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Sépare les trac en amont de l'encodeur flac

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This commit is contained in:
Eric Coissac
2025-11-19 09:04:13 +01:00
parent 77e5eb047e
commit 1ef9a8a932
2 changed files with 188 additions and 81 deletions
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169
pmoaudio-ext/src/sinks/streaming_flac_sink.rs
View File

@@ -603,7 +603,7 @@ struct EncoderState {
struct StreamingFlacSinkLogic {
encoder_options: EncoderOptions,
bits_per_sample: u8,
pcm_tx: mpsc::Sender<PcmChunk>,
pcm_tx: Option<mpsc::Sender<PcmChunk>>,
pcm_rx: Option<mpsc::Receiver<PcmChunk>>,
metadata: Arc<RwLock<MetadataSnapshot>>,
flac_broadcast: timed_broadcast::Sender<Bytes>,
@@ -612,11 +612,15 @@ struct StreamingFlacSinkLogic {
sample_rate: Option<u32>,
broadcast_max_lead_time: f64,
first_chunk_timestamp_checked: bool,
/// Accumulated timestamp offset for maintaining continuity across encoder restarts
timestamp_offset_sec: f64,
/// Shared timestamp reference to read the last broadcast timestamp
current_timestamp: Arc<RwLock<f64>>,
}
impl StreamingFlacSinkLogic {
/// Initialize the FLAC encoder once we know the sample rate.
async fn initialize_encoder(&mut self, sample_rate: u32) -> Result<(), AudioError> {
async fn initialize_encoder(&mut self, sample_rate: u32, timestamp_offset_sec: f64) -> Result<(), AudioError> {
if self.encoder_state.is_some() {
return Ok(()); // Already initialized
}
@@ -633,7 +637,8 @@ impl StreamingFlacSinkLogic {
.ok_or_else(|| AudioError::ProcessingError("PCM receiver already consumed".into()))?;
// Create shared timestamp and duration for pacing
let current_timestamp = Arc::new(RwLock::new(0.0f64));
// Reuse existing current_timestamp Arc to maintain reference for reading later
let current_timestamp = self.current_timestamp.clone();
let current_duration = Arc::new(RwLock::new(0.0f64));
// Create ByteStreamReader for the encoder
@@ -669,6 +674,7 @@ impl StreamingFlacSinkLogic {
current_duration,
max_lead,
sample_rate,
timestamp_offset_sec,
)
.await
{
@@ -724,6 +730,54 @@ impl StreamingFlacSinkLogic {
Ok(())
}
/// Restart the FLAC encoder for a new track.
/// This causes a new "fLaC" header to be emitted and timestamps to reset to 0.
async fn restart_encoder_for_new_track(&mut self) -> Result<(), AudioError> {
let sample_rate = self
.sample_rate
.ok_or_else(|| AudioError::ProcessingError("Sample rate not initialized".into()))?;
debug!("Restarting FLAC encoder for new track");
// 1. Read current timestamp to maintain continuity
let last_timestamp = *self.current_timestamp.read().await;
debug!("Last timestamp before restart: {:.3}s", last_timestamp);
// 2. Close current PCM sender to signal encoder to finish
if let Some(tx) = self.pcm_tx.take() {
drop(tx);
trace!("Dropped PCM sender to signal encoder finish");
}
// 3. Wait for current broadcaster task to complete
if let Some(state) = self.encoder_state.take() {
trace!("Waiting for broadcaster task to finish...");
match state.broadcaster_task.await {
Ok(_) => {
trace!("Broadcaster task finished successfully");
}
Err(e) => {
warn!("Broadcaster task error during restart: {:?}", e);
}
}
}
// 4. Update timestamp offset to maintain continuity across encoder restart
self.timestamp_offset_sec += last_timestamp;
debug!("New timestamp offset: {:.3}s", self.timestamp_offset_sec);
// 5. Create new PCM channel
let (pcm_tx, pcm_rx) = mpsc::channel::<PcmChunk>(16);
self.pcm_tx = Some(pcm_tx);
self.pcm_rx = Some(pcm_rx);
// 6. Initialize new encoder with timestamp offset (will naturally emit new "fLaC" header)
self.initialize_encoder(sample_rate, self.timestamp_offset_sec).await?;
debug!("FLAC encoder restarted successfully for new track");
Ok(())
}
}
#[async_trait]
@@ -773,8 +827,8 @@ impl NodeLogic for StreamingFlacSinkLogic {
self.sample_rate = Some(sample_rate);
debug!("Detected sample rate: {} Hz", sample_rate);
// Initialize the FLAC encoder now
self.initialize_encoder(sample_rate).await?;
// Initialize the FLAC encoder now (first track starts at 0.0)
self.initialize_encoder(sample_rate, 0.0).await?;
}
// Convert chunk to PCM bytes
@@ -801,7 +855,17 @@ impl NodeLogic for StreamingFlacSinkLogic {
duration_sec,
};
let send_start = std::time::Instant::now();
if let Err(e) = self.pcm_tx.send(pcm_chunk).await {
// Get the sender (it should always be Some after initialization)
let pcm_tx = match &self.pcm_tx {
Some(tx) => tx,
None => {
error!("PCM sender not initialized");
break;
}
};
if let Err(e) = pcm_tx.send(pcm_chunk).await {
warn!("Failed to send PCM data to encoder: {}", e);
break;
}
@@ -818,6 +882,18 @@ impl NodeLogic for StreamingFlacSinkLogic {
_AudioSegment::Sync(marker) => {
match marker.as_ref() {
SyncMarker::TrackBoundary { metadata } => {
// Only restart encoder if it's already initialized (not the first track)
if self.sample_rate.is_some() && self.encoder_state.is_some() {
// Restart encoder to emit new header and reset timestamps
if let Err(e) = self.restart_encoder_for_new_track().await {
error!("Failed to restart encoder for new track: {}", e);
break;
}
} else {
trace!("Skipping encoder restart for first track (encoder not yet initialized)");
}
// Update metadata for the new track
if let Err(e) = self.update_metadata(metadata, seg.timestamp_sec).await {
error!("Failed to update metadata: {}", e);
}
@@ -866,6 +942,7 @@ async fn broadcast_flac_stream(
current_duration: Arc<RwLock<f64>>,
broadcast_max_lead_time: f64,
sample_rate: u32,
timestamp_offset_sec: f64,
) -> Result<(), AudioError> {
trace!(
"Broadcaster task started with FLAC frame boundary detection (max_lead={:.3}s)",
@@ -947,41 +1024,10 @@ async fn broadcast_flac_stream(
n
);
// Detect FLAC stream header "fLaC" in accumulator
// If found, we need to:
// 1. Process only frames BEFORE the header (end of previous track)
// 2. After broadcasting those frames, reset sample counter
// 3. Keep header + following data for next iteration
let header_pos = if accumulator.len() >= 4 {
accumulator.windows(4).position(|w| w == b"fLaC")
} else {
None
};
// Locate complete audio frames and total samples
// Note: find_complete_frames_with_samples() searches for AUDIO frames (0xFF 0xF8...),
// not the stream header "fLaC". Audio frames come AFTER the header.
let (boundary, total_samples) = if let Some(pos) = header_pos {
if pos > 0 {
// Header in the middle: process only frames BEFORE header (end of previous track)
trace!(
"New FLAC header detected at position {} in accumulator (size={}), searching frames before header",
pos,
accumulator.len()
);
flac_frame_utils::find_complete_frames_with_samples(&accumulator[..pos])
} else {
// Header at start: this is normal for new tracks, search all (audio frames come AFTER header)
trace!(
"FLAC header at start of accumulator (size={}), searching all frames",
accumulator.len()
);
flac_frame_utils::find_complete_frames_with_samples(&accumulator)
}
} else {
// No header: normal streaming, search all frames
flac_frame_utils::find_complete_frames_with_samples(&accumulator)
};
// Since encoder restarts on TrackBoundary, we only see one header per encoder instance
let (boundary, total_samples) =
flac_frame_utils::find_complete_frames_with_samples(&accumulator);
trace!(
"Buffer state: accumulator={} bytes, boundary={} bytes, total_samples={}, will_send={}",
@@ -1005,10 +1051,10 @@ async fn broadcast_flac_stream(
// ║ permettant de dropper les chunks vraiment périmés. ║
// ╚═══════════════════════════════════════════════════════════════╝
// Calculer le timestamp de cette FLAC frame
// Calculer le timestamp de cette FLAC frame (avec offset pour continuité entre tracks)
let frame_start_samples = encoded_samples;
encoded_samples = encoded_samples.saturating_add(total_samples);
let audio_timestamp = frame_start_samples as f64 / sample_rate_f64;
let audio_timestamp = timestamp_offset_sec + (frame_start_samples as f64 / sample_rate_f64);
let segment_duration = total_samples as f64 / sample_rate_f64;
if stats_last_log.elapsed() >= Duration::from_secs(1) {
@@ -1040,15 +1086,6 @@ async fn broadcast_flac_stream(
let to_send = std::mem::replace(&mut accumulator, remaining);
let bytes = Bytes::from(to_send);
// After splitting, check if accumulator now starts with FLAC header
// If so, we just finished the previous track and need to reset sample counter
if accumulator.len() >= 4 && &accumulator[0..4] == b"fLaC" {
trace!(
"FLAC header now at start of accumulator after split, resetting sample counter for new track"
);
encoded_samples = 0;
}
// Measure broadcast interval for burst detection
let broadcast_interval = last_broadcast_time.elapsed().as_secs_f64();
last_broadcast_time = std::time::Instant::now();
@@ -1074,25 +1111,15 @@ async fn broadcast_flac_stream(
);
}
// Detect FLAC header "fLaC" - indicates new track
if bytes.len() >= 4 && &bytes[0..4] == b"fLaC" {
// New track detected: reset sample counter
encoded_samples = 0;
// Cache FLAC header "fLaC" for late-joining clients
// Each encoder instance emits exactly one header at the start
if !header_captured && bytes.len() >= 4 && &bytes[0..4] == b"fLaC" {
header_captured = true;
*header_cache.write().await = Some(bytes.clone());
if !header_captured {
header_captured = true;
trace!(
"FLAC header captured ({} bytes), sample counter reset",
bytes.len()
);
} else {
trace!(
"New FLAC header detected ({} bytes), sample counter reset for new track",
bytes.len()
);
}
// Also broadcast the header so early-connecting clients receive it
// Later-connecting clients will get it from the cache
trace!(
"FLAC header captured and cached ({} bytes) for late-joining clients",
bytes.len()
);
}
let num_receivers = broadcast_tx.receiver_count();
@@ -1221,7 +1248,7 @@ impl StreamingFlacSink {
let logic = StreamingFlacSinkLogic {
encoder_options,
bits_per_sample,
pcm_tx,
pcm_tx: Some(pcm_tx),
pcm_rx: Some(pcm_rx),
metadata,
flac_broadcast,
@@ -1230,6 +1257,8 @@ impl StreamingFlacSink {
sample_rate: None,
broadcast_max_lead_time: broadcast_max_lead_time.max(0.0),
first_chunk_timestamp_checked: false,
timestamp_offset_sec: 0.0,
current_timestamp: Arc::new(RwLock::new(0.0)),
};
let sink = Self {

100
pmoaudio-ext/src/sinks/streaming_ogg_flac_sink.rs
View File

@@ -275,7 +275,7 @@ struct EncoderState {
struct StreamingOggFlacSinkLogic {
encoder_options: EncoderOptions,
bits_per_sample: u8,
pcm_tx: mpsc::Sender<PcmChunk>,
pcm_tx: Option<mpsc::Sender<PcmChunk>>,
pcm_rx: Option<mpsc::Receiver<PcmChunk>>,
metadata: Arc<RwLock<MetadataSnapshot>>,
ogg_broadcast: timed_broadcast::Sender<Bytes>,
@@ -283,11 +283,15 @@ struct StreamingOggFlacSinkLogic {
encoder_state: Option<EncoderState>,
sample_rate: Option<u32>,
broadcast_max_lead_time: f64,
/// Accumulated timestamp offset for maintaining continuity across encoder restarts
timestamp_offset_sec: f64,
/// Shared timestamp reference to read the last broadcast timestamp
current_timestamp: Arc<RwLock<f64>>,
}
impl StreamingOggFlacSinkLogic {
/// Initialize the FLAC encoder once we know the sample rate.
async fn initialize_encoder(&mut self, sample_rate: u32) -> Result<(), AudioError> {
async fn initialize_encoder(&mut self, sample_rate: u32, timestamp_offset_sec: f64) -> Result<(), AudioError> {
if self.encoder_state.is_some() {
return Ok(()); // Already initialized
}
@@ -304,7 +308,8 @@ impl StreamingOggFlacSinkLogic {
.ok_or_else(|| AudioError::ProcessingError("PCM receiver already consumed".into()))?;
// Create shared timestamp and duration for pacing
let current_timestamp = Arc::new(RwLock::new(0.0f64));
// Reuse existing current_timestamp Arc to maintain reference for reading later
let current_timestamp = self.current_timestamp.clone();
let current_duration = Arc::new(RwLock::new(0.0f64));
// Create ByteStreamReader for the encoder
@@ -339,6 +344,7 @@ impl StreamingOggFlacSinkLogic {
current_timestamp,
current_duration,
max_lead,
timestamp_offset_sec,
)
.await
{
@@ -393,6 +399,54 @@ impl StreamingOggFlacSinkLogic {
Ok(())
}
/// Restart the FLAC encoder for a new track.
/// This causes a new OGG stream header to be emitted and timestamps to reset to 0.
async fn restart_encoder_for_new_track(&mut self) -> Result<(), AudioError> {
let sample_rate = self
.sample_rate
.ok_or_else(|| AudioError::ProcessingError("Sample rate not initialized".into()))?;
debug!("Restarting OGG-FLAC encoder for new track");
// 1. Read current timestamp to maintain continuity
let last_timestamp = *self.current_timestamp.read().await;
debug!("Last timestamp before restart: {:.3}s", last_timestamp);
// 2. Close current PCM sender to signal encoder to finish
if let Some(tx) = self.pcm_tx.take() {
drop(tx);
trace!("Dropped PCM sender to signal encoder finish");
}
// 3. Wait for current broadcaster task to complete
if let Some(state) = self.encoder_state.take() {
trace!("Waiting for OGG broadcaster task to finish...");
match state.broadcaster_task.await {
Ok(_) => {
trace!("OGG broadcaster task finished successfully");
}
Err(e) => {
warn!("OGG broadcaster task error during restart: {:?}", e);
}
}
}
// 4. Update timestamp offset to maintain continuity across encoder restart
self.timestamp_offset_sec += last_timestamp;
debug!("New timestamp offset: {:.3}s", self.timestamp_offset_sec);
// 5. Create new PCM channel
let (pcm_tx, pcm_rx) = mpsc::channel::<PcmChunk>(16);
self.pcm_tx = Some(pcm_tx);
self.pcm_rx = Some(pcm_rx);
// 6. Initialize new encoder with timestamp offset (will naturally emit new OGG stream header)
self.initialize_encoder(sample_rate, self.timestamp_offset_sec).await?;
debug!("OGG-FLAC encoder restarted successfully for new track");
Ok(())
}
}
#[async_trait]
@@ -430,8 +484,8 @@ impl NodeLogic for StreamingOggFlacSinkLogic {
self.sample_rate = Some(sample_rate);
debug!("Detected sample rate: {} Hz", sample_rate);
// Initialize the FLAC encoder now
self.initialize_encoder(sample_rate).await?;
// Initialize the FLAC encoder now (first track starts at 0.0)
self.initialize_encoder(sample_rate, 0.0).await?;
}
// Convert chunk to PCM bytes
@@ -455,8 +509,18 @@ impl NodeLogic for StreamingOggFlacSinkLogic {
timestamp_sec: seg.timestamp_sec,
duration_sec,
};
if let Err(e) = self.pcm_tx.send(pcm_chunk).await {
warn!("Failed to send PCM data to encoder: {}", e);
// Get the sender (it should always be Some after initialization)
let pcm_tx = match &self.pcm_tx {
Some(tx) => tx,
None => {
error!("OGG PCM sender not initialized");
break;
}
};
if let Err(e) = pcm_tx.send(pcm_chunk).await {
warn!("Failed to send PCM data to OGG encoder: {}", e);
break;
}
}
@@ -464,10 +528,21 @@ impl NodeLogic for StreamingOggFlacSinkLogic {
_AudioSegment::Sync(marker) => {
match marker.as_ref() {
SyncMarker::TrackBoundary { metadata } => {
// Only restart encoder if it's already initialized (not the first track)
if self.sample_rate.is_some() && self.encoder_state.is_some() {
// Restart encoder to emit new OGG stream header and reset timestamps
if let Err(e) = self.restart_encoder_for_new_track().await {
error!("Failed to restart OGG encoder for new track: {}", e);
break;
}
} else {
trace!("Skipping OGG encoder restart for first track (encoder not yet initialized)");
}
// Update metadata for the new track
if let Err(e) = self.update_metadata(metadata, seg.timestamp_sec).await {
error!("Failed to update metadata: {}", e);
}
// TODO: Implement OGG chaining (EOS → new BOS)
}
SyncMarker::EndOfStream => {
@@ -577,7 +652,7 @@ impl StreamingOggFlacSink {
let logic = StreamingOggFlacSinkLogic {
encoder_options,
bits_per_sample,
pcm_tx,
pcm_tx: Some(pcm_tx),
pcm_rx: Some(pcm_rx),
metadata,
ogg_broadcast,
@@ -585,6 +660,8 @@ impl StreamingOggFlacSink {
encoder_state: None,
sample_rate: None,
broadcast_max_lead_time: broadcast_max_lead_time.max(0.0),
timestamp_offset_sec: 0.0,
current_timestamp: Arc::new(RwLock::new(0.0)),
};
let sink = Self {
@@ -802,6 +879,7 @@ async fn broadcast_ogg_flac_stream(
current_timestamp: Arc<RwLock<f64>>,
current_duration: Arc<RwLock<f64>>,
broadcast_max_lead_time: f64,
timestamp_offset_sec: f64,
) -> Result<(), AudioError> {
trace!(
"OGG-FLAC broadcaster task started with FLAC frame boundary detection (max_lead={:.3}s)",
@@ -1037,10 +1115,10 @@ async fn broadcast_ogg_flac_stream(
);
}
// Calculer le timestamp de cette FLAC frame
// Calculer le timestamp de cette FLAC frame (avec offset pour continuité entre tracks)
let frame_start_samples = encoded_samples;
encoded_samples = encoded_samples.saturating_add(first_frame_samples as u64);
let audio_timestamp = frame_start_samples as f64 / sample_rate_f64;
let audio_timestamp = timestamp_offset_sec + (frame_start_samples as f64 / sample_rate_f64);
let segment_duration = first_frame_samples as f64 / sample_rate_f64;
// Check timing et apply pacing (skip si en retard)