X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.cpp;h=133fc28108a9b50e68e1e7ffa46848313a5e3d65;hb=67768135b61fe89047836d0443cefcb5cae4b485;hp=936a070129352c4b10a80747e740be1e1ebb2dd0;hpb=0faba3d989c344d3db70f241a06761407d966bf7;p=nageru diff --git a/mixer.cpp b/mixer.cpp index 936a070..133fc28 100644 --- a/mixer.cpp +++ b/mixer.cpp @@ -1,5 +1,3 @@ -#define EXTRAHEIGHT 30 - #undef Success #include "mixer.h" @@ -12,12 +10,12 @@ #include #include #include +#include #include #include #include #include #include -#include #include #include #include @@ -61,17 +59,51 @@ void convert_fixed24_to_fp32(float *dst, size_t out_channels, const uint8_t *src } } +void insert_new_frame(RefCountedFrame frame, unsigned field_num, bool interlaced, unsigned card_index, InputState *input_state) +{ + if (interlaced) { + for (unsigned frame_num = FRAME_HISTORY_LENGTH; frame_num --> 1; ) { // :-) + input_state->buffered_frames[card_index][frame_num] = + input_state->buffered_frames[card_index][frame_num - 1]; + } + input_state->buffered_frames[card_index][0] = { frame, field_num }; + } else { + for (unsigned frame_num = 0; frame_num < FRAME_HISTORY_LENGTH; ++frame_num) { + input_state->buffered_frames[card_index][frame_num] = { frame, field_num }; + } + } +} + +string generate_local_dump_filename(int frame) +{ + time_t now = time(NULL); + tm now_tm; + localtime_r(&now, &now_tm); + + char timestamp[256]; + strftime(timestamp, sizeof(timestamp), "%F-%T%z", &now_tm); + + // Use the frame number to disambiguate between two cuts starting + // on the same second. + char filename[256]; + snprintf(filename, sizeof(filename), "%s%s-f%02d%s", + LOCAL_DUMP_PREFIX, timestamp, frame % 100, LOCAL_DUMP_SUFFIX); + return filename; +} + } // namespace Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards) - : httpd(LOCAL_DUMP_FILE_NAME, WIDTH, HEIGHT), + : httpd(WIDTH, HEIGHT), num_cards(num_cards), mixer_surface(create_surface(format)), h264_encoder_surface(create_surface(format)), + correlation(OUTPUT_FREQUENCY), level_compressor(OUTPUT_FREQUENCY), limiter(OUTPUT_FREQUENCY), compressor(OUTPUT_FREQUENCY) { + httpd.open_output_file(generate_local_dump_filename(/*frame=*/0).c_str()); httpd.start(9095); CHECK(init_movit(MOVIT_SHADER_DIR, MOVIT_DEBUG_OFF)); @@ -107,7 +139,7 @@ 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( @@ -132,8 +164,6 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards) cards[card_index].usb->start_bm_capture(); } - //chain->enable_phase_timing(true); - // Set up stuff for NV12 conversion. // Cb/Cr shader. @@ -145,7 +175,8 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards) "void main() { \n" " gl_FragColor = texture2D(cbcr_tex, tc0); \n" "} \n"; - cbcr_program_num = resource_pool->compile_glsl_program(cbcr_vert_shader, cbcr_frag_shader); + vector frag_shader_outputs; + cbcr_program_num = resource_pool->compile_glsl_program(cbcr_vert_shader, cbcr_frag_shader, frag_shader_outputs); r128.init(2, OUTPUT_FREQUENCY); r128.integr_start(); @@ -154,7 +185,7 @@ 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); + peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0); alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2)); } @@ -192,7 +223,7 @@ float find_peak(const float *samples, size_t num_samples) { float m = fabs(samples[0]); for (size_t i = 1; i < num_samples; ++i) { - m = std::max(m, fabs(samples[i])); + m = max(m, fabs(samples[i])); } return m; } @@ -212,57 +243,6 @@ void deinterleave_samples(const vector &in, vector *out_l, vector< } } -// 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, @@ -271,9 +251,15 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, { CaptureCard *card = &cards[card_index]; - int64_t frame_length = find_frame_length(video_format); + unsigned width, height, second_field_start, frame_rate_nom, frame_rate_den, extra_lines_top, extra_lines_bottom; + bool interlaced; - if (audio_frame.len - audio_offset > 30000) { + decode_video_format(video_format, &width, &height, &second_field_start, &extra_lines_top, &extra_lines_bottom, + &frame_rate_nom, &frame_rate_den, &interlaced); // Ignore return value for now. + int64_t frame_length = int64_t(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); @@ -291,10 +277,8 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, if (card->last_timecode != -1) { dropped_frames = unwrap_timecode(timecode, card->last_timecode) - card->last_timecode - 1; } - 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; vector audio; audio.resize(num_samples * 2); convert_fixed24_to_fp32(&audio[0], 2, audio_frame.data + audio_offset, 8, num_samples); @@ -303,29 +287,38 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, { unique_lock lock(card->audio_mutex); + // 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), resetting resampler\n", - card_index); + 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. (The number of samples could be nonintegral, - // but resampling will save us then.) + // 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 * frame_length / TIMEBASE) * 2); + vector silence(silence_samples * 2, 0.0f); for (int i = 0; i < dropped_frames; ++i) { - card->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), silence.data(), silence.size() / 2); + 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; } } + 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); @@ -338,10 +331,12 @@ 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) { + size_t expected_length = width * (height + extra_lines_top + extra_lines_bottom) * 2; + if (video_frame.len - video_offset == 0 || + video_frame.len - video_offset != expected_length) { if (video_frame.len != 0) { - printf("Card %d: Dropping video frame with wrong length (%ld)\n", - card_index, video_frame.len - video_offset); + printf("Card %d: Dropping video frame with wrong length (%ld; expected %ld)\n", + card_index, video_frame.len - video_offset, expected_length); } if (video_frame.owner) { video_frame.owner->release_frame(video_frame); @@ -354,6 +349,7 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, card->new_data_ready = true; card->new_frame = RefCountedFrame(FrameAllocator::Frame()); card->new_frame_length = frame_length; + card->new_frame_interlaced = false; card->new_data_ready_fence = nullptr; card->dropped_frames = dropped_frames; card->new_data_ready_changed.notify_all(); @@ -361,39 +357,109 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, return; } - const PBOFrameAllocator::Userdata *userdata = (const PBOFrameAllocator::Userdata *)video_frame.userdata; - GLuint pbo = userdata->pbo; - check_error(); - glBindBuffer(GL_PIXEL_UNPACK_BUFFER_ARB, pbo); - check_error(); - glFlushMappedBufferRange(GL_PIXEL_UNPACK_BUFFER, 0, video_frame.size); - check_error(); - //glMemoryBarrier(GL_CLIENT_MAPPED_BUFFER_BARRIER_BIT); - //check_error(); + PBOFrameAllocator::Userdata *userdata = (PBOFrameAllocator::Userdata *)video_frame.userdata; + + unsigned num_fields = interlaced ? 2 : 1; + timespec frame_upload_start; + if (interlaced) { + // Send the two fields along as separate frames; the other side will need to add + // a deinterlacer to actually get this right. + assert(height % 2 == 0); + height /= 2; + assert(frame_length % 2 == 0); + frame_length /= 2; + num_fields = 2; + clock_gettime(CLOCK_MONOTONIC, &frame_upload_start); + } + userdata->last_interlaced = interlaced; + userdata->last_frame_rate_nom = frame_rate_nom; + userdata->last_frame_rate_den = frame_rate_den; + RefCountedFrame new_frame(video_frame); // 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(); - glBindTexture(GL_TEXTURE_2D, 0); - check_error(); - GLsync fence = glFenceSync(GL_SYNC_GPU_COMMANDS_COMPLETE, /*flags=*/0); - check_error(); - assert(fence != nullptr); + size_t cbcr_width = width / 2; + size_t cbcr_offset = video_offset / 2; + size_t y_offset = video_frame.size / 2 + video_offset / 2; + + for (unsigned field = 0; field < num_fields; ++field) { + unsigned field_start_line = (field == 1) ? second_field_start : extra_lines_top + field * (height + 22); + + if (userdata->tex_y[field] == 0 || + userdata->tex_cbcr[field] == 0 || + width != userdata->last_width[field] || + height != userdata->last_height[field]) { + // 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[field]); + check_error(); + glTexImage2D(GL_TEXTURE_2D, 0, GL_RG8, cbcr_width, height, 0, GL_RG, GL_UNSIGNED_BYTE, nullptr); + check_error(); + glBindTexture(GL_TEXTURE_2D, userdata->tex_y[field]); + check_error(); + glTexImage2D(GL_TEXTURE_2D, 0, GL_R8, width, height, 0, GL_RED, GL_UNSIGNED_BYTE, nullptr); + check_error(); + userdata->last_width[field] = width; + userdata->last_height[field] = height; + } - { - 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(); + GLuint pbo = userdata->pbo; + check_error(); + glBindBuffer(GL_PIXEL_UNPACK_BUFFER_ARB, pbo); + check_error(); + glMemoryBarrier(GL_CLIENT_MAPPED_BUFFER_BARRIER_BIT); + check_error(); + + glBindTexture(GL_TEXTURE_2D, userdata->tex_cbcr[field]); + check_error(); + glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, cbcr_width, height, GL_RG, GL_UNSIGNED_BYTE, BUFFER_OFFSET(cbcr_offset + cbcr_width * field_start_line * sizeof(uint16_t))); + check_error(); + glBindTexture(GL_TEXTURE_2D, userdata->tex_y[field]); + check_error(); + glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, width, height, GL_RED, GL_UNSIGNED_BYTE, BUFFER_OFFSET(y_offset + width * field_start_line)); + check_error(); + glBindTexture(GL_TEXTURE_2D, 0); + check_error(); + GLsync fence = glFenceSync(GL_SYNC_GPU_COMMANDS_COMPLETE, /*flags=*/0); + check_error(); + assert(fence != nullptr); + + if (field == 1) { + // Don't upload the second field as fast as we can; wait until + // the field time has approximately passed. (Otherwise, we could + // get timing jitter against the other sources, and possibly also + // against the video display, although the latter is not as critical.) + // This requires our system clock to be reasonably close to the + // video clock, but that's not an unreasonable assumption. + timespec second_field_start; + second_field_start.tv_nsec = frame_upload_start.tv_nsec + + frame_length * 1000000000 / TIMEBASE; + second_field_start.tv_sec = frame_upload_start.tv_sec + + second_field_start.tv_nsec / 1000000000; + second_field_start.tv_nsec %= 1000000000; + + while (clock_nanosleep(CLOCK_MONOTONIC, TIMER_ABSTIME, + &second_field_start, nullptr) == -1 && + errno == EINTR) ; + } + + { + unique_lock lock(bmusb_mutex); + card->new_data_ready = true; + card->new_frame = new_frame; + card->new_frame_length = frame_length; + card->new_frame_field = field; + card->new_frame_interlaced = interlaced; + card->new_data_ready_fence = fence; + card->dropped_frames = dropped_frames; + card->new_data_ready_changed.notify_all(); + + if (field != num_fields - 1) { + // Wait until the previous frame was consumed. + card->new_data_ready_changed.wait(lock, [card]{ return !card->new_data_ready || card->should_quit; }); + if (card->should_quit) return; + } + } } } @@ -410,7 +476,7 @@ 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]; @@ -429,6 +495,8 @@ void Mixer::thread_func() 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_frame_field = card->new_frame_field; + card_copy[card_index].new_frame_interlaced = card->new_frame_interlaced; 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; @@ -437,10 +505,12 @@ void Mixer::thread_func() 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. + assert(num_cards > 0); for (unsigned frame_num = 0; frame_num < card_copy[0].dropped_frames + 1; ++frame_num) { { // Signal to the audio thread to process this frame. @@ -452,12 +522,13 @@ void Mixer::thread_func() // 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; + ++stats_dropped_frames; pts_int += card_copy[0].new_frame_length; } } if (audio_level_callback != nullptr) { + unique_lock lock(compressor_mutex); double loudness_s = r128.loudness_S(); double loudness_i = r128.integrated(); double loudness_range_low = r128.range_min(); @@ -465,7 +536,8 @@ void Mixer::thread_func() audio_level_callback(loudness_s, 20.0 * log10(peak), loudness_i, loudness_range_low, loudness_range_high, - last_gain_staging_db); + gain_staging_db, 20.0 * log10(final_makeup_gain), + correlation.get_correlation()); } for (unsigned card_index = 1; card_index < num_cards; ++card_index) { @@ -481,7 +553,7 @@ 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; + ++stats_dropped_frames; pts_int += card_copy[0].new_frame_length; continue; } @@ -492,7 +564,7 @@ void Mixer::thread_func() continue; assert(card->new_frame != nullptr); - bmusb_current_rendering_frame[card_index] = card->new_frame; + insert_new_frame(card->new_frame, card->new_frame_field, card->new_frame_interlaced, card_index, &input_state); check_error(); // The new texture might still be uploaded, @@ -503,14 +575,13 @@ void Mixer::thread_func() glDeleteSync(card->new_data_ready_fence); 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); } // Get the main chain from the theme, and set its state immediately. - pair> theme_main_chain = theme->get_chain(0, pts(), WIDTH, HEIGHT); - EffectChain *chain = theme_main_chain.first; - theme_main_chain.second(); + Theme::Chain theme_main_chain = theme->get_chain(0, pts(), WIDTH, HEIGHT, input_state); + EffectChain *chain = theme_main_chain.chain; + theme_main_chain.setup_chain(); + //theme_main_chain.chain->enable_phase_timing(true); GLuint y_tex, cbcr_tex; bool got_frame = h264_encoder->begin_frame(&y_tex, &cbcr_tex); @@ -537,15 +608,8 @@ void Mixer::thread_func() RefCountedGLsync fence(GL_SYNC_GPU_COMMANDS_COMPLETE, /*flags=*/0); check_error(); - // Make sure the H.264 gets a reference to all the - // input frames needed, so that they are not released back - // until the rendering is done. - vector input_frames; - 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 resampling_queue.h. TODO: Make less hard-coded. - h264_encoder->end_frame(fence, pts_int + av_delay, input_frames); + h264_encoder->end_frame(fence, pts_int + av_delay, theme_main_chain.input_frames); ++frame; pts_int += card_copy[0].new_frame_length; @@ -564,15 +628,11 @@ void Mixer::thread_func() // Set up preview and any additional channels. for (int i = 1; i < theme->get_num_channels() + 2; ++i) { DisplayFrame display_frame; - pair> chain = theme->get_chain(i, pts(), WIDTH, HEIGHT); // FIXME: dimensions - display_frame.chain = chain.first; - display_frame.setup_chain = chain.second; + Theme::Chain chain = theme->get_chain(i, pts(), WIDTH, HEIGHT, input_state); // FIXME: dimensions + display_frame.chain = chain.chain; + display_frame.setup_chain = chain.setup_chain; display_frame.ready_fence = fence; - - // FIXME: possible to do better? - for (unsigned card_index = 0; card_index < num_cards; ++card_index) { - display_frame.input_frames.push_back(bmusb_current_rendering_frame[card_index]); - } + display_frame.input_frames = chain.input_frames; display_frame.temp_textures = {}; output_channel[i].output_frame(display_frame); } @@ -582,11 +642,20 @@ 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(); } + if (should_cut.exchange(false)) { // Test and clear. + string filename = generate_local_dump_filename(frame); + printf("Starting new recording: %s\n", filename.c_str()); + h264_encoder->shutdown(); + httpd.close_output_file(); + httpd.open_output_file(filename.c_str()); + h264_encoder.reset(new H264Encoder(h264_encoder_surface, WIDTH, HEIGHT, &httpd)); + } + #if 0 // Reset every 100 frames, so that local variations in frame times // (especially for the first few frames, when the shaders are @@ -623,6 +692,11 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples) { vector samples_card; vector samples_out; + + // TODO: Allow mixing audio from several sources. + unsigned selected_audio_card = theme->map_signal(audio_source_channel); + assert(selected_audio_card < num_cards); + for (unsigned card_index = 0; card_index < num_cards; ++card_index) { samples_card.resize(num_samples * 2); { @@ -631,8 +705,7 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples) printf("Card %d reported previous underrun.\n", card_index); } } - // TODO: Allow using audio from the other card(s) as well. - if (card_index == 0) { + if (card_index == selected_audio_card) { samples_out = move(samples_card); } } @@ -641,7 +714,9 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples) // 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); + if (locut_enabled) { + 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 @@ -649,15 +724,23 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples) // 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. - 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); + unique_lock lock(compressor_mutex); + if (level_compressor_enabled) { + 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); + gain_staging_db = 20.0 * log10(level_compressor.get_attenuation() * makeup_gain); + } else { + // Just apply the gain we already had. + float g = pow(10.0f, gain_staging_db / 20.0f); + for (size_t i = 0; i < samples_out.size(); ++i) { + samples_out[i] *= g; + } + } } #if 0 @@ -706,13 +789,58 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples) 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)); + peak_resampler.out_data = nullptr; } - // Find R128 levels. + // At this point, we are most likely close to +0 LU, but all of our + // measurements have been on raw sample values, not R128 values. + // So we have a final makeup gain to get us to +0 LU; the gain + // adjustments required should be relatively small, and also, the + // offset shouldn't change much (only if the type of audio changes + // significantly). Thus, we shoot for updating this value basically + // “whenever we process buffers”, since the R128 calculation isn't exactly + // something we get out per-sample. + // + // Note that there's a feedback loop here, so we choose a very slow filter + // (half-time of 100 seconds). + double target_loudness_factor, alpha; + { + unique_lock lock(compressor_mutex); + double loudness_lu = r128.loudness_M() - ref_level_lufs; + double current_makeup_lu = 20.0f * log10(final_makeup_gain); + target_loudness_factor = pow(10.0f, -loudness_lu / 20.0f); + + // If we're outside +/- 5 LU uncorrected, we don't count it as + // a normal signal (probably silence) and don't change the + // correction factor; just apply what we already have. + if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) { + alpha = 0.0; + } else { + // Formula adapted from + // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter. + const double half_time_s = 100.0; + const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY); + alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0); + } + + double m = final_makeup_gain; + for (size_t i = 0; i < samples_out.size(); i += 2) { + samples_out[i + 0] *= m; + samples_out[i + 1] *= m; + m += (target_loudness_factor - m) * alpha; + } + final_makeup_gain = m; + } + + // Find R128 levels and L/R correlation. vector left, right; deinterleave_samples(samples_out, &left, &right); float *ptrs[] = { left.data(), right.data() }; - r128.process(left.size(), ptrs); + { + unique_lock lock(compressor_mutex); + r128.process(left.size(), ptrs); + correlation.process_samples(samples_out); + } // Send the samples to the sound card. if (alsa) { @@ -816,6 +944,7 @@ void Mixer::reset_meters() peak = 0.0f; r128.reset(); r128.integr_start(); + correlation.reset(); } Mixer::OutputChannel::~OutputChannel()