X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.cpp;h=5fd5d1f7b452f2cda88a56cc6cccc2df62fe00df;hb=c3003b00fa5281e2905a9e2f204d7b31893d01fc;hp=bc349bef275304a57533281667cc5a892b80feee;hpb=9f0fe70f88ed4a9b20ced19d971c0df14cec4069;p=nageru diff --git a/mixer.cpp b/mixer.cpp index bc349be..5fd5d1f 100644 --- a/mixer.cpp +++ b/mixer.cpp @@ -1,7 +1,3 @@ -#define WIDTH 1280 -#define HEIGHT 720 -#define EXTRAHEIGHT 30 - #undef Success #include "mixer.h" @@ -66,7 +62,7 @@ void convert_fixed24_to_fp32(float *dst, size_t out_channels, const uint8_t *src } // 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)), @@ -109,7 +105,7 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards) 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(8 << 20, WIDTH, HEIGHT)); // 8 MB. card->usb->set_video_frame_allocator(card->frame_allocator.get()); card->surface = create_surface(format); card->usb->set_dequeue_thread_callbacks( @@ -157,6 +153,8 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards) // 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() @@ -172,6 +170,8 @@ Mixer::~Mixer() } cards[card_index].usb->stop_dequeue_thread(); } + + h264_encoder.reset(nullptr); } namespace { @@ -218,7 +218,15 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, { CaptureCard *card = &cards[card_index]; - if (audio_frame.len - audio_offset > 30000) { + int width, height, frame_rate_nom, frame_rate_den, extra_lines_top, extra_lines_bottom; + bool interlaced; + + decode_video_format(video_format, &width, &height, &extra_lines_top, &extra_lines_bottom, + &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); @@ -231,16 +239,13 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, 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 audio; audio.resize(num_samples * 2); convert_fixed24_to_fp32(&audio[0], 2, audio_frame.data + audio_offset, 8, num_samples); @@ -249,24 +254,39 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, { unique_lock 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)); + dropped_frames = 0; } 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 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); @@ -279,7 +299,9 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, if (card->should_quit) return; } - if (video_frame.len - video_offset != WIDTH * (HEIGHT+EXTRAHEIGHT) * 2) { + if (video_frame.len - video_offset == 0 || + video_frame.len - video_offset != size_t(width * (height + extra_lines_top + extra_lines_bottom) * 2) || + width != WIDTH || height != HEIGHT) { // TODO: Remove this once the rest of the infrastructure is in place. if (video_frame.len != 0) { printf("Card %d: Dropping video frame with wrong length (%ld)\n", card_index, video_frame.len - video_offset); @@ -294,6 +316,7 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, unique_lock 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(); @@ -312,13 +335,17 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, //check_error(); // Upload the textures. - glBindTexture(GL_TEXTURE_2D, userdata->tex_y); + size_t cbcr_width = width / 2; + size_t cbcr_offset = video_offset / 2; + size_t y_offset = video_frame.size / 2 + video_offset / 2; + + glBindTexture(GL_TEXTURE_2D, userdata->tex_cbcr); check_error(); - glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, WIDTH, HEIGHT, GL_RED, GL_UNSIGNED_BYTE, BUFFER_OFFSET((WIDTH * (HEIGHT+EXTRAHEIGHT) * 2 + 44) / 2 + WIDTH * 25 + 22)); + glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, cbcr_width, height, GL_RG, GL_UNSIGNED_BYTE, BUFFER_OFFSET(cbcr_offset + cbcr_width * extra_lines_top * sizeof(uint16_t))); check_error(); - glBindTexture(GL_TEXTURE_2D, userdata->tex_cbcr); + glBindTexture(GL_TEXTURE_2D, userdata->tex_y); check_error(); - glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, WIDTH/2, HEIGHT, GL_RG, GL_UNSIGNED_BYTE, BUFFER_OFFSET(WIDTH * 25 + 22)); + glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, width, height, GL_RED, GL_UNSIGNED_BYTE, BUFFER_OFFSET(y_offset + width * extra_lines_top)); check_error(); glBindTexture(GL_TEXTURE_2D, 0); check_error(); @@ -330,6 +357,7 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, unique_lock 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(); @@ -349,10 +377,11 @@ void Mixer::thread_func() 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 lock(bmusb_mutex); @@ -366,20 +395,33 @@ void Mixer::thread_func() 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); } } // 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 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. - ++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; } } @@ -407,8 +449,8 @@ void Mixer::thread_func() // 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; } @@ -473,7 +515,7 @@ void Mixer::thread_func() 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. @@ -508,7 +550,7 @@ void Mixer::thread_func() 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(); } @@ -529,15 +571,31 @@ void Mixer::thread_func() resource_pool->clean_context(); } -void Mixer::process_audio_one_frame() +void Mixer::audio_thread_func() +{ + while (!should_quit) { + AudioTask task; + + { + unique_lock 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 samples_card; vector 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 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); } } @@ -547,9 +605,10 @@ void Mixer::process_audio_one_frame() } } - // 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. @@ -578,22 +637,9 @@ void Mixer::process_audio_one_frame() // 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; @@ -602,6 +648,18 @@ void Mixer::process_audio_one_frame() // 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. @@ -624,8 +682,13 @@ void Mixer::process_audio_one_frame() 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) @@ -695,12 +758,14 @@ void Mixer::release_display_frame(DisplayFrame *frame) 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)