mirror of https://github.com/Genymobile/scrcpy
Add audio player
Play the decoded audio using SDL. The audio player frame sink receives the audio frames, resample them and write them to a byte buffer (introduced by this commit). On SDL audio callback (from an internal SDL thread), copy samples from this byte buffer to the SDL audio buffer. The byte buffer is protected by the SDL_AudioDeviceLock(), but it has been designed so that the producer and the consumer may write and read in parallel, provided that they don't access the same slices of the ring-buffer buffer. PR #3757 <https://github.com/Genymobile/scrcpy/pull/3757> Co-authored-by: Simon Chan <1330321+yume-chan@users.noreply.github.com>pull/3757/head
Romain Vimont
1 year ago
parent
e1333f6f3b
commit
fbe0f951e1
@ -0,0 +1,409 @@
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#include "audio_player.h"
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#include <libavutil/opt.h>
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#include "util/log.h"
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#define SC_AUDIO_PLAYER_NDEBUG // comment to debug
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/** Downcast frame_sink to sc_audio_player */
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#define DOWNCAST(SINK) container_of(SINK, struct sc_audio_player, frame_sink)
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#define SC_AV_SAMPLE_FMT AV_SAMPLE_FMT_FLT
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#define SC_SDL_SAMPLE_FMT AUDIO_F32
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#define SC_AUDIO_OUTPUT_BUFFER_SAMPLES 240 // 5ms at 48000Hz
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static inline uint32_t
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bytes_to_samples(struct sc_audio_player *ap, size_t bytes) {
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assert(bytes % (ap->nb_channels * ap->out_bytes_per_sample) == 0);
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return bytes / (ap->nb_channels * ap->out_bytes_per_sample);
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}
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static inline size_t
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samples_to_bytes(struct sc_audio_player *ap, uint32_t samples) {
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return samples * ap->nb_channels * ap->out_bytes_per_sample;
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}
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static void SDLCALL
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sc_audio_player_sdl_callback(void *userdata, uint8_t *stream, int len_int) {
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struct sc_audio_player *ap = userdata;
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// This callback is called with the lock used by SDL_AudioDeviceLock(), so
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// the bytebuf is protected
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assert(len_int > 0);
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size_t len = len_int;
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#ifndef SC_AUDIO_PLAYER_NDEBUG
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LOGD("[Audio] SDL callback requests %" PRIu32 " samples",
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bytes_to_samples(ap, len));
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#endif
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size_t read_avail = sc_bytebuf_read_available(&ap->buf);
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if (!ap->played) {
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uint32_t buffered_samples = bytes_to_samples(ap, read_avail);
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// Part of the buffering is handled by inserting initial silence. The
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// remaining (margin) last samples will be handled by compensation.
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uint32_t margin = 30 * ap->sample_rate / 1000; // 30ms
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if (buffered_samples + margin < ap->target_buffering) {
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LOGV("[Audio] Inserting initial buffering silence: %" PRIu32
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" samples", bytes_to_samples(ap, len));
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// Delay playback starting to reach the target buffering. Fill the
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// whole buffer with silence (len is small compared to the
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// arbitrary margin value).
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memset(stream, 0, len);
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return;
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}
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}
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size_t read = MIN(read_avail, len);
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if (read) {
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sc_bytebuf_read(&ap->buf, stream, read);
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}
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if (read < len) {
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size_t silence_bytes = len - read;
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uint32_t silence_samples = bytes_to_samples(ap, silence_bytes);
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// Insert silence. In theory, the inserted silent samples replace the
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// missing real samples, which will arrive later, so they should be
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// dropped to keep the latency minimal. However, this would cause very
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// audible glitches, so let the clock compensation restore the target
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// latency.
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LOGD("[Audio] Buffer underflow, inserting silence: %" PRIu32 " samples",
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silence_samples);
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memset(stream + read, 0, silence_bytes);
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if (ap->received) {
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// Inserting additional samples immediately increases buffering
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ap->avg_buffering.avg += silence_samples;
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}
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}
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ap->played = true;
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}
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static uint8_t *
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sc_audio_player_get_swr_buf(struct sc_audio_player *ap, uint32_t min_samples) {
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size_t min_buf_size = samples_to_bytes(ap, min_samples);
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if (min_buf_size > ap->swr_buf_alloc_size) {
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size_t new_size = min_buf_size + 4096;
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uint8_t *buf = realloc(ap->swr_buf, new_size);
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if (!buf) {
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LOG_OOM();
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// Could not realloc to the requested size
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return NULL;
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}
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ap->swr_buf = buf;
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ap->swr_buf_alloc_size = new_size;
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}
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return ap->swr_buf;
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}
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static bool
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sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
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const AVFrame *frame) {
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struct sc_audio_player *ap = DOWNCAST(sink);
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SwrContext *swr_ctx = ap->swr_ctx;
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int64_t swr_delay = swr_get_delay(swr_ctx, ap->sample_rate);
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// No need to av_rescale_rnd(), input and output sample rates are the same.
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// Add more space (256) for clock compensation.
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int dst_nb_samples = swr_delay + frame->nb_samples + 256;
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uint8_t *swr_buf = sc_audio_player_get_swr_buf(ap, dst_nb_samples);
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if (!swr_buf) {
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return false;
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}
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int ret = swr_convert(swr_ctx, &swr_buf, dst_nb_samples,
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(const uint8_t **) frame->data, frame->nb_samples);
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if (ret < 0) {
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LOGE("Resampling failed: %d", ret);
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return false;
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}
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// swr_convert() returns the number of samples which would have been
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// written if the buffer was big enough.
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uint32_t samples_written = MIN(ret, dst_nb_samples);
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size_t swr_buf_size = samples_to_bytes(ap, samples_written);
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#ifndef SC_AUDIO_PLAYER_NDEBUG
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LOGD("[Audio] %" PRIu32 " samples written to buffer", samples_written);
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#endif
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// Since this function is the only writer, the current available space is
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// at least the previous available space. In practice, it should almost
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// always be possible to write without lock.
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bool lockless_write = swr_buf_size <= ap->previous_write_avail;
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if (lockless_write) {
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sc_bytebuf_prepare_write(&ap->buf, swr_buf, swr_buf_size);
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}
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SDL_LockAudioDevice(ap->device);
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size_t read_avail = sc_bytebuf_read_available(&ap->buf);
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uint32_t buffered_samples = bytes_to_samples(ap, read_avail);
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if (lockless_write) {
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sc_bytebuf_commit_write(&ap->buf, swr_buf_size);
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} else {
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// Take care to keep full samples
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size_t align = ap->nb_channels * ap->out_bytes_per_sample;
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size_t write_avail =
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sc_bytebuf_write_available(&ap->buf) / align * align;
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if (swr_buf_size > write_avail) {
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// Entering this branch is very unlikely, the ring-buffer (bytebuf)
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// is allocated with a size sufficient to store 1 second more than
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// the target buffering. If this happens, though, we have to skip
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// old samples.
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size_t cap = sc_bytebuf_capacity(&ap->buf) / align * align;
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if (swr_buf_size > cap) {
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// Very very unlikely: a single resampled frame should never
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// exceed the ring-buffer size (or something is very wrong).
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// Ignore the first bytes in swr_buf
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swr_buf += swr_buf_size - cap;
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swr_buf_size = cap;
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// This change in samples_written will impact the
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// instant_compensation below
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samples_written -= bytes_to_samples(ap, swr_buf_size - cap);
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}
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assert(swr_buf_size >= write_avail);
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if (swr_buf_size > write_avail) {
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sc_bytebuf_skip(&ap->buf, swr_buf_size - write_avail);
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uint32_t skip_samples =
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bytes_to_samples(ap, swr_buf_size - write_avail);
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assert(buffered_samples >= skip_samples);
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buffered_samples -= skip_samples;
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if (ap->played) {
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// Dropping input samples instantly decreases buffering
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ap->avg_buffering.avg -= skip_samples;
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}
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}
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// It should remain exactly the expected size to write the new
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// samples.
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assert((sc_bytebuf_write_available(&ap->buf) / align * align)
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== swr_buf_size);
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}
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sc_bytebuf_write(&ap->buf, swr_buf, swr_buf_size);
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}
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buffered_samples += samples_written;
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assert(samples_to_bytes(ap, buffered_samples)
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== sc_bytebuf_read_available(&ap->buf));
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// Read with lock held, to be used after unlocking
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bool played = ap->played;
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if (played) {
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uint32_t max_buffered_samples = ap->target_buffering
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+ 12 * SC_AUDIO_OUTPUT_BUFFER_SAMPLES
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+ ap->target_buffering / 10;
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if (buffered_samples > max_buffered_samples) {
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uint32_t skip_samples = buffered_samples - max_buffered_samples;
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size_t skip_bytes = samples_to_bytes(ap, skip_samples);
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sc_bytebuf_skip(&ap->buf, skip_bytes);
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#ifndef SC_AUDIO_PLAYER_NDEBUG
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LOGD("[Audio] Buffering threshold exceeded, skipping %" PRIu32
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" samples", skip_samples);
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#endif
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}
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// Number of samples added (or removed, if negative) for compensation
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int32_t instant_compensation =
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(int32_t) samples_written - frame->nb_samples;
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// The compensation must apply instantly, it must not be smoothed
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ap->avg_buffering.avg += instant_compensation;
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// However, the buffering level must be smoothed
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sc_average_push(&ap->avg_buffering, buffered_samples);
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#ifndef SC_AUDIO_PLAYER_NDEBUG
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LOGD("[Audio] buffered_samples=%" PRIu32 " avg_buffering=%f",
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buffered_samples, sc_average_get(&ap->avg_buffering));
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#endif
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} else {
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// SDL playback not started yet, do not accumulate more than
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// max_initial_buffering samples, this would cause unnecessary delay
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// (and glitches to compensate) on start.
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uint32_t max_initial_buffering = ap->target_buffering
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+ 2 * SC_AUDIO_OUTPUT_BUFFER_SAMPLES;
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if (buffered_samples > max_initial_buffering) {
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uint32_t skip_samples = buffered_samples - max_initial_buffering;
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size_t skip_bytes = samples_to_bytes(ap, skip_samples);
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sc_bytebuf_skip(&ap->buf, skip_bytes);
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#ifndef SC_AUDIO_PLAYER_NDEBUG
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LOGD("[Audio] Playback not started, skipping %" PRIu32 " samples",
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skip_samples);
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#endif
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}
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}
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ap->previous_write_avail = sc_bytebuf_write_available(&ap->buf);
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ap->received = true;
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SDL_UnlockAudioDevice(ap->device);
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if (played) {
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ap->samples_since_resync += samples_written;
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if (ap->samples_since_resync >= ap->sample_rate) {
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// Recompute compensation every second
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ap->samples_since_resync = 0;
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float avg = sc_average_get(&ap->avg_buffering);
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int diff = ap->target_buffering - avg;
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if (diff < 0 && buffered_samples < ap->target_buffering) {
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// Do not accelerate if the instant buffering level is below
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// the average, this would increase underflow
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diff = 0;
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}
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// Compensate the diff over 4 seconds (but will be recomputed after
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// 1 second)
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int distance = 4 * ap->sample_rate;
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// Limit compensation rate to 2%
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int abs_max_diff = distance / 50;
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diff = CLAMP(diff, -abs_max_diff, abs_max_diff);
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LOGV("[Audio] Buffering: target=%" PRIu32 " avg=%f cur=%" PRIu32
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" compensation=%d", ap->target_buffering, avg,
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buffered_samples, diff);
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int ret = swr_set_compensation(swr_ctx, diff, distance);
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if (ret < 0) {
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LOGW("Resampling compensation failed: %d", ret);
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// not fatal
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}
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}
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}
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return true;
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}
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static bool
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sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
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const AVCodecContext *ctx) {
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struct sc_audio_player *ap = DOWNCAST(sink);
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SDL_AudioSpec desired = {
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.freq = ctx->sample_rate,
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.format = SC_SDL_SAMPLE_FMT,
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.channels = ctx->ch_layout.nb_channels,
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.samples = SC_AUDIO_OUTPUT_BUFFER_SAMPLES,
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.callback = sc_audio_player_sdl_callback,
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.userdata = ap,
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};
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SDL_AudioSpec obtained;
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ap->device = SDL_OpenAudioDevice(NULL, 0, &desired, &obtained, 0);
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if (!ap->device) {
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LOGE("Could not open audio device: %s", SDL_GetError());
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return false;
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}
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SwrContext *swr_ctx = swr_alloc();
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if (!swr_ctx) {
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LOG_OOM();
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goto error_close_audio_device;
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}
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ap->swr_ctx = swr_ctx;
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assert(ctx->sample_rate > 0);
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assert(ctx->ch_layout.nb_channels > 0);
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assert(!av_sample_fmt_is_planar(SC_AV_SAMPLE_FMT));
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int out_bytes_per_sample = av_get_bytes_per_sample(SC_AV_SAMPLE_FMT);
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assert(out_bytes_per_sample > 0);
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av_opt_set_chlayout(swr_ctx, "in_chlayout", &ctx->ch_layout, 0);
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av_opt_set_chlayout(swr_ctx, "out_chlayout", &ctx->ch_layout, 0);
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av_opt_set_int(swr_ctx, "in_sample_rate", ctx->sample_rate, 0);
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av_opt_set_int(swr_ctx, "out_sample_rate", ctx->sample_rate, 0);
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av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", ctx->sample_fmt, 0);
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av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", SC_AV_SAMPLE_FMT, 0);
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int ret = swr_init(swr_ctx);
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if (ret) {
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LOGE("Failed to initialize the resampling context");
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goto error_free_swr_ctx;
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}
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ap->sample_rate = ctx->sample_rate;
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ap->nb_channels = ctx->ch_layout.nb_channels;
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ap->out_bytes_per_sample = out_bytes_per_sample;
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ap->target_buffering = ap->target_buffering_delay * ap->sample_rate
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/ SC_TICK_FREQ;
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// Use a ring-buffer of the target buffering size plus 1 second between the
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// producer and the consumer. It's too big on purpose, to guarantee that
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// the producer and the consumer will be able to access it in parallel
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// without locking.
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size_t bytebuf_samples = ap->target_buffering + ap->sample_rate;
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size_t bytebuf_size = samples_to_bytes(ap, bytebuf_samples);
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bool ok = sc_bytebuf_init(&ap->buf, bytebuf_size);
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if (!ok) {
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goto error_free_swr_ctx;
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}
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size_t initial_swr_buf_size = samples_to_bytes(ap, 4096);
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ap->swr_buf = malloc(initial_swr_buf_size);
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if (!ap->swr_buf) {
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LOG_OOM();
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goto error_destroy_bytebuf;
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}
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ap->swr_buf_alloc_size = initial_swr_buf_size;
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ap->previous_write_avail = sc_bytebuf_write_available(&ap->buf);
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// Samples are produced and consumed by blocks, so the buffering must be
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// smoothed to get a relatively stable value.
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sc_average_init(&ap->avg_buffering, 32);
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ap->samples_since_resync = 0;
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ap->received = false;
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ap->played = false;
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SDL_PauseAudioDevice(ap->device, 0);
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return true;
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error_destroy_bytebuf:
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sc_bytebuf_destroy(&ap->buf);
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error_free_swr_ctx:
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swr_free(&ap->swr_ctx);
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error_close_audio_device:
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SDL_CloseAudioDevice(ap->device);
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return false;
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}
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static void
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sc_audio_player_frame_sink_close(struct sc_frame_sink *sink) {
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struct sc_audio_player *ap = DOWNCAST(sink);
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assert(ap->device);
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SDL_PauseAudioDevice(ap->device, 1);
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SDL_CloseAudioDevice(ap->device);
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free(ap->swr_buf);
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sc_bytebuf_destroy(&ap->buf);
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swr_free(&ap->swr_ctx);
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}
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void
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sc_audio_player_init(struct sc_audio_player *ap, sc_tick target_buffering) {
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ap->target_buffering_delay = target_buffering;
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static const struct sc_frame_sink_ops ops = {
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.open = sc_audio_player_frame_sink_open,
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.close = sc_audio_player_frame_sink_close,
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.push = sc_audio_player_frame_sink_push,
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};
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ap->frame_sink.ops = &ops;
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}
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@ -0,0 +1,78 @@
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#ifndef SC_AUDIO_PLAYER_H
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#define SC_AUDIO_PLAYER_H
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#include "common.h"
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#include <stdbool.h>
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#include "trait/frame_sink.h"
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#include <util/average.h>
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#include <util/bytebuf.h>
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#include <util/thread.h>
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#include <util/tick.h>
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#include <libavformat/avformat.h>
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#include <libswresample/swresample.h>
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#include <SDL2/SDL.h>
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struct sc_audio_player {
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struct sc_frame_sink frame_sink;
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SDL_AudioDeviceID device;
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// The target buffering between the producer and the consumer. This value
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// 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
|
||||
|
||||
// Audio buffer to communicate between the receiver and the SDL audio
|
||||
// callback (protected by SDL_AudioDeviceLock())
|
||||
struct sc_bytebuf buf;
|
||||
|
||||
// The previous number of bytes available in the buffer (only used by the
|
||||
// receiver thread)
|
||||
size_t previous_write_avail;
|
||||
|
||||
// Resampler (only used from the receiver thread)
|
||||
struct SwrContext *swr_ctx;
|
||||
|
||||
// The sample rate is the same for input and output
|
||||
unsigned sample_rate;
|
||||
// The number of channels is the same for input and output
|
||||
unsigned nb_channels;
|
||||
// The number of bytes per sample for a single channel
|
||||
unsigned out_bytes_per_sample;
|
||||
|
||||
// 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;
|
||||
|
||||
// Set to true the first time a sample is received (protected by
|
||||
// SDL_AudioDeviceLock())
|
||||
bool received;
|
||||
|
||||
// Set to true the first time the SDL callback is called (protected by
|
||||
// SDL_AudioDeviceLock())
|
||||
bool played;
|
||||
|
||||
const struct sc_audio_player_callbacks *cbs;
|
||||
void *cbs_userdata;
|
||||
};
|
||||
|
||||
struct sc_audio_player_callbacks {
|
||||
void (*on_ended)(struct sc_audio_player *ap, bool success, void *userdata);
|
||||
};
|
||||
|
||||
void
|
||||
sc_audio_player_init(struct sc_audio_player *ap, sc_tick target_buffering);
|
||||
|
||||
#endif
|
||||
@ -0,0 +1,26 @@
|
||||
#include "average.h"
|
||||
|
||||
#include <assert.h>
|
||||
|
||||
void
|
||||
sc_average_init(struct sc_average *avg, unsigned range) {
|
||||
avg->range = range;
|
||||
avg->avg = 0;
|
||||
avg->count = 0;
|
||||
}
|
||||
|
||||
void
|
||||
sc_average_push(struct sc_average *avg, float value) {
|
||||
if (avg->count < avg->range) {
|
||||
++avg->count;
|
||||
}
|
||||
|
||||
assert(avg->count);
|
||||
avg->avg = ((avg->count - 1) * avg->avg + value) / avg->count;
|
||||
}
|
||||
|
||||
float
|
||||
sc_average_get(struct sc_average *avg) {
|
||||
assert(avg->count);
|
||||
return avg->avg;
|
||||
}
|
||||
@ -0,0 +1,40 @@
|
||||
#ifndef SC_AVERAGE
|
||||
#define SC_AVERAGE
|
||||
|
||||
#include "common.h"
|
||||
|
||||
#include <stdbool.h>
|
||||
#include <stdint.h>
|
||||
|
||||
struct sc_average {
|
||||
// Current average value
|
||||
float avg;
|
||||
|
||||
// Target range, to update the average as follow:
|
||||
// avg = ((range - 1) * avg + new_value) / range
|
||||
unsigned range;
|
||||
|
||||
// Number of values pushed when less than range (count <= range).
|
||||
// The purpose is to handle the first (range - 1) values properly.
|
||||
unsigned count;
|
||||
};
|
||||
|
||||
void
|
||||
sc_average_init(struct sc_average *avg, unsigned range);
|
||||
|
||||
/**
|
||||
* Push a new value to update the "rolling" average
|
||||
*/
|
||||
void
|
||||
sc_average_push(struct sc_average *avg, float value);
|
||||
|
||||
/**
|
||||
* Get the current average value
|
||||
*
|
||||
* It is an error to call this function if sc_average_push() has not been
|
||||
* called at least once.
|
||||
*/
|
||||
float
|
||||
sc_average_get(struct sc_average *avg);
|
||||
|
||||
#endif
|
||||
Loading…
Reference in New Issue