September 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda: hdmi: Hint matching between input devices and pcm devices
by David Henningsson 21 Sep '11

21 Sep '11

Hi Takashi, You went on vacation when this patch was posted and discussed, but would you mind reviewing/committing this patch now that you're back? It was reviewed by Stephen Warren who thought the patch was okay, but had some thoughts on doing other changes to the format of that string at the same time [1]. I had to distro patch a 3.0 adapted version of this patch into Ubuntu in order to meet the Ubuntu 11.10 kernel deadline, so for that reason I'd say that the ",pcm=x" format would be preferrable for me, and while I don't mind further improvements to the string format in the future, this would be a step in the right direction and IMO this patch should not block on such future improvements. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-August/043258.html

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Re: [alsa-devel] [PATCH 4/4] ASoC: Staticise nuc900_dma_getposition()
by Wan ZongShun 21 Sep '11

21 Sep '11

2011/9/21 Axel Lin <axel.lin(a)gmail.com>: > It is not used outside this driver so no need to make the symbol global. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/nuc900/nuc900-pcm.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c > index e46d551..4e3626b 100644 > --- a/sound/soc/nuc900/nuc900-pcm.c > +++ b/sound/soc/nuc900/nuc900-pcm.c > @@ -227,7 +227,7 @@ static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd) > return ret; > } > > -int nuc900_dma_getposition(struct snd_pcm_substream *substream, > +static int nuc900_dma_getposition(struct snd_pcm_substream *substream, > dma_addr_t *src, dma_addr_t *dst) > { > struct snd_pcm_runtime *runtime = substream->runtime; > -- > 1.7.4.1 > thanks! Acked- by: Wan Zongshun <mcuos.com(a)gmail.com> > > > -- Wan ZongShun. www.mcuos.com

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[alsa-devel] [RFC] Remove non-exist device and control from snd-ymfpci
by Raymond Yau 21 Sep '11

21 Sep '11

My ymf724f does not have spdif, only with ac97 stac9704 Is it feasible to remove playback device 2 (rear) and control (e.g. 4 ch Duplication) by checking sdac bit of ac97 registers to know that it does not support 4 channels ? How about device 1 spdif since this card does not have spdif connector ? what is the difference of capture device 0 and device 3 ? !!PCI Soundcards installed in the system !!-------------------------------------- 04:02.0 Multimedia audio controller: Yamaha Corporation YMF-724F [DS-1 Audio Controller] (rev 03) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 04:02.0 0401: 1073:000d (rev 03) Subsystem: 1073:000d Simple mixer control 'IEC958',0 Capabilities: cvolume pswitch pswitch-joined cswitch cswitch-joined penum Playback channels: Mono Capture channels: Front Left - Front Right Limits: Capture 0 - 16383 Mono: Playback [off] Front Left: Capture 0 [0%] [-99999.99dB] [off] Front Right: Capture 0 [0%] [-99999.99dB] [off] Simple mixer control 'IEC958 AC97',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 16383 Mono: Front Left: Playback 16383 [100%] [0.00dB] Front Right: Playback 16383 [100%] [0.00dB] Simple mixer control 'IEC958 AC97',1 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 16383 Mono: Front Left: Playback 0 [0%] [-99999.99dB] Front Right: Playback 0 [0%] [-99999.99dB] Simple mixer control 'IEC958 Loop',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958',1 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 16383 Front Left: Capture 0 [0%] [-99999.99dB] Front Right: Capture 0 [0%] [-99999.99dB] Simple mixer control '4ch Duplication',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'ADC',0 Capabilities: pvolume cvolume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 16383 Capture 0 - 16383 Front Left: Playback 0 [0%] [-99999.99dB] Capture 16383 [100%] [0.00dB] Front Right: Playback 0 [0%] [-99999.99dB] Capture 16383 [100%] [0.00dB] Simple mixer control 'ADC',1 Capabilities: pvolume cvolume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 16383 Capture 0 - 16383 Front Left: Playback 0 [0%] [-99999.99dB] Capture 0 [0%] [-99999.99dB] Front Right: Playback 0 [0%] [-99999.99dB] Capture 0 [0%] [-99999.99dB] Simple mixer control 'Digital',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 16383 Front Left: Capture 16383 [100%] [0.00dB] Front Right: Capture 16383 [100%] [0.00dB] Simple mixer control 'Digital',1 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 16383 Front Left: Capture 16383 [100%] [0.00dB] Front Right: Capture 16383 [100%] [0.00dB] **** List of PLAYBACK Hardware Devices **** card 0: YMF724F [Yamaha DS-1 (YMF724F)], device 0: YMFPCI [YMFPCI] Subdevices: 32/32 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 Subdevice #8: subdevice #8 Subdevice #9: subdevice #9 Subdevice #10: subdevice #10 Subdevice #11: subdevice #11 Subdevice #12: subdevice #12 Subdevice #13: subdevice #13 Subdevice #14: subdevice #14 Subdevice #15: subdevice #15 Subdevice #16: subdevice #16 Subdevice #17: subdevice #17 Subdevice #18: subdevice #18 Subdevice #19: subdevice #19 Subdevice #20: subdevice #20 Subdevice #21: subdevice #21 Subdevice #22: subdevice #22 Subdevice #23: subdevice #23 Subdevice #24: subdevice #24 Subdevice #25: subdevice #25 Subdevice #26: subdevice #26 Subdevice #27: subdevice #27 Subdevice #28: subdevice #28 Subdevice #29: subdevice #29 Subdevice #30: subdevice #30 Subdevice #31: subdevice #31 card 0: YMF724F [Yamaha DS-1 (YMF724F)], device 1: YMFPCI - IEC958 [YMFPCI - IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: YMF724F [Yamaha DS-1 (YMF724F)], device 2: YMFPCI - Rear [YMFPCI - Rear PCM] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: YMF724F [Yamaha DS-1 (YMF724F)], device 0: YMFPCI [YMFPCI] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: YMF724F [Yamaha DS-1 (YMF724F)], device 3: YMFPCI - PCM2 [YMFPCI - AC'97] Subdevices: 1/1 Subdevice #0: subdevice #0 !!AC97 Codec information !!--------------------------- 0-0/0: SigmaTel STAC9704 PCI Subsys Vendor: 0x0000 PCI Subsys Device: 0x0000 Flags: 0 Capabilities : DAC resolution : 18-bit ADC resolution : 18-bit 3D enhancement : SigmaTel 3D Enhancement Current setup Mic gain : +0dB [+0dB] POP path : pre 3D Sim. stereo : off 3D enhancement : off Loudness : off Mono output : MIX Mic select : Mic1 ADC/DAC loopback : off 0:00 = 6940 0:02 = 1313 0:04 = 8000 0:06 = 801f 0:08 = ffff 0:0a = 9f1f 0:0c = 801f 0:0e = 801f 0:10 = 9f1f 0:12 = 9f1f 0:14 = 9f1f 0:16 = 9f1f 0:18 = 0808 0:1a = 9818 0:1c = 0808 0:1e = 0000 0:20 = 0000 0:22 = 0000 0:24 = 0000 0:26 = 000f 0:28 = ffff 0:2a = ffff 0:2c = ffff 0:2e = ffff 0:30 = ffff 0:32 = ffff 0:34 = ffff 0:36 = ffff 0:38 = ffff 0:3a = ffff 0:3c = 0000 0:3e = 0000 0:40 = 0000 0:42 = 0000 0:44 = 0000 0:46 = 0000 0:48 = 0000 0:4a = 0000 0:4c = 0000 0:4e = 0000 0:50 = 0000 0:52 = 0000 0:54 = 0000 0:56 = 0000 0:58 = 0000 0:5a = 0000 0:5c = 0000 0:5e = 0000 0:60 = 0000 0:62 = 0000 0:64 = 0000 0:66 = 0000 0:68 = 0000 0:6a = 0000 0:6c = 0000 0:6e = 0000 0:70 = ffff 0:72 = ffff 0:74 = ffff 0:76 = ffff 0:78 = ffff 0:7a = ffff 0:7c = 8384 0:7e = 7605

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[alsa-devel] [RFC PATCH] ALSA: hdmi: expose ELD control
by Pierre-Louis Bossart 21 Sep '11

21 Sep '11

Applications may want to read ELD information to understand what codecs are supported on the HDMI receiver and handle the a-v delay for better lip-sync. ELD information is exposed in a BYTES control named "ELD <device name>". Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_eld.c | 9 ++-- sound/pci/hda/hda_local.h | 2 + sound/pci/hda/patch_hdmi.c | 91 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 97 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 28ce17d..8f16b20 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -335,21 +335,20 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n"); size = 128; } - if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) { + if (size < ELD_FIXED_BYTES || size > ELD_MAX_SIZE || size > PAGE_SIZE) { snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size); return -ERANGE; } - buf = kmalloc(size, GFP_KERNEL); - if (!buf) - return -ENOMEM; + /* update info */ + eld->eld_size = size; + buf = eld->eld_buffer; for (i = 0; i < size; i++) buf[i] = hdmi_get_eld_byte(codec, nid, i); ret = hdmi_update_eld(eld, buf, size); - kfree(buf); return ret; } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 9ed4b0d..cd579a6 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -609,6 +609,7 @@ struct cea_sad { }; #define ELD_FIXED_BYTES 20 +#define ELD_MAX_SIZE 256 #define ELD_MAX_MNL 16 #define ELD_MAX_SAD 16 @@ -633,6 +634,7 @@ struct hdmi_eld { int spk_alloc; int sad_count; struct cea_sad sad[ELD_MAX_SAD]; + char eld_buffer[ELD_MAX_SIZE]; #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; #endif diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 19cb72d..a5b7245 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -324,6 +324,89 @@ static int cvt_nid_to_cvt_index(struct hdmi_spec *spec, hda_nid_t cvt_nid) return -EINVAL; } +static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + int pin_nid; + int pin_idx; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + + pin_nid = kcontrol->private_value; + spec = codec->spec; + + pin_idx = pin_nid_to_pin_index(spec, pin_nid); + if (pin_idx < 0) { + uinfo->count = 0; + } else { + per_pin = &spec->pins[pin_idx]; + eld = &per_pin->sink_eld; + uinfo->count = eld->eld_size; + } + return 0; +} + +static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + int pin_nid; + int pin_idx; + + spec = codec->spec; + pin_nid = kcontrol->private_value; + + pin_idx = pin_nid_to_pin_index(spec, pin_nid); + if (pin_idx < 0) { + memset(ucontrol->value.bytes.data, 0, ELD_MAX_SIZE); + } else { + per_pin = &spec->pins[pin_idx]; + eld = &per_pin->sink_eld; + memcpy(ucontrol->value.bytes.data, eld->eld_buffer, + (eld->eld_size > ELD_MAX_SIZE) ? + ELD_MAX_SIZE : eld->eld_size); + } + return 0; +} + +static struct snd_kcontrol_new eld_bytes_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ELD ", + .info = hdmi_eld_ctl_info, + .get = hdmi_eld_ctl_get, +}; + +static int hdmi_create_eld_bytes_ctl(struct hda_codec *codec, hda_nid_t nid, + char *name) +{ + struct snd_kcontrol *kctl; + struct hdmi_spec *spec = codec->spec; + int pin_idx; + int err; + + pin_idx = pin_nid_to_pin_index(spec, nid); + if (pin_idx < 0) + return -EINVAL; + + kctl = snd_ctl_new1(&eld_bytes_ctl, codec); + if (!kctl) + return -ENOMEM; + err = snd_hda_ctl_add(codec, nid, kctl); + if (err < 0) + return err; + kctl->private_value = nid; + strcat(kctl->id.name, name); + + return 0; +} + #ifdef BE_PARANOID static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, int *packet_index, int *byte_index) @@ -1145,6 +1228,7 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hda_pcm *info; struct hda_pcm_stream *pstr; + int ret; info = &spec->pcm_rec[pin_idx]; info->name = generic_hdmi_pcm_names[pin_idx]; @@ -1154,6 +1238,13 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) pstr->substreams = 1; pstr->ops = generic_ops; /* other pstr fields are set in open */ + + /* add control for ELD Bytes */ + ret = hdmi_create_eld_bytes_ctl(codec, + spec->pins[pin_idx].pin_nid, + info->name); + if (ret < 0) + return ret; } codec->num_pcms = spec->num_pins; -- 1.7.6.2

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[PATCH] ALSA: hdmi: expose ELD control
by Pierre-Louis Bossart 21 Sep '11

21 Sep '11

Applications may want to read ELD information to understand what codecs are supported on the HDMI receiver and handle the a-v delay for better lip-sync. ELD information is exposed in a device-specific IFACE_PCM kcontrol. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_eld.c | 9 ++--- sound/pci/hda/hda_local.h | 2 + sound/pci/hda/patch_hdmi.c | 85 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 91 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 28ce17d..8f16b20 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -335,21 +335,20 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n"); size = 128; } - if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) { + if (size < ELD_FIXED_BYTES || size > ELD_MAX_SIZE || size > PAGE_SIZE) { snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size); return -ERANGE; } - buf = kmalloc(size, GFP_KERNEL); - if (!buf) - return -ENOMEM; + /* update info */ + eld->eld_size = size; + buf = eld->eld_buffer; for (i = 0; i < size; i++) buf[i] = hdmi_get_eld_byte(codec, nid, i); ret = hdmi_update_eld(eld, buf, size); - kfree(buf); return ret; } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 9ed4b0d..cd579a6 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -609,6 +609,7 @@ struct cea_sad { }; #define ELD_FIXED_BYTES 20 +#define ELD_MAX_SIZE 256 #define ELD_MAX_MNL 16 #define ELD_MAX_SAD 16 @@ -633,6 +634,7 @@ struct hdmi_eld { int spk_alloc; int sad_count; struct cea_sad sad[ELD_MAX_SAD]; + char eld_buffer[ELD_MAX_SIZE]; #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; #endif diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 19cb72d..5ea9b55 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -324,6 +324,83 @@ static int cvt_nid_to_cvt_index(struct hdmi_spec *spec, hda_nid_t cvt_nid) return -EINVAL; } +static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + int pin_idx; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + pin_idx = kcontrol->private_value; + spec = codec->spec; + + if (pin_idx < 0) { + uinfo->count = 0; + } else { + per_pin = &spec->pins[pin_idx]; + eld = &per_pin->sink_eld; + uinfo->count = eld->eld_size; + } + return 0; +} + +static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + int pin_idx; + + spec = codec->spec; + pin_idx = kcontrol->private_value; + + if (pin_idx < 0) { + memset(ucontrol->value.bytes.data, 0, ELD_MAX_SIZE); + } else { + per_pin = &spec->pins[pin_idx]; + eld = &per_pin->sink_eld; + memcpy(ucontrol->value.bytes.data, eld->eld_buffer, + (eld->eld_size > ELD_MAX_SIZE) ? + ELD_MAX_SIZE : eld->eld_size); + } + return 0; +} + +static struct snd_kcontrol_new eld_bytes_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdmi_eld_ctl_info, + .get = hdmi_eld_ctl_get, +}; + +static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx, + int device) +{ + struct snd_kcontrol *kctl; + struct hdmi_spec *spec = codec->spec; + int err; + + if (pin_idx < 0) + return -EINVAL; + + kctl = snd_ctl_new1(&eld_bytes_ctl, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = pin_idx; + kctl->id.device = device; + + err = snd_hda_ctl_add(codec, spec->pins[pin_idx].pin_nid, kctl); + if (err < 0) + return err; + + return 0; +} + #ifdef BE_PARANOID static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, int *packet_index, int *byte_index) @@ -1176,6 +1253,14 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) if (err < 0) return err; snd_hda_spdif_ctls_unassign(codec, pin_idx); + + /* add control for ELD Bytes */ + err = hdmi_create_eld_ctl(codec, + pin_idx, + spec->pcm_rec[pin_idx].device); + + if (err < 0) + return err; } return 0; -- 1.7.6.2 --=-TOAARVHGJMGbcf5NX2ZN--

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[alsa-devel] [PATCH] ASoC: Factor out per-widget DAPM power checks
by Mark Brown 20 Sep '11

20 Sep '11

The indentation is getting a little deep. Should be straight code motion, no functional changes. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 112 +++++++++++++++++++++++++++----------------------- 1 files changed, 60 insertions(+), 52 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6a1e13e..84d1d79 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1191,6 +1191,65 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) } } +static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, + struct list_head *up_list, + struct list_head *down_list) +{ + struct snd_soc_dapm_context *d; + int power; + + switch (w->id) { + case snd_soc_dapm_pre: + dapm_seq_insert(w, down_list, false); + break; + case snd_soc_dapm_post: + dapm_seq_insert(w, up_list, true); + break; + + default: + if (!w->power_check) + break; + + if (!w->force) + power = w->power_check(w); + else + power = 1; + + if (power) { + d = w->dapm; + + /* Supplies and micbiases only bring the + * context up to STANDBY as unless something + * else is active and passing audio they + * generally don't require full power. + */ + switch (w->id) { + case snd_soc_dapm_supply: + case snd_soc_dapm_micbias: + if (d->target_bias_level < SND_SOC_BIAS_STANDBY) + d->target_bias_level = SND_SOC_BIAS_STANDBY; + break; + default: + d->target_bias_level = SND_SOC_BIAS_ON; + break; + } + } + + if (w->power == power) + break; + + trace_snd_soc_dapm_widget_power(w, power); + + if (power) + dapm_seq_insert(w, up_list, true); + else + dapm_seq_insert(w, down_list, false); + + w->power = power; + break; + } +} + /* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- @@ -1209,7 +1268,6 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) LIST_HEAD(down_list); LIST_HEAD(async_domain); enum snd_soc_bias_level bias; - int power; trace_snd_soc_dapm_start(card); @@ -1228,57 +1286,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) * lists indicating if they should be powered up or down. */ list_for_each_entry(w, &card->widgets, list) { - switch (w->id) { - case snd_soc_dapm_pre: - dapm_seq_insert(w, &down_list, false); - break; - case snd_soc_dapm_post: - dapm_seq_insert(w, &up_list, true); - break; - - default: - if (!w->power_check) - continue; - - if (!w->force) - power = w->power_check(w); - else - power = 1; - - if (power) { - d = w->dapm; - - /* Supplies and micbiases only bring - * the context up to STANDBY as unless - * something else is active and - * passing audio they generally don't - * require full power. - */ - switch (w->id) { - case snd_soc_dapm_supply: - case snd_soc_dapm_micbias: - if (d->target_bias_level < SND_SOC_BIAS_STANDBY) - d->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - default: - d->target_bias_level = SND_SOC_BIAS_ON; - break; - } - } - - if (w->power == power) - continue; - - trace_snd_soc_dapm_widget_power(w, power); - - if (power) - dapm_seq_insert(w, &up_list, true); - else - dapm_seq_insert(w, &down_list, false); - - w->power = power; - break; - } + dapm_power_one_widget(w, &up_list, &down_list); } /* If there are no DAPM widgets then try to figure out power from the -- 1.7.6.3

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[alsa-devel] [PATCH] ASoC: Trace and collect statistics for DAPM graph walking
by Mark Brown 20 Sep '11

20 Sep '11

One of the longest standing areas for improvement in ASoC has been the DAPM algorithm - it repeats the same checks many times whenever it is run and makes no effort to limit the areas of the graph it checks meaning we do an awful lot of walks over the full graph. This has never mattered too much as the size of the graph has generally been small in relation to the size of the devices supported and the speed of CPUs but it is annoying. In preparation for work on improving this insert a trace point after the graph walk has been done. This gives us specific timing information for the walk, and in order to give quantifiable (non-benchmark) numbers also count every time we check a link or check the power for a widget and report those numbers. Substantial changes in the algorithm may require tweaks to the stats but they should be useful for simpler things. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 5 +++++ include/sound/soc.h | 1 + include/trace/events/asoc.h | 22 ++++++++++++++++++++++ sound/soc/soc-dapm.c | 17 +++++++++++++++++ 4 files changed, 45 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 350b1b3..0e2d017 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -537,4 +537,9 @@ struct snd_soc_dapm_widget_list { struct snd_soc_dapm_widget *widgets[0]; }; +struct snd_soc_dapm_stats { + int power_checks; + int path_checks; +}; + #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index 24e17be..006f4f6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -828,6 +828,7 @@ struct snd_soc_card { /* Generic DAPM context for the card */ struct snd_soc_dapm_context dapm; + struct snd_soc_dapm_stats dapm_stats; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_card_root; diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 603f5a0..2e1adf6 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -216,6 +216,28 @@ DEFINE_EVENT(snd_soc_dapm_widget, snd_soc_dapm_widget_event_done, ); +TRACE_EVENT(snd_soc_dapm_walk_done, + + TP_PROTO(struct snd_soc_card *card), + + TP_ARGS(card), + + TP_STRUCT__entry( + __string( name, card->name ) + __field( int, power_checks ) + __field( int, path_checks ) + ), + + TP_fast_assign( + __assign_str(name, card->name); + __entry->power_checks = card->dapm_stats.power_checks; + __entry->path_checks = card->dapm_stats.path_checks; + ), + + TP_printk("%s: %d power checks, %d path checks", __get_str(name), + (int)__entry->power_checks, (int)__entry->path_checks) +); + TRACE_EVENT(snd_soc_jack_irq, TP_PROTO(const char *name), diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4a440b5..6a1e13e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -48,6 +48,8 @@ #include <trace/events/asoc.h> +#define DAPM_UPDATE_STAT(widget, val) widget->dapm->card->dapm_stats.val++; + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -649,6 +651,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) struct snd_soc_dapm_path *path; int con = 0; + DAPM_UPDATE_STAT(widget, path_checks); + if (widget->id == snd_soc_dapm_supply) return 0; @@ -697,6 +701,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) struct snd_soc_dapm_path *path; int con = 0; + DAPM_UPDATE_STAT(widget, path_checks); + if (widget->id == snd_soc_dapm_supply) return 0; @@ -767,6 +773,8 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) { int in, out; + DAPM_UPDATE_STAT(w, power_checks); + in = is_connected_input_ep(w); dapm_clear_walk(w->dapm); out = is_connected_output_ep(w); @@ -779,6 +787,8 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) { int in; + DAPM_UPDATE_STAT(w, power_checks); + if (w->active) { in = is_connected_input_ep(w); dapm_clear_walk(w->dapm); @@ -793,6 +803,8 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) { int out; + DAPM_UPDATE_STAT(w, power_checks); + if (w->active) { out = is_connected_output_ep(w); dapm_clear_walk(w->dapm); @@ -808,6 +820,8 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) struct snd_soc_dapm_path *path; int power = 0; + DAPM_UPDATE_STAT(w, power_checks); + /* Check if one of our outputs is connected */ list_for_each_entry(path, &w->sinks, list_source) { if (path->weak) @@ -1208,6 +1222,8 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } } + memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); + /* Check which widgets we need to power and store them in * lists indicating if they should be powered up or down. */ @@ -1299,6 +1315,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) list_for_each_entry(d, &card->dapm_list, list) d->target_bias_level = bias; + trace_snd_soc_dapm_walk_done(card); /* Run all the bias changes in parallel */ list_for_each_entry(d, &dapm->card->dapm_list, list) -- 1.7.6.3

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[alsa-devel] [PATCH] ALSA: HDA: Refactor Realtek's automute
by David Henningsson 20 Sep '11

20 Sep '11

Ok, my automute refactoring has been rebased to your latest tree. I've tested it on my own machine, but I don't know if wider testing is needed to ensure there are no regressions. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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Re: [alsa-devel] [PATCH] ASoC: fsl: Fix error handling if platform_device_add fails
by Timur Tabi 20 Sep '11

20 Sep '11

Axel Lin wrote: > Call platform_device_put() instead of platform_device_unregister() if > platform_device_add() fails. What's wrong with calling platform_device_unregister()? -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] ASoC updates for 3.2
by Mark Brown 20 Sep '11

20 Sep '11

The following changes since commit 30ab1e78864ca5781de5b1fb501bed9df2c215f1: ASoC: ad193x: Setup regmap read and write flag masks for SPI (2011-09-09 11:13:31 -0700) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Arnd Bergmann (1): ASoC: Fix trivial build regression in Kirkwood I2S Axel Lin (7): ASoC: samsung: Fix checking return value of clk_get ASoC: bf5xx-ad73311: Fix prototype for bf5xx_probe ASoC: tpa6130a2: Remove obsolete cleanup for clientdata ASoC: wm8995: Return -EINVAL if device ID mismatch ASoC: wm8995: Remove unused i2c variable in wm8995_remove() ASoC: sn95031: Staticize sn95031_pcm_hw_params ASoC: fsl: Fix error handling if platform_device_add fails Ben Gardiner (1): ASoC: davinci-pcm: trivial: replace link with actual chan/link Dong Aisheng (1): ASoC: mxs-saif: add record function Lars-Peter Clausen (1): ASoC: ssm2602: Do not dereference codec->control_data Mark Brown (13): regulator: Implement deferred disable support ASoC: Disable WM8996 CPVDD supply when not in use ASoC: Correct channel numbers for WM8996 AIF2 ASoC: Fix WM8996 DC servo operation without IRQ Merge branch 'for-3.1' into for-3.2 ASoC: Display the error code when we fail to add a DAPM control ASoC: WM8996 only needs bandgap for analogue functionality ASoC: Initial WM8996 headphone impedance measurement support ASoC: Add line loads to the list of supported detections for Speyside ASoC: Fix unused variable warning in WM8996 ASoC: Refcount WM8996 bandgap from FLL too ASoC: Remove bitrotted wm8962_resume() Merge branch 'for-3.1' into for-3.2 Mika Westerberg (7): ASoC: snappercl15: convert to use snd_soc_register_card() ARM: ep93xx: simone: register audio platform device ARM: ep93xx: edb93xx: register audio platform device ARM: ep93xx: snappercl15: register audio platform device ASoC: ep93xx-pcm: add MODULE_ALIAS ASoC: simone: convert to use snd_soc_register_card() ASoC: edb93xx: convert to use snd_soc_register_card() Peter Ujfalusi (8): MFD: twl6040: Remove global pointer for platform_device MFD: twl6040: Add accessor for revision ID Input: twl6040-vibra: Use accessor to get revision information MFD: twl6040: Fix power on GPIO handling ASoC: twl6040: Chip initialization cleanup ASoC: twl6040: Use chip defaults in the initial reg_cache ASoC: twl6040: Fix the number of channels for vibra ASoC: twl6040: Correct supported number of playback channels Timur Tabi (3): ASoC: improve asynchronous mode support in the fsl_ssi driver ASoC: support sample sizes properly in the WM8776 codec driver ASoC: support all possible sample rates in the WM8776 driver arch/arm/mach-ep93xx/edb93xx.c | 6 + arch/arm/mach-ep93xx/simone.c | 13 ++- arch/arm/mach-ep93xx/snappercl15.c | 13 ++- drivers/input/misc/twl6040-vibra.c | 2 +- drivers/mfd/twl6040-core.c | 38 +++--- drivers/regulator/core.c | 59 ++++++++ include/linux/mfd/twl6040.h | 8 +- include/linux/regulator/consumer.h | 7 + include/linux/regulator/driver.h | 3 + include/sound/saif.h | 16 ++ sound/soc/blackfin/bf5xx-ad73311.c | 2 +- sound/soc/codecs/sn95031.c | 2 +- sound/soc/codecs/ssm2602.c | 3 +- sound/soc/codecs/tpa6130a2.c | 1 - sound/soc/codecs/twl6040.c | 114 +++------------ sound/soc/codecs/wm8776.c | 38 +++--- sound/soc/codecs/wm8962.c | 26 ---- sound/soc/codecs/wm8995.c | 3 +- sound/soc/codecs/wm8996.c | 263 +++++++++++++++++++++++++++------- sound/soc/davinci/davinci-pcm.c | 123 ++++++++--------- sound/soc/ep93xx/edb93xx.c | 60 +++++---- sound/soc/ep93xx/ep93xx-pcm.c | 1 + sound/soc/ep93xx/simone.c | 64 +++++---- sound/soc/ep93xx/snappercl15.c | 53 +++++--- sound/soc/fsl/fsl_ssi.c | 145 +++++++++---------- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/mxs/mxs-saif.c | 145 +++++++++++++++++-- sound/soc/mxs/mxs-saif.h | 4 + sound/soc/samsung/s3c2412-i2s.c | 4 +- sound/soc/samsung/s3c24xx-i2s.c | 4 +- sound/soc/samsung/s3c24xx_uda134x.c | 8 +- sound/soc/samsung/speyside.c | 3 +- sound/soc/soc-dapm.c | 4 +- 35 files changed, 775 insertions(+), 466 deletions(-) create mode 100644 include/sound/saif.h

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