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Add audio sample ring-buffer

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Add a thin wrapper around bytebuf to handle samples instead of bytes.
This simplifies the audio player, which mostly handles samples.
pull/3774/head
Romain Vimont 1 year ago
parent bb509d9317
commit 14f9d82fda
3 changed files with 148 additions and 76 deletions
Show Stats Download Patch File Download Diff File

120
app/src/audio_player.c
Unescape Escape View File

@ -15,42 +15,32 @@
#define SC_AUDIO_OUTPUT_BUFFER_MS 5
static inline uint32_t
bytes_to_samples(struct sc_audio_player *ap, size_t bytes) {
assert(bytes % (ap->nb_channels * ap->out_bytes_per_sample) == 0);
return bytes / (ap->nb_channels * ap->out_bytes_per_sample);
}
static inline size_t
samples_to_bytes(struct sc_audio_player *ap, uint32_t samples) {
return samples * ap->nb_channels * ap->out_bytes_per_sample;
}
#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;
// This callback is called with the lock used by SDL_AudioDeviceLock(), so
// the bytebuf is protected
// the audiobuf is protected
assert(len_int > 0);
size_t len = len_int;
uint32_t count = TO_SAMPLES(len);
#ifndef SC_AUDIO_PLAYER_NDEBUG
LOGD("[Audio] SDL callback requests %" PRIu32 " samples",
bytes_to_samples(ap, len));
LOGD("[Audio] SDL callback requests %" PRIu32 " samples", count);
#endif
size_t read_avail = sc_bytebuf_read_available(&ap->buf);
uint32_t buffered_samples = sc_audiobuf_read_available(&ap->buf);
if (!ap->played) {
uint32_t buffered_samples = bytes_to_samples(ap, read_avail);
// Part of the buffering is handled by inserting initial silence. The
// remaining (margin) last samples will be handled by compensation.
uint32_t margin = 30 * ap->sample_rate / 1000; // 30ms
if (buffered_samples + margin < ap->target_buffering) {
LOGV("[Audio] Inserting initial buffering silence: %" PRIu32
" samples", bytes_to_samples(ap, len));
" 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).
@ -59,26 +49,25 @@ sc_audio_player_sdl_callback(void *userdata, uint8_t *stream, int len_int) {
}
}
size_t read = MIN(read_avail, len);
uint32_t read = MIN(buffered_samples, count);
if (read) {
sc_bytebuf_read(&ap->buf, stream, read);
sc_audiobuf_read(&ap->buf, stream, read);
}
if (read < len) {
size_t silence_bytes = len - read;
uint32_t silence_samples = bytes_to_samples(ap, silence_bytes);
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_samples);
memset(stream + read, 0, silence_bytes);
silence);
memset(stream + read, 0, TO_BYTES(silence));
if (ap->received) {
// Inserting additional samples immediately increases buffering
ap->avg_buffering.avg += silence_samples;
ap->avg_buffering.avg += silence;
}
}
@ -87,7 +76,7 @@ sc_audio_player_sdl_callback(void *userdata, uint8_t *stream, int len_int) {
static uint8_t *
sc_audio_player_get_swr_buf(struct sc_audio_player *ap, uint32_t min_samples) {
size_t min_buf_size = samples_to_bytes(ap, 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);
@ -130,7 +119,6 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
// swr_convert() returns the number of samples which would have been
// written if the buffer was big enough.
uint32_t samples_written = MIN(ret, dst_nb_samples);
size_t swr_buf_size = samples_to_bytes(ap, samples_written);
#ifndef SC_AUDIO_PLAYER_NDEBUG
LOGD("[Audio] %" PRIu32 " samples written to buffer", samples_written);
#endif
@ -138,46 +126,40 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
// Since this function is the only writer, the current available space is
// at least the previous available space. In practice, it should almost
// always be possible to write without lock.
bool lockless_write = swr_buf_size <= ap->previous_write_avail;
bool lockless_write = samples_written <= ap->previous_write_avail;
if (lockless_write) {
sc_bytebuf_prepare_write(&ap->buf, swr_buf, swr_buf_size);
sc_audiobuf_prepare_write(&ap->buf, swr_buf, samples_written);
}
SDL_LockAudioDevice(ap->device);
size_t read_avail = sc_bytebuf_read_available(&ap->buf);
uint32_t buffered_samples = bytes_to_samples(ap, read_avail);
uint32_t buffered_samples = sc_audiobuf_read_available(&ap->buf);
if (lockless_write) {
sc_bytebuf_commit_write(&ap->buf, swr_buf_size);
sc_audiobuf_commit_write(&ap->buf, samples_written);
} else {
// Take care to keep full samples
size_t align = ap->nb_channels * ap->out_bytes_per_sample;
size_t write_avail =
sc_bytebuf_write_available(&ap->buf) / align * align;
if (swr_buf_size > write_avail) {
// Entering this branch is very unlikely, the ring-buffer (bytebuf)
// is allocated with a size sufficient to store 1 second more than
// the target buffering. If this happens, though, we have to skip
// old samples.
size_t cap = sc_bytebuf_capacity(&ap->buf) / align * align;
if (swr_buf_size > cap) {
uint32_t write_avail = sc_audiobuf_write_available(&ap->buf);
if (samples_written > write_avail) {
// Entering this branch is very unlikely, the audio buffer is
// allocated with a size sufficient to store 1 second more than the
// target buffering. If this happens, though, we have to skip old
// samples.
uint32_t cap = sc_audiobuf_capacity(&ap->buf);
if (samples_written > cap) {
// Very very unlikely: a single resampled frame should never
// exceed the ring-buffer size (or something is very wrong).
// exceed the audio buffer size (or something is very wrong).
// Ignore the first bytes in swr_buf
swr_buf += swr_buf_size - cap;
swr_buf_size = cap;
swr_buf += TO_BYTES(samples_written - cap);
// This change in samples_written will impact the
// instant_compensation below
samples_written -= bytes_to_samples(ap, swr_buf_size - cap);
samples_written = cap;
}
assert(swr_buf_size >= write_avail);
if (swr_buf_size > write_avail) {
sc_bytebuf_skip(&ap->buf, swr_buf_size - write_avail);
uint32_t skip_samples =
bytes_to_samples(ap, swr_buf_size - write_avail);
assert(samples_written >= write_avail);
if (samples_written > write_avail) {
uint32_t skip_samples = samples_written - write_avail;
assert(buffered_samples >= skip_samples);
sc_audiobuf_skip(&ap->buf, skip_samples);
buffered_samples -= skip_samples;
if (ap->played) {
// Dropping input samples instantly decreases buffering
@ -187,16 +169,14 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
// It should remain exactly the expected size to write the new
// samples.
assert((sc_bytebuf_write_available(&ap->buf) / align * align)
== swr_buf_size);
assert(sc_audiobuf_write_available(&ap->buf) == samples_written);
}
sc_bytebuf_write(&ap->buf, swr_buf, swr_buf_size);
sc_audiobuf_write(&ap->buf, swr_buf, samples_written);
}
buffered_samples += samples_written;
assert(samples_to_bytes(ap, buffered_samples)
== sc_bytebuf_read_available(&ap->buf));
assert(buffered_samples == sc_audiobuf_read_available(&ap->buf));
// Read with lock held, to be used after unlocking
bool played = ap->played;
@ -206,8 +186,7 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
+ ap->target_buffering / 10;
if (buffered_samples > max_buffered_samples) {
uint32_t skip_samples = buffered_samples - max_buffered_samples;
size_t skip_bytes = samples_to_bytes(ap, skip_samples);
sc_bytebuf_skip(&ap->buf, skip_bytes);
sc_audiobuf_skip(&ap->buf, skip_samples);
LOGD("[Audio] Buffering threshold exceeded, skipping %" PRIu32
" samples", skip_samples);
}
@ -234,8 +213,7 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
+ 2 * SC_AUDIO_OUTPUT_BUFFER_MS * ap->sample_rate / 1000;
if (buffered_samples > max_initial_buffering) {
uint32_t skip_samples = buffered_samples - max_initial_buffering;
size_t skip_bytes = samples_to_bytes(ap, skip_samples);
sc_bytebuf_skip(&ap->buf, skip_bytes);
sc_audiobuf_skip(&ap->buf, skip_samples);
#ifndef SC_AUDIO_PLAYER_NDEBUG
LOGD("[Audio] Playback not started, skipping %" PRIu32 " samples",
skip_samples);
@ -243,7 +221,7 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
}
}
ap->previous_write_avail = sc_bytebuf_write_available(&ap->buf);
ap->previous_write_avail = sc_audiobuf_write_available(&ap->buf);
ap->received = true;
SDL_UnlockAudioDevice(ap->device);
@ -355,23 +333,23 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
// 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.
size_t bytebuf_samples = ap->target_buffering + ap->sample_rate;
size_t bytebuf_size = samples_to_bytes(ap, bytebuf_samples);
size_t audiobuf_samples = ap->target_buffering + ap->sample_rate;
bool ok = sc_bytebuf_init(&ap->buf, bytebuf_size);
size_t sample_size = ap->nb_channels * ap->out_bytes_per_sample;
bool ok = sc_audiobuf_init(&ap->buf, sample_size, audiobuf_samples);
if (!ok) {
goto error_free_swr_ctx;
}
size_t initial_swr_buf_size = samples_to_bytes(ap, 4096);
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_bytebuf;
goto error_destroy_audiobuf;
}
ap->swr_buf_alloc_size = initial_swr_buf_size;
ap->previous_write_avail = sc_bytebuf_write_available(&ap->buf);
ap->previous_write_avail = sc_audiobuf_write_available(&ap->buf);
// Samples are produced and consumed by blocks, so the buffering must be
// smoothed to get a relatively stable value.
@ -393,8 +371,8 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
return true;
error_destroy_bytebuf:
sc_bytebuf_destroy(&ap->buf);
error_destroy_audiobuf:
sc_audiobuf_destroy(&ap->buf);
error_free_swr_ctx:
swr_free(&ap->swr_ctx);
error_close_audio_device:
@ -412,7 +390,7 @@ sc_audio_player_frame_sink_close(struct sc_frame_sink *sink) {
SDL_CloseAudioDevice(ap->device);
free(ap->swr_buf);
sc_bytebuf_destroy(&ap->buf);
sc_audiobuf_destroy(&ap->buf);
swr_free(&ap->swr_ctx);
}

10
app/src/audio_player.h
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@ -5,8 +5,8 @@
#include <stdbool.h>
#include "trait/frame_sink.h"
#include <util/audiobuf.h>
#include <util/average.h>
#include <util/bytebuf.h>
#include <util/thread.h>
#include <util/tick.h>
@ -29,11 +29,11 @@ struct sc_audio_player {
// Audio buffer to communicate between the receiver and the SDL audio
// callback (protected by SDL_AudioDeviceLock())
struct sc_bytebuf buf;
struct sc_audiobuf buf;
// The previous number of bytes available in the buffer (only used by the
// receiver thread)
size_t previous_write_avail;
// The previous empty space in the buffer (only used by the receiver
// thread)
uint32_t previous_write_avail;
// Resampler (only used from the receiver thread)
struct SwrContext *swr_ctx;

94
app/src/util/audiobuf.h
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@ -0,0 +1,94 @@
#ifndef SC_AUDIOBUF_H
#define SC_AUDIOBUF_H
#include "common.h"
#include <stdbool.h>
#include <stdint.h>
#include "util/bytebuf.h"
/**
* Wrapper around bytebuf to read and write samples
*
* Each sample takes sample_size bytes.
*/
struct sc_audiobuf {
struct sc_bytebuf buf;
size_t sample_size;
};
static inline uint32_t
sc_audiobuf_to_samples(struct sc_audiobuf *buf, size_t bytes) {
assert(bytes % buf->sample_size == 0);
return bytes / buf->sample_size;
}
static inline size_t
sc_audiobuf_to_bytes(struct sc_audiobuf *buf, uint32_t samples) {
return samples * buf->sample_size;
}
static inline bool
sc_audiobuf_init(struct sc_audiobuf *buf, size_t sample_size,
uint32_t capacity) {
buf->sample_size = sample_size;
return sc_bytebuf_init(&buf->buf, capacity * sample_size + 1);
}
static inline void
sc_audiobuf_read(struct sc_audiobuf *buf, uint8_t *to, uint32_t samples) {
size_t bytes = sc_audiobuf_to_bytes(buf, samples);
sc_bytebuf_read(&buf->buf, to, bytes);
}
static inline void
sc_audiobuf_skip(struct sc_audiobuf *buf, uint32_t samples) {
size_t bytes = sc_audiobuf_to_bytes(buf, samples);
sc_bytebuf_skip(&buf->buf, bytes);
}
static inline void
sc_audiobuf_write(struct sc_audiobuf *buf, const uint8_t *from,
uint32_t samples) {
size_t bytes = sc_audiobuf_to_bytes(buf, samples);
sc_bytebuf_write(&buf->buf, from, bytes);
}
static inline void
sc_audiobuf_prepare_write(struct sc_audiobuf *buf, const uint8_t *from,
uint32_t samples) {
size_t bytes = sc_audiobuf_to_bytes(buf, samples);
sc_bytebuf_prepare_write(&buf->buf, from, bytes);
}
static inline void
sc_audiobuf_commit_write(struct sc_audiobuf *buf, uint32_t samples) {
size_t bytes = sc_audiobuf_to_bytes(buf, samples);
sc_bytebuf_commit_write(&buf->buf, bytes);
}
static inline uint32_t
sc_audiobuf_read_available(struct sc_audiobuf *buf) {
size_t bytes = sc_bytebuf_read_available(&buf->buf);
return sc_audiobuf_to_samples(buf, bytes);
}
static inline uint32_t
sc_audiobuf_write_available(struct sc_audiobuf *buf) {
size_t bytes = sc_bytebuf_write_available(&buf->buf);
return sc_audiobuf_to_samples(buf, bytes);
}
static inline uint32_t
sc_audiobuf_capacity(struct sc_audiobuf *buf) {
size_t bytes = sc_bytebuf_capacity(&buf->buf);
return sc_audiobuf_to_samples(buf, bytes);
}
static inline void
sc_audiobuf_destroy(struct sc_audiobuf *buf) {
sc_bytebuf_destroy(&buf->buf);
}
#endif
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