X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.cpp;h=4b92041396bb896f67fa806c171d5edc7340c88b;hb=fb5360d92864760d70d8a69c8dd4a9e738bcc0f0;hp=f28ea378f8f48410ad890f24df1c144da07a6162;hpb=ab30e757e8a5f39acae77e833e168d732ae37073;p=nageru diff --git a/mixer.cpp b/mixer.cpp index f28ea37..4b92041 100644 --- a/mixer.cpp +++ b/mixer.cpp @@ -74,18 +74,36 @@ void insert_new_frame(RefCountedFrame frame, unsigned field_num, bool interlaced } } +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)); @@ -146,8 +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. @@ -169,7 +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)); } @@ -207,7 +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; } @@ -240,7 +256,7 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, 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 = TIMEBASE * frame_rate_den / frame_rate_nom; + 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) { @@ -315,11 +331,12 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, if (card->should_quit) return; } + 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 != size_t(width * (height + extra_lines_top + extra_lines_bottom) * 2)) { + 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); @@ -345,9 +362,8 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, unsigned num_fields = interlaced ? 2 : 1; timespec frame_upload_start; if (interlaced) { - // NOTE: This isn't deinterlacing. This is just sending the two fields along - // as separate frames without considering anything like the half-field offset. - // We'll need to add a proper deinterlacer on the receiving side to get this right. + // 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); @@ -355,6 +371,9 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, 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. @@ -491,6 +510,7 @@ void Mixer::thread_func() } // 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. @@ -508,7 +528,7 @@ void Mixer::thread_func() } if (audio_level_callback != nullptr) { - unique_lock lock(r128_mutex); + unique_lock lock(compressor_mutex); double loudness_s = r128.loudness_S(); double loudness_i = r128.integrated(); double loudness_range_low = r128.range_min(); @@ -516,7 +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) { @@ -560,6 +581,7 @@ void Mixer::thread_func() 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); @@ -625,6 +647,15 @@ void Mixer::thread_func() // 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 @@ -679,7 +710,9 @@ 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 @@ -687,15 +720,23 @@ 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 @@ -744,15 +785,57 @@ 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; + } + + // 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. + // Find R128 levels and L/R correlation. vector left, right; deinterleave_samples(samples_out, &left, &right); float *ptrs[] = { left.data(), right.data() }; { - unique_lock lock(r128_mutex); + unique_lock lock(compressor_mutex); r128.process(left.size(), ptrs); + correlation.process_samples(samples_out); } // Send the samples to the sound card. @@ -857,6 +940,7 @@ void Mixer::reset_meters() peak = 0.0f; r128.reset(); r128.integr_start(); + correlation.reset(); } Mixer::OutputChannel::~OutputChannel()