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Simplify bytebuf naming

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Rename read_available to can_read and write_available to can_write.
This is more readable.
This commit is contained in:
Romain Vimont
2023-03-11 10:16:43 +01:00
parent 14f9d82fda
commit e06acc1ba2
6 changed files with 47 additions and 47 deletions
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24
app/src/audio_player.c
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@@ -33,7 +33,7 @@ sc_audio_player_sdl_callback(void *userdata, uint8_t *stream, int len_int) {
LOGD("[Audio] SDL callback requests %" PRIu32 " samples", count);
#endif
uint32_t buffered_samples = sc_audiobuf_read_available(&ap->buf);
uint32_t buffered_samples = sc_audiobuf_can_read(&ap->buf);
if (!ap->played) {
// Part of the buffering is handled by inserting initial silence. The
// remaining (margin) last samples will be handled by compensation.
@@ -126,20 +126,20 @@ 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 = samples_written <= ap->previous_write_avail;
bool lockless_write = samples_written <= ap->previous_can_write;
if (lockless_write) {
sc_audiobuf_prepare_write(&ap->buf, swr_buf, samples_written);
}
SDL_LockAudioDevice(ap->device);
uint32_t buffered_samples = sc_audiobuf_read_available(&ap->buf);
uint32_t buffered_samples = sc_audiobuf_can_read(&ap->buf);
if (lockless_write) {
sc_audiobuf_commit_write(&ap->buf, samples_written);
} else {
uint32_t write_avail = sc_audiobuf_write_available(&ap->buf);
if (samples_written > write_avail) {
uint32_t can_write = sc_audiobuf_can_write(&ap->buf);
if (samples_written > can_write) {
// 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
@@ -155,9 +155,9 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
samples_written = cap;
}
assert(samples_written >= write_avail);
if (samples_written > write_avail) {
uint32_t skip_samples = samples_written - write_avail;
assert(samples_written >= can_write);
if (samples_written > can_write) {
uint32_t skip_samples = samples_written - can_write;
assert(buffered_samples >= skip_samples);
sc_audiobuf_skip(&ap->buf, skip_samples);
buffered_samples -= skip_samples;
@@ -169,14 +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_audiobuf_write_available(&ap->buf) == samples_written);
assert(sc_audiobuf_can_write(&ap->buf) == samples_written);
}
sc_audiobuf_write(&ap->buf, swr_buf, samples_written);
}
buffered_samples += samples_written;
assert(buffered_samples == sc_audiobuf_read_available(&ap->buf));
assert(buffered_samples == sc_audiobuf_can_read(&ap->buf));
// Read with lock held, to be used after unlocking
bool played = ap->played;
@@ -221,7 +221,7 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
}
}
ap->previous_write_avail = sc_audiobuf_write_available(&ap->buf);
ap->previous_can_write = sc_audiobuf_can_write(&ap->buf);
ap->received = true;
SDL_UnlockAudioDevice(ap->device);
@@ -349,7 +349,7 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
}
ap->swr_buf_alloc_size = initial_swr_buf_size;
ap->previous_write_avail = sc_audiobuf_write_available(&ap->buf);
ap->previous_can_write = sc_audiobuf_can_write(&ap->buf);
// Samples are produced and consumed by blocks, so the buffering must be
// smoothed to get a relatively stable value.