-#define WIDTH 1280
-#define HEIGHT 720
#define EXTRAHEIGHT 30
#undef Success
} // namespace
Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
- : httpd("test.ts", WIDTH, HEIGHT),
+ : httpd(LOCAL_DUMP_FILE_NAME, WIDTH, HEIGHT),
num_cards(num_cards),
mixer_surface(create_surface(format)),
h264_encoder_surface(create_surface(format)),
// hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
// and there's a limit to how important the peak meter is.
peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16);
+
+ alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2));
}
Mixer::~Mixer()
}
cards[card_index].usb->stop_dequeue_thread();
}
+
+ h264_encoder.reset(nullptr);
}
namespace {
}
}
+// Returns length of a frame with the given format, in TIMEBASE units.
+int64_t find_frame_length(uint16_t video_format)
+{
+ if (video_format == 0x0800) {
+ // No video signal. These green pseudo-frames seem to come at about 30.13 Hz.
+ // It's a strange thing, but what can you do.
+ return TIMEBASE * 100 / 3013;
+ }
+ if ((video_format & 0xe800) != 0xe800) {
+ printf("Video format 0x%04x does not appear to be a video format. Assuming 60 Hz.\n",
+ video_format);
+ return TIMEBASE / 60;
+ }
+
+ // 0x8 seems to be a flag about availability of deep color on the input,
+ // except when it's not (e.g. it's the only difference between NTSC 23.98
+ // and PAL). Rather confusing. But we clear it here nevertheless, because
+ // usually it doesn't mean anything.
+ //
+ // We don't really handle interlaced formats at all yet.
+ uint16_t normalized_video_format = video_format & ~0xe808;
+ if (normalized_video_format == 0x0143) { // 720p50.
+ return TIMEBASE / 50;
+ } else if (normalized_video_format == 0x0103) { // 720p60.
+ return TIMEBASE / 60;
+ } else if (normalized_video_format == 0x0121) { // 720p59.94.
+ return TIMEBASE * 1001 / 60000;
+ } else if (normalized_video_format == 0x01c3 || // 1080p30.
+ normalized_video_format == 0x0003) { // 1080i60.
+ return TIMEBASE / 30;
+ } else if (normalized_video_format == 0x01e1 || // 1080p29.97.
+ normalized_video_format == 0x0021 || // 1080i59.94.
+ video_format == 0xe901 || // NTSC (480i59.94, I suppose).
+ video_format == 0xe9c1 || // Ditto.
+ video_format == 0xe801) { // Ditto.
+ return TIMEBASE * 1001 / 30000;
+ } else if (normalized_video_format == 0x0063 || // 1080p25.
+ normalized_video_format == 0x0043 || // 1080i50.
+ video_format == 0xe909) { // PAL (576i50, I suppose).
+ return TIMEBASE / 25;
+ } else if (normalized_video_format == 0x008e) { // 1080p24.
+ return TIMEBASE / 24;
+ } else if (normalized_video_format == 0x00a1) { // 1080p23.98.
+ return TIMEBASE * 1001 / 24000;
+ return TIMEBASE / 25;
+ } else {
+ printf("Unknown video format 0x%04x. Assuming 60 Hz.\n", video_format);
+ return TIMEBASE / 60;
+ }
+}
+
} // namespace
void Mixer::bm_frame(unsigned card_index, uint16_t timecode,
{
CaptureCard *card = &cards[card_index];
+ int64_t frame_length = find_frame_length(video_format);
+
if (audio_frame.len - audio_offset > 30000) {
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),
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;
+ card->last_timecode = 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;
{
unique_lock<mutex> lock(card->audio_mutex);
- int unwrapped_timecode = timecode;
- if (dropped_frames > FPS * 2) {
+ if (dropped_frames > MAX_FPS * 2) {
fprintf(stderr, "Card %d lost more than two seconds (or time code jumping around), resetting resampler\n",
card_index);
card->resampling_queue.reset(new ResamplingQueue(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2));
} else if (dropped_frames > 0) {
- // Insert silence as needed.
+ // Insert silence as needed. (The number of samples could be nonintegral,
+ // but resampling will save us then.)
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((OUTPUT_FREQUENCY * frame_length / TIMEBASE) * 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.size() / 2);
+ // 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);
+ card->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
+ card->next_local_pts = local_pts + frame_length;
}
// Done with the audio, so release it.
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();
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;
}
}
// Resample the audio as needed, including from previously dropped frames.
for (unsigned frame_num = 0; frame_num < card_copy[0].dropped_frames + 1; ++frame_num) {
- process_audio_one_frame();
+ {
+ // Signal to the audio thread to process this frame.
+ unique_lock<mutex> lock(audio_mutex);
+ 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.
+ // 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.
++dropped_frames;
- pts_int += TIMEBASE / FPS;
+ pts_int += card_copy[0].new_frame_length;
}
}
// 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;
+ 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.
resource_pool->clean_context();
}
-void Mixer::process_audio_one_frame()
+void Mixer::audio_thread_func()
+{
+ while (!should_quit) {
+ AudioTask task;
+
+ {
+ unique_lock<mutex> lock(audio_mutex);
+ 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(task.pts_int, task.num_samples);
+ }
+}
+
+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(pts(), &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);
}
}
}
}
- // Cut away everything under 150 Hz; we don't need it for voice,
- // and it will reduce headroom and confuse the compressor.
- // (In particular, any hums at 50 or 60 Hz should be dampened.)
+ // Cut away everything under 120 Hz (or whatever the cutoff is);
+ // we don't need it for voice, and it will reduce headroom
+ // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
+ // should be dampened.)
locut.render(samples_out.data(), samples_out.size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
// Apply a level compressor to get the general level right.
// float limiter_att, compressor_att;
- // Then a limiter at +0 dB (so, -14 dBFS) to take out the worst peaks only.
- // Note that since ratio is not infinite, we could go slightly higher than this.
- // Probably more tuning is warranted here.
- {
- float threshold = pow(10.0f, (ref_level_dbfs + 0.0f) / 20.0f); // +0 dB.
- float ratio = 30.0f;
- float attack_time = 0.0f; // Instant.
- float release_time = 0.005f;
- float makeup_gain = 1.0f; // 0 dB.
- limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
-// limiter_att = limiter.get_attenuation();
- }
-
- // Finally, the real compressor.
- {
- float threshold = pow(10.0f, (ref_level_dbfs - 12.0f) / 20.0f); // -12 dB.
+ // The real compressor.
+ if (compressor_enabled) {
+ float threshold = pow(10.0f, compressor_threshold_dbfs / 20.0f);
float ratio = 20.0f;
float attack_time = 0.005f;
float release_time = 0.040f;
// compressor_att = compressor.get_attenuation();
}
+ // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
+ // Note that since ratio is not infinite, we could go slightly higher than this.
+ if (limiter_enabled) {
+ float threshold = pow(10.0f, limiter_threshold_dbfs / 20.0f);
+ float ratio = 30.0f;
+ float attack_time = 0.0f; // Instant.
+ float release_time = 0.020f;
+ float makeup_gain = 1.0f; // 0 dB.
+ limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
+// limiter_att = limiter.get_attenuation();
+ }
+
// printf("limiter=%+5.1f compressor=%+5.1f\n", 20.0*log10(limiter_att), 20.0*log10(compressor_att));
// Upsample 4x to find interpolated peak.
float *ptrs[] = { left.data(), right.data() };
r128.process(left.size(), ptrs);
- // Actually add the samples to the output.
- h264_encoder->add_audio(pts_int, move(samples_out));
+ // Send the samples to the sound card.
+ if (alsa) {
+ alsa->write(samples_out);
+ }
+
+ // And finally add them to the output.
+ h264_encoder->add_audio(frame_pts_int, move(samples_out));
}
void Mixer::subsample_chroma(GLuint src_tex, GLuint dst_tex)
void Mixer::start()
{
mixer_thread = thread(&Mixer::thread_func, this);
+ audio_thread = thread(&Mixer::audio_thread_func, this);
}
void Mixer::quit()
{
should_quit = true;
mixer_thread.join();
+ audio_thread.join();
}
void Mixer::transition_clicked(int transition_num)
projects / nageru / blobdiff