-#define WIDTH 1280
-#define HEIGHT 720
#define EXTRAHEIGHT 30
#undef Success
CaptureCard *card = &cards[card_index];
card->usb = new BMUSBCapture(card_index);
card->usb->set_frame_callback(bind(&Mixer::bm_frame, this, card_index, _1, _2, _3, _4, _5, _6, _7));
- card->frame_allocator.reset(new PBOFrameAllocator(WIDTH * (HEIGHT+EXTRAHEIGHT) * 2 + 44, WIDTH, HEIGHT));
+ card->frame_allocator.reset(new PBOFrameAllocator(WIDTH * (HEIGHT+EXTRAHEIGHT) * 2 + 44 + 1, WIDTH, HEIGHT));
card->usb->set_video_frame_allocator(card->frame_allocator.get());
card->surface = create_surface(format);
card->usb->set_dequeue_thread_callbacks(
{
CaptureCard *card = &cards[card_index];
- if (audio_frame.len - audio_offset > 30000) {
+ int width, height, frame_rate_nom, frame_rate_den;
+ bool interlaced;
+
+ decode_video_format(video_format, &width, &height, &frame_rate_nom, &frame_rate_den, &interlaced); // Ignore return value for now.
+ int64_t frame_length = TIMEBASE * frame_rate_den / frame_rate_nom;
+
+ size_t num_samples = (audio_frame.len >= audio_offset) ? (audio_frame.len - audio_offset) / 8 / 3 : 0;
+ if (num_samples > OUTPUT_FREQUENCY / 10) {
printf("Card %d: Dropping frame with implausible audio length (len=%d, offset=%d) [timecode=0x%04x video_len=%d video_offset=%d video_format=%x)\n",
card_index, int(audio_frame.len), int(audio_offset),
timecode, int(video_frame.len), int(video_offset), video_format);
return;
}
- int unwrapped_timecode = timecode;
+ int64_t local_pts = card->next_local_pts;
int dropped_frames = 0;
if (card->last_timecode != -1) {
- unwrapped_timecode = unwrap_timecode(unwrapped_timecode, card->last_timecode);
- dropped_frames = unwrapped_timecode - card->last_timecode - 1;
+ dropped_frames = unwrap_timecode(timecode, card->last_timecode) - card->last_timecode - 1;
}
- card->last_timecode = unwrapped_timecode;
// Convert the audio to stereo fp32 and add it.
- size_t num_samples = (audio_frame.len >= audio_offset) ? (audio_frame.len - audio_offset) / 8 / 3 : 0;
vector<float> audio;
audio.resize(num_samples * 2);
convert_fixed24_to_fp32(&audio[0], 2, audio_frame.data + audio_offset, 8, num_samples);
{
unique_lock<mutex> lock(card->audio_mutex);
- int unwrapped_timecode = timecode;
- if (dropped_frames > FPS * 2) {
- fprintf(stderr, "Card %d lost more than two seconds (or time code jumping around), resetting resampler\n",
- card_index);
+ // Number of samples per frame if we need to insert silence.
+ // (Could be nonintegral, but resampling will save us then.)
+ int silence_samples = OUTPUT_FREQUENCY * frame_rate_den / frame_rate_nom;
+
+ if (dropped_frames > MAX_FPS * 2) {
+ fprintf(stderr, "Card %d lost more than two seconds (or time code jumping around; from 0x%04x to 0x%04x), resetting resampler\n",
+ card_index, card->last_timecode, timecode);
card->resampling_queue.reset(new ResamplingQueue(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2));
} else if (dropped_frames > 0) {
// Insert silence as needed.
fprintf(stderr, "Card %d dropped %d frame(s) (before timecode 0x%04x), inserting silence.\n",
card_index, dropped_frames, timecode);
vector<float> silence;
- silence.resize((OUTPUT_FREQUENCY / FPS) * 2);
+ silence.resize(silence_samples * 2);
for (int i = 0; i < dropped_frames; ++i) {
- card->resampling_queue->add_input_samples((unwrapped_timecode - dropped_frames + i) / double(FPS), silence.data(), (OUTPUT_FREQUENCY / FPS));
+ card->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), silence.data(), silence_samples);
+ // Note that if the format changed in the meantime, we have
+ // no way of detecting that; we just have to assume the frame length
+ // is always the same.
+ local_pts += frame_length;
}
}
- card->resampling_queue->add_input_samples(unwrapped_timecode / double(FPS), audio.data(), num_samples);
+ if (num_samples == 0) {
+ audio.resize(silence_samples * 2);
+ num_samples = silence_samples;
+ }
+ card->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
+ card->next_local_pts = local_pts + frame_length;
}
+ card->last_timecode = timecode;
+
// Done with the audio, so release it.
if (audio_frame.owner) {
audio_frame.owner->release_frame(audio_frame);
unique_lock<mutex> lock(bmusb_mutex);
card->new_data_ready = true;
card->new_frame = RefCountedFrame(FrameAllocator::Frame());
+ card->new_frame_length = frame_length;
card->new_data_ready_fence = nullptr;
card->dropped_frames = dropped_frames;
card->new_data_ready_changed.notify_all();
unique_lock<mutex> lock(bmusb_mutex);
card->new_data_ready = true;
card->new_frame = RefCountedFrame(video_frame);
+ card->new_frame_length = frame_length;
card->new_data_ready_fence = fence;
card->dropped_frames = dropped_frames;
card->new_data_ready_changed.notify_all();
clock_gettime(CLOCK_MONOTONIC, &start);
int frame = 0;
- int dropped_frames = 0;
+ int stats_dropped_frames = 0;
while (!should_quit) {
CaptureCard card_copy[MAX_CARDS];
+ int num_samples[MAX_CARDS];
{
unique_lock<mutex> lock(bmusb_mutex);
card_copy[card_index].usb = card->usb;
card_copy[card_index].new_data_ready = card->new_data_ready;
card_copy[card_index].new_frame = card->new_frame;
+ card_copy[card_index].new_frame_length = card->new_frame_length;
card_copy[card_index].new_data_ready_fence = card->new_data_ready_fence;
card_copy[card_index].dropped_frames = card->dropped_frames;
card->new_data_ready = false;
card->new_data_ready_changed.notify_all();
+
+ int num_samples_times_timebase = OUTPUT_FREQUENCY * card->new_frame_length + card->fractional_samples;
+ num_samples[card_index] = num_samples_times_timebase / TIMEBASE;
+ card->fractional_samples = num_samples_times_timebase % TIMEBASE;
+ assert(num_samples[card_index] >= 0);
}
}
{
// Signal to the audio thread to process this frame.
unique_lock<mutex> lock(audio_mutex);
- audio_pts_queue.push(pts_int);
- audio_pts_queue_changed.notify_one();
+ audio_task_queue.push(AudioTask{pts_int, num_samples[0]});
+ audio_task_queue_changed.notify_one();
}
if (frame_num != card_copy[0].dropped_frames) {
- // For dropped frames, increase the pts.
- ++dropped_frames;
- pts_int += TIMEBASE / FPS;
+ // For dropped frames, increase the pts. Note that if the format changed
+ // in the meantime, we have no way of detecting that; we just have to
+ // assume the frame length is always the same.
+ ++stats_dropped_frames;
+ pts_int += card_copy[0].new_frame_length;
}
}
// If the first card is reporting a corrupted or otherwise dropped frame,
// just increase the pts (skipping over this frame) and don't try to compute anything new.
if (card_copy[0].new_frame->len == 0) {
- ++dropped_frames;
- pts_int += TIMEBASE / FPS;
+ ++stats_dropped_frames;
+ pts_int += card_copy[0].new_frame_length;
continue;
}
const int64_t av_delay = TIMEBASE / 10; // Corresponds to the fixed delay in resampling_queue.h. TODO: Make less hard-coded.
h264_encoder->end_frame(fence, pts_int + av_delay, input_frames);
++frame;
- pts_int += TIMEBASE / FPS;
+ pts_int += card_copy[0].new_frame_length;
// The live frame just shows the RGBA texture we just rendered.
// It owns rgba_tex now.
1e-9 * (now.tv_nsec - start.tv_nsec);
if (frame % 100 == 0) {
printf("%d frames (%d dropped) in %.3f seconds = %.1f fps (%.1f ms/frame)\n",
- frame, dropped_frames, elapsed, frame / elapsed,
+ frame, stats_dropped_frames, elapsed, frame / elapsed,
1e3 * elapsed / frame);
// chain->print_phase_timing();
}
void Mixer::audio_thread_func()
{
while (!should_quit) {
- int64_t frame_pts_int;
+ AudioTask task;
{
unique_lock<mutex> lock(audio_mutex);
- audio_pts_queue_changed.wait(lock, [this]{ return !audio_pts_queue.empty(); });
- frame_pts_int = audio_pts_queue.front();
- audio_pts_queue.pop();
+ audio_task_queue_changed.wait(lock, [this]{ return !audio_task_queue.empty(); });
+ task = audio_task_queue.front();
+ audio_task_queue.pop();
}
- process_audio_one_frame(frame_pts_int);
+ process_audio_one_frame(task.pts_int, task.num_samples);
}
}
-void Mixer::process_audio_one_frame(int64_t frame_pts_int)
+void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
{
vector<float> samples_card;
vector<float> samples_out;
for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
- samples_card.resize((OUTPUT_FREQUENCY / FPS) * 2);
+ samples_card.resize(num_samples * 2);
{
unique_lock<mutex> lock(cards[card_index].audio_mutex);
- if (!cards[card_index].resampling_queue->get_output_samples(double(frame_pts_int) / TIMEBASE, &samples_card[0], OUTPUT_FREQUENCY / FPS)) {
+ if (!cards[card_index].resampling_queue->get_output_samples(double(frame_pts_int) / TIMEBASE, &samples_card[0], num_samples)) {
printf("Card %d reported previous underrun.\n", card_index);
}
}