/* TODO: optimize mixing */
static void mix_float(struct audio_output *audio, struct audio_line *line,
size_t size, size_t time_offset, size_t plane)
{
float *mixes[MAX_AUDIO_MIXES];
float vals[MIX_BUFFER_SIZE];
for (size_t mix_idx = 0; mix_idx < MAX_AUDIO_MIXES; mix_idx++) {
uint8_t *bytes = audio->mixes[mix_idx].mix_buffers[plane].array;
mixes[mix_idx] = (float*)&bytes[time_offset];
}
while (size) {
size_t pop_count = min_size(size, sizeof(vals));
size -= pop_count;
circlebuf_pop_front(&line->buffers[plane], vals, pop_count);
pop_count /= sizeof(float);
for (size_t mix_idx = 0; mix_idx < MAX_AUDIO_MIXES; mix_idx++) {
/* only include this audio line in this mix if it's set
* via the line's 'mixes' variable */
if ((line->mixers & (1 << mix_idx)) == 0)
continue;
for (size_t i = 0; i < pop_count; i++) {
*(mixes[mix_idx]++) += vals[i];
}
}
}
}
static void droptest_cap_data_rate(struct rtmp_stream *stream, size_t size)
{
uint64_t ts = os_gettime_ns();
struct droptest_info info;
info.ts = ts;
info.size = size;
circlebuf_push_back(&stream->droptest_info, &info, sizeof(info));
stream->droptest_size += size;
if (stream->droptest_info.size) {
circlebuf_peek_front(&stream->droptest_info,
&info, sizeof(info));
if (stream->droptest_size > DROPTEST_MAX_BYTES) {
uint64_t elapsed = ts - info.ts;
if (elapsed < 1000000000ULL) {
elapsed = 1000000000ULL - elapsed;
os_sleepto_ns(ts + elapsed);
}
while (stream->droptest_size > DROPTEST_MAX_BYTES) {
circlebuf_pop_front(&stream->droptest_info,
&info, sizeof(info));
stream->droptest_size -= info.size;
}
}
}
}
/* TODO: optimize mixing */
static void mix_float(uint8_t *mix_in, struct circlebuf *buf, size_t size)
{
float *mix = (float*)mix_in;
float vals[MIX_BUFFER_SIZE];
register float mix_val;
while (size) {
size_t pop_count = min_size(size, sizeof(vals));
size -= pop_count;
circlebuf_pop_front(buf, vals, pop_count);
pop_count /= sizeof(float);
/* This sequence provides hints for MSVC to use packed SSE
* instructions addps, minps, maxps, etc. */
for (size_t i = 0; i < pop_count; i++) {
mix_val = *mix + vals[i];
/* clamp confuses the optimisation */
mix_val = (mix_val > 1.0f) ? 1.0f : mix_val;
mix_val = (mix_val < -1.0f) ? -1.0f : mix_val;
*(mix++) = mix_val;
}
}
}
/* this only really happens with the very initial data insertion. can be
* ignored safely. */
static inline void clear_excess_audio_data(struct audio_line *line,
uint64_t prev_time)
{
size_t size = (size_t)ts_diff_bytes(line->audio, prev_time,
line->base_timestamp);
/*blog(LOG_DEBUG, "Excess audio data for audio line '%s', somehow "
"audio data went back in time by %"PRIu32" bytes. "
"prev_time: %"PRIu64", line->base_timestamp: %"PRIu64,
line->name, (uint32_t)size,
prev_time, line->base_timestamp);*/
if (!line->audio_getting_cut_off) {
blog(LOG_WARNING, "Audio line '%s' audio data currently "
"getting cut off. This could be due to a "
"negative sync offset that's larger than "
"the current audio buffering time.",
line->name);
line->audio_getting_cut_off = true;
}
for (size_t i = 0; i < line->audio->planes; i++) {
size_t clear_size = (size < line->buffers[i].size) ?
size : line->buffers[i].size;
circlebuf_pop_front(&line->buffers[i], NULL, clear_size);
}
}
static bool discard_if_stopped(obs_source_t *source, size_t channels)
{
size_t last_size;
size_t size;
last_size = source->last_audio_input_buf_size;
size = source->audio_input_buf[0].size;
if (!size)
return false;
/* if perpetually pending data, it means the audio has stopped,
* so clear the audio data */
if (last_size == size) {
for (size_t ch = 0; ch < channels; ch++)
circlebuf_pop_front(&source->audio_input_buf[ch], NULL,
source->audio_input_buf[ch].size);
source->audio_ts = 0;
source->last_audio_input_buf_size = 0;
#if DEBUG_AUDIO == 1
blog(LOG_DEBUG, "source audio data appears to have "
"stopped, clearing");
#endif
return true;
} else {
source->last_audio_input_buf_size = size;
return false;
}
}
static inline void output_frame(uint64_t *cur_time, uint64_t interval)
{
struct obs_core_video *video = &obs->video;
int cur_texture = video->cur_texture;
int prev_texture = cur_texture == 0 ? NUM_TEXTURES-1 : cur_texture-1;
struct video_data frame;
bool frame_ready;
memset(&frame, 0, sizeof(struct video_data));
gs_enter_context(video->graphics);
render_video(video, cur_texture, prev_texture);
frame_ready = download_frame(video, prev_texture, &frame);
gs_flush();
gs_leave_context();
if (frame_ready) {
struct obs_vframe_info vframe_info;
circlebuf_pop_front(&video->vframe_info_buffer, &vframe_info,
sizeof(vframe_info));
frame.timestamp = vframe_info.timestamp;
output_video_data(video, &frame, vframe_info.count);
}
if (++video->cur_texture == NUM_TEXTURES)
video->cur_texture = 0;
video_sleep(video, cur_time, interval);
}
static inline void free_packets(struct rtmp_stream *stream)
{
while (stream->packets.size) {
struct encoder_packet packet;
circlebuf_pop_front(&stream->packets, &packet, sizeof(packet));
obs_free_encoder_packet(&packet);
}
}
static inline bool queue_frame(struct obs_core_video *video, bool raw_active,
struct obs_vframe_info *vframe_info, int prev_texture)
{
bool duplicate = !video->gpu_encoder_avail_queue.size ||
(video->gpu_encoder_queue.size && vframe_info->count > 1);
if (duplicate) {
struct obs_tex_frame *tf = circlebuf_data(
&video->gpu_encoder_queue,
video->gpu_encoder_queue.size - sizeof(*tf));
/* texture-based encoding is stopping */
if (!tf) {
return false;
}
tf->count++;
os_sem_post(video->gpu_encode_semaphore);
goto finish;
}
struct obs_tex_frame tf;
circlebuf_pop_front(&video->gpu_encoder_avail_queue, &tf, sizeof(tf));
if (tf.released) {
gs_texture_acquire_sync(tf.tex, tf.lock_key, GS_WAIT_INFINITE);
tf.released = false;
}
/* the vframe_info->count > 1 case causing a copy can only happen if by
* some chance the very first frame has to be duplicated for whatever
* reason. otherwise, it goes to the 'duplicate' case above, which
* will ensure better performance. */
if (raw_active || vframe_info->count > 1) {
gs_copy_texture(tf.tex, video->convert_textures[prev_texture]);
} else {
gs_texture_t *tex = video->convert_textures[prev_texture];
gs_texture_t *tex_uv = video->convert_uv_textures[prev_texture];
video->convert_textures[prev_texture] = tf.tex;
video->convert_uv_textures[prev_texture] = tf.tex_uv;
tf.tex = tex;
tf.tex_uv = tex_uv;
tf.handle = gs_texture_get_shared_handle(tex);
}
tf.count = 1;
tf.timestamp = vframe_info->timestamp;
tf.released = true;
gs_texture_release_sync(tf.tex, ++tf.lock_key);
circlebuf_push_back(&video->gpu_encoder_queue, &tf, sizeof(tf));
os_sem_post(video->gpu_encode_semaphore);
finish:
return --vframe_info->count;
}
void mp_decode_clear_packets(struct mp_decode *d)
{
if (d->packet_pending) {
av_packet_unref(&d->orig_pkt);
d->packet_pending = false;
}
while (d->packets.size) {
AVPacket pkt;
circlebuf_pop_front(&d->packets, &pkt, sizeof(pkt));
av_packet_unref(&pkt);
}
}
static void ignore_audio(obs_source_t *source, size_t channels,
size_t sample_rate)
{
size_t num_floats = source->audio_input_buf[0].size / sizeof(float);
if (num_floats) {
for (size_t ch = 0; ch < channels; ch++)
circlebuf_pop_front(&source->audio_input_buf[ch], NULL,
source->audio_input_buf[ch].size);
source->last_audio_input_buf_size = 0;
source->audio_ts += (uint64_t)num_floats * 1000000000ULL /
(uint64_t)sample_rate;
}
}
static inline bool get_next_packet(struct rtmp_stream *stream,
struct encoder_packet *packet)
{
bool new_packet = false;
pthread_mutex_lock(&stream->packets_mutex);
if (stream->packets.size) {
circlebuf_pop_front(&stream->packets, packet,
sizeof(struct encoder_packet));
new_packet = true;
}
pthread_mutex_unlock(&stream->packets_mutex);
return new_packet;
}
static void drop_frames(struct rtmp_stream *stream, const char *name,
int highest_priority, int64_t *p_min_dts_usec)
{
struct circlebuf new_buf = {0};
uint64_t last_drop_dts_usec = 0;
int num_frames_dropped = 0;
#ifdef _DEBUG
int start_packets = (int)num_buffered_packets(stream);
#else
UNUSED_PARAMETER(name);
#endif
circlebuf_reserve(&new_buf, sizeof(struct encoder_packet) * 8);
while (stream->packets.size) {
struct encoder_packet packet;
circlebuf_pop_front(&stream->packets, &packet, sizeof(packet));
last_drop_dts_usec = packet.dts_usec;
/* do not drop audio data or video keyframes */
if (packet.type == OBS_ENCODER_AUDIO ||
packet.drop_priority >= highest_priority) {
circlebuf_push_back(&new_buf, &packet, sizeof(packet));
} else {
num_frames_dropped++;
obs_free_encoder_packet(&packet);
}
}
circlebuf_free(&stream->packets);
stream->packets = new_buf;
if (stream->min_priority < highest_priority)
stream->min_priority = highest_priority;
*p_min_dts_usec = last_drop_dts_usec;
stream->dropped_frames += num_frames_dropped;
#ifdef _DEBUG
debug("Dropped %s, prev packet count: %d, new packet count: %d",
name,
start_packets,
(int)num_buffered_packets(stream));
#endif
}
static inline void free_packets(struct rtmp_stream *stream)
{
size_t num_packets;
pthread_mutex_lock(&stream->packets_mutex);
num_packets = num_buffered_packets(stream);
if (num_packets)
info("Freeing %d remaining packets", (int)num_packets);
while (stream->packets.size) {
struct encoder_packet packet;
circlebuf_pop_front(&stream->packets, &packet, sizeof(packet));
obs_free_encoder_packet(&packet);
}
pthread_mutex_unlock(&stream->packets_mutex);
}
static inline void output_frame(bool raw_active, const bool gpu_active)
{
struct obs_core_video *video = &obs->video;
int cur_texture = video->cur_texture;
int prev_texture = cur_texture == 0 ? NUM_TEXTURES-1 : cur_texture-1;
struct video_data frame;
bool active = raw_active || gpu_active;
bool frame_ready;
memset(&frame, 0, sizeof(struct video_data));
profile_start(output_frame_gs_context_name);
gs_enter_context(video->graphics);
profile_start(output_frame_render_video_name);
render_video(video, raw_active, gpu_active, cur_texture, prev_texture);
profile_end(output_frame_render_video_name);
if (raw_active) {
profile_start(output_frame_download_frame_name);
frame_ready = download_frame(video, prev_texture, &frame);
profile_end(output_frame_download_frame_name);
}
profile_start(output_frame_gs_flush_name);
gs_flush();
profile_end(output_frame_gs_flush_name);
gs_leave_context();
profile_end(output_frame_gs_context_name);
if (raw_active && frame_ready) {
struct obs_vframe_info vframe_info;
circlebuf_pop_front(&video->vframe_info_buffer, &vframe_info,
sizeof(vframe_info));
frame.timestamp = vframe_info.timestamp;
profile_start(output_frame_output_video_data_name);
output_video_data(video, &frame, vframe_info.count);
profile_end(output_frame_output_video_data_name);
}
if (++video->cur_texture == NUM_TEXTURES)
video->cur_texture = 0;
}
/* this only really happens with the very initial data insertion. can be
* ignored safely. */
static inline void clear_excess_audio_data(struct audio_line *line,
uint64_t prev_time)
{
size_t size = ts_diff_bytes(line->audio, prev_time,
line->base_timestamp);
/*blog(LOG_DEBUG, "Excess audio data for audio line '%s', somehow "
"audio data went back in time by %"PRIu32" bytes. "
"prev_time: %"PRIu64", line->base_timestamp: %"PRIu64,
line->name, (uint32_t)size,
prev_time, line->base_timestamp);*/
for (size_t i = 0; i < line->audio->planes; i++) {
size_t clear_size = (size < line->buffers[i].size) ?
size : line->buffers[i].size;
circlebuf_pop_front(&line->buffers[i], NULL, clear_size);
}
}
static void output_gpu_encoders(struct obs_core_video *video, bool raw_active,
int prev_texture)
{
profile_start(output_gpu_encoders_name);
if (!video->textures_converted[prev_texture])
goto end;
if (!video->vframe_info_buffer_gpu.size)
goto end;
struct obs_vframe_info vframe_info;
circlebuf_pop_front(&video->vframe_info_buffer_gpu, &vframe_info,
sizeof(vframe_info));
pthread_mutex_lock(&video->gpu_encoder_mutex);
encode_gpu(video, raw_active, &vframe_info, prev_texture);
pthread_mutex_unlock(&video->gpu_encoder_mutex);
end:
profile_end(output_gpu_encoders_name);
}
static void drop_frames(struct rtmp_stream *stream)
{
struct circlebuf new_buf = {0};
int drop_priority = 0;
uint64_t last_drop_dts_usec = 0;
int num_frames_dropped = 0;
debug("Previous packet count: %d", (int)num_buffered_packets(stream));
circlebuf_reserve(&new_buf, sizeof(struct encoder_packet) * 8);
while (stream->packets.size) {
struct encoder_packet packet;
circlebuf_pop_front(&stream->packets, &packet, sizeof(packet));
last_drop_dts_usec = packet.dts_usec;
/* do not drop audio data or video keyframes */
if (packet.type == OBS_ENCODER_AUDIO ||
packet.drop_priority == OBS_NAL_PRIORITY_HIGHEST) {
circlebuf_push_back(&new_buf, &packet, sizeof(packet));
} else {
if (drop_priority < packet.drop_priority)
drop_priority = packet.drop_priority;
num_frames_dropped++;
obs_free_encoder_packet(&packet);
}
}
circlebuf_free(&stream->packets);
stream->packets = new_buf;
stream->min_priority = drop_priority;
stream->min_drop_dts_usec = last_drop_dts_usec;
stream->dropped_frames += num_frames_dropped;
debug("New packet count: %d", (int)num_buffered_packets(stream));
}
static void send_audio_data(struct obs_encoder *encoder)
{
struct encoder_frame enc_frame;
memset(&enc_frame, 0, sizeof(struct encoder_frame));
for (size_t i = 0; i < encoder->planes; i++) {
circlebuf_pop_front(&encoder->audio_input_buffer[i],
encoder->audio_output_buffer[i],
encoder->framesize_bytes);
enc_frame.data[i] = encoder->audio_output_buffer[i];
enc_frame.linesize[i] = (uint32_t)encoder->framesize_bytes;
}
enc_frame.frames = (uint32_t)encoder->framesize;
enc_frame.pts = encoder->cur_pts;
do_encode(encoder, &enc_frame);
encoder->cur_pts += encoder->framesize;
}
static void drop_frames(struct rtmp_stream *stream)
{
struct circlebuf new_buf = {0};
int drop_priority = 0;
uint64_t last_drop_dts_usec = 0;
blog(LOG_DEBUG, "Previous packet count: %d",
(int)num_buffered_packets(stream));
circlebuf_reserve(&new_buf, sizeof(struct encoder_packet) * 8);
while (stream->packets.size) {
struct encoder_packet packet;
circlebuf_pop_front(&stream->packets, &packet, sizeof(packet));
last_drop_dts_usec = packet.dts_usec;
if (packet.type == OBS_ENCODER_AUDIO) {
circlebuf_push_back(&new_buf, &packet, sizeof(packet));
} else {
if (drop_priority < packet.drop_priority)
drop_priority = packet.drop_priority;
obs_free_encoder_packet(&packet);
}
}
circlebuf_free(&stream->packets);
stream->packets = new_buf;
stream->min_priority = drop_priority;
stream->min_drop_dts_usec = last_drop_dts_usec;
blog(LOG_DEBUG, "New packet count: %d",
(int)num_buffered_packets(stream));
}
bool mp_decode_next(struct mp_decode *d)
{
bool eof = d->m->eof;
int got_frame;
int ret;
d->frame_ready = false;
if (!eof && !d->packets.size)
return true;
while (!d->frame_ready) {
if (!d->packet_pending) {
if (!d->packets.size) {
if (eof) {
d->pkt.data = NULL;
d->pkt.size = 0;
} else {
return true;
}
} else {
circlebuf_pop_front(&d->packets, &d->orig_pkt,
sizeof(d->orig_pkt));
d->pkt = d->orig_pkt;
d->packet_pending = true;
}
}
ret = decode_packet(d, &got_frame);
if (!got_frame && ret == 0) {
d->eof = true;
return true;
}
if (ret < 0) {
#ifdef DETAILED_DEBUG_INFO
blog(LOG_DEBUG, "MP: decode failed: %s",
av_err2str(ret));
#endif
if (d->packet_pending) {
av_packet_unref(&d->orig_pkt);
av_init_packet(&d->orig_pkt);
av_init_packet(&d->pkt);
d->packet_pending = false;
}
return true;
}
d->frame_ready = !!got_frame;
if (d->packet_pending) {
if (d->pkt.size) {
d->pkt.data += ret;
d->pkt.size -= ret;
}
if (d->pkt.size <= 0) {
av_packet_unref(&d->orig_pkt);
av_init_packet(&d->orig_pkt);
av_init_packet(&d->pkt);
d->packet_pending = false;
}
}
}
if (d->frame_ready) {
int64_t last_pts = d->frame_pts;
if (d->frame->best_effort_timestamp == AV_NOPTS_VALUE)
d->frame_pts = d->next_pts;
else
d->frame_pts = av_rescale_q(
d->frame->best_effort_timestamp,
d->stream->time_base,
(AVRational){1, 1000000000});
int64_t duration = d->frame->pkt_duration;
if (!duration)
duration = get_estimated_duration(d, last_pts);
else
duration = av_rescale_q(duration,
d->stream->time_base,
(AVRational){1, 1000000000});
d->last_duration = duration;
d->next_pts = d->frame_pts + duration;
}
return true;
}
bool mp_decode_next(struct mp_decode *d)
{
bool eof = d->m->eof;
int got_frame;
int ret;
d->frame_ready = false;
if (!eof && !d->packets.size)
return true;
while (!d->frame_ready) {
if (!d->packet_pending) {
if (!d->packets.size) {
if (eof) {
d->pkt.data = NULL;
d->pkt.size = 0;
} else {
return true;
}
} else {
circlebuf_pop_front(&d->packets, &d->orig_pkt,
sizeof(d->orig_pkt));
d->pkt = d->orig_pkt;
d->packet_pending = true;
}
}
if (d->audio) {
ret = avcodec_decode_audio4(d->decoder,
d->frame, &got_frame, &d->pkt);
} else {
if (d->m->is_network && !d->got_first_keyframe) {
if (d->pkt.flags & AV_PKT_FLAG_KEY) {
d->got_first_keyframe = true;
} else {
av_packet_unref(&d->orig_pkt);
av_init_packet(&d->orig_pkt);
av_init_packet(&d->pkt);
d->packet_pending = false;
return true;
}
}
ret = avcodec_decode_video2(d->decoder,
d->frame, &got_frame, &d->pkt);
}
if (!got_frame && ret == 0) {
d->eof = true;
return true;
}
if (ret < 0) {
blog(LOG_DEBUG, "MP: decode failed: %s",
av_err2str(ret));
if (d->packet_pending) {
av_packet_unref(&d->orig_pkt);
av_init_packet(&d->orig_pkt);
av_init_packet(&d->pkt);
d->packet_pending = false;
}
return true;
}
d->frame_ready = !!got_frame;
if (d->packet_pending) {
if (d->pkt.size) {
d->pkt.data += ret;
d->pkt.size -= ret;
}
if (d->pkt.size == 0) {
av_packet_unref(&d->orig_pkt);
av_init_packet(&d->orig_pkt);
av_init_packet(&d->pkt);
d->packet_pending = false;
}
}
}
if (d->frame_ready) {
int64_t last_pts = d->frame_pts;
d->frame_pts = av_rescale_q(d->frame->best_effort_timestamp,
d->stream->time_base,
(AVRational){1, 1000000000});
int64_t duration = d->frame->pkt_duration;
if (!duration)
duration = get_estimated_duration(d, last_pts);
else
duration = av_rescale_q(duration,
d->stream->time_base,
(AVRational){1, 1000000000});
d->last_duration = duration;
d->next_pts = d->frame_pts + duration;
}
return true;
}
bool audio_callback(void *param,
uint64_t start_ts_in, uint64_t end_ts_in, uint64_t *out_ts,
uint32_t mixers, struct audio_output_data *mixes)
{
struct obs_core_data *data = &obs->data;
struct obs_core_audio *audio = &obs->audio;
struct obs_source *source;
size_t sample_rate = audio_output_get_sample_rate(audio->audio);
size_t channels = audio_output_get_channels(audio->audio);
struct ts_info ts = {start_ts_in, end_ts_in};
size_t audio_size;
uint64_t min_ts;
da_resize(audio->render_order, 0);
da_resize(audio->root_nodes, 0);
circlebuf_push_back(&audio->buffered_timestamps, &ts, sizeof(ts));
circlebuf_peek_front(&audio->buffered_timestamps, &ts, sizeof(ts));
min_ts = ts.start;
audio_size = AUDIO_OUTPUT_FRAMES * sizeof(float);
#if DEBUG_AUDIO == 1
blog(LOG_DEBUG, "ts %llu-%llu", ts.start, ts.end);
#endif
/* ------------------------------------------------ */
/* build audio render order
* NOTE: these are source channels, not audio channels */
for (uint32_t i = 0; i < MAX_CHANNELS; i++) {
obs_source_t *source = obs_get_output_source(i);
if (source) {
obs_source_enum_active_tree(source, push_audio_tree,
audio);
push_audio_tree(NULL, source, audio);
da_push_back(audio->root_nodes, &source);
obs_source_release(source);
}
}
pthread_mutex_lock(&data->audio_sources_mutex);
source = data->first_audio_source;
while (source) {
push_audio_tree(NULL, source, audio);
source = (struct obs_source*)source->next_audio_source;
}
pthread_mutex_unlock(&data->audio_sources_mutex);
/* ------------------------------------------------ */
/* render audio data */
for (size_t i = 0; i < audio->render_order.num; i++) {
obs_source_t *source = audio->render_order.array[i];
obs_source_audio_render(source, mixers, channels, sample_rate,
audio_size);
}
/* ------------------------------------------------ */
/* get minimum audio timestamp */
pthread_mutex_lock(&data->audio_sources_mutex);
calc_min_ts(data, sample_rate, &min_ts);
pthread_mutex_unlock(&data->audio_sources_mutex);
/* ------------------------------------------------ */
/* if a source has gone backward in time, buffer */
if (min_ts < ts.start)
add_audio_buffering(audio, sample_rate, &ts, min_ts);
/* ------------------------------------------------ */
/* mix audio */
if (!audio->buffering_wait_ticks) {
for (size_t i = 0; i < audio->root_nodes.num; i++) {
obs_source_t *source = audio->root_nodes.array[i];
if (source->audio_pending)
continue;
pthread_mutex_lock(&source->audio_buf_mutex);
if (source->audio_output_buf[0][0] && source->audio_ts)
mix_audio(mixes, source, channels, sample_rate,
&ts);
pthread_mutex_unlock(&source->audio_buf_mutex);
}
}
/* ------------------------------------------------ */
/* discard audio */
pthread_mutex_lock(&data->audio_sources_mutex);
source = data->first_audio_source;
while (source) {
pthread_mutex_lock(&source->audio_buf_mutex);
discard_audio(audio, source, channels, sample_rate, &ts);
pthread_mutex_unlock(&source->audio_buf_mutex);
source = (struct obs_source*)source->next_audio_source;
}
pthread_mutex_unlock(&data->audio_sources_mutex);
/* ------------------------------------------------ */
/* release audio sources */
release_audio_sources(audio);
circlebuf_pop_front(&audio->buffered_timestamps, NULL, sizeof(ts));
*out_ts = ts.start;
if (audio->buffering_wait_ticks) {
audio->buffering_wait_ticks--;
return false;
}
UNUSED_PARAMETER(param);
return true;
}
static bool process_audio_delay(struct audio_monitor *monitor,
float **data, uint32_t *frames, uint64_t ts, uint32_t pad)
{
obs_source_t *s = monitor->source;
uint64_t last_frame_ts = s->last_frame_ts;
uint64_t cur_time = os_gettime_ns();
uint64_t front_ts;
uint64_t cur_ts;
int64_t diff;
uint32_t blocksize = monitor->channels * sizeof(float);
/* cut off audio if long-since leftover audio in delay buffer */
if (cur_time - monitor->last_recv_time > 1000000000)
circlebuf_free(&monitor->delay_buffer);
monitor->last_recv_time = cur_time;
ts += monitor->source->sync_offset;
circlebuf_push_back(&monitor->delay_buffer, &ts, sizeof(ts));
circlebuf_push_back(&monitor->delay_buffer, frames, sizeof(*frames));
circlebuf_push_back(&monitor->delay_buffer, *data,
*frames * blocksize);
if (!monitor->prev_video_ts) {
monitor->prev_video_ts = last_frame_ts;
} else if (monitor->prev_video_ts == last_frame_ts) {
monitor->time_since_prev += (uint64_t)*frames *
1000000000ULL / (uint64_t)monitor->sample_rate;
} else {
monitor->time_since_prev = 0;
}
while (monitor->delay_buffer.size != 0) {
size_t size;
bool bad_diff;
circlebuf_peek_front(&monitor->delay_buffer, &cur_ts,
sizeof(ts));
front_ts = cur_ts -
((uint64_t)pad * 1000000000ULL /
(uint64_t)monitor->sample_rate);
diff = (int64_t)front_ts - (int64_t)last_frame_ts;
bad_diff = !last_frame_ts ||
llabs(diff) > 5000000000 ||
monitor->time_since_prev > 100000000ULL;
/* delay audio if rushing */
if (!bad_diff && diff > 75000000) {
#ifdef DEBUG_AUDIO
blog(LOG_INFO, "audio rushing, cutting audio, "
"diff: %lld, delay buffer size: %lu, "
"v: %llu: a: %llu",
diff, (int)monitor->delay_buffer.size,
last_frame_ts, front_ts);
#endif
return false;
}
circlebuf_pop_front(&monitor->delay_buffer, NULL, sizeof(ts));
circlebuf_pop_front(&monitor->delay_buffer, frames,
sizeof(*frames));
size = *frames * blocksize;
da_resize(monitor->buf, size);
circlebuf_pop_front(&monitor->delay_buffer,
monitor->buf.array, size);
/* cut audio if dragging */
if (!bad_diff && diff < -75000000 && monitor->delay_buffer.size > 0) {
#ifdef DEBUG_AUDIO
blog(LOG_INFO, "audio dragging, cutting audio, "
"diff: %lld, delay buffer size: %lu, "
"v: %llu: a: %llu",
diff, (int)monitor->delay_buffer.size,
last_frame_ts, front_ts);
#endif
continue;
}
*data = monitor->buf.array;
return true;
}
return false;
}
static inline void discard_audio(struct obs_core_audio *audio,
obs_source_t *source, size_t channels, size_t sample_rate,
struct ts_info *ts)
{
size_t total_floats = AUDIO_OUTPUT_FRAMES;
size_t size;
#if DEBUG_AUDIO == 1
bool is_audio_source = source->info.output_flags & OBS_SOURCE_AUDIO;
#endif
if (source->info.audio_render) {
source->audio_ts = 0;
return;
}
if (ts->end <= source->audio_ts) {
#if DEBUG_AUDIO == 1
blog(LOG_DEBUG, "can't discard, source "
"timestamp (%"PRIu64") >= "
"end timestamp (%"PRIu64")",
source->audio_ts, ts->end);
#endif
return;
}
if (source->audio_ts < (ts->start - 1)) {
if (discard_if_stopped(source, channels))
return;
#if DEBUG_AUDIO == 1
if (is_audio_source) {
blog(LOG_DEBUG, "can't discard, source "
"timestamp (%"PRIu64") < "
"start timestamp (%"PRIu64")",
source->audio_ts, ts->start);
}
#endif
if (audio->total_buffering_ticks == MAX_BUFFERING_TICKS)
ignore_audio(source, channels, sample_rate);
return;
}
if (source->audio_ts != ts->start &&
source->audio_ts != (ts->start - 1)) {
size_t start_point = convert_time_to_frames(sample_rate,
source->audio_ts - ts->start);
if (start_point == AUDIO_OUTPUT_FRAMES) {
#if DEBUG_AUDIO == 1
if (is_audio_source)
blog(LOG_DEBUG, "can't dicard, start point is "
"at audio frame count");
#endif
return;
}
total_floats -= start_point;
}
size = total_floats * sizeof(float);
if (source->audio_input_buf[0].size < size) {
if (discard_if_stopped(source, channels))
return;
#if DEBUG_AUDIO == 1
if (is_audio_source)
blog(LOG_DEBUG, "can't discard, data still pending");
#endif
return;
}
for (size_t ch = 0; ch < channels; ch++)
circlebuf_pop_front(&source->audio_input_buf[ch], NULL, size);
source->last_audio_input_buf_size = 0;
#if DEBUG_AUDIO == 1
if (is_audio_source)
blog(LOG_DEBUG, "audio discarded, new ts: %"PRIu64,
ts->end);
#endif
source->audio_ts = ts->end;
}