X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.cpp;h=42a02521258e93f34c71518c4de0917135c2fad4;hb=6800acad34e7b16d07da9d2dc7f2ecb93ee24b23;hp=cf2ed8311a001080641fabb2afc2ae077e7409cb;hpb=c7969079ddaa7ac4aced304f039420435933715e;p=nageru diff --git a/mixer.cpp b/mixer.cpp index cf2ed83..42a0252 100644 --- a/mixer.cpp +++ b/mixer.cpp @@ -1,5 +1,3 @@ -#define WIDTH 1280 -#define HEIGHT 720 #define EXTRAHEIGHT 30 #undef Success @@ -7,27 +5,20 @@ #include "mixer.h" #include -#include -#include -#include #include -#include -#include #include -#include -#include -#include -#include -#include +#include +#include +#include +#include +#include #include #include #include #include #include #include -#include -#include -#include +#include #include #include #include @@ -35,10 +26,12 @@ #include #include #include +#include #include #include "bmusb/bmusb.h" #include "context.h" +#include "defs.h" #include "h264encode.h" #include "pbo_frame_allocator.h" #include "ref_counted_gl_sync.h" @@ -71,10 +64,13 @@ 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)) + h264_encoder_surface(create_surface(format)), + level_compressor(OUTPUT_FREQUENCY), + limiter(OUTPUT_FREQUENCY), + compressor(OUTPUT_FREQUENCY) { httpd.start(9095); @@ -87,10 +83,9 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards) resource_pool.reset(new ResourcePool); theme.reset(new Theme("theme.lua", resource_pool.get(), num_cards)); - output_channel[OUTPUT_LIVE].parent = this; - output_channel[OUTPUT_PREVIEW].parent = this; - output_channel[OUTPUT_INPUT0].parent = this; - output_channel[OUTPUT_INPUT1].parent = this; + for (unsigned i = 0; i < NUM_OUTPUTS; ++i) { + output_channel[i].parent = this; + } ImageFormat inout_format; inout_format.color_space = COLORSPACE_sRGB; @@ -118,7 +113,7 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards) card->usb->set_dequeue_thread_callbacks( [card]{ eglBindAPI(EGL_OPENGL_API); - card->context = create_context(); + card->context = create_context(card->surface); if (!make_current(card->context, card->surface)) { printf("failed to create bmusb context\n"); exit(1); @@ -127,7 +122,7 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards) [this]{ resource_pool->clean_context(); }); - card->resampler.reset(new Resampler(48000.0, 48000.0, 2)); + card->resampling_queue.reset(new ResamplingQueue(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2)); card->usb->configure_card(); } @@ -152,8 +147,16 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards) "} \n"; cbcr_program_num = resource_pool->compile_glsl_program(cbcr_vert_shader, cbcr_frag_shader); - r128.init(2, 48000); + r128.init(2, OUTPUT_FREQUENCY); 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() @@ -169,6 +172,8 @@ Mixer::~Mixer() } cards[card_index].usb->stop_dequeue_thread(); } + + h264_encoder.reset(nullptr); } namespace { @@ -183,10 +188,10 @@ 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; @@ -247,21 +252,21 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, unique_lock lock(card->audio_mutex); int unwrapped_timecode = timecode; - if (dropped_frames > 60 * 2) { + 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(48000.0, 48000.0, 2)); + card->resampling_queue.reset(new ResamplingQueue(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2)); } 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((48000 / 60) * 2); + silence.resize((OUTPUT_FREQUENCY / FPS) * 2); for (int i = 0; i < dropped_frames; ++i) { - card->resampler->add_input_samples((unwrapped_timecode - dropped_frames + i) / 60.0, silence.data(), (48000 / 60)); + card->resampling_queue->add_input_samples((unwrapped_timecode - dropped_frames + i) / double(FPS), silence.data(), (OUTPUT_FREQUENCY / FPS)); } } - card->resampler->add_input_samples(unwrapped_timecode / 60.0, audio.data(), num_samples); + card->resampling_queue->add_input_samples(unwrapped_timecode / double(FPS), audio.data(), num_samples); } // Done with the audio, so release it. @@ -336,7 +341,7 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode, void Mixer::thread_func() { eglBindAPI(EGL_OPENGL_API); - QOpenGLContext *context = create_context(); + QOpenGLContext *context = create_context(mixer_surface); if (!make_current(context, mixer_surface)) { printf("oops\n"); exit(1); @@ -364,7 +369,6 @@ 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_data_ready_fence = card->new_data_ready_fence; - card_copy[card_index].new_frame_audio = move(card->new_frame_audio); card_copy[card_index].dropped_frames = card->dropped_frames; card->new_data_ready = false; card->new_data_ready_changed.notify_all(); @@ -372,30 +376,17 @@ void Mixer::thread_func() } // Resample the audio as needed, including from previously dropped frames. - vector samples_out; - // TODO: Allow using audio from the other card(s) as well. for (unsigned frame_num = 0; frame_num < card_copy[0].dropped_frames + 1; ++frame_num) { - for (unsigned card_index = 0; card_index < num_cards; ++card_index) { - samples_out.resize((48000 / 60) * 2); - { - unique_lock lock(cards[card_index].audio_mutex); - if (!cards[card_index].resampler->get_output_samples(pts(), &samples_out[0], 48000 / 60)) { - printf("Card %d reported previous underrun.\n", card_index); - } - } - if (card_index == 0) { - vector left, right; - peak = std::max(peak, find_peak(samples_out)); - deinterleave_samples(samples_out, &left, &right); - float *ptrs[] = { left.data(), right.data() }; - r128.process(left.size(), ptrs); - h264_encoder->add_audio(pts_int, move(samples_out)); - } + { + // Signal to the audio thread to process this frame. + unique_lock lock(audio_mutex); + audio_pts_queue.push(pts_int); + audio_pts_queue_changed.notify_one(); } if (frame_num != card_copy[0].dropped_frames) { // For dropped frames, increase the pts. ++dropped_frames; - pts_int += TIMEBASE / 60; + pts_int += TIMEBASE / FPS; } } @@ -406,7 +397,8 @@ void Mixer::thread_func() double loudness_range_high = r128.range_max(); audio_level_callback(loudness_s, 20.0 * log10(peak), - loudness_i, loudness_range_low, loudness_range_high); + loudness_i, loudness_range_low, loudness_range_high, + last_gain_staging_db); } for (unsigned card_index = 1; card_index < num_cards; ++card_index) { @@ -423,7 +415,7 @@ void Mixer::thread_func() // 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 / 60; + pts_int += TIMEBASE / FPS; continue; } @@ -485,10 +477,10 @@ 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 / 60; + pts_int += TIMEBASE / FPS; // The live frame just shows the RGBA texture we just rendered. // It owns rgba_tex now. @@ -544,6 +536,126 @@ void Mixer::thread_func() resource_pool->clean_context(); } +void Mixer::audio_thread_func() +{ + while (!should_quit) { + int64_t frame_pts_int; + + { + unique_lock lock(audio_mutex); + audio_pts_queue_changed.wait(lock, [this]{ return !audio_pts_queue.empty(); }); + frame_pts_int = audio_pts_queue.front(); + audio_pts_queue.pop(); + } + + process_audio_one_frame(frame_pts_int); + } +} + +void Mixer::process_audio_one_frame(int64_t frame_pts_int) +{ + vector samples_card; + vector samples_out; + for (unsigned card_index = 0; card_index < num_cards; ++card_index) { + samples_card.resize((OUTPUT_FREQUENCY / FPS) * 2); + { + unique_lock lock(cards[card_index].audio_mutex); + if (!cards[card_index].resampling_queue->get_output_samples(double(frame_pts_int) / TIMEBASE, &samples_card[0], OUTPUT_FREQUENCY / FPS)) { + printf("Card %d reported previous underrun.\n", card_index); + } + } + // TODO: Allow using audio from the other card(s) as well. + if (card_index == 0) { + samples_out = move(samples_card); + } + } + + // 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 + // (or more precisely, near it, since we don't use infinite ratio), + // 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); + } + +#if 0 + printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n", + level_compressor.get_level(), 20.0 * log10(level_compressor.get_level()), + level_compressor.get_attenuation(), 20.0 * log10(level_compressor.get_attenuation()), + 20.0 * log10(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain)); +#endif + +// 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); + + // 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) { GLuint vao; @@ -611,12 +723,14 @@ 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) @@ -629,6 +743,14 @@ 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) {