From 15c0cee6c809a137e0fc7f1d2b0867cc03473c0c Mon Sep 17 00:00:00 2001 From: Ben Collins Date: Fri, 28 May 2010 11:43:45 -0400 Subject: ALSA: pcm: Define G723 3-bit and 5-bit formats This defines the 24bps and 40bps (8khz sample rate) G.723 codec formats. They are going to be used once I submit the driver for an mpeg4/g723 compression card. I've updated the signed value to -1 as per Takashi's comments since these are non-linear formats. Signed-off-by: Ben Collins Signed-off-by: Takashi Iwai --- include/sound/asound.h | 6 +++++- include/sound/pcm.h | 4 ++++ 2 files changed, 9 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/asound.h b/include/sound/asound.h index 9f1eecf99e6b..a1803ecea34d 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -212,7 +212,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_S18_3BE ((__force snd_pcm_format_t) 41) /* in three bytes */ #define SNDRV_PCM_FORMAT_U18_3LE ((__force snd_pcm_format_t) 42) /* in three bytes */ #define SNDRV_PCM_FORMAT_U18_3BE ((__force snd_pcm_format_t) 43) /* in three bytes */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_U18_3BE +#define SNDRV_PCM_FORMAT_G723_24 ((__force snd_pcm_format_t) 44) /* 8 samples in 3 bytes */ +#define SNDRV_PCM_FORMAT_G723_24_1B ((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */ +#define SNDRV_PCM_FORMAT_G723_40 ((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */ +#define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_G723_40_1B #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/include/sound/pcm.h b/include/sound/pcm.h index dd76cdede64d..07fd630db88d 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -174,6 +174,10 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_U18_3LE (1ULL << SNDRV_PCM_FORMAT_U18_3LE) #define SNDRV_PCM_FMTBIT_S18_3BE (1ULL << SNDRV_PCM_FORMAT_S18_3BE) #define SNDRV_PCM_FMTBIT_U18_3BE (1ULL << SNDRV_PCM_FORMAT_U18_3BE) +#define SNDRV_PCM_FMTBIT_G723_24 (1ULL << SNDRV_PCM_FORMAT_G723_24) +#define SNDRV_PCM_FMTBIT_G723_24_1B (1ULL << SNDRV_PCM_FORMAT_G723_24_1B) +#define SNDRV_PCM_FMTBIT_G723_40 (1ULL << SNDRV_PCM_FORMAT_G723_40) +#define SNDRV_PCM_FMTBIT_G723_40_1B (1ULL << SNDRV_PCM_FORMAT_G723_40_1B) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE -- cgit v1.2.3 From 69da9bcb98ccbfb5d5f751bc13418f1307332925 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 16 Jun 2010 17:57:28 +0200 Subject: ALSA: usb-audio: unify UAC macros and struct names Get rid of the last occurances of _v1 suffixes, and move the version number right after the "uac" string. Now things are consitent again. Sorry for the forth and back, but it just looks much nicer this way. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- drivers/usb/gadget/f_audio.c | 6 +++--- drivers/usb/gadget/gmidi.c | 2 +- include/linux/usb/audio-v2.h | 2 +- include/linux/usb/audio.h | 12 ++++++------ sound/usb/card.c | 2 +- sound/usb/endpoint.c | 4 ++-- sound/usb/mixer.c | 14 +++++++------- 7 files changed, 21 insertions(+), 21 deletions(-) (limited to 'include') diff --git a/drivers/usb/gadget/f_audio.c b/drivers/usb/gadget/f_audio.c index b91115f84b13..1f48ceb55a77 100644 --- a/drivers/usb/gadget/f_audio.c +++ b/drivers/usb/gadget/f_audio.c @@ -61,7 +61,7 @@ DECLARE_UAC_AC_HEADER_DESCRIPTOR(2); #define UAC_DT_TOTAL_LENGTH (UAC_DT_AC_HEADER_LENGTH + UAC_DT_INPUT_TERMINAL_SIZE \ + UAC_DT_OUTPUT_TERMINAL_SIZE + UAC_DT_FEATURE_UNIT_SIZE(0)) /* B.3.2 Class-Specific AC Interface Descriptor */ -static struct uac_ac_header_descriptor_v1_2 ac_header_desc = { +static struct uac1_ac_header_descriptor_2 ac_header_desc = { .bLength = UAC_DT_AC_HEADER_LENGTH, .bDescriptorType = USB_DT_CS_INTERFACE, .bDescriptorSubtype = UAC_HEADER, @@ -125,7 +125,7 @@ static struct usb_audio_control_selector feature_unit = { }; #define OUTPUT_TERMINAL_ID 3 -static struct uac_output_terminal_descriptor_v1 output_terminal_desc = { +static struct uac1_output_terminal_descriptor output_terminal_desc = { .bLength = UAC_DT_OUTPUT_TERMINAL_SIZE, .bDescriptorType = USB_DT_CS_INTERFACE, .bDescriptorSubtype = UAC_OUTPUT_TERMINAL, @@ -155,7 +155,7 @@ static struct usb_interface_descriptor as_interface_alt_1_desc = { }; /* B.4.2 Class-Specific AS Interface Descriptor */ -static struct uac_as_header_descriptor_v1 as_header_desc = { +static struct uac1_as_header_descriptor as_header_desc = { .bLength = UAC_DT_AS_HEADER_SIZE, .bDescriptorType = USB_DT_CS_INTERFACE, .bDescriptorSubtype = UAC_AS_GENERAL, diff --git a/drivers/usb/gadget/gmidi.c b/drivers/usb/gadget/gmidi.c index 2b56ce621852..b7bf88019b06 100644 --- a/drivers/usb/gadget/gmidi.c +++ b/drivers/usb/gadget/gmidi.c @@ -238,7 +238,7 @@ static const struct usb_interface_descriptor ac_interface_desc = { }; /* B.3.2 Class-Specific AC Interface Descriptor */ -static const struct uac_ac_header_descriptor_v1_1 ac_header_desc = { +static const struct uac1_ac_header_descriptor_1 ac_header_desc = { .bLength = UAC_DT_AC_HEADER_SIZE(1), .bDescriptorType = USB_DT_CS_INTERFACE, .bDescriptorSubtype = USB_MS_HEADER, diff --git a/include/linux/usb/audio-v2.h b/include/linux/usb/audio-v2.h index 383b94ba8c20..716aebe339e8 100644 --- a/include/linux/usb/audio-v2.h +++ b/include/linux/usb/audio-v2.h @@ -121,7 +121,7 @@ struct uac2_feature_unit_descriptor { /* 4.9.2 Class-Specific AS Interface Descriptor */ -struct uac_as_header_descriptor_v2 { +struct uac2_as_header_descriptor { __u8 bLength; __u8 bDescriptorType; __u8 bDescriptorSubtype; diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index c51200c715e5..a54b8255d75f 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -39,8 +39,8 @@ #define UAC_MIXER_UNIT 0x04 #define UAC_SELECTOR_UNIT 0x05 #define UAC_FEATURE_UNIT 0x06 -#define UAC_PROCESSING_UNIT_V1 0x07 -#define UAC_EXTENSION_UNIT_V1 0x08 +#define UAC1_PROCESSING_UNIT 0x07 +#define UAC1_EXTENSION_UNIT 0x08 /* A.6 Audio Class-Specific AS Interface Descriptor Subtypes */ #define UAC_AS_GENERAL 0x01 @@ -151,7 +151,7 @@ /* Terminal Control Selectors */ /* 4.3.2 Class-Specific AC Interface Descriptor */ -struct uac_ac_header_descriptor_v1 { +struct uac1_ac_header_descriptor { __u8 bLength; /* 8 + n */ __u8 bDescriptorType; /* USB_DT_CS_INTERFACE */ __u8 bDescriptorSubtype; /* UAC_MS_HEADER */ @@ -165,7 +165,7 @@ struct uac_ac_header_descriptor_v1 { /* As above, but more useful for defining your own descriptors: */ #define DECLARE_UAC_AC_HEADER_DESCRIPTOR(n) \ -struct uac_ac_header_descriptor_v1_##n { \ +struct uac1_ac_header_descriptor_##n { \ __u8 bLength; \ __u8 bDescriptorType; \ __u8 bDescriptorSubtype; \ @@ -205,7 +205,7 @@ struct uac_input_terminal_descriptor { #define UAC_TERMINAL_CS_COPY_PROTECT_CONTROL 0x01 /* 4.3.2.2 Output Terminal Descriptor */ -struct uac_output_terminal_descriptor_v1 { +struct uac1_output_terminal_descriptor { __u8 bLength; /* in bytes: 9 */ __u8 bDescriptorType; /* CS_INTERFACE descriptor type */ __u8 bDescriptorSubtype; /* OUTPUT_TERMINAL descriptor subtype */ @@ -395,7 +395,7 @@ static inline __u8 *uac_processing_unit_specific(struct uac_processing_unit_desc } /* 4.5.2 Class-Specific AS Interface Descriptor */ -struct uac_as_header_descriptor_v1 { +struct uac1_as_header_descriptor { __u8 bLength; /* in bytes: 7 */ __u8 bDescriptorType; /* USB_DT_CS_INTERFACE */ __u8 bDescriptorSubtype; /* AS_GENERAL */ diff --git a/sound/usb/card.c b/sound/usb/card.c index 7a8ac1d81be7..9feb00c831a0 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -217,7 +217,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) switch (protocol) { case UAC_VERSION_1: { - struct uac_ac_header_descriptor_v1 *h1 = control_header; + struct uac1_ac_header_descriptor *h1 = control_header; if (!h1->bInCollection) { snd_printk(KERN_INFO "skipping empty audio interface (v1)\n"); diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 6f6596cf2b19..2af0f9e3dcdf 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -275,7 +275,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* get audio formats */ switch (protocol) { case UAC_VERSION_1: { - struct uac_as_header_descriptor_v1 *as = + struct uac1_as_header_descriptor *as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); if (!as) { @@ -297,7 +297,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) case UAC_VERSION_2: { struct uac2_input_terminal_descriptor *input_term; struct uac2_output_terminal_descriptor *output_term; - struct uac_as_header_descriptor_v2 *as = + struct uac2_as_header_descriptor *as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); if (!as) { diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 736d134cc03c..ba54eb6bb0c9 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -582,9 +582,9 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm switch (iterm->type >> 16) { case UAC_SELECTOR_UNIT: strcpy(name, "Selector"); return 8; - case UAC_PROCESSING_UNIT_V1: + case UAC1_PROCESSING_UNIT: strcpy(name, "Process Unit"); return 12; - case UAC_EXTENSION_UNIT_V1: + case UAC1_EXTENSION_UNIT: strcpy(name, "Ext Unit"); return 8; case UAC_MIXER_UNIT: strcpy(name, "Mixer"); return 5; @@ -672,8 +672,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->name = uac_selector_unit_iSelector(d); return 0; } - case UAC_PROCESSING_UNIT_V1: - case UAC_EXTENSION_UNIT_V1: { + case UAC1_PROCESSING_UNIT: + case UAC1_EXTENSION_UNIT: { struct uac_processing_unit_descriptor *d = p1; if (d->bNrInPins) { id = d->baSourceID[0]; @@ -1855,13 +1855,13 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return parse_audio_selector_unit(state, unitid, p1); case UAC_FEATURE_UNIT: return parse_audio_feature_unit(state, unitid, p1); - case UAC_PROCESSING_UNIT_V1: + case UAC1_PROCESSING_UNIT: /* UAC2_EFFECT_UNIT has the same value */ if (state->mixer->protocol == UAC_VERSION_1) return parse_audio_processing_unit(state, unitid, p1); else return 0; /* FIXME - effect units not implemented yet */ - case UAC_EXTENSION_UNIT_V1: + case UAC1_EXTENSION_UNIT: /* UAC2_PROCESSING_UNIT_V2 has the same value */ if (state->mixer->protocol == UAC_VERSION_1) return parse_audio_extension_unit(state, unitid, p1); @@ -1925,7 +1925,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) p = NULL; while ((p = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) { if (mixer->protocol == UAC_VERSION_1) { - struct uac_output_terminal_descriptor_v1 *desc = p; + struct uac1_output_terminal_descriptor *desc = p; if (desc->bLength < sizeof(*desc)) continue; /* invalid descriptor? */ -- cgit v1.2.3 From 157a57b6fae7d3c6d24b7623dcc6679c6d244621 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 16 Jun 2010 17:57:30 +0200 Subject: ALSA: usb-audio: move and add some comments Also add a list of open topics. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- include/linux/usb/audio-v2.h | 15 +++++++++++++++ sound/usb/clock.c | 16 ++++++++++++++-- sound/usb/mixer.c | 24 ++++++++++++++++-------- 3 files changed, 45 insertions(+), 10 deletions(-) (limited to 'include') diff --git a/include/linux/usb/audio-v2.h b/include/linux/usb/audio-v2.h index 716aebe339e8..964cb603f7c7 100644 --- a/include/linux/usb/audio-v2.h +++ b/include/linux/usb/audio-v2.h @@ -18,6 +18,21 @@ /* v1.0 and v2.0 of this standard have many things in common. For the rest * of the definitions, please refer to audio.h */ +/* + * bmControl field decoders + * + * From the USB Audio spec v2.0: + * + * bmaControls() is a (ch+1)-element array of 4-byte bitmaps, + * each containing a set of bit pairs. If a Control is present, + * it must be Host readable. If a certain Control is not + * present then the bit pair must be set to 0b00. + * If a Control is present but read-only, the bit pair must be + * set to 0b01. If a Control is also Host programmable, the bit + * pair must be set to 0b11. The value 0b10 is not allowed. + * + */ + static inline bool uac2_control_is_readable(u32 bmControls, u8 control) { return (bmControls >> (control * 2)) & 0x1; diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 386b09c5ce73..7279d6190875 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -120,8 +120,6 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id) return !!data; } -/* Try to find the clock source ID of a given clock entity */ - static int __uac_clock_find_source(struct snd_usb_audio *chip, struct usb_host_interface *host_iface, int entity_id, unsigned long *visited) @@ -154,6 +152,8 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, if (ret < 0) return ret; + /* Selector values are one-based */ + if (ret > selector->bNrInPins || ret < 1) { printk(KERN_ERR "%s(): selector reported illegal value, id %d, ret %d\n", @@ -176,6 +176,17 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, return -EINVAL; } +/* + * For all kinds of sample rate settings and other device queries, + * the clock source (end-leaf) must be used. However, clock selectors, + * clock multipliers and sample rate converters may be specified as + * clock source input to terminal. This functions walks the clock path + * to its end and tries to find the source. + * + * The 'visited' bitfield is used internally to detect recursive loops. + * + * Returns the clock source UnitID (>=0) on success, or an error. + */ int snd_usb_clock_find_source(struct snd_usb_audio *chip, struct usb_host_interface *host_iface, int entity_id) @@ -246,6 +257,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return clock; if (!uac_clock_source_is_valid(chip, clock)) { + /* TODO: should we try to find valid clock setups by ourself? */ snd_printk(KERN_ERR "%d:%d:%d: clock source %d is not valid, cannot use\n", dev->devnum, iface, fmt->altsetting, clock); return -ENXIO; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ba54eb6bb0c9..1163ec3ca8a0 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -26,6 +26,22 @@ * */ +/* + * TODOs, for both the mixer and the streaming interfaces: + * + * - support for UAC2 effect units + * - support for graphical equalizers + * - RANGE and MEM set commands (UAC2) + * - RANGE and MEM interrupt dispatchers (UAC2) + * - audio channel clustering (UAC2) + * - audio sample rate converter units (UAC2) + * - proper handling of clock multipliers (UAC2) + * - dispatch clock change notifications (UAC2) + * - stop PCM streams which use a clock that became invalid + * - stop PCM streams which use a clock selector that has changed + * - parse available sample rates again when clock sources changed + */ + #include #include #include @@ -1199,14 +1215,6 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void } } else { /* UAC_VERSION_2 */ for (i = 0; i < 30/2; i++) { - /* From the USB Audio spec v2.0: - bmaControls() is a (ch+1)-element array of 4-byte bitmaps, - each containing a set of bit pairs. If a Control is present, - it must be Host readable. If a certain Control is not - present then the bit pair must be set to 0b00. - If a Control is present but read-only, the bit pair must be - set to 0b01. If a Control is also Host programmable, the bit - pair must be set to 0b11. The value 0b10 is not allowed. */ unsigned int ch_bits = 0; unsigned int ch_read_only = 0; -- cgit v1.2.3 From 5daeba34d2aab669aea07abee13d53cd116578fb Mon Sep 17 00:00:00 2001 From: David Dillow Date: Sun, 27 Jun 2010 00:13:20 +0200 Subject: ALSA: pcm_lib: avoid timing jitter in snd_pcm_read/write() When using poll() to wait for the next period -- or avail_min samples -- one gets a consistent delay for each system call that is usually just a little short of the selected period time. However, When using snd_pcm_read/write(), one gets a jittery delay that alternates between less than a millisecond and approximately two period times. This is caused by snd_pcm_lib_{read,write}1() transferring any available samples to the user's buffer and adjusting the application pointer prior to sleeping to the end of the current period. When the next period interrupt occurs, there is then less than avail_min samples remaining to be transferred in the period, so we end up sleeping until a second period occurs. This is solved by using runtime->twake as the number of samples needed for a wakeup in addition to selecting the proper wait queue to wake in snd_pcm_update_state(). This requires twake to be non-zero when used by snd_pcm_lib_{read,write}1() even if avail_min is zero. Signed-off-by: Dave Dillow Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 2 +- sound/core/pcm_lib.c | 23 +++++++++++++++-------- 2 files changed, 16 insertions(+), 9 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index dd76cdede64d..83c6fa6aac43 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -313,7 +313,7 @@ struct snd_pcm_runtime { struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ - unsigned int twake: 1; /* do transfer (!poll) wakeup */ + snd_pcm_uframes_t twake; /* do transfer (!poll) wakeup if non-zero */ wait_queue_head_t sleep; /* poll sleep */ wait_queue_head_t tsleep; /* transfer sleep */ struct fasync_struct *fasync; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index e9d98be190c5..bcf95d3ff5c7 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -287,8 +287,11 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return -EPIPE; } } - if (avail >= runtime->control->avail_min) - wake_up(runtime->twake ? &runtime->tsleep : &runtime->sleep); + if (runtime->twake) { + if (avail >= runtime->twake) + wake_up(&runtime->tsleep); + } else if (avail >= runtime->control->avail_min) + wake_up(&runtime->sleep); return 0; } @@ -1707,7 +1710,7 @@ EXPORT_SYMBOL(snd_pcm_period_elapsed); * The available space is stored on availp. When err = 0 and avail = 0 * on the capture stream, it indicates the stream is in DRAINING state. */ -static int wait_for_avail_min(struct snd_pcm_substream *substream, +static int wait_for_avail(struct snd_pcm_substream *substream, snd_pcm_uframes_t *availp) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -1757,7 +1760,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, avail = snd_pcm_playback_avail(runtime); else avail = snd_pcm_capture_avail(runtime); - if (avail >= runtime->control->avail_min) + if (avail >= runtime->twake) break; } _endloop: @@ -1820,7 +1823,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->twake = 1; + runtime->twake = runtime->control->avail_min ? : 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -1833,7 +1836,9 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, err = -EAGAIN; goto _end_unlock; } - err = wait_for_avail_min(substream, &avail); + runtime->twake = min_t(snd_pcm_uframes_t, size, + runtime->control->avail_min ? : 1); + err = wait_for_avail(substream, &avail); if (err < 0) goto _end_unlock; } @@ -2042,7 +2047,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->twake = 1; + runtime->twake = runtime->control->avail_min ? : 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -2060,7 +2065,9 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, err = -EAGAIN; goto _end_unlock; } - err = wait_for_avail_min(substream, &avail); + runtime->twake = min_t(snd_pcm_uframes_t, size, + runtime->control->avail_min ? : 1); + err = wait_for_avail(substream, &avail); if (err < 0) goto _end_unlock; if (!avail) -- cgit v1.2.3