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

[alsa-devel] [PATCH v2] davinci-mcasp: Add support for multichannel playback
by Michal Bachraty 10 Apr '13

10 Apr '13

Davinci McASP has support for I2S multichannel playback. For I2S playback/receive, each serializer is capable to play 2 channels (L/R) audio data.Serializer function (Playback-receive-none) is configured in DT, depending on hardware specification. It is possible to play less channels than configured in DT. For that purpose,only specific number of active serializers are enabled. McASP FIFO need to have DMA transfer Bcnt set to number of enabled serializers, otherwise no data are transfered to McASP and Alsa generates "DMA/IRQ playback write error (DMA or IRQ trouble?)" error. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 56 ++++++++++++++++++++++++++++++++----- sound/soc/davinci/davinci-pcm.c | 16 ++++++----- sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 59 insertions(+), 14 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index afef3fb..b84bb73 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -235,6 +235,10 @@ #define DISMOD (val)(val<<2) #define TXSTATE BIT(4) #define RXSTATE BIT(5) +#define SRMOD_MASK 3 +#define SRMOD_INACTIVE 0 +#define SRMOD_TX 1 +#define SRMOD_RX 2 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -657,12 +661,15 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, return 0; } -static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, + int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; - + u8 ser; + u8 slots = dev->tdm_slots; + u8 max_active_serializers = (channels + slots - 1) / slots; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -680,16 +687,42 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) } for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) + tx_ser++; + if (dev->serial_dir[i] == RX_MODE) + rx_ser++; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ser = tx_ser; + else + ser = rx_ser; + + if (ser < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + return -EINVAL; + } + + tx_ser = 0; + rx_ser = 0; + + for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) { + if (dev->serial_dir[i] == TX_MODE && + tx_ser < max_active_serializers) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE) { + } else if (dev->serial_dir[i] == RX_MODE && + rx_ser < max_active_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + SRMOD_INACTIVE, SRMOD_MASK); } } @@ -729,6 +762,8 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } } + + return 0; } static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) @@ -812,8 +847,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + u8 slots = dev->tdm_slots; + int channels; + struct snd_interval *pcm_channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels = pcm_channels->min; - davinci_hw_common_param(dev, substream->stream); + if (davinci_hw_common_param(dev, substream->stream, channels)) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; else @@ -862,6 +903,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; + dma_params->active_serializers = (channels + slots - 1) / slots; davinci_config_channel_size(dev, word_length); return 0; @@ -936,13 +978,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bb57552..3af8b50 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,6 +182,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; + unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -195,14 +196,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) data_type = prtd->params->data_type; count = period_size / data_type; if (fifo_level) - count /= fifo_level; + count /= fifo_level * serializers; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; - dst_bidx = 0; - src_cidx = data_type * fifo_level; + dst_bidx = 4; + src_cidx = data_type * fifo_level * serializers; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -210,7 +211,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level; + dst_cidx = data_type * fifo_level * serializers; } acnt = prtd->params->acnt; @@ -224,9 +225,10 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, ASYNC); else - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, - ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, + fifo_level * serializers, + count, fifo_level * serializers, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fbb710c..0d84d32 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,6 +27,7 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; + unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

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Re: [alsa-devel] [alsa-cvslog] [ALSA GIT]ALSA utilities repository branch master updated. v1.0.26-26-ge05b903
by Takashi Iwai 10 Apr '13

10 Apr '13

At Mon, 8 Apr 2013 14:51:54 +0200 (CEST), GIT server wrote: > > This is an automated email from the git hooks/post-receive script. It was > generated because a ref change was pushed to the repository containing > the project "ALSA utilities repository". > > The branch, master has been updated > via e05b903b1fb16e967d838edac408304cd4470fee (commit) > from 46cb355d4ff41c8065b02644b0534ab42d16cb72 (commit) > > Those revisions listed above that are new to this repository have > not appeared on any other notification email; so we list those > revisions in full, below. > > - Log ----------------------------------------------------------------- > commit e05b903b1fb16e967d838edac408304cd4470fee > Author: Jaroslav Kysela <perex(a)perex.cz> > Date: Mon Apr 8 14:49:31 2013 +0200 > > alsactl: move systemd config to the daemon mode > > Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> I'm not thrilled by the silent default behavior change like this. This will affect all systems using systemd from now on. I understand the reason behind it, but I wonder whether it's an overkill. Yet another daemon, unconditionally no matter whether hotplug or not? Hmm... Takashi

4 13

[alsa-devel] [PATCH] ALSA: snd-usb: Playback Design: use usb_set_inferface quirk from more locations
by Daniel Mack 10 Apr '13

10 Apr '13

It turns out the devices from Playback Design need the delay quirk after usb_set_interface from clocks.c as well. Make it a proper quirks function and factor out the code to quirks.c. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/clock.c | 3 +++ sound/usb/pcm.c | 7 +------ sound/usb/quirks.c | 10 ++++++++++ sound/usb/quirks.h | 1 + 4 files changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index ae35e7d..b0ec364 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -32,6 +32,7 @@ #include "card.h" #include "helper.h" #include "clock.h" +#include "quirks.h" static struct uac_clock_source_descriptor * snd_usb_find_clock_source(struct usb_host_interface *ctrl_iface, @@ -392,7 +393,9 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, * interface is active. */ if (rate != prev_rate) { usb_set_interface(dev, iface, 0); + snd_usb_set_interface_quirk(dev); usb_set_interface(dev, iface, fmt->altsetting); + snd_usb_set_interface_quirk(dev); } return 0; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 754cb5b..815a37d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -349,12 +349,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; - /* - * "Playback Design" products need a 50ms delay after setting the - * USB interface. - */ - if (le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) - mdelay(50); + snd_usb_set_interface_quirk(dev); } subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7882a82..551ae1a 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -892,6 +892,16 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) ep->skip_packets = 16; } +void snd_usb_set_interface_quirk(struct usb_device *dev) +{ + /* + * "Playback Design" products need a 50ms delay after setting the + * USB interface. + */ + if (le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) + mdelay(50); +} + void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size) diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index f10d00e..665e972 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -26,6 +26,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep); +void snd_usb_set_interface_quirk(struct usb_device *dev); void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size); -- 1.7.10.4

2 1

Re: [alsa-devel] [PATCH] ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is running
by Mark Brown 09 Apr '13

09 Apr '13

On Tue, Apr 09, 2013 at 05:13:59PM +0200, Alban Bedel wrote: > The Charge Pump needs the DSP clock to work properly, without it the > bypass to HP/LINEOUT is not working properly. This requirement is not > mentioned in the datasheet but has been confirmed by Mark Brown from > Wolfson. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is running
by Mark Brown 09 Apr '13

09 Apr '13

On Wed, Mar 20, 2013 at 05:37:33PM +0100, Alban Bedel wrote: > output enable sequence requiere the DSP clock. Some input from Wolfson > will be needed to find out if this is just an undocumented requirement > or a chip bug. It seems it's an undocumented requirement. > + { "Left Output Mixer", "Left Bypass Switch", "CLK_DSP" }, > + { "Left Output Mixer", "Right Bypass Switch", "CLK_DSP" }, > + { "Right Output Mixer", "Left Bypass Switch", "CLK_DSP" }, > + { "Right Output Mixer", "Right Bypass Switch", "CLK_DSP" }, The actual change here should be to add a route: { "Charge Pump", NULL, "CLK_DSP" }, Can you submit an update for this or should I do one?

1 0

[alsa-devel] [PATCH v3 3/3] ALSA: hda - delay resuming Haswell HDMI codec in system resume
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> In system resume, Haswell codec cannot be programmed to D0 before Gfx driver initializes the display pipeline and audio, which will trigger an unsol event on the pin with HDMI/DP cable connected. Otherwise, the connected pin will stay in D3 with right channel muted and thus no sound can be heard. In this patch, Haswell codec will claim it needs "delaying resume" by setting flag "support_delay_resume", so system resume will skip it and mark it as "delay_resumed". On later codec access, Haswell codec will be resumed and hda_call_codec_resume() will call its set_power_states ops intel_haswell_set_ power_state(). For a delayed resume, this ops will enable and wait for the unsol event, and then resume the codec. A 300ms timeout is set in case unsol event is lost. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ede8215..22f276c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -89,6 +89,14 @@ struct hdmi_spec { */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; + +#ifdef CONFIG_PM + /* + * Non-generic Intel Haswell specific + */ + unsigned int ready_to_resume:1; + wait_queue_head_t resume_wq; +#endif }; @@ -975,6 +983,13 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; +#ifdef CONFIG_PM + if (codec->resume_delayed) { + spec->ready_to_resume = 1; + wake_up(&spec->resume_wq); + } +#endif + hdmi_present_sense(get_pin(spec, pin_idx), 1); snd_hda_jack_report_sync(codec); } @@ -1725,6 +1740,11 @@ static int generic_hdmi_init(struct hda_codec *codec) hdmi_init_pin(codec, pin_nid); snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); } + +#ifdef CONFIG_PM + spec->ready_to_resume = 0; +#endif + return 0; } @@ -1855,6 +1875,60 @@ static const struct hda_fixup hdmi_fixups[] = { }; +#ifdef CONFIG_PM +static void intel_haswell_wait_ready_to_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin; + hda_nid_t pin_nid; + struct hda_jack_tbl *jack; + + per_pin = get_pin(spec, pin_idx); + pin_nid = per_pin->pin_nid; + jack = snd_hda_jack_tbl_get(codec, pin_nid); + if (jack) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | jack->tag); + } + + wait_event_timeout(spec->resume_wq, + spec->ready_to_resume, msecs_to_jiffies(300)); + if (!spec->ready_to_resume) + snd_printk(KERN_WARNING "HDMI: Haswell not ready to resume\n"); +} + +static void intel_haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (codec->resume_delayed && power_state == AC_PWRST_D0) { + intel_haswell_wait_ready_to_resume(codec); + codec->resume_delayed = 0; + } + + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} + +static void intel_haswell_allow_delay_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + init_waitqueue_head(&spec->resume_wq); + codec->support_delay_resume = 1; + + codec->patch_ops.set_power_state = intel_haswell_set_power_state; +} +#else +define intel_haswell_allow_delay_resume do { } while (0) +#endif + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1878,6 +1952,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + intel_haswell_allow_delay_resume(codec); + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

1 0

[alsa-devel] [PATCH v2 2/3] ALSA: hda - system resume may delay resuming a codec
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> On system resume, a codec may not be ready to resume for some external dependency. e.g. Intel Haswell HDMI codec depends on Gfx to initialize the display pipeline before the codec can resume. To avoid the external dependency blocking system resume, this patch allow system resume ops "snd_hda_resume" skip a not ready codec and mark it as "resume delayed". So the codec resume is delayed until later codec access, assuming the external dependency may be resolved and waiting is acceptable at that time. A codec that needs delaying resume shall explicitly set "support_delay_resume" flag on initialization in its patch ops. And it's up to the codec when it's ready and how to finish the delayed resume. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ba147d7..81e91b8 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5518,7 +5518,10 @@ int snd_hda_resume(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - hda_call_codec_resume(codec); + if (codec->support_delay_resume) + codec->resume_delayed = 1; + else + hda_call_codec_resume(codec); } return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6061360..5fe7362 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -889,6 +889,8 @@ struct hda_codec { unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ unsigned int pm_down_notified:1; /* PM notified to controller */ unsigned int in_pm:1; /* suspend/resume being performed */ + unsigned int support_delay_resume:1; /* codec support delay resume */ + unsigned int resume_delayed:1; /* resume delayed by PM */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ -- 1.7.10.4

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[alsa-devel] [PATCH v2 1/3] ALSA: hda - add new function snd_hda_bus_reset() for codec communitation error
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> A new function snd_hda_bus_reset() is defined to reset all codecs on the bus. This is implemented as temporary suspend-and-resume calls to codecs as an ad hoc solution. It will be called by bus reset ops azx_bus_reset() as a rescue of codec communitation error, replacing calls to snd_hda_suspend and snd_hda_resume. It's because snd_hda_resume need change for system resume: It will delay resuming a codec that is not ready to resume for some external dependency, to avoid blocking system resume. But this change of "delay resume" is not suitable for recovering from codec communication error, which need resume the codec immediately after suspend to simulate a bus reset. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 17286b3..ba147d7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5523,6 +5523,29 @@ int snd_hda_resume(struct hda_bus *bus) return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); + +/** + * snd_hda_bus_reset - reset the codecs on the bus. + * @bus: the HDA bus + * + * This is implemented as temporary suspend-and-resume calls + * to codecs as an ad hoc solution. + * + * Returns 0 if successful. + */ +int snd_hda_bus_reset(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + cancel_delayed_work_sync(&codec->jackpoll_work); + if (hda_codec_is_power_on(codec)) + hda_call_codec_suspend(codec, false); + hda_call_codec_resume(codec); + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_bus_reset); #endif /* CONFIG_PM */ /* diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c93f902..6061360 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1076,6 +1076,7 @@ void snd_hda_unlock_devices(struct hda_bus *bus); #ifdef CONFIG_PM int snd_hda_suspend(struct hda_bus *bus); int snd_hda_resume(struct hda_bus *bus); +int snd_hda_bus_reset(struct hda_bus *bus); #endif static inline diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..6f11c7a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1659,8 +1659,7 @@ static void azx_bus_reset(struct hda_bus *bus) struct azx_pcm *p; list_for_each_entry(p, &chip->pcm_list, list) snd_pcm_suspend_all(p->pcm); - snd_hda_suspend(chip->bus); - snd_hda_resume(chip->bus); + snd_hda_bus_reset(chip->bus); } #endif bus->in_reset = 0; -- 1.7.10.4

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[alsa-devel] [PATCH 3/3] ALSA: hda - delay resuming Haswell HDMI codec in system resume
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> In system resume, Haswell codec cannot be programmed to D0 before Gfx driver initializes the display pipeline and audio, which will trigger an unsol event on the pin with HDMI/DP cable connected. Otherwise, the connected pin will stay in D3 with right channel muted and thus no sound can be heard. In this patch, Haswell codec will claim it needs "delaying resume" by setting flag "support_delay_resume", so system resume will skip it and mark it as "delay_resumed". On later codec access, Haswell codec will be resumed and hda_call_codec_resume() will call its set_power_states ops intel_haswell_set_ power_state(). For a delayed resume, this ops will enable and wait for the unsol event, and then resume the codec. A 300ms timeout is set in case unsol event is lost. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ede8215..f39e374 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -89,6 +89,14 @@ struct hdmi_spec { */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; + +#ifdef CONFIG_PM + /* + * Non-generic Intel Haswell specific + */ + unsigned int ready_to_resume:1; + wait_queue_head_t resume_wq; +#endif }; @@ -975,6 +983,13 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; +#ifdef CONFIG_PM + if (codec->resume_delayed) { + spec->ready_to_resume = 1; + wake_up(&spec->resume_wq); + } +#endif + hdmi_present_sense(get_pin(spec, pin_idx), 1); snd_hda_jack_report_sync(codec); } @@ -1725,6 +1740,11 @@ static int generic_hdmi_init(struct hda_codec *codec) hdmi_init_pin(codec, pin_nid); snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); } + +#ifdef CONFIG_PM + spec->ready_to_resume = 0; +#endif + return 0; } @@ -1855,6 +1875,61 @@ static const struct hda_fixup hdmi_fixups[] = { }; +#ifdef CONFIG_PM +static void intel_haswell_wait_ready_to_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin; + hda_nid_t pin_nid; + struct hda_jack_tbl *jack; + + per_pin = get_pin(spec, pin_idx); + pin_nid = per_pin->pin_nid; + jack = snd_hda_jack_tbl_get(codec, pin_nid); + if (jack) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | jack->tag); + } + + wait_event_timeout(spec->resume_wq, + spec->ready_to_resume, msecs_to_jiffies(300)); + if (!spec->ready_to_resume) + snd_printd(KERN_WARNING "HDMI: Haswell not ready to resume\n"); +} + +static void intel_haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (codec->resume_delayed && power_state == AC_PWRST_D0) { + intel_haswell_wait_ready_to_resume(codec); + codec->resume_delayed = 0; + } + + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} + +static inline void intel_haswell_allow_delay_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + init_waitqueue_head(&spec->resume_wq); + codec->support_delay_resume = 1; + + codec->patch_ops.set_power_state = + intel_haswell_set_power_state; +} +#else +define intel_haswell_allow_delay_resume NULL +#endif + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1878,6 +1953,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + intel_haswell_allow_delay_resume(codec); + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: hda - Apply codec delay to wallclock.
by Dylan Reid 09 Apr '13

09 Apr '13

For playback add the codec-side delay to the timestamp, for capture subtract it. This brings the timestamps in line with the time that was recently added to the delay reporting. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_intel.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..ec8ac71 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1889,6 +1889,27 @@ static void azx_timecounter_init(struct snd_pcm_substream *substream, tc->cycle_last = last; } +static u64 azx_add_codec_delay_nsec(struct snd_pcm_substream *substream, + u64 nsec) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + int stream = substream->stream; + struct hda_pcm_stream *hinfo = apcm->hinfo[stream]; + struct hda_codec *codec = apcm->codec; + u64 codec_nsec; + + if (hinfo->ops.get_delay) { + codec_nsec = + hinfo->ops.get_delay(hinfo, codec, substream) * 1000000; + if (stream == SNDRV_PCM_STREAM_CAPTURE) + nsec = (nsec > codec_nsec) ? nsec - codec_nsec : 0; + else if (stream == SNDRV_PCM_STREAM_PLAYBACK) + nsec += codec_nsec; + } + + return nsec; +} + static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, struct timespec *ts) { @@ -1897,6 +1918,7 @@ static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, nsec = timecounter_read(&azx_dev->azx_tc); nsec = div_u64(nsec, 3); /* can be optimized */ + nsec = azx_add_codec_delay_nsec(substream, nsec); *ts = ns_to_timespec(nsec); -- 1.8.1.3.605.g02339dd

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