Reduce the default cutoff a bit; we have no instruments below, so better to give...

[nageru] / mixer.cpp

diff --git a/mixer.cpp b/mixer.cpp
index 0e362f069395874c837dac1bf732c9e1d32d25a0..72ae215f54cb993967de0e8a9ea8eb2bb3f0d3e7 100644 (file)
--- a/mixer.cpp
+++ b/mixer.cpp
@@ -66,7 +66,7 @@ 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)),
@@ -157,6 +157,8 @@ 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);
+
+       alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2));
 }
 
 Mixer::~Mixer()
@@ -172,6 +174,8 @@ Mixer::~Mixer()
                }
                cards[card_index].usb->stop_dequeue_thread();
        }
+
+       h264_encoder.reset(nullptr);
 }
 
 namespace {
@@ -375,7 +379,12 @@ void Mixer::thread_func()
 
                // Resample the audio as needed, including from previously dropped frames.
                for (unsigned frame_num = 0; frame_num < card_copy[0].dropped_frames + 1; ++frame_num) {
-                       process_audio_one_frame();
+                       {
+                               // Signal to the audio thread to process this frame.
+                               unique_lock<mutex> 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;
@@ -529,7 +538,23 @@ void Mixer::thread_func()
        resource_pool->clean_context();
 }
 
-void Mixer::process_audio_one_frame()
+void Mixer::audio_thread_func()
+{
+       while (!should_quit) {
+               int64_t frame_pts_int;
+
+               {
+                       unique_lock<mutex> 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<float> samples_card;
        vector<float> samples_out;
@@ -537,7 +562,7 @@ void Mixer::process_audio_one_frame()
                samples_card.resize((OUTPUT_FREQUENCY / FPS) * 2);
                {
                        unique_lock<mutex> lock(cards[card_index].audio_mutex);
-                       if (!cards[card_index].resampling_queue->get_output_samples(pts(), &samples_card[0], OUTPUT_FREQUENCY / FPS)) {
+                       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);
                        }
                }
@@ -547,9 +572,10 @@ void Mixer::process_audio_one_frame()
                }
        }
 
-       // Cut away everything under 150 Hz; 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.)
+       // 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.
@@ -623,8 +649,13 @@ void Mixer::process_audio_one_frame()
        float *ptrs[] = { left.data(), right.data() };
        r128.process(left.size(), ptrs);
 
-       // Actually add the samples to the output.
-       h264_encoder->add_audio(pts_int, move(samples_out));
+       // 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)
@@ -694,12 +725,14 @@ 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)