DAPM_STREAM_SUSPEND); } + /* Recheck all analogue paths too */ + dapm_mark_io_dirty(&card->dapm); + snd_soc_dapm_sync(&card->dapm); + /* suspend all CODECs */ list_for_each_entry(codec, &card->codec_dev_list, card_list) { /* If there are paths active then the CODEC will be held with @@ -756,6 +760,10 @@ static void soc_resume_deferred(struct work_struct *work) /* userspace can access us now we are back as we were before */ snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0); + + /* Recheck all analogue paths too */ + dapm_mark_io_dirty(&card->dapm); + snd_soc_dapm_sync(&card->dapm); } /* powers up audio subsystem after a suspend */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 53f69b5..fd97dd7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -141,6 +141,28 @@ void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) } EXPORT_SYMBOL_GPL(dapm_mark_dirty); +void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm) +{ + struct snd_soc_card *card = dapm->card; + struct snd_soc_dapm_widget *w; + + mutex_lock(&card->dapm_mutex); + + list_for_each_entry(w, &card->widgets, list) { + switch (w->id) { + case snd_soc_dapm_input: + case snd_soc_dapm_output: + dapm_mark_dirty(w, "Rechecking inputs and outputs"); + break; + default: + break; + } + } + + mutex_unlock(&card->dapm_mutex); +} +EXPORT_SYMBOL_GPL(dapm_mark_io_dirty); + /* create a new dapm widget */ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( const struct snd_soc_dapm_widget *_widget) -- 1.7.10.4

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Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Regression fixes for 3.5
by Daniel Mack 01 Sep '12

01 Sep '12

(please don't drop mailing lists from Cc:) On 31.08.2012 18:18, ☈king wrote: > On 08/31/2012 12:15 PM, Daniel Mack wrote: >> What kernel do you work on? Did you try the other patch that I sent >> some days ago and that already went mainline? >> https://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commit;h=01561… > > Hrm. > > I am using Gentoo's kernel from gentoo-sources. When I wrote the other > mail I was on 3.5.1, but now I'm using 3.5.3, and when I apply that > patch, I get: > > 1 out of 5 hunks FAILED -- saving rejects to file sound/usb/pcm.c.rej > % cat sound/usb/pcm.c.rej > --- sound/usb/pcm.c > +++ sound/usb/pcm.c > @@ -544,13 +544,10 @@ > subs->last_frame_number = 0; > runtime->delay = 0; > - /* clear the pending deactivation on the target EPs */ > - deactivate_endpoints(subs); > - > /* for playback, submit the URBs now; otherwise, the first hwptr_done > * updates for all URBs would happen at the same time when starting */ > if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) > - return start_endpoints(subs); > + return start_endpoints(subs, 1); > return 0; > } > > > It already doesn't have that deactivate_endpoints() line. > > So, I manually adding the 1 arg to start_endpoints() for "can_sleep". The important thing is not to call deactivate_endpoints() from the prepare callback. > Rebooting to this kernel gives the exact same results, though. Please try a kernel built from Takashi's branch: https://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git That alone should fix your problem. The three patches of this series fix some occasional stream dropouts. Thanks for testing, Daniel

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[alsa-devel] [PATCH] ASoC: samsung dma - Don't indicate support for pause/resume.
by Dylan Reid 01 Sep '12

01 Sep '12

The pause and resume operations indicate that the stream can be un-paused/resumed from the exact location they were paused/suspended. This is not true for this driver, the pause and suspend triggers share the same code path with stop, they flush all pending DMA transfers. This drops all pending samples. The pause_release/resume triggers are the same as start, except that prepare won't be called beforehand, nothing will be enqueued to the DMA engine and nothing will happen (no audio). Removing the pause flag will let apps know that it isn't supported. Removing the resume flag will cause user space to call prepare and start instead of resume, so audio will continue playing when the system wakes up. Before removing the pause and resume flags, I tested this on an exynos 5250, using 'aplay -i'. Pause/un-pause leads to silence followed by a write error. Suspend/resume testing led to the same result. Removing the two flags fixes suspend/resume (since snd_pcm_prepare is called again). And leads to a proper reporting of pause not supported. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/samsung/dma.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index f3ebc38..56a507a 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -34,9 +34,7 @@ static const struct snd_pcm_hardware dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_MMAP_VALID, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U8 | -- 1.7.12.146.g16d26b1

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Re: [alsa-devel] no 5.1 surround sound on 3stack-6ch motherboard
by Raymond Yau 01 Sep '12

01 Sep '12

> https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1044150 > > you need to switch channel mode from 2ch to 6ch and restore this > setting by alsactl before pulseaudio start > > Simple mixer control 'Channel Mode',0 > Capabilities: enum > Items: '2ch' '4ch' '6ch' > Item0: '2ch > > but how can i do this?? amixer -c0 set "Channel Mode" "6ch" alsactl store restart pulseaudio server http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=blob_plain;f=Do… Realtek codecs -------------- * Channel Mode This is an enum control to change the surround-channel setup, appears only when the surround channels are available. It gives the number of channels to be used, "2ch", "4ch", "6ch", and "8ch". According to the configuration, this also controls the jack-retasking of multi-I/O jacks. can you post the pulseaudio log ? I don't understand how can pulseaudio server switch to active profile surround51 when blue and pink jacks had not been retasked as output ( Pin-ctls are still IN ) and channel mode is still 2ch. analog-input rear mic and line-in ports should not be available too. http://www.intel.com/support/motherboards/desktop/sb/cs-020642.htm#6ch Jack retasking is a feature that allows an audio jack to support more than one function. For example, the pink audio jack can support microphone input when the system configuration is set to Stereo Mode, and as Center/LFE or Rear Surround (depending on the audio solution used) when set as 6 channel. Node 0x18 [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Rear Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00003736: IN OUT Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19c50: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=03, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00003736: IN OUT Detect Trigger Vref caps: HIZ 50 GRD 80 100 Pin Default 0x0181345f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x5, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=05, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 index: 1 name: <alsa_card.pci-0000_00_1b.0> driver: <module-alsa-card.c> owner module: 5 properties: alsa.card = "0" alsa.card_name = "HDA Intel PCH" alsa.long_card_name = "HDA Intel PCH at 0xf7d00000 irq 44" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:00:1b.0" sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" device.bus = "pci" device.vendor.id = "8086" device.vendor.name = "Intel Corporation" device.product.name = "6 Series/C200 Series Chipset Family High Definition Audio Controller" device.form_factor = "internal" device.string = "0" device.description = "Built-in Audio" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" profiles: output:analog-stereo: Analogue Stereo Output (priority 1000) output:analog-stereo+input:analog-stereo: Analogue Stereo Duplex (priority 1010) output:analog-surround-40: Analogue Surround 4.0 Output (priority 700) output:analog-surround-40+input:analog-stereo: Analogue Surround 4.0 Output + Analogue Stereo Input (priority 710) output:analog-surround-41: Analogue Surround 4.1 Output (priority 800) output:analog-surround-41+input:analog-stereo: Analogue Surround 4.1 Output + Analogue Stereo Input (priority 810) output:analog-surround-50: Analogue Surround 5.0 Output (priority 700) output:analog-surround-50+input:analog-stereo: Analogue Surround 5.0 Output + Analogue Stereo Input (priority 710) output:analog-surround-51: Analogue Surround 5.1 Output (priority 3800) output:analog-surround-51+input:analog-stereo: Analogue Surround 5.1 Output + Analogue Stereo Input (priority 3810) input:analog-stereo: Analogue Stereo Input (priority 10) off: Off (priority 0) active profile: <output:analog-surround-51+input:analog-stereo> sinks: alsa_output.pci-0000_00_1b.0.analog-surround-51/#0: Built-in Audio Analogue Surround 5.1 sources: alsa_output.pci-0000_00_1b.0.analog-surround-51.monitor/#1: Monitor of Built-in Audio Analogue Surround 5.1 alsa_input.pci-0000_00_1b.0.analog-stereo/#2: Built-in Audio Analogue Stereo ports: analog-output: Analogue Output (priority 9900, available: unknown) properties: analog-output-headphones: Headphones (priority 9000, available: unknown) properties: analog-input-microphone-front: Front Microphone (priority 8500, available: no) properties: analog-input-microphone-rear: Rear Microphone (priority 8200, available: yes) properties: analog-input-linein: Line In (priority 8100, available: yes) properties:

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