8 #include <movit/effect_chain.h>
9 #include <movit/effect_util.h>
10 #include <movit/flat_input.h>
11 #include <movit/image_format.h>
12 #include <movit/resource_pool.h>
13 #include <movit/util.h>
21 #include <condition_variable>
30 #include "bmusb/bmusb.h"
33 #include "h264encode.h"
34 #include "pbo_frame_allocator.h"
35 #include "ref_counted_gl_sync.h"
40 using namespace movit;
42 using namespace std::placeholders;
44 Mixer *global_mixer = nullptr;
48 void convert_fixed24_to_fp32(float *dst, size_t out_channels, const uint8_t *src, size_t in_channels, size_t num_samples)
50 for (size_t i = 0; i < num_samples; ++i) {
51 for (size_t j = 0; j < out_channels; ++j) {
55 uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
56 dst[i * out_channels + j] = int(s) * (1.0f / 4294967296.0f);
58 src += 3 * (in_channels - out_channels);
62 void insert_new_frame(RefCountedFrame frame, unsigned field_num, bool interlaced, unsigned card_index, InputState *input_state)
65 for (unsigned frame_num = FRAME_HISTORY_LENGTH; frame_num --> 1; ) { // :-)
66 input_state->buffered_frames[card_index][frame_num] =
67 input_state->buffered_frames[card_index][frame_num - 1];
69 input_state->buffered_frames[card_index][0] = { frame, field_num };
71 for (unsigned frame_num = 0; frame_num < FRAME_HISTORY_LENGTH; ++frame_num) {
72 input_state->buffered_frames[card_index][frame_num] = { frame, field_num };
77 string generate_local_dump_filename(int frame)
79 time_t now = time(NULL);
81 localtime_r(&now, &now_tm);
84 strftime(timestamp, sizeof(timestamp), "%F-%T%z", &now_tm);
86 // Use the frame number to disambiguate between two cuts starting
87 // on the same second.
89 snprintf(filename, sizeof(filename), "%s%s-f%02d%s",
90 LOCAL_DUMP_PREFIX, timestamp, frame % 100, LOCAL_DUMP_SUFFIX);
96 Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
97 : httpd(WIDTH, HEIGHT),
99 mixer_surface(create_surface(format)),
100 h264_encoder_surface(create_surface(format)),
101 correlation(OUTPUT_FREQUENCY),
102 level_compressor(OUTPUT_FREQUENCY),
103 limiter(OUTPUT_FREQUENCY),
104 compressor(OUTPUT_FREQUENCY)
106 httpd.open_output_file(generate_local_dump_filename(/*frame=*/0).c_str());
109 CHECK(init_movit(MOVIT_SHADER_DIR, MOVIT_DEBUG_OFF));
112 // Since we allow non-bouncing 4:2:2 YCbCrInputs, effective subpixel precision
113 // will be halved when sampling them, and we need to compensate here.
114 movit_texel_subpixel_precision /= 2.0;
116 resource_pool.reset(new ResourcePool);
117 theme.reset(new Theme("theme.lua", resource_pool.get(), num_cards));
118 for (unsigned i = 0; i < NUM_OUTPUTS; ++i) {
119 output_channel[i].parent = this;
122 ImageFormat inout_format;
123 inout_format.color_space = COLORSPACE_sRGB;
124 inout_format.gamma_curve = GAMMA_sRGB;
126 // Display chain; shows the live output produced by the main chain (its RGBA version).
127 display_chain.reset(new EffectChain(WIDTH, HEIGHT, resource_pool.get()));
129 display_input = new FlatInput(inout_format, FORMAT_RGB, GL_UNSIGNED_BYTE, WIDTH, HEIGHT); // FIXME: GL_UNSIGNED_BYTE is really wrong.
130 display_chain->add_input(display_input);
131 display_chain->add_output(inout_format, OUTPUT_ALPHA_FORMAT_POSTMULTIPLIED);
132 display_chain->set_dither_bits(0); // Don't bother.
133 display_chain->finalize();
135 h264_encoder.reset(new H264Encoder(h264_encoder_surface, WIDTH, HEIGHT, &httpd));
137 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
138 printf("Configuring card %d...\n", card_index);
139 CaptureCard *card = &cards[card_index];
140 card->usb = new BMUSBCapture(card_index);
141 card->usb->set_frame_callback(bind(&Mixer::bm_frame, this, card_index, _1, _2, _3, _4, _5, _6, _7));
142 card->frame_allocator.reset(new PBOFrameAllocator(8 << 20, WIDTH, HEIGHT)); // 8 MB.
143 card->usb->set_video_frame_allocator(card->frame_allocator.get());
144 card->surface = create_surface(format);
145 card->usb->set_dequeue_thread_callbacks(
147 eglBindAPI(EGL_OPENGL_API);
148 card->context = create_context(card->surface);
149 if (!make_current(card->context, card->surface)) {
150 printf("failed to create bmusb context\n");
155 resource_pool->clean_context();
157 card->resampling_queue.reset(new ResamplingQueue(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2));
158 card->usb->configure_card();
161 BMUSBCapture::start_bm_thread();
163 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
164 cards[card_index].usb->start_bm_capture();
167 // Set up stuff for NV12 conversion.
170 string cbcr_vert_shader = read_file("vs-cbcr.130.vert");
171 string cbcr_frag_shader =
174 "uniform sampler2D cbcr_tex; \n"
176 " gl_FragColor = texture2D(cbcr_tex, tc0); \n"
178 vector<string> frag_shader_outputs;
179 cbcr_program_num = resource_pool->compile_glsl_program(cbcr_vert_shader, cbcr_frag_shader, frag_shader_outputs);
181 r128.init(2, OUTPUT_FREQUENCY);
184 locut.init(FILTER_HPF, 2);
186 // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
187 // and there's a limit to how important the peak meter is.
188 peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
190 alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2));
195 resource_pool->release_glsl_program(cbcr_program_num);
196 BMUSBCapture::stop_bm_thread();
198 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
200 unique_lock<mutex> lock(bmusb_mutex);
201 cards[card_index].should_quit = true; // Unblock thread.
202 cards[card_index].new_data_ready_changed.notify_all();
204 cards[card_index].usb->stop_dequeue_thread();
207 h264_encoder.reset(nullptr);
212 int unwrap_timecode(uint16_t current_wrapped, int last)
214 uint16_t last_wrapped = last & 0xffff;
215 if (current_wrapped > last_wrapped) {
216 return (last & ~0xffff) | current_wrapped;
218 return 0x10000 + ((last & ~0xffff) | current_wrapped);
222 float find_peak(const float *samples, size_t num_samples)
224 float m = fabs(samples[0]);
225 for (size_t i = 1; i < num_samples; ++i) {
226 m = max(m, fabs(samples[i]));
231 void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
233 size_t num_samples = in.size() / 2;
234 out_l->resize(num_samples);
235 out_r->resize(num_samples);
237 const float *inptr = in.data();
238 float *lptr = &(*out_l)[0];
239 float *rptr = &(*out_r)[0];
240 for (size_t i = 0; i < num_samples; ++i) {
248 void Mixer::bm_frame(unsigned card_index, uint16_t timecode,
249 FrameAllocator::Frame video_frame, size_t video_offset, uint16_t video_format,
250 FrameAllocator::Frame audio_frame, size_t audio_offset, uint16_t audio_format)
252 CaptureCard *card = &cards[card_index];
254 unsigned width, height, second_field_start, frame_rate_nom, frame_rate_den, extra_lines_top, extra_lines_bottom;
257 decode_video_format(video_format, &width, &height, &second_field_start, &extra_lines_top, &extra_lines_bottom,
258 &frame_rate_nom, &frame_rate_den, &interlaced); // Ignore return value for now.
259 int64_t frame_length = TIMEBASE * frame_rate_den / frame_rate_nom;
261 size_t num_samples = (audio_frame.len >= audio_offset) ? (audio_frame.len - audio_offset) / 8 / 3 : 0;
262 if (num_samples > OUTPUT_FREQUENCY / 10) {
263 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",
264 card_index, int(audio_frame.len), int(audio_offset),
265 timecode, int(video_frame.len), int(video_offset), video_format);
266 if (video_frame.owner) {
267 video_frame.owner->release_frame(video_frame);
269 if (audio_frame.owner) {
270 audio_frame.owner->release_frame(audio_frame);
275 int64_t local_pts = card->next_local_pts;
276 int dropped_frames = 0;
277 if (card->last_timecode != -1) {
278 dropped_frames = unwrap_timecode(timecode, card->last_timecode) - card->last_timecode - 1;
281 // Convert the audio to stereo fp32 and add it.
283 audio.resize(num_samples * 2);
284 convert_fixed24_to_fp32(&audio[0], 2, audio_frame.data + audio_offset, 8, num_samples);
288 unique_lock<mutex> lock(card->audio_mutex);
290 // Number of samples per frame if we need to insert silence.
291 // (Could be nonintegral, but resampling will save us then.)
292 int silence_samples = OUTPUT_FREQUENCY * frame_rate_den / frame_rate_nom;
294 if (dropped_frames > MAX_FPS * 2) {
295 fprintf(stderr, "Card %d lost more than two seconds (or time code jumping around; from 0x%04x to 0x%04x), resetting resampler\n",
296 card_index, card->last_timecode, timecode);
297 card->resampling_queue.reset(new ResamplingQueue(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2));
299 } else if (dropped_frames > 0) {
300 // Insert silence as needed.
301 fprintf(stderr, "Card %d dropped %d frame(s) (before timecode 0x%04x), inserting silence.\n",
302 card_index, dropped_frames, timecode);
303 vector<float> silence(silence_samples * 2, 0.0f);
304 for (int i = 0; i < dropped_frames; ++i) {
305 card->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), silence.data(), silence_samples);
306 // Note that if the format changed in the meantime, we have
307 // no way of detecting that; we just have to assume the frame length
308 // is always the same.
309 local_pts += frame_length;
312 if (num_samples == 0) {
313 audio.resize(silence_samples * 2);
314 num_samples = silence_samples;
316 card->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
317 card->next_local_pts = local_pts + frame_length;
320 card->last_timecode = timecode;
322 // Done with the audio, so release it.
323 if (audio_frame.owner) {
324 audio_frame.owner->release_frame(audio_frame);
328 // Wait until the previous frame was consumed.
329 unique_lock<mutex> lock(bmusb_mutex);
330 card->new_data_ready_changed.wait(lock, [card]{ return !card->new_data_ready || card->should_quit; });
331 if (card->should_quit) return;
334 size_t expected_length = width * (height + extra_lines_top + extra_lines_bottom) * 2;
335 if (video_frame.len - video_offset == 0 ||
336 video_frame.len - video_offset != expected_length) {
337 if (video_frame.len != 0) {
338 printf("Card %d: Dropping video frame with wrong length (%ld; expected %ld)\n",
339 card_index, video_frame.len - video_offset, expected_length);
341 if (video_frame.owner) {
342 video_frame.owner->release_frame(video_frame);
345 // Still send on the information that we _had_ a frame, even though it's corrupted,
346 // so that pts can go up accordingly.
348 unique_lock<mutex> lock(bmusb_mutex);
349 card->new_data_ready = true;
350 card->new_frame = RefCountedFrame(FrameAllocator::Frame());
351 card->new_frame_length = frame_length;
352 card->new_frame_interlaced = false;
353 card->new_data_ready_fence = nullptr;
354 card->dropped_frames = dropped_frames;
355 card->new_data_ready_changed.notify_all();
360 PBOFrameAllocator::Userdata *userdata = (PBOFrameAllocator::Userdata *)video_frame.userdata;
362 unsigned num_fields = interlaced ? 2 : 1;
363 timespec frame_upload_start;
365 // Send the two fields along as separate frames; the other side will need to add
366 // a deinterlacer to actually get this right.
367 assert(height % 2 == 0);
369 assert(frame_length % 2 == 0);
372 clock_gettime(CLOCK_MONOTONIC, &frame_upload_start);
374 userdata->last_interlaced = interlaced;
375 userdata->last_frame_rate_nom = frame_rate_nom;
376 userdata->last_frame_rate_den = frame_rate_den;
377 RefCountedFrame new_frame(video_frame);
379 // Upload the textures.
380 size_t cbcr_width = width / 2;
381 size_t cbcr_offset = video_offset / 2;
382 size_t y_offset = video_frame.size / 2 + video_offset / 2;
384 for (unsigned field = 0; field < num_fields; ++field) {
385 unsigned field_start_line = (field == 1) ? second_field_start : extra_lines_top + field * (height + 22);
387 if (userdata->tex_y[field] == 0 ||
388 userdata->tex_cbcr[field] == 0 ||
389 width != userdata->last_width[field] ||
390 height != userdata->last_height[field]) {
391 // We changed resolution since last use of this texture, so we need to create
392 // a new object. Note that this each card has its own PBOFrameAllocator,
393 // we don't need to worry about these flip-flopping between resolutions.
394 glBindTexture(GL_TEXTURE_2D, userdata->tex_cbcr[field]);
396 glTexImage2D(GL_TEXTURE_2D, 0, GL_RG8, cbcr_width, height, 0, GL_RG, GL_UNSIGNED_BYTE, nullptr);
398 glBindTexture(GL_TEXTURE_2D, userdata->tex_y[field]);
400 glTexImage2D(GL_TEXTURE_2D, 0, GL_R8, width, height, 0, GL_RED, GL_UNSIGNED_BYTE, nullptr);
402 userdata->last_width[field] = width;
403 userdata->last_height[field] = height;
406 GLuint pbo = userdata->pbo;
408 glBindBuffer(GL_PIXEL_UNPACK_BUFFER_ARB, pbo);
410 glMemoryBarrier(GL_CLIENT_MAPPED_BUFFER_BARRIER_BIT);
413 glBindTexture(GL_TEXTURE_2D, userdata->tex_cbcr[field]);
415 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)));
417 glBindTexture(GL_TEXTURE_2D, userdata->tex_y[field]);
419 glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, width, height, GL_RED, GL_UNSIGNED_BYTE, BUFFER_OFFSET(y_offset + width * field_start_line));
421 glBindTexture(GL_TEXTURE_2D, 0);
423 GLsync fence = glFenceSync(GL_SYNC_GPU_COMMANDS_COMPLETE, /*flags=*/0);
425 assert(fence != nullptr);
428 // Don't upload the second field as fast as we can; wait until
429 // the field time has approximately passed. (Otherwise, we could
430 // get timing jitter against the other sources, and possibly also
431 // against the video display, although the latter is not as critical.)
432 // This requires our system clock to be reasonably close to the
433 // video clock, but that's not an unreasonable assumption.
434 timespec second_field_start;
435 second_field_start.tv_nsec = frame_upload_start.tv_nsec +
436 frame_length * 1000000000 / TIMEBASE;
437 second_field_start.tv_sec = frame_upload_start.tv_sec +
438 second_field_start.tv_nsec / 1000000000;
439 second_field_start.tv_nsec %= 1000000000;
441 while (clock_nanosleep(CLOCK_MONOTONIC, TIMER_ABSTIME,
442 &second_field_start, nullptr) == -1 &&
447 unique_lock<mutex> lock(bmusb_mutex);
448 card->new_data_ready = true;
449 card->new_frame = new_frame;
450 card->new_frame_length = frame_length;
451 card->new_frame_field = field;
452 card->new_frame_interlaced = interlaced;
453 card->new_data_ready_fence = fence;
454 card->dropped_frames = dropped_frames;
455 card->new_data_ready_changed.notify_all();
457 if (field != num_fields - 1) {
458 // Wait until the previous frame was consumed.
459 card->new_data_ready_changed.wait(lock, [card]{ return !card->new_data_ready || card->should_quit; });
460 if (card->should_quit) return;
466 void Mixer::thread_func()
468 eglBindAPI(EGL_OPENGL_API);
469 QOpenGLContext *context = create_context(mixer_surface);
470 if (!make_current(context, mixer_surface)) {
475 struct timespec start, now;
476 clock_gettime(CLOCK_MONOTONIC, &start);
479 int stats_dropped_frames = 0;
481 while (!should_quit) {
482 CaptureCard card_copy[MAX_CARDS];
483 int num_samples[MAX_CARDS];
486 unique_lock<mutex> lock(bmusb_mutex);
488 // The first card is the master timer, so wait for it to have a new frame.
489 // TODO: Make configurable, and with a timeout.
490 cards[0].new_data_ready_changed.wait(lock, [this]{ return cards[0].new_data_ready; });
492 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
493 CaptureCard *card = &cards[card_index];
494 card_copy[card_index].usb = card->usb;
495 card_copy[card_index].new_data_ready = card->new_data_ready;
496 card_copy[card_index].new_frame = card->new_frame;
497 card_copy[card_index].new_frame_length = card->new_frame_length;
498 card_copy[card_index].new_frame_field = card->new_frame_field;
499 card_copy[card_index].new_frame_interlaced = card->new_frame_interlaced;
500 card_copy[card_index].new_data_ready_fence = card->new_data_ready_fence;
501 card_copy[card_index].dropped_frames = card->dropped_frames;
502 card->new_data_ready = false;
503 card->new_data_ready_changed.notify_all();
505 int num_samples_times_timebase = OUTPUT_FREQUENCY * card->new_frame_length + card->fractional_samples;
506 num_samples[card_index] = num_samples_times_timebase / TIMEBASE;
507 card->fractional_samples = num_samples_times_timebase % TIMEBASE;
508 assert(num_samples[card_index] >= 0);
512 // Resample the audio as needed, including from previously dropped frames.
513 for (unsigned frame_num = 0; frame_num < card_copy[0].dropped_frames + 1; ++frame_num) {
515 // Signal to the audio thread to process this frame.
516 unique_lock<mutex> lock(audio_mutex);
517 audio_task_queue.push(AudioTask{pts_int, num_samples[0]});
518 audio_task_queue_changed.notify_one();
520 if (frame_num != card_copy[0].dropped_frames) {
521 // For dropped frames, increase the pts. Note that if the format changed
522 // in the meantime, we have no way of detecting that; we just have to
523 // assume the frame length is always the same.
524 ++stats_dropped_frames;
525 pts_int += card_copy[0].new_frame_length;
529 if (audio_level_callback != nullptr) {
530 unique_lock<mutex> lock(compressor_mutex);
531 double loudness_s = r128.loudness_S();
532 double loudness_i = r128.integrated();
533 double loudness_range_low = r128.range_min();
534 double loudness_range_high = r128.range_max();
536 audio_level_callback(loudness_s, 20.0 * log10(peak),
537 loudness_i, loudness_range_low, loudness_range_high,
538 gain_staging_db, 20.0 * log10(final_makeup_gain),
539 correlation.get_correlation());
542 for (unsigned card_index = 1; card_index < num_cards; ++card_index) {
543 if (card_copy[card_index].new_data_ready && card_copy[card_index].new_frame->len == 0) {
544 ++card_copy[card_index].dropped_frames;
546 if (card_copy[card_index].dropped_frames > 0) {
547 printf("Card %u dropped %d frames before this\n",
548 card_index, int(card_copy[card_index].dropped_frames));
552 // If the first card is reporting a corrupted or otherwise dropped frame,
553 // just increase the pts (skipping over this frame) and don't try to compute anything new.
554 if (card_copy[0].new_frame->len == 0) {
555 ++stats_dropped_frames;
556 pts_int += card_copy[0].new_frame_length;
560 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
561 CaptureCard *card = &card_copy[card_index];
562 if (!card->new_data_ready || card->new_frame->len == 0)
565 assert(card->new_frame != nullptr);
566 insert_new_frame(card->new_frame, card->new_frame_field, card->new_frame_interlaced, card_index, &input_state);
569 // The new texture might still be uploaded,
570 // tell the GPU to wait until it's there.
571 if (card->new_data_ready_fence) {
572 glWaitSync(card->new_data_ready_fence, /*flags=*/0, GL_TIMEOUT_IGNORED);
574 glDeleteSync(card->new_data_ready_fence);
579 // Get the main chain from the theme, and set its state immediately.
580 Theme::Chain theme_main_chain = theme->get_chain(0, pts(), WIDTH, HEIGHT, input_state);
581 EffectChain *chain = theme_main_chain.chain;
582 theme_main_chain.setup_chain();
583 //theme_main_chain.chain->enable_phase_timing(true);
585 GLuint y_tex, cbcr_tex;
586 bool got_frame = h264_encoder->begin_frame(&y_tex, &cbcr_tex);
589 // Render main chain.
590 GLuint cbcr_full_tex = resource_pool->create_2d_texture(GL_RG8, WIDTH, HEIGHT);
591 GLuint rgba_tex = resource_pool->create_2d_texture(GL_RGB565, WIDTH, HEIGHT); // Saves texture bandwidth, although dithering gets messed up.
592 GLuint fbo = resource_pool->create_fbo(y_tex, cbcr_full_tex, rgba_tex);
594 chain->render_to_fbo(fbo, WIDTH, HEIGHT);
595 resource_pool->release_fbo(fbo);
597 subsample_chroma(cbcr_full_tex, cbcr_tex);
598 resource_pool->release_2d_texture(cbcr_full_tex);
600 // Set the right state for rgba_tex.
601 glBindFramebuffer(GL_FRAMEBUFFER, 0);
602 glBindTexture(GL_TEXTURE_2D, rgba_tex);
603 glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
604 glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
605 glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
607 RefCountedGLsync fence(GL_SYNC_GPU_COMMANDS_COMPLETE, /*flags=*/0);
610 const int64_t av_delay = TIMEBASE / 10; // Corresponds to the fixed delay in resampling_queue.h. TODO: Make less hard-coded.
611 h264_encoder->end_frame(fence, pts_int + av_delay, theme_main_chain.input_frames);
613 pts_int += card_copy[0].new_frame_length;
615 // The live frame just shows the RGBA texture we just rendered.
616 // It owns rgba_tex now.
617 DisplayFrame live_frame;
618 live_frame.chain = display_chain.get();
619 live_frame.setup_chain = [this, rgba_tex]{
620 display_input->set_texture_num(rgba_tex);
622 live_frame.ready_fence = fence;
623 live_frame.input_frames = {};
624 live_frame.temp_textures = { rgba_tex };
625 output_channel[OUTPUT_LIVE].output_frame(live_frame);
627 // Set up preview and any additional channels.
628 for (int i = 1; i < theme->get_num_channels() + 2; ++i) {
629 DisplayFrame display_frame;
630 Theme::Chain chain = theme->get_chain(i, pts(), WIDTH, HEIGHT, input_state); // FIXME: dimensions
631 display_frame.chain = chain.chain;
632 display_frame.setup_chain = chain.setup_chain;
633 display_frame.ready_fence = fence;
634 display_frame.input_frames = chain.input_frames;
635 display_frame.temp_textures = {};
636 output_channel[i].output_frame(display_frame);
639 clock_gettime(CLOCK_MONOTONIC, &now);
640 double elapsed = now.tv_sec - start.tv_sec +
641 1e-9 * (now.tv_nsec - start.tv_nsec);
642 if (frame % 100 == 0) {
643 printf("%d frames (%d dropped) in %.3f seconds = %.1f fps (%.1f ms/frame)\n",
644 frame, stats_dropped_frames, elapsed, frame / elapsed,
645 1e3 * elapsed / frame);
646 // chain->print_phase_timing();
649 if (should_cut.exchange(false)) { // Test and clear.
650 string filename = generate_local_dump_filename(frame);
651 printf("Starting new recording: %s\n", filename.c_str());
652 h264_encoder->shutdown();
653 httpd.close_output_file();
654 httpd.open_output_file(filename.c_str());
655 h264_encoder.reset(new H264Encoder(h264_encoder_surface, WIDTH, HEIGHT, &httpd));
659 // Reset every 100 frames, so that local variations in frame times
660 // (especially for the first few frames, when the shaders are
661 // compiled etc.) don't make it hard to measure for the entire
662 // remaining duration of the program.
663 if (frame == 10000) {
671 resource_pool->clean_context();
674 void Mixer::audio_thread_func()
676 while (!should_quit) {
680 unique_lock<mutex> lock(audio_mutex);
681 audio_task_queue_changed.wait(lock, [this]{ return !audio_task_queue.empty(); });
682 task = audio_task_queue.front();
683 audio_task_queue.pop();
686 process_audio_one_frame(task.pts_int, task.num_samples);
690 void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
692 vector<float> samples_card;
693 vector<float> samples_out;
694 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
695 samples_card.resize(num_samples * 2);
697 unique_lock<mutex> lock(cards[card_index].audio_mutex);
698 if (!cards[card_index].resampling_queue->get_output_samples(double(frame_pts_int) / TIMEBASE, &samples_card[0], num_samples)) {
699 printf("Card %d reported previous underrun.\n", card_index);
702 // TODO: Allow using audio from the other card(s) as well.
703 if (card_index == 0) {
704 samples_out = move(samples_card);
708 // Cut away everything under 120 Hz (or whatever the cutoff is);
709 // we don't need it for voice, and it will reduce headroom
710 // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
711 // should be dampened.)
712 locut.render(samples_out.data(), samples_out.size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
714 // Apply a level compressor to get the general level right.
715 // Basically, if it's over about -40 dBFS, we squeeze it down to that level
716 // (or more precisely, near it, since we don't use infinite ratio),
717 // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
718 // entirely arbitrary, but from practical tests with speech, it seems to
719 // put ut around -23 LUFS, so it's a reasonable starting point for later use.
721 unique_lock<mutex> lock(compressor_mutex);
722 if (level_compressor_enabled) {
723 float threshold = 0.01f; // -40 dBFS.
725 float attack_time = 0.5f;
726 float release_time = 20.0f;
727 float makeup_gain = pow(10.0f, (ref_level_dbfs - (-40.0f)) / 20.0f); // +26 dB.
728 level_compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
729 gain_staging_db = 20.0 * log10(level_compressor.get_attenuation() * makeup_gain);
731 // Just apply the gain we already had.
732 float g = pow(10.0f, gain_staging_db / 20.0f);
733 for (size_t i = 0; i < samples_out.size(); ++i) {
740 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
741 level_compressor.get_level(), 20.0 * log10(level_compressor.get_level()),
742 level_compressor.get_attenuation(), 20.0 * log10(level_compressor.get_attenuation()),
743 20.0 * log10(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
746 // float limiter_att, compressor_att;
748 // The real compressor.
749 if (compressor_enabled) {
750 float threshold = pow(10.0f, compressor_threshold_dbfs / 20.0f);
752 float attack_time = 0.005f;
753 float release_time = 0.040f;
754 float makeup_gain = 2.0f; // +6 dB.
755 compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
756 // compressor_att = compressor.get_attenuation();
759 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
760 // Note that since ratio is not infinite, we could go slightly higher than this.
761 if (limiter_enabled) {
762 float threshold = pow(10.0f, limiter_threshold_dbfs / 20.0f);
764 float attack_time = 0.0f; // Instant.
765 float release_time = 0.020f;
766 float makeup_gain = 1.0f; // 0 dB.
767 limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
768 // limiter_att = limiter.get_attenuation();
771 // printf("limiter=%+5.1f compressor=%+5.1f\n", 20.0*log10(limiter_att), 20.0*log10(compressor_att));
773 // Upsample 4x to find interpolated peak.
774 peak_resampler.inp_data = samples_out.data();
775 peak_resampler.inp_count = samples_out.size() / 2;
777 vector<float> interpolated_samples_out;
778 interpolated_samples_out.resize(samples_out.size());
779 while (peak_resampler.inp_count > 0) { // About four iterations.
780 peak_resampler.out_data = &interpolated_samples_out[0];
781 peak_resampler.out_count = interpolated_samples_out.size() / 2;
782 peak_resampler.process();
783 size_t out_stereo_samples = interpolated_samples_out.size() / 2 - peak_resampler.out_count;
784 peak = max<float>(peak, find_peak(interpolated_samples_out.data(), out_stereo_samples * 2));
787 // At this point, we are most likely close to +0 LU, but all of our
788 // measurements have been on raw sample values, not R128 values.
789 // So we have a final makeup gain to get us to +0 LU; the gain
790 // adjustments required should be relatively small, and also, the
791 // offset shouldn't change much (only if the type of audio changes
792 // significantly). Thus, we shoot for updating this value basically
793 // “whenever we process buffers”, since the R128 calculation isn't exactly
794 // something we get out per-sample.
796 // Note that there's a feedback loop here, so we choose a very slow filter
797 // (half-time of 100 seconds).
798 double target_loudness_factor, alpha;
800 unique_lock<mutex> lock(compressor_mutex);
801 double loudness_lu = r128.loudness_M() - ref_level_lufs;
802 double current_makeup_lu = 20.0f * log10(final_makeup_gain);
803 target_loudness_factor = pow(10.0f, -loudness_lu / 20.0f);
805 // If we're outside +/- 5 LU uncorrected, we don't count it as
806 // a normal signal (probably silence) and don't change the
807 // correction factor; just apply what we already have.
808 if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) {
811 // Formula adapted from
812 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
813 const double half_time_s = 100.0;
814 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
815 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
818 double m = final_makeup_gain;
819 for (size_t i = 0; i < samples_out.size(); i += 2) {
820 samples_out[i + 0] *= m;
821 samples_out[i + 1] *= m;
822 m += (target_loudness_factor - m) * alpha;
824 final_makeup_gain = m;
827 // Find R128 levels and L/R correlation.
828 vector<float> left, right;
829 deinterleave_samples(samples_out, &left, &right);
830 float *ptrs[] = { left.data(), right.data() };
832 unique_lock<mutex> lock(compressor_mutex);
833 r128.process(left.size(), ptrs);
834 correlation.process_samples(samples_out);
837 // Send the samples to the sound card.
839 alsa->write(samples_out);
842 // And finally add them to the output.
843 h264_encoder->add_audio(frame_pts_int, move(samples_out));
846 void Mixer::subsample_chroma(GLuint src_tex, GLuint dst_tex)
849 glGenVertexArrays(1, &vao);
858 glBindVertexArray(vao);
862 GLuint fbo = resource_pool->create_fbo(dst_tex);
863 glBindFramebuffer(GL_FRAMEBUFFER, fbo);
864 glViewport(0, 0, WIDTH/2, HEIGHT/2);
867 glUseProgram(cbcr_program_num);
870 glActiveTexture(GL_TEXTURE0);
872 glBindTexture(GL_TEXTURE_2D, src_tex);
874 glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
876 glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
878 glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
881 float chroma_offset_0[] = { -0.5f / WIDTH, 0.0f };
882 set_uniform_vec2(cbcr_program_num, "foo", "chroma_offset_0", chroma_offset_0);
884 GLuint position_vbo = fill_vertex_attribute(cbcr_program_num, "position", 2, GL_FLOAT, sizeof(vertices), vertices);
885 GLuint texcoord_vbo = fill_vertex_attribute(cbcr_program_num, "texcoord", 2, GL_FLOAT, sizeof(vertices), vertices); // Same as vertices.
887 glDrawArrays(GL_TRIANGLES, 0, 3);
890 cleanup_vertex_attribute(cbcr_program_num, "position", position_vbo);
891 cleanup_vertex_attribute(cbcr_program_num, "texcoord", texcoord_vbo);
896 resource_pool->release_fbo(fbo);
897 glDeleteVertexArrays(1, &vao);
900 void Mixer::release_display_frame(DisplayFrame *frame)
902 for (GLuint texnum : frame->temp_textures) {
903 resource_pool->release_2d_texture(texnum);
905 frame->temp_textures.clear();
906 frame->ready_fence.reset();
907 frame->input_frames.clear();
912 mixer_thread = thread(&Mixer::thread_func, this);
913 audio_thread = thread(&Mixer::audio_thread_func, this);
923 void Mixer::transition_clicked(int transition_num)
925 theme->transition_clicked(transition_num, pts());
928 void Mixer::channel_clicked(int preview_num)
930 theme->channel_clicked(preview_num);
933 void Mixer::reset_meters()
935 peak_resampler.reset();
942 Mixer::OutputChannel::~OutputChannel()
944 if (has_current_frame) {
945 parent->release_display_frame(¤t_frame);
947 if (has_ready_frame) {
948 parent->release_display_frame(&ready_frame);
952 void Mixer::OutputChannel::output_frame(DisplayFrame frame)
954 // Store this frame for display. Remove the ready frame if any
955 // (it was seemingly never used).
957 unique_lock<mutex> lock(frame_mutex);
958 if (has_ready_frame) {
959 parent->release_display_frame(&ready_frame);
962 has_ready_frame = true;
965 if (has_new_frame_ready_callback) {
966 new_frame_ready_callback();
970 bool Mixer::OutputChannel::get_display_frame(DisplayFrame *frame)
972 unique_lock<mutex> lock(frame_mutex);
973 if (!has_current_frame && !has_ready_frame) {
977 if (has_current_frame && has_ready_frame) {
978 // We have a new ready frame. Toss the current one.
979 parent->release_display_frame(¤t_frame);
980 has_current_frame = false;
982 if (has_ready_frame) {
983 assert(!has_current_frame);
984 current_frame = ready_frame;
985 ready_frame.ready_fence.reset(); // Drop the refcount.
986 ready_frame.input_frames.clear(); // Drop the refcounts.
987 has_current_frame = true;
988 has_ready_frame = false;
991 *frame = current_frame;
995 void Mixer::OutputChannel::set_frame_ready_callback(Mixer::new_frame_ready_callback_t callback)
997 new_frame_ready_callback = callback;
998 has_new_frame_ready_callback = true;