fix: preserve audio sample continuity across timestamp rounding

Checks/audio-continuity.cpp

@@ -0,0 +1,96 @@
+#include "../Sources/Core/player.h"
+#include <algorithm>
+#include <array>
+#include <chrono>
+#include <cmath>
+#include <cstdio>
+#include <cstdlib>
+#include <cstring>
+#include <fstream>
+#include <memory>
+#include <string>
+#include <thread>
+#include <vector>
+
+namespace {
+std::array<float, 48000 * 8 * 2> capture{};
+size_t capturedFrames = 0;
+void require(bool condition, const char *message) {
+    if (!condition) { std::fprintf(stderr, "FAIL audio: %s\n", message); std::exit(1); }
+}
+void play(const std::string &path) {
+    capturedFrames = 0;
+    std::unique_ptr<Player, decltype(&player_destroy)> player(player_create(), player_destroy);
+    require(player && player_open(player.get(), path.c_str()), "open fixture");
+    bool started = false, ended = false;
+    auto deadline = std::chrono::steady_clock::now() + std::chrono::seconds(12);
+    while (std::chrono::steady_clock::now() < deadline) {
+        auto state = player_get_state(player.get());
+        if (state.status == PLAYER_ERROR) {
+            std::fprintf(stderr, "%s\n", state.error);
+            require(false, "playback error");
+        }
+        if (state.status == PLAYER_PAUSED && !started) {
+            require(player_play(player.get()), "play fixture");
+            started = true;
+        }
+        if (state.status == PLAYER_ENDED) { ended = true; break; }
+        std::this_thread::sleep_for(std::chrono::milliseconds(2));
+    }
+    require(ended, "playback completes");
+    player.reset(); // Stop/join before inspecting callback-owned capture storage.
+}
+size_t firstActive(const float *samples, size_t frames) {
+    size_t first = 0;
+    while (first < frames && std::abs(samples[first * 2]) < .002f) ++first;
+    require(first < frames, "tone is present");
+    return first;
+}
+}
+
+// Linked only into the check build. No allocation, locks, or file I/O in callback.
+void player_check_audio(const float *samples, size_t frames) {
+    size_t count = std::min(frames, capture.size() / 2 - capturedFrames);
+    std::memcpy(capture.data() + capturedFrames * 2, samples, count * 2 * sizeof(float));
+    capturedFrames += count;
+}
+
+int main(int argc, char **argv) {
+    require(argc == 2, "fixture directory required");
+    std::string directory = argv[1];
+    for (const char *name : {"tone441.wav", "tone480.wav", "tone441.mka", "tone-aac.m4a"}) {
+        std::string path = directory + "/" + name;
+        std::ifstream input(path + ".f32", std::ios::binary | std::ios::ate);
+        require(bool(input), "PCM reference exists");
+        auto bytes = input.tellg();
+        require(bytes > 0 && bytes % (2 * sizeof(float)) == 0, "stereo reference format");
+        std::vector<float> reference(static_cast<size_t>(bytes) / sizeof(float));
+        input.seekg(0);
+        require(bool(input.read(reinterpret_cast<char *>(reference.data()), bytes)), "read PCM reference");
+        play(path);
+        size_t frames = reference.size() / 2;
+        size_t start = firstActive(capture.data(), capturedFrames);
+        size_t referenceStart = firstActive(reference.data(), frames);
+        require(start >= referenceStart, "capture alignment");
+        size_t offset = start - referenceStart;
+        require(capturedFrames >= offset + frames, "all output samples reach callback");
+        float maximumError = 0;
+        for (size_t i = 0; i < reference.size(); ++i)
+            maximumError = std::max(maximumError, std::abs(capture[offset * 2 + i] - reference[i]));
+        std::printf("%s: max sample error %.8f\n", name, maximumError);
+        require(maximumError < .00003f, "continuous callback PCM matches reference (no inserted/dropped samples)");
+    }
+    // Continuity must not erase a real 250 ms PTS gap in the source.
+    play(directory + "/tone-gap.mka");
+    size_t offset = firstActive(capture.data(), capturedFrames) - 1;
+    require(capturedFrames > offset + 48000 * 2, "gapped media is presented");
+    for (size_t i = 51840; i < 57600; ++i)
+        require(std::abs(capture[(offset + i) * 2]) < 1e-6f, "real timestamp gap remains silent");
+    double energy = 0;
+    for (size_t i = 62000; i < 66800; ++i) {
+        double value = capture[(offset + i) * 2];
+        energy += value * value;
+    }
+    require(energy / 4800 > .001, "audible tone resumes after real gap");
+    std::puts("PASS callback PCM continuity at 44.1/48 kHz, coarse timestamps, and real timestamp gap");
+}

Makefile

@@ -9,7 +9,7 @@ FRAMEWORKS := -framework AppKit -framework AudioToolbox -framework CoreAudio -fr
 OBJECTS := .build/player.o .build/Output.o
 APP := .build/Player.app
 
-.PHONY: all run check
+.PHONY: all run check check-audio
 all: $(APP)/Contents/MacOS/Player $(APP)/Contents/Info.plist
 
 .build:
@@ -35,10 +35,29 @@ run: all
 .build/playback-check: Checks/playback.mm $(OBJECTS) Makefile
 	$(CXX) $(CXXFLAGS) -fobjc-arc $< $(OBJECTS) $(FF_LIBS) $(FRAMEWORKS) -o $@
 
-check: .build/playback-check
+check: .build/playback-check check-audio
 	mkdir -p .build/fixtures
 	ffmpeg -hide_banner -loglevel error -y -f lavfi -i testsrc2=size=320x180:rate=24 -f lavfi -i sine=frequency=440:sample_rate=44100 -t 4 -c:v mpeg4 -bf 2 -c:a aac .build/fixtures/av.mp4
 	ffmpeg -hide_banner -loglevel error -y -i .build/fixtures/av.mp4 -an -c:v copy .build/fixtures/video.mp4
 	ffmpeg -hide_banner -loglevel error -y -i .build/fixtures/av.mp4 -vn -c:a pcm_s16le .build/fixtures/audio.wav
 	ffmpeg -hide_banner -loglevel error -y -f lavfi -i testsrc2=size=160x90:rate=12 -t 1 -vf 'format=yuv420p10le,setparams=color_primaries=bt2020:color_trc=smpte2084:colorspace=bt2020nc' -c:v ffv1 .build/fixtures/hdr.mkv
 	.build/playback-check .build/fixtures
+
+.build/Output-check.o: Sources/macOS/Output.mm Sources/Core/output.hpp Makefile | .build
+	$(CXX) $(CXXFLAGS) -fobjc-arc -DPLAYER_AUDIO_CHECK -c $< -o $@
+
+.build/audio-continuity-check: Checks/audio-continuity.cpp .build/player.o .build/Output-check.o Makefile
+	$(CXX) $(CXXFLAGS) $< .build/player.o .build/Output-check.o $(FF_LIBS) $(FRAMEWORKS) -o $@
+
+check-audio: .build/audio-continuity-check
+	mkdir -p .build/fixtures
+	ffmpeg -hide_banner -loglevel error -y -f lavfi -i sine=frequency=440:sample_rate=44100 -t 2 -c:a pcm_s16le .build/fixtures/tone441.wav
+	ffmpeg -hide_banner -loglevel error -y -f lavfi -i sine=frequency=440:sample_rate=48000 -t 2 -c:a pcm_s16le .build/fixtures/tone480.wav
+	ffmpeg -hide_banner -loglevel error -y -i .build/fixtures/tone441.wav -c:a copy .build/fixtures/tone441.mka
+	ffmpeg -hide_banner -loglevel error -y -i .build/fixtures/tone441.wav -c:a aac .build/fixtures/tone-aac.m4a
+	ffmpeg -hide_banner -loglevel error -y -i .build/fixtures/tone441.wav -af "asetpts='PTS+gte(T,1)*0.25/TB'" -c:a pcm_s16le .build/fixtures/tone-gap.mka
+	ffmpeg -hide_banner -loglevel error -y -i .build/fixtures/tone441.wav -ar 48000 -ac 2 -f f32le .build/fixtures/tone441.wav.f32
+	ffmpeg -hide_banner -loglevel error -y -i .build/fixtures/tone480.wav -ar 48000 -ac 2 -f f32le .build/fixtures/tone480.wav.f32
+	ffmpeg -hide_banner -loglevel error -y -i .build/fixtures/tone441.mka -ar 48000 -ac 2 -f f32le .build/fixtures/tone441.mka.f32
+	ffmpeg -hide_banner -loglevel error -y -i .build/fixtures/tone-aac.m4a -ar 48000 -ac 2 -f f32le .build/fixtures/tone-aac.m4a.f32
+	.build/audio-continuity-check .build/fixtures

README.md

@@ -32,6 +32,10 @@ make check
 
 Generates short fixtures and exercises the real playback interface and native outputs: audio/video, video-only, audio-only, pause, paused/playing seek, completion, replay, close, HDR rejection, failed-open recovery, and destruction while playing. It initializes the actual Metal shaders and uses an offscreen native window. It requires a working audio device and graphical macOS session, and plays short sine-wave audio. It is a runtime smoke check, not a mocked clock/device test.
 
+`make check-audio` runs just the audio-quality check. A check-only hook captures actual Core Audio callback PCM into preallocated storage and compares every sample against FFmpeg's reference output for PCM/AAC, 44.1/48 kHz, and coarse container timestamps. A separate gapped fixture verifies that real timestamp gaps remain silent. Release builds contain no capture hook.
+
+Continuous audio uses resampled sample counts for its output timeline; packet timestamp quantization and rounded resampler delay must not insert silence or drop samples between blocks.
+
 Visual correctness, perceived A/V synchronization, and control usability still need a check in the visible app.
 
 ## Current limits

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;
+        const double tick = av_q2d(format->streams[track.index]->time_base);
-        pcmPts = timestamp - static_cast<double>(swr_get_delay(resampler, rate)) / rate;
+        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);