X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.cpp;h=ec333ef1124357dfffa3554557f0a6650839ff7a;hb=dc8948dc58ec1d212d544930967f6e4dda0114bd;hp=f28ea378f8f48410ad890f24df1c144da07a6162;hpb=ab30e757e8a5f39acae77e833e168d732ae37073;p=nageru diff --git a/mixer.cpp b/mixer.cpp index f28ea37..ec333ef 100644 --- a/mixer.cpp +++ b/mixer.cpp @@ -4,11 +4,11 @@ #include #include -#include #include #include #include #include +#include #include #include #include @@ -30,6 +30,7 @@ #include "bmusb/bmusb.h" #include "context.h" #include "defs.h" +#include "flags.h" #include "h264encode.h" #include "pbo_frame_allocator.h" #include "ref_counted_gl_sync.h" @@ -74,18 +75,36 @@ 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)); @@ -114,7 +133,7 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards) display_chain->set_dither_bits(0); // Don't bother. display_chain->finalize(); - h264_encoder.reset(new H264Encoder(h264_encoder_surface, WIDTH, HEIGHT, &httpd)); + h264_encoder.reset(new H264Encoder(h264_encoder_surface, global_flags.va_display, WIDTH, HEIGHT, &httpd)); for (unsigned card_index = 0; card_index < num_cards; ++card_index) { printf("Configuring card %d...\n", card_index); @@ -146,22 +165,49 @@ 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. - string cbcr_vert_shader = read_file("vs-cbcr.130.vert"); + string cbcr_vert_shader = + "#version 130 \n" + " \n" + "in vec2 position; \n" + "in vec2 texcoord; \n" + "out vec2 tc0; \n" + "uniform vec2 foo_chroma_offset_0; \n" + " \n" + "void main() \n" + "{ \n" + " // The result of glOrtho(0.0, 1.0, 0.0, 1.0, 0.0, 1.0) is: \n" + " // \n" + " // 2.000 0.000 0.000 -1.000 \n" + " // 0.000 2.000 0.000 -1.000 \n" + " // 0.000 0.000 -2.000 -1.000 \n" + " // 0.000 0.000 0.000 1.000 \n" + " gl_Position = vec4(2.0 * position.x - 1.0, 2.0 * position.y - 1.0, -1.0, 1.0); \n" + " vec2 flipped_tc = texcoord; \n" + " tc0 = flipped_tc + foo_chroma_offset_0; \n" + "} \n"; string cbcr_frag_shader = "#version 130 \n" "in vec2 tc0; \n" "uniform sampler2D cbcr_tex; \n" + "out vec4 FragColor; \n" "void main() { \n" - " gl_FragColor = texture2D(cbcr_tex, tc0); \n" + " FragColor = texture(cbcr_tex, tc0); \n" "} \n"; vector frag_shader_outputs; cbcr_program_num = resource_pool->compile_glsl_program(cbcr_vert_shader, cbcr_frag_shader, frag_shader_outputs); + float vertices[] = { + 0.0f, 2.0f, + 0.0f, 0.0f, + 2.0f, 0.0f + }; + cbcr_vbo = generate_vbo(2, GL_FLOAT, sizeof(vertices), vertices); + cbcr_position_attribute_index = glGetAttribLocation(cbcr_program_num, "position"); + cbcr_texcoord_attribute_index = glGetAttribLocation(cbcr_program_num, "texcoord"); + r128.init(2, OUTPUT_FREQUENCY); r128.integr_start(); @@ -169,7 +215,7 @@ 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)); } @@ -177,6 +223,7 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards) Mixer::~Mixer() { resource_pool->release_glsl_program(cbcr_program_num); + glDeleteBuffers(1, &cbcr_vbo); BMUSBCapture::stop_bm_thread(); for (unsigned card_index = 0; card_index < num_cards; ++card_index) { @@ -207,7 +254,7 @@ 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 +287,7 @@ 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 +362,12 @@ 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 +393,8 @@ 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 +402,9 @@ 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. @@ -401,6 +451,8 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, check_error(); glBindTexture(GL_TEXTURE_2D, 0); check_error(); + glBindBuffer(GL_PIXEL_UNPACK_BUFFER_ARB, 0); + check_error(); GLsync fence = glFenceSync(GL_SYNC_GPU_COMMANDS_COMPLETE, /*flags=*/0); check_error(); assert(fence != nullptr); @@ -491,6 +543,7 @@ 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 +561,7 @@ 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 +569,8 @@ 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 +614,7 @@ 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 +680,15 @@ 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, global_flags.va_display, 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 @@ -661,6 +725,11 @@ 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); { @@ -669,8 +738,7 @@ 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); } } @@ -679,7 +747,9 @@ 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 +757,23 @@ 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 +822,57 @@ 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; } - // 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() }; { - 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. @@ -770,12 +890,6 @@ void Mixer::subsample_chroma(GLuint src_tex, GLuint dst_tex) glGenVertexArrays(1, &vao); check_error(); - float vertices[] = { - 0.0f, 2.0f, - 0.0f, 0.0f, - 2.0f, 0.0f - }; - glBindVertexArray(vao); check_error(); @@ -802,17 +916,28 @@ void Mixer::subsample_chroma(GLuint src_tex, GLuint dst_tex) float chroma_offset_0[] = { -0.5f / WIDTH, 0.0f }; set_uniform_vec2(cbcr_program_num, "foo", "chroma_offset_0", chroma_offset_0); - GLuint position_vbo = fill_vertex_attribute(cbcr_program_num, "position", 2, GL_FLOAT, sizeof(vertices), vertices); - GLuint texcoord_vbo = fill_vertex_attribute(cbcr_program_num, "texcoord", 2, GL_FLOAT, sizeof(vertices), vertices); // Same as vertices. + glBindBuffer(GL_ARRAY_BUFFER, cbcr_vbo); + check_error(); + + for (GLint attr_index : { cbcr_position_attribute_index, cbcr_texcoord_attribute_index }) { + glEnableVertexAttribArray(attr_index); + check_error(); + glVertexAttribPointer(attr_index, 2, GL_FLOAT, GL_FALSE, 0, BUFFER_OFFSET(0)); + check_error(); + } glDrawArrays(GL_TRIANGLES, 0, 3); check_error(); - cleanup_vertex_attribute(cbcr_program_num, "position", position_vbo); - cleanup_vertex_attribute(cbcr_program_num, "texcoord", texcoord_vbo); + for (GLint attr_index : { cbcr_position_attribute_index, cbcr_texcoord_attribute_index }) { + glDisableVertexAttribArray(attr_index); + check_error(); + } glUseProgram(0); check_error(); + glBindFramebuffer(GL_FRAMEBUFFER, 0); + check_error(); resource_pool->release_fbo(fbo); glDeleteVertexArrays(1, &vao); @@ -857,6 +982,7 @@ void Mixer::reset_meters() peak = 0.0f; r128.reset(); r128.integr_start(); + correlation.reset(); } Mixer::OutputChannel::~OutputChannel()