Add a paranoia assert.

[nageru] / mixer.cpp

diff --git a/mixer.cpp b/mixer.cpp
index 1d1a60cdea8f96576d202730bfdf763f7deb0712..1603b8dc47f8ea894ef48c51232c4b6f7c6edcb3 100644 (file)
--- a/mixer.cpp
+++ b/mixer.cpp
@@ -245,7 +245,6 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode,
        if (card->last_timecode != -1) {
                dropped_frames = unwrap_timecode(timecode, card->last_timecode) - card->last_timecode - 1;
        }
-       card->last_timecode = timecode;
 
        // Convert the audio to stereo fp32 and add it.
        vector<float> audio;
@@ -256,29 +255,38 @@ 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), resetting resampler\n",
-                               card_index);
+                       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;
        }
 
+       card->last_timecode = timecode;
+
        // Done with the audio, so release it.
        if (audio_frame.owner) {
                audio_frame.owner->release_frame(audio_frame);
@@ -390,6 +398,7 @@ void Mixer::thread_func()
                                int num_samples_times_timebase = OUTPUT_FREQUENCY * card->new_frame_length + card->fractional_samples;
                                num_samples[card_index] = num_samples_times_timebase / TIMEBASE;
                                card->fractional_samples = num_samples_times_timebase % TIMEBASE;
+                               assert(num_samples[card_index] >= 0);
                        }
                }