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Split audio player

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The audio player had 2 roles:
 - handle the SDL audio output device;
 - resample input samples to maintain a target latency.

Extract the latter to a separate component (an "audio regulator"),
independent of SDL.
This commit is contained in:
Romain Vimont 2024-09-23 23:58:02 +02:00
parent 62776fb261
commit 0bb3955b95
5 changed files with 507 additions and 440 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
app/meson.build
View File

@ -5,6 +5,7 @@ src = [
'src/adb/adb_parser.c',
'src/adb/adb_tunnel.c',
'src/audio_player.c',
'src/audio_regulator.c',
'src/cli.c',
'src/clock.c',
'src/compat.c',

413
app/src/audio_player.c
View File

@ -1,143 +1,23 @@
#include "audio_player.h"
#include <libavcodec/avcodec.h>
#include <libavutil/opt.h>
#include "util/log.h"
//#define SC_AUDIO_PLAYER_DEBUG // uncomment to debug
/**
* Real-time audio player with configurable latency
*
* As input, the player regularly receives AVFrames of decoded audio samples.
* As output, an SDL callback regularly requests audio samples to be played.
* In the middle, an audio buffer stores the samples produced but not consumed
* yet.
*
* The goal of the player is to feed the audio output with a latency as low as
* possible while avoiding buffer underrun (i.e. not being able to provide
* samples when requested).
*
* The player aims to feed the audio output with as little latency as possible
* while avoiding buffer underrun. To achieve this, it attempts to maintain the
* average buffering (the number of samples present in the buffer) around a
* target value. If this target buffering is too low, then buffer underrun will
* occur frequently. If it is too high, then latency will become unacceptable.
* This target value is configured using the scrcpy option --audio-buffer.
*
* The player cannot adjust the sample input rate (it receives samples produced
* in real-time) or the sample output rate (it must provide samples as
* requested by the audio output callback). Therefore, it may only apply
* compensation by resampling (converting _m_ input samples to _n_ output
* samples).
*
* The compensation itself is applied by libswresample (FFmpeg). It is
* configured using swr_set_compensation(). An important work for the player
* is to estimate the compensation value regularly and apply it.
*
* The estimated buffering level is the result of averaging the "natural"
* buffering (samples are produced and consumed by blocks, so it must be
* smoothed), and making instant adjustments resulting of its own actions
* (explicit compensation and silence insertion on underflow), which are not
* smoothed.
*
* Buffer underflow events can occur when packets arrive too late. In that case,
* the player inserts silence. Once the packets finally arrive (late), one
* strategy could be to drop the samples that were replaced by silence, in
* order to keep a minimal latency. However, dropping samples in case of buffer
* underflow is inadvisable, as it would temporarily increase the underflow
* even more and cause very noticeable audio glitches.
*
* Therefore, the player doesn't drop any sample on underflow. The compensation
* mechanism will absorb the delay introduced by the inserted silence.
*/
/** Downcast frame_sink to sc_audio_player */
#define DOWNCAST(SINK) container_of(SINK, struct sc_audio_player, frame_sink)
#define SC_AV_SAMPLE_FMT AV_SAMPLE_FMT_FLT
#define SC_SDL_SAMPLE_FMT AUDIO_F32
#define TO_BYTES(SAMPLES) sc_audiobuf_to_bytes(&ap->buf, (SAMPLES))
#define TO_SAMPLES(BYTES) sc_audiobuf_to_samples(&ap->buf, (BYTES))
static void SDLCALL
sc_audio_player_sdl_callback(void *userdata, uint8_t *stream, int len_int) {
struct sc_audio_player *ap = userdata;
assert(len_int > 0);
size_t len = len_int;
uint32_t count = TO_SAMPLES(len);
#ifdef SC_AUDIO_PLAYER_DEBUG
LOGD("[Audio] SDL callback requests %" PRIu32 " samples", count);
#endif
assert(len % ap->audioreg.sample_size == 0);
uint32_t out_samples = len / ap->audioreg.sample_size;
// A lock is necessary in the rare case where the producer needs to drop
// samples already pushed (when the buffer is full)
sc_mutex_lock(&ap->mutex);
bool played = atomic_load_explicit(&ap->played, memory_order_relaxed);
if (!played) {
uint32_t buffered_samples = sc_audiobuf_can_read(&ap->buf);
// Wait until the buffer is filled up to at least target_buffering
// before playing
if (buffered_samples < ap->target_buffering) {
LOGV("[Audio] Inserting initial buffering silence: %" PRIu32
" samples", count);
// Delay playback starting to reach the target buffering. Fill the
// whole buffer with silence (len is small compared to the
// arbitrary margin value).
memset(stream, 0, len);
sc_mutex_unlock(&ap->mutex);
return;
}
}
uint32_t read = sc_audiobuf_read(&ap->buf, stream, count);
sc_mutex_unlock(&ap->mutex);
if (read < count) {
uint32_t silence = count - read;
// Insert silence. In theory, the inserted silent samples replace the
// missing real samples, which will arrive later, so they should be
// dropped to keep the latency minimal. However, this would cause very
// audible glitches, so let the clock compensation restore the target
// latency.
LOGD("[Audio] Buffer underflow, inserting silence: %" PRIu32 " samples",
silence);
memset(stream + TO_BYTES(read), 0, TO_BYTES(silence));
bool received = atomic_load_explicit(&ap->received,
memory_order_relaxed);
if (received) {
// Inserting additional samples immediately increases buffering
atomic_fetch_add_explicit(&ap->underflow, silence,
memory_order_relaxed);
}
}
atomic_store_explicit(&ap->played, true, memory_order_relaxed);
}
static uint8_t *
sc_audio_player_get_swr_buf(struct sc_audio_player *ap, uint32_t min_samples) {
size_t min_buf_size = TO_BYTES(min_samples);
if (min_buf_size > ap->swr_buf_alloc_size) {
size_t new_size = min_buf_size + 4096;
uint8_t *buf = realloc(ap->swr_buf, new_size);
if (!buf) {
LOG_OOM();
// Could not realloc to the requested size
return NULL;
}
ap->swr_buf = buf;
ap->swr_buf_alloc_size = new_size;
}
return ap->swr_buf;
sc_audio_regulator_pull(&ap->audioreg, stream, out_samples);
}
static bool
@ -145,202 +25,14 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
const AVFrame *frame) {
struct sc_audio_player *ap = DOWNCAST(sink);
SwrContext *swr_ctx = ap->swr_ctx;
int64_t swr_delay = swr_get_delay(swr_ctx, ap->sample_rate);
// No need to av_rescale_rnd(), input and output sample rates are the same.
// Add more space (256) for clock compensation.
int dst_nb_samples = swr_delay + frame->nb_samples + 256;
uint8_t *swr_buf = sc_audio_player_get_swr_buf(ap, dst_nb_samples);
if (!swr_buf) {
return false;
}
int ret = swr_convert(swr_ctx, &swr_buf, dst_nb_samples,
(const uint8_t **) frame->data, frame->nb_samples);
if (ret < 0) {
LOGE("Resampling failed: %d", ret);
return false;
}
// swr_convert() returns the number of samples which would have been
// written if the buffer was big enough.
uint32_t samples = MIN(ret, dst_nb_samples);
#ifdef SC_AUDIO_PLAYER_DEBUG
LOGD("[Audio] %" PRIu32 " samples written to buffer", samples);
#endif
uint32_t cap = sc_audiobuf_capacity(&ap->buf);
if (samples > cap) {
// Very very unlikely: a single resampled frame should never
// exceed the audio buffer size (or something is very wrong).
// Ignore the first bytes in swr_buf to avoid memory corruption anyway.
swr_buf += TO_BYTES(samples - cap);
samples = cap;
}
uint32_t skipped_samples = 0;
uint32_t written = sc_audiobuf_write(&ap->buf, swr_buf, samples);
if (written < samples) {
uint32_t remaining = samples - written;
// All samples that could be written without locking have been written,
// now we need to lock to drop/consume old samples
sc_mutex_lock(&ap->mutex);
// Retry with the lock
written += sc_audiobuf_write(&ap->buf,
swr_buf + TO_BYTES(written),
remaining);
if (written < samples) {
remaining = samples - written;
// Still insufficient, drop old samples to make space
skipped_samples = sc_audiobuf_read(&ap->buf, NULL, remaining);
assert(skipped_samples == remaining);
}
sc_mutex_unlock(&ap->mutex);
if (written < samples) {
// Now there is enough space
uint32_t w = sc_audiobuf_write(&ap->buf,
swr_buf + TO_BYTES(written),
remaining);
assert(w == remaining);
(void) w;
}
}
uint32_t underflow = 0;
uint32_t max_buffered_samples;
bool played = atomic_load_explicit(&ap->played, memory_order_relaxed);
if (played) {
underflow = atomic_exchange_explicit(&ap->underflow, 0,
memory_order_relaxed);
max_buffered_samples = ap->target_buffering * 11 / 10
+ 60 * ap->sample_rate / 1000 /* 60 ms */;
} else {
// SDL playback not started yet, do not accumulate more than
// max_initial_buffering samples, this would cause unnecessary delay
// (and glitches to compensate) on start.
max_buffered_samples = ap->target_buffering
+ 10 * ap->sample_rate / 1000 /* 10 ms */;
}
uint32_t can_read = sc_audiobuf_can_read(&ap->buf);
if (can_read > max_buffered_samples) {
uint32_t skip_samples = 0;
sc_mutex_lock(&ap->mutex);
can_read = sc_audiobuf_can_read(&ap->buf);
if (can_read > max_buffered_samples) {
skip_samples = can_read - max_buffered_samples;
uint32_t r = sc_audiobuf_read(&ap->buf, NULL, skip_samples);
assert(r == skip_samples);
(void) r;
skipped_samples += skip_samples;
}
sc_mutex_unlock(&ap->mutex);
if (skip_samples) {
if (played) {
LOGD("[Audio] Buffering threshold exceeded, skipping %" PRIu32
" samples", skip_samples);
#ifdef SC_AUDIO_PLAYER_DEBUG
} else {
LOGD("[Audio] Playback not started, skipping %" PRIu32
" samples", skip_samples);
#endif
}
}
}
atomic_store_explicit(&ap->received, true, memory_order_relaxed);
if (!played) {
// Nothing more to do
return true;
}
// Number of samples added (or removed, if negative) for compensation
int32_t instant_compensation = (int32_t) written - frame->nb_samples;
// Inserting silence instantly increases buffering
int32_t inserted_silence = (int32_t) underflow;
// Dropping input samples instantly decreases buffering
int32_t dropped = (int32_t) skipped_samples;
// The compensation must apply instantly, it must not be smoothed
ap->avg_buffering.avg += instant_compensation + inserted_silence - dropped;
if (ap->avg_buffering.avg < 0) {
// Since dropping samples instantly reduces buffering, the difference
// is applied immediately to the average value, assuming that the delay
// between the producer and the consumer will be caught up.
//
// However, when this assumption is not valid, the average buffering
// may decrease indefinitely. Prevent it to become negative to limit
// the consequences.
ap->avg_buffering.avg = 0;
}
// However, the buffering level must be smoothed
sc_average_push(&ap->avg_buffering, can_read);
#ifdef SC_AUDIO_PLAYER_DEBUG
LOGD("[Audio] can_read=%" PRIu32 " avg_buffering=%f",
can_read, sc_average_get(&ap->avg_buffering));
#endif
ap->samples_since_resync += written;
if (ap->samples_since_resync >= ap->sample_rate) {
// Recompute compensation every second
ap->samples_since_resync = 0;
float avg = sc_average_get(&ap->avg_buffering);
int diff = ap->target_buffering - avg;
// Enable compensation when the difference exceeds +/- 4ms.
// Disable compensation when the difference is lower than +/- 1ms.
int threshold = ap->compensation != 0
? ap->sample_rate / 1000 /* 1ms */
: ap->sample_rate * 4 / 1000; /* 4ms */
if (abs(diff) < threshold) {
// Do not compensate for small values, the error is just noise
diff = 0;
} else if (diff < 0 && can_read < ap->target_buffering) {
// Do not accelerate if the instant buffering level is below the
// target, this would increase underflow
diff = 0;
}
// Compensate the diff over 4 seconds (but will be recomputed after 1
// second)
int distance = 4 * ap->sample_rate;
// Limit compensation rate to 2%
int abs_max_diff = distance / 50;
diff = CLAMP(diff, -abs_max_diff, abs_max_diff);
LOGV("[Audio] Buffering: target=%" PRIu32 " avg=%f cur=%" PRIu32
" compensation=%d", ap->target_buffering, avg, can_read, diff);
if (diff != ap->compensation) {
int ret = swr_set_compensation(swr_ctx, diff, distance);
if (ret < 0) {
LOGW("Resampling compensation failed: %d", ret);
// not fatal
} else {
ap->compensation = diff;
}
}
}
return true;
return sc_audio_regulator_push(&ap->audioreg, frame);
}
static bool
sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
const AVCodecContext *ctx) {
struct sc_audio_player *ap = DOWNCAST(sink);
#ifdef SCRCPY_LAVU_HAS_CHLAYOUT
assert(ctx->ch_layout.nb_channels > 0 && ctx->ch_layout.nb_channels < 256);
uint8_t nb_channels = ctx->ch_layout.nb_channels;
@ -355,12 +47,17 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
int out_bytes_per_sample = av_get_bytes_per_sample(SC_AV_SAMPLE_FMT);
assert(out_bytes_per_sample > 0);
ap->sample_rate = ctx->sample_rate;
uint32_t target_buffering_samples =
ap->target_buffering_delay * ctx->sample_rate / SC_TICK_FREQ;
ap->target_buffering = ap->target_buffering_delay * ap->sample_rate
/ SC_TICK_FREQ;
size_t sample_size = nb_channels * out_bytes_per_sample;
bool ok = sc_audio_regulator_init(&ap->audioreg, sample_size, ctx,
target_buffering_samples);
if (!ok) {
return false;
}
uint64_t aout_samples = ap->output_buffer_duration * ap->sample_rate
uint64_t aout_samples = ap->output_buffer_duration * ctx->sample_rate
/ SC_TICK_FREQ;
assert(aout_samples <= 0xFFFF);
@ -377,74 +74,10 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
ap->device = SDL_OpenAudioDevice(NULL, 0, &desired, &obtained, 0);
if (!ap->device) {
LOGE("Could not open audio device: %s", SDL_GetError());
sc_audio_regulator_destroy(&ap->audioreg);
return false;
}
SwrContext *swr_ctx = swr_alloc();
if (!swr_ctx) {
LOG_OOM();
goto error_close_audio_device;
}
ap->swr_ctx = swr_ctx;
#ifdef SCRCPY_LAVU_HAS_CHLAYOUT
av_opt_set_chlayout(swr_ctx, "in_chlayout", &ctx->ch_layout, 0);
av_opt_set_chlayout(swr_ctx, "out_chlayout", &ctx->ch_layout, 0);
#else
av_opt_set_channel_layout(swr_ctx, "in_channel_layout",
ctx->channel_layout, 0);
av_opt_set_channel_layout(swr_ctx, "out_channel_layout",
ctx->channel_layout, 0);
#endif
av_opt_set_int(swr_ctx, "in_sample_rate", ctx->sample_rate, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", ctx->sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", ctx->sample_fmt, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", SC_AV_SAMPLE_FMT, 0);
int ret = swr_init(swr_ctx);
if (ret) {
LOGE("Failed to initialize the resampling context");
goto error_free_swr_ctx;
}
// Use a ring-buffer of the target buffering size plus 1 second between the
// producer and the consumer. It's too big on purpose, to guarantee that
// the producer and the consumer will be able to access it in parallel
// without locking.
uint32_t audiobuf_samples = ap->target_buffering + ap->sample_rate;
bool ok = sc_mutex_init(&ap->mutex);
if (!ok) {
goto error_free_swr_ctx;
}
size_t sample_size = nb_channels * out_bytes_per_sample;
ok = sc_audiobuf_init(&ap->buf, sample_size, audiobuf_samples);
if (!ok) {
goto error_destroy_mutex;
}
size_t initial_swr_buf_size = TO_BYTES(4096);
ap->swr_buf = malloc(initial_swr_buf_size);
if (!ap->swr_buf) {
LOG_OOM();
goto error_destroy_audiobuf;
}
ap->swr_buf_alloc_size = initial_swr_buf_size;
// Samples are produced and consumed by blocks, so the buffering must be
// smoothed to get a relatively stable value.
sc_average_init(&ap->avg_buffering, 128);
ap->samples_since_resync = 0;
ap->received = false;
atomic_init(&ap->played, false);
atomic_init(&ap->received, false);
atomic_init(&ap->underflow, 0);
ap->compensation = 0;
// The thread calling open() is the thread calling push(), which fills the
// audio buffer consumed by the SDL audio thread.
ok = sc_thread_set_priority(SC_THREAD_PRIORITY_TIME_CRITICAL);
@ -456,17 +89,6 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
SDL_PauseAudioDevice(ap->device, 0);
return true;
error_destroy_audiobuf:
sc_audiobuf_destroy(&ap->buf);
error_destroy_mutex:
sc_mutex_destroy(&ap->mutex);
error_free_swr_ctx:
swr_free(&ap->swr_ctx);
error_close_audio_device:
SDL_CloseAudioDevice(ap->device);
return false;
}
static void
@ -477,10 +99,7 @@ sc_audio_player_frame_sink_close(struct sc_frame_sink *sink) {
SDL_PauseAudioDevice(ap->device, 1);
SDL_CloseAudioDevice(ap->device);
free(ap->swr_buf);
sc_audiobuf_destroy(&ap->buf);
sc_mutex_destroy(&ap->mutex);
swr_free(&ap->swr_ctx);
sc_audio_regulator_destroy(&ap->audioreg);
}
void

47
app/src/audio_player.h
View File

@ -5,66 +5,27 @@
#include <stdatomic.h>
#include <stdbool.h>
#include <libavformat/avformat.h>
#include <libswresample/swresample.h>
#include <SDL2/SDL.h>
#include "audio_regulator.h"
#include "trait/frame_sink.h"
#include "util/audiobuf.h"
#include "util/average.h"
#include "util/thread.h"
#include "util/tick.h"
struct sc_audio_player {
struct sc_frame_sink frame_sink;
SDL_AudioDeviceID device;
sc_mutex mutex;
// The target buffering between the producer and the consumer. This value
// is directly use for compensation.
// Since audio capture and/or encoding on the device typically produce
// blocks of 960 samples (20ms) or 1024 samples (~21.3ms), this target
// value should be higher.
sc_tick target_buffering_delay;
uint32_t target_buffering; // in samples
// SDL audio output buffer size.
// SDL audio output buffer size
sc_tick output_buffer_duration;
// Audio buffer to communicate between the receiver and the SDL audio
// callback
struct sc_audiobuf buf;
// Resampler (only used from the receiver thread)
struct SwrContext *swr_ctx;
// The sample rate is the same for input and output
uint32_t sample_rate;
// Target buffer for resampling (only used by the receiver thread)
uint8_t *swr_buf;
size_t swr_buf_alloc_size;
// Number of buffered samples (may be negative on underflow) (only used by
// the receiver thread)
struct sc_average avg_buffering;
// Count the number of samples to trigger a compensation update regularly
// (only used by the receiver thread)
uint32_t samples_since_resync;
// Number of silence samples inserted since the last received packet
atomic_uint_least32_t underflow;
// Current applied compensation value (only used by the receiver thread)
int compensation;
// Set to true the first time a sample is received
atomic_bool received;
// Set to true the first time the SDL callback is called
atomic_bool played;
SDL_AudioDeviceID device;
struct sc_audio_regulator audioreg;
};
void

415
app/src/audio_regulator.c Normal file
View File

@ -0,0 +1,415 @@
#include "audio_regulator.h"
#include <libavcodec/avcodec.h>
#include <libavutil/opt.h>
#include "util/log.h"
//#define SC_AUDIO_REGULATOR_DEBUG // uncomment to debug
/**
* Real-time audio regulator with configurable latency
*
* As input, the regulator regularly receives AVFrames of decoded audio samples.
* As output, the audio player regularly requests audio samples to be played.
* In the middle, an audio buffer stores the samples produced but not consumed
* yet.
*
* The goal of the regulator is to feed the audio player with a latency as low
* as possible while avoiding buffer underrun (i.e. not being able to provide
* samples when requested).
*
* To achieve this, it attempts to maintain the average buffering (the number
* of samples present in the buffer) around a target value. If this target
* buffering is too low, then buffer underrun will occur frequently. If it is
* too high, then latency will become unacceptable. This target value is
* configured using the scrcpy option --audio-buffer.
*
* The regulator cannot adjust the sample input rate (it receives samples
* produced in real-time) or the sample output rate (it must provide samples as
* requested by the audio player). Therefore, it may only apply compensation by
* resampling (converting _m_ input samples to _n_ output samples).
*
* The compensation itself is applied by libswresample (FFmpeg). It is
* configured using swr_set_compensation(). An important work for the regulator
* is to estimate the compensation value regularly and apply it.
*
* The estimated buffering level is the result of averaging the "natural"
* buffering (samples are produced and consumed by blocks, so it must be
* smoothed), and making instant adjustments resulting of its own actions
* (explicit compensation and silence insertion on underflow), which are not
* smoothed.
*
* Buffer underflow events can occur when packets arrive too late. In that case,
* the regulator inserts silence. Once the packets finally arrive (late), one
* strategy could be to drop the samples that were replaced by silence, in
* order to keep a minimal latency. However, dropping samples in case of buffer
* underflow is inadvisable, as it would temporarily increase the underflow
* even more and cause very noticeable audio glitches.
*
* Therefore, the regulator doesn't drop any sample on underflow. The
* compensation mechanism will absorb the delay introduced by the inserted
* silence.
*/
#define TO_BYTES(SAMPLES) sc_audiobuf_to_bytes(&ar->buf, (SAMPLES))
#define TO_SAMPLES(BYTES) sc_audiobuf_to_samples(&ar->buf, (BYTES))
void
sc_audio_regulator_pull(struct sc_audio_regulator *ar, uint8_t *out,
uint32_t out_samples) {
#ifdef SC_AUDIO_REGULATOR_DEBUG
LOGD("[Audio] Audio regulator pulls %" PRIu32 " samples", out_samples);
#endif
// A lock is necessary in the rare case where the producer needs to drop
// samples already pushed (when the buffer is full)
sc_mutex_lock(&ar->mutex);
bool played = atomic_load_explicit(&ar->played, memory_order_relaxed);
if (!played) {
uint32_t buffered_samples = sc_audiobuf_can_read(&ar->buf);
// Wait until the buffer is filled up to at least target_buffering
// before playing
if (buffered_samples < ar->target_buffering) {
LOGV("[Audio] Inserting initial buffering silence: %" PRIu32
" samples", out_samples);
// Delay playback starting to reach the target buffering. Fill the
// whole buffer with silence (len is small compared to the
// arbitrary margin value).
memset(out, 0, out_samples * ar->sample_size);
sc_mutex_unlock(&ar->mutex);
return;
}
}
uint32_t read = sc_audiobuf_read(&ar->buf, out, out_samples);
sc_mutex_unlock(&ar->mutex);
if (read < out_samples) {
uint32_t silence = out_samples - read;
// Insert silence. In theory, the inserted silent samples replace the
// missing real samples, which will arrive later, so they should be
// dropped to keep the latency minimal. However, this would cause very
// audible glitches, so let the clock compensation restore the target
// latency.
LOGD("[Audio] Buffer underflow, inserting silence: %" PRIu32 " samples",
silence);
memset(out + TO_BYTES(read), 0, TO_BYTES(silence));
bool received = atomic_load_explicit(&ar->received,
memory_order_relaxed);
if (received) {
// Inserting additional samples immediately increases buffering
atomic_fetch_add_explicit(&ar->underflow, silence,
memory_order_relaxed);
}
}
atomic_store_explicit(&ar->played, true, memory_order_relaxed);
}
static uint8_t *
sc_audio_regulator_get_swr_buf(struct sc_audio_regulator *ar,
uint32_t min_samples) {
size_t min_buf_size = TO_BYTES(min_samples);
if (min_buf_size > ar->swr_buf_alloc_size) {
size_t new_size = min_buf_size + 4096;
uint8_t *buf = realloc(ar->swr_buf, new_size);
if (!buf) {
LOG_OOM();
// Could not realloc to the requested size
return NULL;
}
ar->swr_buf = buf;
ar->swr_buf_alloc_size = new_size;
}
return ar->swr_buf;
}
bool
sc_audio_regulator_push(struct sc_audio_regulator *ar, const AVFrame *frame) {
SwrContext *swr_ctx = ar->swr_ctx;
int64_t swr_delay = swr_get_delay(swr_ctx, ar->sample_rate);
// No need to av_rescale_rnd(), input and output sample rates are the same.
// Add more space (256) for clock compensation.
int dst_nb_samples = swr_delay + frame->nb_samples + 256;
uint8_t *swr_buf = sc_audio_regulator_get_swr_buf(ar, dst_nb_samples);
if (!swr_buf) {
return false;
}
int ret = swr_convert(swr_ctx, &swr_buf, dst_nb_samples,
(const uint8_t **) frame->data, frame->nb_samples);
if (ret < 0) {
LOGE("Resampling failed: %d", ret);
return false;
}
// swr_convert() returns the number of samples which would have been
// written if the buffer was big enough.
uint32_t samples = MIN(ret, dst_nb_samples);
#ifdef SC_AUDIO_REGULATOR_DEBUG
LOGD("[Audio] %" PRIu32 " samples written to buffer", samples);
#endif
uint32_t cap = sc_audiobuf_capacity(&ar->buf);
if (samples > cap) {
// Very very unlikely: a single resampled frame should never
// exceed the audio buffer size (or something is very wrong).
// Ignore the first bytes in swr_buf to avoid memory corruption anyway.
swr_buf += TO_BYTES(samples - cap);
samples = cap;
}
uint32_t skipped_samples = 0;
uint32_t written = sc_audiobuf_write(&ar->buf, swr_buf, samples);
if (written < samples) {
uint32_t remaining = samples - written;
// All samples that could be written without locking have been written,
// now we need to lock to drop/consume old samples
sc_mutex_lock(&ar->mutex);
// Retry with the lock
written += sc_audiobuf_write(&ar->buf,
swr_buf + TO_BYTES(written),
remaining);
if (written < samples) {
remaining = samples - written;
// Still insufficient, drop old samples to make space
skipped_samples = sc_audiobuf_read(&ar->buf, NULL, remaining);
assert(skipped_samples == remaining);
}
sc_mutex_unlock(&ar->mutex);
if (written < samples) {
// Now there is enough space
uint32_t w = sc_audiobuf_write(&ar->buf,
swr_buf + TO_BYTES(written),
remaining);
assert(w == remaining);
(void) w;
}
}
uint32_t underflow = 0;
uint32_t max_buffered_samples;
bool played = atomic_load_explicit(&ar->played, memory_order_relaxed);
if (played) {
underflow = atomic_exchange_explicit(&ar->underflow, 0,
memory_order_relaxed);
max_buffered_samples = ar->target_buffering * 11 / 10
+ 60 * ar->sample_rate / 1000 /* 60 ms */;
} else {
// Playback not started yet, do not accumulate more than
// max_initial_buffering samples, this would cause unnecessary delay
// (and glitches to compensate) on start.
max_buffered_samples = ar->target_buffering
+ 10 * ar->sample_rate / 1000 /* 10 ms */;
}
uint32_t can_read = sc_audiobuf_can_read(&ar->buf);
if (can_read > max_buffered_samples) {
uint32_t skip_samples = 0;
sc_mutex_lock(&ar->mutex);
can_read = sc_audiobuf_can_read(&ar->buf);
if (can_read > max_buffered_samples) {
skip_samples = can_read - max_buffered_samples;
uint32_t r = sc_audiobuf_read(&ar->buf, NULL, skip_samples);
assert(r == skip_samples);
(void) r;
skipped_samples += skip_samples;
}
sc_mutex_unlock(&ar->mutex);
if (skip_samples) {
if (played) {
LOGD("[Audio] Buffering threshold exceeded, skipping %" PRIu32
" samples", skip_samples);
#ifdef SC_AUDIO_REGULATOR_DEBUG
} else {
LOGD("[Audio] Playback not started, skipping %" PRIu32
" samples", skip_samples);
#endif
}
}
}
atomic_store_explicit(&ar->received, true, memory_order_relaxed);
if (!played) {
// Nothing more to do
return true;
}
// Number of samples added (or removed, if negative) for compensation
int32_t instant_compensation = (int32_t) written - frame->nb_samples;
// Inserting silence instantly increases buffering
int32_t inserted_silence = (int32_t) underflow;
// Dropping input samples instantly decreases buffering
int32_t dropped = (int32_t) skipped_samples;
// The compensation must apply instantly, it must not be smoothed
ar->avg_buffering.avg += instant_compensation + inserted_silence - dropped;
if (ar->avg_buffering.avg < 0) {
// Since dropping samples instantly reduces buffering, the difference
// is applied immediately to the average value, assuming that the delay
// between the producer and the consumer will be caught up.
//
// However, when this assumption is not valid, the average buffering
// may decrease indefinitely. Prevent it to become negative to limit
// the consequences.
ar->avg_buffering.avg = 0;
}
// However, the buffering level must be smoothed
sc_average_push(&ar->avg_buffering, can_read);
#ifdef SC_AUDIO_REGULATOR_DEBUG
LOGD("[Audio] can_read=%" PRIu32 " avg_buffering=%f",
can_read, sc_average_get(&ar->avg_buffering));
#endif
ar->samples_since_resync += written;
if (ar->samples_since_resync >= ar->sample_rate) {
// Recompute compensation every second
ar->samples_since_resync = 0;
float avg = sc_average_get(&ar->avg_buffering);
int diff = ar->target_buffering - avg;
// Enable compensation when the difference exceeds +/- 4ms.
// Disable compensation when the difference is lower than +/- 1ms.
int threshold = ar->compensation != 0
? ar->sample_rate / 1000 /* 1ms */
: ar->sample_rate * 4 / 1000; /* 4ms */
if (abs(diff) < threshold) {
// Do not compensate for small values, the error is just noise
diff = 0;
} else if (diff < 0 && can_read < ar->target_buffering) {
// Do not accelerate if the instant buffering level is below the
// target, this would increase underflow
diff = 0;
}
// Compensate the diff over 4 seconds (but will be recomputed after 1
// second)
int distance = 4 * ar->sample_rate;
// Limit compensation rate to 2%
int abs_max_diff = distance / 50;
diff = CLAMP(diff, -abs_max_diff, abs_max_diff);
LOGV("[Audio] Buffering: target=%" PRIu32 " avg=%f cur=%" PRIu32
" compensation=%d", ar->target_buffering, avg, can_read, diff);
if (diff != ar->compensation) {
int ret = swr_set_compensation(swr_ctx, diff, distance);
if (ret < 0) {
LOGW("Resampling compensation failed: %d", ret);
// not fatal
} else {
ar->compensation = diff;
}
}
}
return true;
}
bool
sc_audio_regulator_init(struct sc_audio_regulator *ar, size_t sample_size,
const AVCodecContext *ctx, uint32_t target_buffering) {
SwrContext *swr_ctx = swr_alloc();
if (!swr_ctx) {
LOG_OOM();
return false;
}
ar->swr_ctx = swr_ctx;
#ifdef SCRCPY_LAVU_HAS_CHLAYOUT
av_opt_set_chlayout(swr_ctx, "in_chlayout", &ctx->ch_layout, 0);
av_opt_set_chlayout(swr_ctx, "out_chlayout", &ctx->ch_layout, 0);
#else
av_opt_set_channel_layout(swr_ctx, "in_channel_layout",
ctx->channel_layout, 0);
av_opt_set_channel_layout(swr_ctx, "out_channel_layout",
ctx->channel_layout, 0);
#endif
av_opt_set_int(swr_ctx, "in_sample_rate", ctx->sample_rate, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", ctx->sample_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", ctx->sample_fmt, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", SC_AV_SAMPLE_FMT, 0);
int ret = swr_init(swr_ctx);
if (ret) {
LOGE("Failed to initialize the resampling context");
goto error_free_swr_ctx;
}
bool ok = sc_mutex_init(&ar->mutex);
if (!ok) {
goto error_free_swr_ctx;
}
ar->target_buffering = target_buffering;
ar->sample_size = sample_size;
ar->sample_rate = ctx->sample_rate;
// Use a ring-buffer of the target buffering size plus 1 second between the
// producer and the consumer. It's too big on purpose, to guarantee that
// the producer and the consumer will be able to access it in parallel
// without locking.
uint32_t audiobuf_samples = target_buffering + ar->sample_rate;
ok = sc_audiobuf_init(&ar->buf, sample_size, audiobuf_samples);
if (!ok) {
goto error_destroy_mutex;
}
size_t initial_swr_buf_size = TO_BYTES(4096);
ar->swr_buf = malloc(initial_swr_buf_size);
if (!ar->swr_buf) {
LOG_OOM();
goto error_destroy_audiobuf;
}
ar->swr_buf_alloc_size = initial_swr_buf_size;
// Samples are produced and consumed by blocks, so the buffering must be
// smoothed to get a relatively stable value.
sc_average_init(&ar->avg_buffering, 128);
ar->samples_since_resync = 0;
ar->received = false;
atomic_init(&ar->played, false);
atomic_init(&ar->received, false);
atomic_init(&ar->underflow, 0);
ar->compensation = 0;
return true;
error_destroy_audiobuf:
sc_audiobuf_destroy(&ar->buf);
error_destroy_mutex:
sc_mutex_destroy(&ar->mutex);
error_free_swr_ctx:
swr_free(&ar->swr_ctx);
return false;
}
void
sc_audio_regulator_destroy(struct sc_audio_regulator *ar) {
free(ar->swr_buf);
sc_audiobuf_destroy(&ar->buf);
sc_mutex_destroy(&ar->mutex);
swr_free(&ar->swr_ctx);
}

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#ifndef SC_AUDIO_REGULATOR_H
#define SC_AUDIO_REGULATOR_H
#include "common.h"
#include <stdatomic.h>
#include <stdbool.h>
#include <libavcodec/avcodec.h>
#include <libswresample/swresample.h>
#include "util/audiobuf.h"
#include "util/average.h"
#include "util/thread.h"
#define SC_AV_SAMPLE_FMT AV_SAMPLE_FMT_FLT
struct sc_audio_regulator {
sc_mutex mutex;
// Target buffering between the producer and the consumer (in samples)
uint32_t target_buffering;
// Audio buffer to communicate between the receiver and the player
struct sc_audiobuf buf;
// Resampler (only used from the receiver thread)
struct SwrContext *swr_ctx;
// The sample rate is the same for input and output
uint32_t sample_rate;
// The number of bytes per sample (for all channels)
size_t sample_size;
// Target buffer for resampling (only used by the receiver thread)
uint8_t *swr_buf;
size_t swr_buf_alloc_size;
// Number of buffered samples (may be negative on underflow) (only used by
// the receiver thread)
struct sc_average avg_buffering;
// Count the number of samples to trigger a compensation update regularly
// (only used by the receiver thread)
uint32_t samples_since_resync;
// Number of silence samples inserted since the last received packet
atomic_uint_least32_t underflow;
// Current applied compensation value (only used by the receiver thread)
int compensation;
// Set to true the first time a sample is received
atomic_bool received;
// Set to true the first time samples are pulled by the player
atomic_bool played;
};
bool
sc_audio_regulator_init(struct sc_audio_regulator *ar, size_t sample_size,
const AVCodecContext *ctx, uint32_t target_buffering);
void
sc_audio_regulator_destroy(struct sc_audio_regulator *ar);
bool
sc_audio_regulator_push(struct sc_audio_regulator *ar, const AVFrame *frame);
void
sc_audio_regulator_pull(struct sc_audio_regulator *ar, uint8_t *out,
uint32_t samples);
#endif