]> git.sesse.net Git - nageru/blobdiff - mixer.cpp
Make it possible to override the level compressor with the gain staging knob.
[nageru] / mixer.cpp
index 384047a6b63b2df30f056b7daf0ec1d73dc98d37..72022026f771056e0211ccac874e69c8ee0ed22a 100644 (file)
--- a/mixer.cpp
+++ b/mixer.cpp
@@ -74,11 +74,27 @@ 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)),
@@ -86,6 +102,7 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
          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 +163,6 @@ 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.
@@ -207,7 +222,7 @@ 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;
 }
@@ -315,11 +330,12 @@ 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 +361,8 @@ 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 +370,9 @@ 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.
@@ -388,10 +406,8 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode,
                check_error();
                glBindBuffer(GL_PIXEL_UNPACK_BUFFER_ARB, pbo);
                check_error();
-               glFlushMappedBufferRange(GL_PIXEL_UNPACK_BUFFER, 0, video_frame.size);
+               glMemoryBarrier(GL_CLIENT_MAPPED_BUFFER_BARRIER_BIT);
                check_error();
-               //glMemoryBarrier(GL_CLIENT_MAPPED_BUFFER_BARRIER_BIT);
-               //check_error();
 
                glBindTexture(GL_TEXTURE_2D, userdata->tex_cbcr[field]);
                check_error();
@@ -510,7 +526,7 @@ void Mixer::thread_func()
                }
 
                if (audio_level_callback != nullptr) {
-                       unique_lock<mutex> lock(r128_mutex);
+                       unique_lock<mutex> lock(compressor_mutex);
                        double loudness_s = r128.loudness_S();
                        double loudness_i = r128.integrated();
                        double loudness_range_low = r128.range_min();
@@ -518,7 +534,7 @@ 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);
                }
 
                for (unsigned card_index = 1; card_index < num_cards; ++card_index) {
@@ -562,6 +578,7 @@ 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);
@@ -627,6 +644,15 @@ 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
@@ -689,15 +715,23 @@ 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<mutex> lock(level_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
@@ -753,7 +787,7 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
        deinterleave_samples(samples_out, &left, &right);
        float *ptrs[] = { left.data(), right.data() };
        {
-               unique_lock<mutex> lock(r128_mutex);
+               unique_lock<mutex> lock(compressor_mutex);
                r128.process(left.size(), ptrs);
        }