Rework silence handling a bit; in particular, if num_samples == 0, add silence.

diff --git a/mixer.cpp b/mixer.cpp
index 3fd264615dbc0aab32bbf8c82c4e970c2fb07751..e305376e9a86febcd9f044be95d9a159c80d7b42 100644 (file)
--- a/mixer.cpp
+++ b/mixer.cpp
@@ -255,25 +255,32 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode,
        {
                unique_lock<mutex> lock(card->audio_mutex);
 
+               // Number of samples per frame if we need to insert silence.
+               // (Could be nonintegral, but resampling will save us then.)
+               int silence_samples = OUTPUT_FREQUENCY * frame_rate_den / frame_rate_nom;
+
                if (dropped_frames > MAX_FPS * 2) {
                        fprintf(stderr, "Card %d lost more than two seconds (or time code jumping around; from 0x%04x to 0x%04x), resetting resampler\n",
                                card_index, card->last_timecode, timecode);
                        card->resampling_queue.reset(new ResamplingQueue(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY, 2));
                } else if (dropped_frames > 0) {
-                       // Insert silence as needed. (The number of samples could be nonintegral,
-                       // but resampling will save us then.)
+                       // 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<float> silence;
-                       silence.resize((OUTPUT_FREQUENCY * frame_length / TIMEBASE) * 2);
+                       silence.resize(silence_samples * 2);
                        for (int i = 0; i < dropped_frames; ++i) {
-                               card->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), silence.data(), silence.size() / 2);
+                               card->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), silence.data(), silence_samples);
                                // Note that if the format changed in the meantime, we have
                                // no way of detecting that; we just have to assume the frame length
                                // is always the same.
                                local_pts += frame_length;
                        }
                }
+               if (num_samples == 0) {
+                       audio.resize(silence_samples * 2);
+                       num_samples = silence_samples;
+               }
                card->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
                card->next_local_pts = local_pts + frame_length;
        }