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author | Linus Torvalds <torvalds@linux-foundation.org> | 2012-12-13 23:51:23 +0400 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2012-12-13 23:51:23 +0400 |
commit | 046e7d685bc370fd4c879ab6635ad3f69e6673d1 (patch) | |
tree | 36b981f8d1f2bfd348c1479acbe3a9426d35c377 /sound/usb/pcm.c | |
parent | fe504c5c745aeb767d978fbedeb94775fd4cb69c (diff) | |
parent | 6eb827d23577a4efec2b10a9c4cc9ded268a1d1c (diff) | |
download | linux-046e7d685bc370fd4c879ab6635ad3f69e6673d1.tar.xz |
Merge tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This update contains a fairly wide range of changes all over in sound
subdirectory, mainly because of UAPI header moves by David and __dev*
annotation removals by Bill. Other highlights are:
- Introduced the support for wallclock timestamps in ALSA PCM core
- Add the poll loop implementation for HD-audio jack detection
- Yet more VGA-switcheroo fixes for HD-audio
- New VIA HD-audio codec support
- More fixes on resource management in USB audio and MIDI drivers
- More quirks for USB-audio ASUS Xonar U3, Reloop Play, Focusrite,
Roland VG-99, etc
- Add support for FastTrack C400 usb-audio
- Clean ups in many drivers regarding firmware loading
- Add PSC724 Ultiimate Edge support to ice1712
- A few hdspm driver updates
- New Stanton SCS.1d/1m FireWire driver
- Standardisation of the logging in ASoC codes
- DT and dmaengine support for ASoC Atmel
- Support for Wolfson ADSP cores
- New drivers for Freescale/iVeia P1022 and Maxim MAX98090
- Lots of other ASoC driver fixes and developments"
Fix up trivial conflicts. And go out on a limb and assume the dts file
'status' field of one of the conflicting things was supposed to be
"disabled", not "disable" like in pretty much all other cases.
* tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (341 commits)
ALSA: hda - Move runtime PM check to runtime_idle callback
ALSA: hda - Add stereo-dmic fixup for Acer Aspire One 522
ALSA: hda - Avoid doubly suspend after vga switcheroo
ALSA: usb-audio: Enable S/PDIF on the ASUS Xonar U3
ALSA: hda - Check validity of CORB/RIRB WP reads
ALSA: hda - use usleep_range in link reset and change timeout check
ALSA: HDA: VIA: Add support for codec VT1808.
ALSA: HDA: VIA Add support for codec VT1705CF.
ASoC: codecs: remove __dev* attributes
ASoC: utils: remove __dev* attributes
ASoC: ux500: remove __dev* attributes
ASoC: txx9: remove __dev* attributes
ASoC: tegra: remove __dev* attributes
ASoC: spear: remove __dev* attributes
ASoC: sh: remove __dev* attributes
ASoC: s6000: remove __dev* attributes
ASoC: OMAP: remove __dev* attributes
ASoC: nuc900: remove __dev* attributes
ASoC: mxs: remove __dev* attributes
ASoC: kirkwood: remove __dev* attributes
...
Diffstat (limited to 'sound/usb/pcm.c')
-rw-r--r-- | sound/usb/pcm.c | 175 |
1 files changed, 148 insertions, 27 deletions
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index ef6fa24fc473..c6593101c049 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -46,6 +46,9 @@ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, int frame_diff; int est_delay; + if (!subs->last_delay) + return 0; /* short path */ + current_frame_number = usb_get_current_frame_number(subs->dev); /* * HCD implementations use different widths, use lower 8 bits. @@ -75,7 +78,8 @@ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; - substream->runtime->delay = snd_usb_pcm_delay(subs, + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + substream->runtime->delay = snd_usb_pcm_delay(subs, substream->runtime->rate); spin_unlock(&subs->lock); return hwptr_done / (substream->runtime->frame_bits >> 3); @@ -173,11 +177,8 @@ static int init_pitch_v2(struct snd_usb_audio *chip, int iface, { struct usb_device *dev = chip->dev; unsigned char data[1]; - unsigned int ep; int err; - ep = get_endpoint(alts, 0)->bEndpointAddress; - data[0] = 1; if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, @@ -214,7 +215,7 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, } } -static int start_endpoints(struct snd_usb_substream *subs, int can_sleep) +static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep) { int err; @@ -266,16 +267,18 @@ static int start_endpoints(struct snd_usb_substream *subs, int can_sleep) return 0; } -static void stop_endpoints(struct snd_usb_substream *subs, - int force, int can_sleep, int wait) +static void stop_endpoints(struct snd_usb_substream *subs, bool wait) { if (test_and_clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) - snd_usb_endpoint_stop(subs->sync_endpoint, - force, can_sleep, wait); + snd_usb_endpoint_stop(subs->sync_endpoint); if (test_and_clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags)) - snd_usb_endpoint_stop(subs->data_endpoint, - force, can_sleep, wait); + snd_usb_endpoint_stop(subs->data_endpoint); + + if (wait) { + snd_usb_endpoint_sync_pending_stop(subs->sync_endpoint); + snd_usb_endpoint_sync_pending_stop(subs->data_endpoint); + } } static int deactivate_endpoints(struct snd_usb_substream *subs) @@ -359,6 +362,19 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; switch (subs->stream->chip->usb_id) { + case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ + if (is_playback) { + implicit_fb = 1; + ep = 0x81; + iface = usb_ifnum_to_if(dev, 3); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; + } + break; case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ case USB_ID(0x0763, 0x2081): if (is_playback) { @@ -381,7 +397,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) /* ... and check descriptor size before accessing bSynchAddress because there is a version of the SB Audigy 2 NX firmware lacking the audio fields in the endpoint descriptors */ - if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != 0x01 || + if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC || (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && get_endpoint(alts, 1)->bSynchAddress != 0 && !implicit_fb)) { @@ -438,6 +454,103 @@ add_sync_ep: } /* + * Return the score of matching two audioformats. + * Veto the audioformat if: + * - It has no channels for some reason. + * - Requested PCM format is not supported. + * - Requested sample rate is not supported. + */ +static int match_endpoint_audioformats(struct audioformat *fp, + struct audioformat *match, int rate, + snd_pcm_format_t pcm_format) +{ + int i; + int score = 0; + + if (fp->channels < 1) { + snd_printdd("%s: (fmt @%p) no channels\n", __func__, fp); + return 0; + } + + if (!(fp->formats & (1ULL << pcm_format))) { + snd_printdd("%s: (fmt @%p) no match for format %d\n", __func__, + fp, pcm_format); + return 0; + } + + for (i = 0; i < fp->nr_rates; i++) { + if (fp->rate_table[i] == rate) { + score++; + break; + } + } + if (!score) { + snd_printdd("%s: (fmt @%p) no match for rate %d\n", __func__, + fp, rate); + return 0; + } + + if (fp->channels == match->channels) + score++; + + snd_printdd("%s: (fmt @%p) score %d\n", __func__, fp, score); + + return score; +} + +/* + * Configure the sync ep using the rate and pcm format of the data ep. + */ +static int configure_sync_endpoint(struct snd_usb_substream *subs) +{ + int ret; + struct audioformat *fp; + struct audioformat *sync_fp = NULL; + int cur_score = 0; + int sync_period_bytes = subs->period_bytes; + struct snd_usb_substream *sync_subs = + &subs->stream->substream[subs->direction ^ 1]; + + /* Try to find the best matching audioformat. */ + list_for_each_entry(fp, &sync_subs->fmt_list, list) { + int score = match_endpoint_audioformats(fp, subs->cur_audiofmt, + subs->cur_rate, subs->pcm_format); + + if (score > cur_score) { + sync_fp = fp; + cur_score = score; + } + } + + if (unlikely(sync_fp == NULL)) { + snd_printk(KERN_ERR "%s: no valid audioformat for sync ep %x found\n", + __func__, sync_subs->ep_num); + return -EINVAL; + } + + /* + * Recalculate the period bytes if channel number differ between + * data and sync ep audioformat. + */ + if (sync_fp->channels != subs->channels) { + sync_period_bytes = (subs->period_bytes / subs->channels) * + sync_fp->channels; + snd_printdd("%s: adjusted sync ep period bytes (%d -> %d)\n", + __func__, subs->period_bytes, sync_period_bytes); + } + + ret = snd_usb_endpoint_set_params(subs->sync_endpoint, + subs->pcm_format, + sync_fp->channels, + sync_period_bytes, + subs->cur_rate, + sync_fp, + NULL); + + return ret; +} + +/* * configure endpoint params * * called during initial setup and upon resume @@ -447,7 +560,7 @@ static int configure_endpoint(struct snd_usb_substream *subs) int ret; /* format changed */ - stop_endpoints(subs, 0, 0, 0); + stop_endpoints(subs, true); ret = snd_usb_endpoint_set_params(subs->data_endpoint, subs->pcm_format, subs->channels, @@ -459,13 +572,8 @@ static int configure_endpoint(struct snd_usb_substream *subs) return ret; if (subs->sync_endpoint) - ret = snd_usb_endpoint_set_params(subs->sync_endpoint, - subs->pcm_format, - subs->channels, - subs->period_bytes, - subs->cur_rate, - subs->cur_audiofmt, - NULL); + ret = configure_sync_endpoint(subs); + return ret; } @@ -533,7 +641,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) subs->period_bytes = 0; down_read(&subs->stream->chip->shutdown_rwsem); if (!subs->stream->chip->shutdown) { - stop_endpoints(subs, 0, 1, 1); + stop_endpoints(subs, true); deactivate_endpoints(subs); } up_read(&subs->stream->chip->shutdown_rwsem); @@ -608,7 +716,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) /* 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) - ret = start_endpoints(subs, 1); + ret = start_endpoints(subs, true); unlock: up_read(&subs->stream->chip->shutdown_rwsem); @@ -1013,7 +1121,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; - stop_endpoints(subs, 0, 0, 0); + stop_endpoints(subs, true); if (!as->chip->shutdown && subs->interface >= 0) { usb_set_interface(subs->dev, subs->interface, 0); @@ -1195,6 +1303,9 @@ static void retire_playback_urb(struct snd_usb_substream *subs, return; spin_lock_irqsave(&subs->lock, flags); + if (!subs->last_delay) + goto out; /* short path */ + est_delay = snd_usb_pcm_delay(subs, runtime->rate); /* update delay with exact number of samples played */ if (processed > subs->last_delay) @@ -1212,6 +1323,15 @@ static void retire_playback_urb(struct snd_usb_substream *subs, snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n", est_delay, subs->last_delay); + if (!subs->running) { + /* update last_frame_number for delay counting here since + * prepare_playback_urb won't be called during pause + */ + subs->last_frame_number = + usb_get_current_frame_number(subs->dev) & 0xff; + } + + out: spin_unlock_irqrestore(&subs->lock, flags); } @@ -1248,12 +1368,13 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea subs->running = 1; return 0; case SNDRV_PCM_TRIGGER_STOP: - stop_endpoints(subs, 0, 0, 0); + stop_endpoints(subs, false); subs->running = 0; return 0; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: subs->data_endpoint->prepare_data_urb = NULL; - subs->data_endpoint->retire_data_urb = NULL; + /* keep retire_data_urb for delay calculation */ + subs->data_endpoint->retire_data_urb = retire_playback_urb; subs->running = 0; return 0; } @@ -1269,7 +1390,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream switch (cmd) { case SNDRV_PCM_TRIGGER_START: - err = start_endpoints(subs, 0); + err = start_endpoints(subs, false); if (err < 0) return err; @@ -1277,7 +1398,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream subs->running = 1; return 0; case SNDRV_PCM_TRIGGER_STOP: - stop_endpoints(subs, 0, 0, 0); + stop_endpoints(subs, false); subs->running = 0; return 0; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: |