X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.cpp;h=e3af8789530f06186c527c31334d071cd49572f6;hb=a98732df9454028ddaa54a9d07b5d7513767bfe9;hp=506af4aeabb95fa8d554cc32baf80dfe5c61b3ff;hpb=fae8d2ae053d580ad27a7a0bd71031e3df8f618f;p=nageru diff --git a/mixer.cpp b/mixer.cpp index 506af4a..e3af878 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" @@ -66,11 +62,13 @@ 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)), - level_compressor(OUTPUT_FREQUENCY) + level_compressor(OUTPUT_FREQUENCY), + limiter(OUTPUT_FREQUENCY), + compressor(OUTPUT_FREQUENCY) { httpd.start(9095); @@ -107,7 +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( @@ -122,7 +120,7 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards) [this]{ resource_pool->clean_context(); }); - card->resampler.reset(new Resampler(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2)); + card->resampling_queue.reset(new ResamplingQueue(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2)); card->usb->configure_card(); } @@ -151,6 +149,12 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards) r128.integr_start(); locut.init(FILTER_HPF, 2); + + // 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() @@ -166,6 +170,8 @@ Mixer::~Mixer() } cards[card_index].usb->stop_dequeue_thread(); } + + h264_encoder.reset(nullptr); } namespace { @@ -180,10 +186,10 @@ int unwrap_timecode(uint16_t current_wrapped, int last) } } -float find_peak(const vector &samples) +float find_peak(const float *samples, size_t num_samples) { float m = fabs(samples[0]); - for (size_t i = 1; i < samples.size(); ++i) { + for (size_t i = 1; i < num_samples; ++i) { m = std::max(m, fabs(samples[i])); } return m; @@ -212,7 +218,15 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, { CaptureCard *card = &cards[card_index]; - if (audio_frame.len - audio_offset > 30000) { + unsigned 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); @@ -225,16 +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); @@ -243,24 +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); - card->resampler.reset(new Resampler(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2)); + // 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->resampler->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->resampler->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); @@ -273,7 +299,8 @@ 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)) { if (video_frame.len != 0) { printf("Card %d: Dropping video frame with wrong length (%ld)\n", card_index, video_frame.len - video_offset); @@ -288,6 +315,7 @@ 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(); @@ -295,7 +323,7 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, return; } - const PBOFrameAllocator::Userdata *userdata = (const PBOFrameAllocator::Userdata *)video_frame.userdata; + PBOFrameAllocator::Userdata *userdata = (PBOFrameAllocator::Userdata *)video_frame.userdata; GLuint pbo = userdata->pbo; check_error(); glBindBuffer(GL_PIXEL_UNPACK_BUFFER_ARB, pbo); @@ -306,14 +334,34 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, //check_error(); // Upload the textures. - glBindTexture(GL_TEXTURE_2D, userdata->tex_y); - 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)); - check_error(); - glBindTexture(GL_TEXTURE_2D, userdata->tex_cbcr); - check_error(); - glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, WIDTH/2, HEIGHT, GL_RG, GL_UNSIGNED_BYTE, BUFFER_OFFSET(WIDTH * 25 + 22)); - check_error(); + size_t cbcr_width = width / 2; + size_t cbcr_offset = video_offset / 2; + size_t y_offset = video_frame.size / 2 + video_offset / 2; + + if (width != userdata->last_width || height != userdata->last_height) { + // We changed resolution since last use of this texture, so we need to create + // a new object. Note that this each card has its own PBOFrameAllocator, + // we don't need to worry about these flip-flopping between resolutions. + glBindTexture(GL_TEXTURE_2D, userdata->tex_cbcr); + check_error(); + glTexImage2D(GL_TEXTURE_2D, 0, GL_RG8, cbcr_width, height, 0, 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_y); + check_error(); + glTexImage2D(GL_TEXTURE_2D, 0, GL_R8, width, height, 0, GL_RED, GL_UNSIGNED_BYTE, BUFFER_OFFSET(y_offset + width * extra_lines_top)); + check_error(); + userdata->last_width = width; + userdata->last_height = height; + } else { + glBindTexture(GL_TEXTURE_2D, userdata->tex_cbcr); + check_error(); + 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_y); + check_error(); + 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(); GLsync fence = glFenceSync(GL_SYNC_GPU_COMMANDS_COMPLETE, /*flags=*/0); @@ -324,6 +372,7 @@ 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(); @@ -343,10 +392,11 @@ 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); @@ -360,20 +410,33 @@ 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; } } @@ -401,8 +464,8 @@ 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; } @@ -424,7 +487,7 @@ void Mixer::thread_func() check_error(); } const PBOFrameAllocator::Userdata *userdata = (const PBOFrameAllocator::Userdata *)card->new_frame->userdata; - theme->set_input_textures(card_index, userdata->tex_y, userdata->tex_cbcr); + theme->set_input_textures(card_index, userdata->tex_y, userdata->tex_cbcr, userdata->last_width, userdata->last_height); } // Get the main chain from the theme, and set its state immediately. @@ -464,10 +527,10 @@ void Mixer::thread_func() for (unsigned card_index = 0; card_index < num_cards; ++card_index) { input_frames.push_back(bmusb_current_rendering_frame[card_index]); } - const int64_t av_delay = TIMEBASE / 10; // Corresponds to the fixed delay in resampler.h. TODO: Make less hard-coded. + 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. @@ -502,7 +565,7 @@ 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(); } @@ -523,15 +586,31 @@ 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].resampler->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); } } @@ -541,10 +620,11 @@ 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.) - locut.render(samples_out.data(), samples_out.size() / 2, 150.0 * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f); + // 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. // Basically, if it's over about -40 dBFS, we squeeze it down to that level @@ -552,16 +632,16 @@ void Mixer::process_audio_one_frame() // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course, // entirely arbitrary, but from practical tests with speech, it seems to // put ut around -23 LUFS, so it's a reasonable starting point for later use. - // - // TODO: Add the actual compressors/limiters (for taking care of transients) - // later in the chain. - float threshold = 0.01f; // -40 dBFS. - float ratio = 20.0f; - float attack_time = 0.5f; - float release_time = 20.0f; - float makeup_gain = pow(10.0f, 26.0f / 20.0f); // +26 dB takes us to -14 dBFS. - level_compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain); - last_gain_staging_db = 20.0 * log10(level_compressor.get_attenuation() * makeup_gain); + float ref_level_dbfs = -14.0f; + { + float threshold = 0.01f; // -40 dBFS. + float ratio = 20.0f; + float attack_time = 0.5f; + float release_time = 20.0f; + float makeup_gain = pow(10.0f, (ref_level_dbfs - (-40.0f)) / 20.0f); // +26 dB. + level_compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain); + last_gain_staging_db = 20.0 * log10(level_compressor.get_attenuation() * makeup_gain); + } #if 0 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n", @@ -570,15 +650,60 @@ void Mixer::process_audio_one_frame() 20.0 * log10(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain)); #endif - // Find peak and R128 levels. - peak = std::max(peak, find_peak(samples_out)); +// float limiter_att, compressor_att; + + // 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; + float makeup_gain = 2.0f; // +6 dB. + compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain); +// 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. + peak_resampler.inp_data = samples_out.data(); + peak_resampler.inp_count = samples_out.size() / 2; + + vector interpolated_samples_out; + interpolated_samples_out.resize(samples_out.size()); + while (peak_resampler.inp_count > 0) { // About four iterations. + peak_resampler.out_data = &interpolated_samples_out[0]; + peak_resampler.out_count = interpolated_samples_out.size() / 2; + peak_resampler.process(); + size_t out_stereo_samples = interpolated_samples_out.size() / 2 - peak_resampler.out_count; + peak = max(peak, find_peak(interpolated_samples_out.data(), out_stereo_samples * 2)); + } + + // Find R128 levels. vector left, right; deinterleave_samples(samples_out, &left, &right); 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) @@ -648,12 +773,14 @@ 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) @@ -666,6 +793,14 @@ void Mixer::channel_clicked(int preview_num) theme->channel_clicked(preview_num); } +void Mixer::reset_meters() +{ + peak_resampler.reset(); + peak = 0.0f; + r128.reset(); + r128.integr_start(); +} + Mixer::OutputChannel::~OutputChannel() { if (has_current_frame) {