fix: preserve audio sample continuity across timestamp rounding

Sources/Core/player.cpp

@@ -404,6 +404,7 @@ struct Player {
     }
 
     void convertAudio() {
+        const bool continuing = resampler != nullptr;
         auto &track = tracks[0];
         const int rate = decoded->sample_rate > 0 ? decoded->sample_rate : track.codec->sample_rate;
         const AVChannelLayout *layout = decoded->ch_layout.nb_channels > 0 ?
@@ -424,8 +425,13 @@ struct Player {
             throw std::runtime_error("Audio format changes within a track are not supported");
         }
         const double timestamp = framePts(track, decoded);
-        track.nextPts = timestamp + static_cast<double>(decoded->nb_samples) / rate;
-        pcmPts = timestamp - static_cast<double>(swr_get_delay(resampler, rate)) / rate;
+        const double tick = av_q2d(format->streams[track.index]->time_base);
+        const bool contiguous = continuing && std::abs(timestamp - track.nextPts) <= tick + 1.0 / rate;
+        // Quantized packet PTS and rounded resampler delay must not insert/drop PCM samples.
+        const int64_t delayBase = static_cast<int64_t>(rate) * sampleRate;
+        pcmPts = contiguous ? resampleNextPts :
+            timestamp - static_cast<double>(swr_get_delay(resampler, delayBase)) / delayBase;
+        track.nextPts = (contiguous ? track.nextPts : timestamp) + static_cast<double>(decoded->nb_samples) / rate;
         const int capacity = swr_get_out_samples(resampler, decoded->nb_samples);
         check(capacity, "Sizing resampled audio");
         if (capacity > audioFrameLimit)

Sources/macOS/Output.mm

@@ -17,6 +17,11 @@ extern "C" {
 #include <stdexcept>
 #include <vector>
 
+#ifdef PLAYER_AUDIO_CHECK
+// Check builds observe the actual callback samples; release builds have no hook.
+extern void player_check_audio(const float *samples, size_t frames);
+#endif
+
 namespace {
 constexpr size_t capacity = 48000;
 double host_seconds(uint64_t ticks) {
@@ -258,6 +263,9 @@ struct Output::Impl {
         size_t first = std::min(frames, capacity - size_t(r % capacity));
         std::memcpy(destination, self.samples.data() + (r % capacity) * 2, first * 2 * sizeof(float));
         std::memcpy(destination + first * 2, self.samples.data(), (frames - first) * 2 * sizeof(float));
+        #ifdef PLAYER_AUDIO_CHECK
+        player_check_audio(destination, count);
+        #endif
         if (frames) {
             self.clockVersion.fetch_add(1, std::memory_order_acq_rel);
             self.anchorPTS.store(self.origin + double(r) / 48000, std::memory_order_relaxed);