Rename Resampler to ResamplingQueue, to avoid conflicts with zita-resampler.

[nageru] / mixer.cpp

diff --git a/mixer.cpp b/mixer.cpp
index 08c562f7ab448250ba36f48e48b4805f01edc707..6bf81192fdfd99421398dedc189e23be75344a5d 100644 (file)
--- a/mixer.cpp
+++ b/mixer.cpp
@@ -124,7 +124,7 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
                        [this]{
                                resource_pool->clean_context();
                        });
-               card->resampler.reset(new Resampler(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2));
+               card->resampling_queue.reset(new ResamplingQueue(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2));
                card->usb->configure_card();
        }
 
@@ -249,7 +249,7 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode,
                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(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 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",
@@ -257,10 +257,10 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode,
                        vector<float> silence;
                        silence.resize((OUTPUT_FREQUENCY / FPS) * 2);
                        for (int i = 0; i < dropped_frames; ++i) {
-                               card->resampler->add_input_samples((unwrapped_timecode - dropped_frames + i) / double(FPS), silence.data(), (OUTPUT_FREQUENCY / FPS));
+                               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 / double(FPS), 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.
@@ -466,7 +466,7 @@ 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 / FPS;
@@ -533,7 +533,7 @@ 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].resampler->get_output_samples(pts(), &samples_card[0], OUTPUT_FREQUENCY / FPS)) {
+                       if (!cards[card_index].resampling_queue->get_output_samples(pts(), &samples_card[0], OUTPUT_FREQUENCY / FPS)) {
                                printf("Card %d reported previous underrun.\n", card_index);
                        }
                }