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

[alsa-devel] [PATCH 1/2] ALSA: hda - Introduce bit flags to snd_hda_codec_read/write()
by Takashi Iwai 06 Jun '13

06 Jun '13

snd_hda_codec_read(), snd_hda_codec_write() & co take the argument "direct" that indicates whether the given NID is a direct reference or an indirect reference. However, the indirect reference is practically unimplemented and never exists. And moreover, we don't need the indication of indirect reference at this high level, as NID can be represented in 16bit values at this point. Meanwhile, there are some cases where it'd be nice to give some operational options to these functions. So, we can reuse this argument as a new bit flag! Pretty frugal, eh? All callers so far pass zero to this argument, thus there is no behavior change by this replacement. The real usage of this new bit option will be added in the following patches. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 41 ++++++++++++++++++++--------------------- sound/pci/hda/hda_codec.h | 8 ++++---- 2 files changed, 24 insertions(+), 25 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 679fba4..503869a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -185,20 +185,19 @@ EXPORT_SYMBOL_HDA(snd_hda_get_jack_type); * Compose a 32bit command word to be sent to the HD-audio controller */ static inline unsigned int -make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, +make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm) { u32 val; - if ((codec->addr & ~0xf) || (direct & ~1) || (nid & ~0x7f) || + if ((codec->addr & ~0xf) || (nid & ~0x7f) || (verb & ~0xfff) || (parm & ~0xffff)) { - printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x:%x\n", - codec->addr, direct, nid, verb, parm); + printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x\n", + codec->addr, nid, verb, parm); return ~0; } val = (u32)codec->addr << 28; - val |= (u32)direct << 27; val |= (u32)nid << 20; val |= verb << 8; val |= parm; @@ -209,7 +208,7 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, * Send and receive a verb */ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, - unsigned int *res) + int flags, unsigned int *res) { struct hda_bus *bus = codec->bus; int err; @@ -255,7 +254,7 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, * snd_hda_codec_read - send a command and get the response * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -264,12 +263,12 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, * Returns the obtained response value, or -1 for an error. */ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, - int direct, + int flags, unsigned int verb, unsigned int parm) { - unsigned cmd = make_codec_cmd(codec, nid, direct, verb, parm); + unsigned cmd = make_codec_cmd(codec, nid, flags, verb, parm); unsigned int res; - if (codec_exec_verb(codec, cmd, &res)) + if (codec_exec_verb(codec, cmd, flags, &res)) return -1; return res; } @@ -279,7 +278,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_read); * snd_hda_codec_write - send a single command without waiting for response * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -287,12 +286,12 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_read); * * Returns 0 if successful, or a negative error code. */ -int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, - unsigned int verb, unsigned int parm) +int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, + unsigned int verb, unsigned int parm) { - unsigned int cmd = make_codec_cmd(codec, nid, direct, verb, parm); + unsigned int cmd = make_codec_cmd(codec, nid, flags, verb, parm); unsigned int res; - return codec_exec_verb(codec, cmd, + return codec_exec_verb(codec, cmd, flags, codec->bus->sync_write ? &res : NULL); } EXPORT_SYMBOL_HDA(snd_hda_codec_write); @@ -3582,7 +3581,7 @@ EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); * snd_hda_codec_write_cache - send a single command with caching * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -3591,7 +3590,7 @@ EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); * Returns 0 if successful, or a negative error code. */ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm) + int flags, unsigned int verb, unsigned int parm) { int err; struct hda_cache_head *c; @@ -3600,7 +3599,7 @@ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, cache_only = codec->cached_write; if (!cache_only) { - err = snd_hda_codec_write(codec, nid, direct, verb, parm); + err = snd_hda_codec_write(codec, nid, flags, verb, parm); if (err < 0) return err; } @@ -3624,7 +3623,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); * snd_hda_codec_update_cache - check cache and write the cmd only when needed * @codec: the HDA codec * @nid: NID to send the command - * @direct: direct flag + * @flags: optional bit flags * @verb: the verb to send * @parm: the parameter for the verb * @@ -3635,7 +3634,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); * Returns 0 if successful, or a negative error code. */ int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm) + int flags, unsigned int verb, unsigned int parm) { struct hda_cache_head *c; u32 key; @@ -3651,7 +3650,7 @@ int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, return 0; } mutex_unlock(&codec->bus->cmd_mutex); - return snd_hda_codec_write_cache(codec, nid, direct, verb, parm); + return snd_hda_codec_write_cache(codec, nid, flags, verb, parm); } EXPORT_SYMBOL_HDA(snd_hda_codec_update_cache); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c93f902..39a658e 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -945,9 +945,9 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec); * low level functions */ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, - int direct, + int flags, unsigned int verb, unsigned int parm); -int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, +int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm); #define snd_hda_param_read(codec, nid, param) \ snd_hda_codec_read(codec, nid, 0, AC_VERB_PARAMETERS, param) @@ -986,11 +986,11 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex); /* cached write */ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm); + int flags, unsigned int verb, unsigned int parm); void snd_hda_sequence_write_cache(struct hda_codec *codec, const struct hda_verb *seq); int snd_hda_codec_update_cache(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, unsigned int parm); + int flags, unsigned int verb, unsigned int parm); void snd_hda_codec_resume_cache(struct hda_codec *codec); /* both for cmd & amp caches */ void snd_hda_codec_flush_cache(struct hda_codec *codec); -- 1.8.3

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[alsa-devel] [PATCH] ALSA: hda - Drop hard dependency on CONFIG_SND_DYNAMIC_MINORS
by Takashi Iwai 06 Jun '13

06 Jun '13

Currently HDMI codec driver sets the hard dependency (reverse selection) on CONFIG_SND_DYNAMIC_MINORS because the recent codecs may support more than two PCMs. But, this doesn't mean that we need always this option, since there can be a single PCM stream even with the modern codecs. This patch drops the hard dependency again but give more sensible error message when no enough PCMs are available due to the lack of this option. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 1 - sound/pci/hda/hda_codec.c | 19 +++++++++++++++---- 2 files changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 80a7d44..0c5371a 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -140,7 +140,6 @@ config SND_HDA_CODEC_VIA config SND_HDA_CODEC_HDMI bool "Build HDMI/DisplayPort HD-audio codec support" - select SND_DYNAMIC_MINORS default y help Say Y here to include HDMI and DisplayPort HD-audio codec diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 55108b5..679fba4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4461,12 +4461,13 @@ const char *snd_hda_pcm_type_name[HDA_PCM_NTYPES] = { /* * get the empty PCM device number to assign - * - * note the max device number is limited by HDA_MAX_PCMS, currently 10 */ -static int get_empty_pcm_device(struct hda_bus *bus, int type) +static int get_empty_pcm_device(struct hda_bus *bus, unsigned int type) { /* audio device indices; not linear to keep compatibility */ + /* assigned to static slots up to dev#10; if more needed, assign + * the later slot dynamically (when CONFIG_SND_DYNAMIC_MINORS=y) + */ static int audio_idx[HDA_PCM_NTYPES][5] = { [HDA_PCM_TYPE_AUDIO] = { 0, 2, 4, 5, -1 }, [HDA_PCM_TYPE_SPDIF] = { 1, -1 }, @@ -4480,18 +4481,28 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) return -EINVAL; } - for (i = 0; audio_idx[type][i] >= 0 ; i++) + for (i = 0; audio_idx[type][i] >= 0; i++) { +#ifndef CONFIG_SND_DYNAMIC_MINORS + if (audio_idx[type][i] >= 8) + break; +#endif if (!test_and_set_bit(audio_idx[type][i], bus->pcm_dev_bits)) return audio_idx[type][i]; + } +#ifdef CONFIG_SND_DYNAMIC_MINORS /* non-fixed slots starting from 10 */ for (i = 10; i < 32; i++) { if (!test_and_set_bit(i, bus->pcm_dev_bits)) return i; } +#endif snd_printk(KERN_WARNING "Too many %s devices\n", snd_hda_pcm_type_name[type]); +#ifndef CONFIG_SND_DYNAMIC_MINORS + snd_printk(KERN_WARNING "Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y\n"); +#endif return -EAGAIN; } -- 1.8.3

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[alsa-devel] hda question: 'Headphone Playback Switch' cannot unmute the pin's output amplifier?
by Lin, Mengdong 06 Jun '13

06 Jun '13

Hi, I'm using a Realtek HD-A codec ALC282. But the headphone output amplifier is always muted and I cannot unmute it by amixer. I tried to unmute the pin via control 'Headphone Playback Switch', but snd_hda_mixer_amp_switch_put() gets a ZERO value from ucontrol->value.integer.value, thus the pin will be muted. Could somebody give me some tips to further check this issue? Many thanks! Here are the headphone pin info: Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000001c: OUT HP Detect Pin Default 0x0321401f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c 0x0d* Here is the pin's controls listed by amixer: numid=2,iface=MIXER,name='Headphone Playback Switch' Then I use this command to unmute the pin's amplifier: amixer -c 1 cset numid=2,iface=MIXER,name='Headphone Playback Switch' on My debug message show that snd_hda_mixer_amp_switch_put() gets zero value , indicating to 'mute': [ 346.263519] snd_hda_mixer_amp_switch_put: nid 0x0021, chs 3, control value [0x00000000, 0x00000000] [ 346.263527] snd_hda_codec_amp_update: nid 0x0021 [ 346.263532] codec_amp_update: nid 0x0021, ch 0, direction 1 (OUT 1), index 0, val 0x00000080 [ 346.263536] codec_amp_update: amp val not change [ 346.263538] snd_hda_codec_amp_update: nid 0x0021 [ 346.263542] codec_amp_update: nid 0x0021, ch 1, direction 1 (OUT 1), index 0, val 0x00000080 [ 346.263544] codec_amp_update: amp val not change Thanks Mengdong

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[alsa-devel] [PATCH 0/2] Add device tree binding support to WM8962 codec driver
by Nicolin Chen 06 Jun '13

06 Jun '13

*** These two patch added devicetree binding support to WM8962 codec driver *** so that it can fetch pdata not only from old platform_data but also from *** newer dts binding. Nicolin Chen (2): ASoC: WM8962: Create default platform data structure ASoC: WM8962: Add device tree binding Documentation/devicetree/bindings/sound/wm8962.txt | 20 +++++++ sound/soc/codecs/wm8962.c | 56 ++++++++++++++++++-- 2 files changed, 72 insertions(+), 4 deletions(-)

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Re: [alsa-devel] [PATCH V5 4/5] ARM: dts: add clock provider information for i2s controllers in Exynos5250
by Padma Venkat 06 Jun '13

06 Jun '13

Hi Doug, On Tue, Jun 4, 2013 at 10:49 PM, Doug Anderson <dianders(a)chromium.org> wrote: > Padmavathi, > > On Tue, Jun 4, 2013 at 5:28 AM, Padmavathi Venna <padma.v(a)samsung.com> wrote: >> @@ -471,6 +477,8 @@ >> dmas = <&pdma1 12 >> &pdma1 11>; >> dma-names = "tx", "rx"; >> + clocks = <&clock 307>; >> + clock-names = "iis"; > > ...actually, glancing at the driver I'm a little surprised that you > don't need to list "i2s_opclk0". Did you test i2s1? I didn't test i2s1 but i2s_opclk0 is also required. I will resubmit the patch. > >> pinctrl-names = "default"; >> pinctrl-0 = <&i2s1_bus>; >> }; >> @@ -481,6 +489,8 @@ >> dmas = <&pdma0 12 >> &pdma0 11>; >> dma-names = "tx", "rx"; >> + clocks = <&clock 308>; >> + clock-names = "iis"; > > ...and here. > > > -Doug Thanks Padma

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[alsa-devel] [PATCH] ASoC: dapm: Treat DAI widgets like AIF widgets for power
by Mark Brown 06 Jun '13

06 Jun '13

Even though they are virtual widgets DAI widgets still get counted for the DAPM context power management so we can't just use the active state to check if they should be powered as they may not be part of a complete path. Instead split them into input and output widgets and do the same power checks as we perform on AIFs. Reported-by: Stephen Warren <swarren(a)nvidia.com> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- Compile tested only thus far. include/sound/soc-dapm.h | 3 ++- sound/soc/soc-dapm.c | 49 ++++++++++++++++++++++++---------------------- sound/soc/soc-pcm.c | 7 ++++++- 3 files changed, 34 insertions(+), 25 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index f4f85dd..69e8e07 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -452,7 +452,8 @@ enum snd_soc_dapm_type { snd_soc_dapm_aif_in, /* audio interface input */ snd_soc_dapm_aif_out, /* audio interface output */ snd_soc_dapm_siggen, /* signal generator */ - snd_soc_dapm_dai, /* link to DAI structure */ + snd_soc_dapm_dai_in, /* link to DAI structure */ + snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ }; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3507346..fdb14e4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -55,7 +55,8 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_micbias] = 2, [snd_soc_dapm_dai_link] = 2, - [snd_soc_dapm_dai] = 3, + [snd_soc_dapm_dai_in] = 3, + [snd_soc_dapm_dai_out] = 3, [snd_soc_dapm_aif_in] = 3, [snd_soc_dapm_aif_out] = 3, [snd_soc_dapm_mic] = 4, @@ -92,7 +93,8 @@ static int dapm_down_seq[] = { [snd_soc_dapm_value_mux] = 9, [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, - [snd_soc_dapm_dai] = 10, + [snd_soc_dapm_dai_in] = 10, + [snd_soc_dapm_dai_out] = 10, [snd_soc_dapm_dai_link] = 11, [snd_soc_dapm_clock_supply] = 12, [snd_soc_dapm_regulator_supply] = 12, @@ -419,7 +421,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: case snd_soc_dapm_hp: case snd_soc_dapm_mic: case snd_soc_dapm_spk: @@ -813,7 +816,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_out: if (widget->active) { widget->outputs = snd_soc_dapm_suspend_check(widget); return widget->outputs; @@ -909,7 +912,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: if (widget->active) { widget->inputs = snd_soc_dapm_suspend_check(widget); return widget->inputs; @@ -1128,16 +1131,6 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) return out != 0 && in != 0; } -static int dapm_dai_check_power(struct snd_soc_dapm_widget *w) -{ - DAPM_UPDATE_STAT(w, power_checks); - - if (w->active) - return w->active; - - return dapm_generic_check_power(w); -} - /* Check to see if an ADC has power */ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) { @@ -2327,7 +2320,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: case snd_soc_dapm_dai_link: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); @@ -3134,10 +3128,12 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: + case snd_soc_dapm_dai_out: w->power_check = dapm_adc_check_power; break; case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: + case snd_soc_dapm_dai_in: w->power_check = dapm_dac_check_power; break; case snd_soc_dapm_pga: @@ -3157,9 +3153,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_clock_supply: w->power_check = dapm_supply_check_power; break; - case snd_soc_dapm_dai: - w->power_check = dapm_dai_check_power; - break; default: w->power_check = dapm_always_on_check_power; break; @@ -3380,7 +3373,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.reg = SND_SOC_NOPM; if (dai->driver->playback.stream_name) { - template.id = snd_soc_dapm_dai; + template.id = snd_soc_dapm_dai_in; template.name = dai->driver->playback.stream_name; template.sname = dai->driver->playback.stream_name; @@ -3398,7 +3391,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, } if (dai->driver->capture.stream_name) { - template.id = snd_soc_dapm_dai; + template.id = snd_soc_dapm_dai_out; template.name = dai->driver->capture.stream_name; template.sname = dai->driver->capture.stream_name; @@ -3428,8 +3421,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) /* For each DAI widget... */ list_for_each_entry(dai_w, &card->widgets, list) { - if (dai_w->id != snd_soc_dapm_dai) + switch (w->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + break; + default: continue; + } dai = dai_w->priv; @@ -3438,8 +3436,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) if (w->dapm != dai_w->dapm) continue; - if (w->id == snd_soc_dapm_dai) + switch (w->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: continue; + default: + break; + } if (!w->sname) continue; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a14335e..b6c6403 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -931,8 +931,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, /* Create any new FE <--> BE connections */ for (i = 0; i < list->num_widgets; i++) { - if (list->widgets[i]->id != snd_soc_dapm_dai) + switch (list->widgets[i]->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + break; + default: continue; + } /* is there a valid BE rtd for this widget */ be = dpcm_get_be(card, list->widgets[i], stream); -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: max98090: Guard runtime PM callbacks
by Mark Brown 05 Jun '13

05 Jun '13

Otherwise the functions will be defined but unreferenced when runtime PM is disabled, generating warnings. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max98090.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 1bf96bd..ad5313f 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2350,6 +2350,7 @@ static int max98090_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME static int max98090_runtime_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); @@ -2369,6 +2370,7 @@ static int max98090_runtime_suspend(struct device *dev) return 0; } +#endif static const struct dev_pm_ops max98090_pm = { SET_RUNTIME_PM_OPS(max98090_runtime_suspend, -- 1.7.10.4

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[alsa-devel] [PATCH (alsa-lib)] chmap: Always succeed setting the map to what it already is
by David Henningsson 05 Jun '13

05 Jun '13

If we try to set the channel map to what it already is, this should always succeed. E g, speaker-test can do this sometimes. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/pcm/pcm.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 7ec1f0e..ca4d416 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -7361,6 +7361,13 @@ OBSOLETE1(snd_pcm_sw_params_get_silence_size, ALSA_0.9, ALSA_0.9.0rc4); #endif /* DOC_HIDDEN */ +static int chmap_equal(const snd_pcm_chmap_t *a, const snd_pcm_chmap_t *b) +{ + if (a->channels != b->channels) + return 0; + return !memcmp(a->pos, b->pos, a->channels * sizeof(a->pos[0])); +} + /** * \!brief Query the available channel maps * \param pcm PCM handle to query @@ -7415,6 +7422,10 @@ snd_pcm_chmap_t *snd_pcm_get_chmap(snd_pcm_t *pcm) */ int snd_pcm_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map) { + const snd_pcm_chmap_t *oldmap = snd_pcm_get_chmap(pcm); + if (oldmap && chmap_equal(oldmap, map)) + return 0; + if (!pcm->ops->set_chmap) return -ENXIO; return pcm->ops->set_chmap(pcm, map); -- 1.7.9.5

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Re: [alsa-devel] CONFIG_SND_DEBUG
by David Henningsson 05 Jun '13

05 Jun '13

On 06/05/2013 12:53 PM, Andy Whitcroft wrote: > On Wed, Jun 05, 2013 at 09:47:30AM +0200, David Henningsson wrote: >>>>> On Wed, 2013-06-05 at 08:54 +0200, Takashi Iwai wrote: >>>>>> At Tue, 04 Jun 2013 23:52:01 -0700, Joe Perches wrote: >>>>>>> On Wed, 2013-06-05 at 08:32 +0200, Takashi Iwai wrote: >>>>>>>> Yes. These are snd_printd() just to be conditionally built in. >>>>>>>> But in most cases it's rather useful to print them (as most distros >>>>>>>> set CONFIG_SND_DEBUG=y). >>>>>>> Ubuntu doesn't, I believe Fedora doesn't. >>>>>> Then they should have done so :) >>>>> But they don't, so what distros do? >>>> >>>> RedHat (including Fedora) and SUSE do at least. >>> >>> Mandriva does too. (still looking around for others) >>> >>> We can ask Ubuntu to enable CONFIG_SND_DEBUG. >>> (cc'd Andy Whitcroft, Leann Ogasawara and David Henningsson) >>> >>> Maybe there are others Canonical folk that >>> should be cc'd? >> >> Adding kernel team mailing list to CC. >> >> Andy/Leann - apparently CONFIG_SND_DEBUG is on by default from >> upstream, and we explicitly disable it. Is there any reason why we >> do that? > > config SND_DEBUG > bool "Debug" > help > Say Y here to enable ALSA debug code. > > It is off by default in upstream, and the really helpful description > would cirtainly tend to lead to it being disabled. But if it is helpful > to your debugging efforts David then I suspect we can enable it in Saucy > and see what happens. Okay, so then the ball is back in Takashi's area - if we're recommended to turn CONFIG_SND_DEBUG on, why is it off by default in the upstream Linux kernel? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [PATCH 1/2] ASoC: arizona: Correct AEC loopback enable
by Charles Keepax 05 Jun '13

05 Jun '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 3 ++- sound/soc/codecs/wm5110.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index e895d39..100fdad 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1120,7 +1120,8 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5102_aec_loopback_mux), SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index ba38f06..88ad7db 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -503,7 +503,8 @@ SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5110_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5110_aec_loopback_mux), SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), -- 1.7.2.5

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