Mike-Solar
9dab9efc38
retrieve_audio_to decodes whole codec frames but copies only the part inside the requested chunk; the tail of the frame crossing the chunk end (up to 1023 samples for AAC) was consumed by the decoder and lost, so the next chunk started with a hole. On the playback grid (1920 samples at 25fps/48kHz) the hole cycles 128..896 samples and hits 7 of 8 chunk starts -- the heavy stutter/noise heard during playback. Keep the overflow (decode and resampler-flush tails) in a per-session carry buffer and serve it at the start of the next contiguous chunk; clear it on seek, format change and chunk failure. Also make seek() actually drop the cached resampler as its comment claimed. Verified sample-exact: chunked decode of a 440Hz tone now matches a one-shot decode bit for bit, and chunked renders of real media line up with the ffmpeg CLI reference at correlation 1.0 / drift 0. Adds a regression test (playback_sized_chunks_match_oneshot_sample_exact) with a new tone fixture -- demo.mp4's audio is -90dB digital silence and cannot expose the holes -- plus a render_audio_wav example used to diagnose chunk-boundary artifacts offline.
93 lines
3.0 KiB
Rust
93 lines
3.0 KiB
Rust
// Temporary diagnostic: render a media file's audio exactly like the
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// playback prefetch does (one sequence frame per chunk through
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// `eval::render_audio_samples`) and write the result as a PCM s16 WAV,
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// so render-side artifacts can be analyzed offline.
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//
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// Usage: render_audio_wav <media> <stream_index> <start_sec> <end_sec> <out.wav> [fps]
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use oak_core::{Rational, TimeRange};
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use oak_render::eval::render_audio_samples;
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use oak_render::ticket::{AudioTicketParams, MontageClip, TicketPayload};
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fn main() {
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let args: Vec<String> = std::env::args().collect();
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if args.len() < 6 {
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eprintln!("usage: render_audio_wav <media> <stream_index> <start_sec> <end_sec> <out.wav> [fps]");
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std::process::exit(64);
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}
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let media = &args[1];
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let stream_index: i32 = args[2].parse().unwrap();
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let start_sec: f64 = args[3].parse().unwrap();
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let end_sec: f64 = args[4].parse().unwrap();
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let out_path = &args[5];
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let fps: i64 = if args.len() > 6 { args[6].parse().unwrap() } else { 25 };
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let sample_rate = 48000i32;
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let channel_layout = 0x3u64; // stereo
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let channels = 2usize;
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let start_frame = (start_sec * fps as f64).round() as i64;
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let end_frame = (end_sec * fps as f64).round() as i64;
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let montage = vec![MontageClip {
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filename: media.clone(),
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stream_index,
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in_time: Rational::new(start_frame, fps),
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out_time: Rational::new(end_frame, fps),
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media_in: Rational::new(0, 1),
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gain: 1.0,
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effects: Vec::new(),
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}];
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let mut pcm: Vec<i16> = Vec::new();
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for f in start_frame..end_frame {
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let range = TimeRange::new(Rational::new(f, fps), Rational::new(f + 1, fps));
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let params = AudioTicketParams {
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viewer: 1,
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range,
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sample_rate,
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channel_layout,
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montage: montage.clone(),
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};
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match render_audio_samples(¶ms) {
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Ok(TicketPayload::Audio(samples)) => {
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for s in &samples.samples {
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pcm.push((s.clamp(-1.0, 1.0) * 32767.0) as i16);
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}
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}
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other => {
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eprintln!("frame {f}: render failed: {:?}", other.is_err());
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let frames = ((sample_rate as f64) / fps as f64).round() as usize;
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pcm.extend(std::iter::repeat(0i16).take(frames * channels));
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}
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}
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}
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// Minimal 16-bit PCM WAV.
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let data_len = (pcm.len() * 2) as u32;
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let mut w = Vec::with_capacity(44 + data_len as usize);
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w.extend_from_slice(b"RIFF");
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w.extend_from_slice(&(36 + data_len).to_le_bytes());
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w.extend_from_slice(b"WAVEfmt ");
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w.extend_from_slice(&16u32.to_le_bytes());
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w.extend_from_slice(&1u16.to_le_bytes()); // PCM
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w.extend_from_slice(&(channels as u16).to_le_bytes());
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w.extend_from_slice(&(sample_rate as u32).to_le_bytes());
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w.extend_from_slice(&(sample_rate as u32 * channels as u32 * 2).to_le_bytes());
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w.extend_from_slice(&(channels as u16 * 2).to_le_bytes());
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w.extend_from_slice(&16u16.to_le_bytes());
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w.extend_from_slice(b"data");
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w.extend_from_slice(&data_len.to_le_bytes());
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for s in &pcm {
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w.extend_from_slice(&s.to_le_bytes());
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}
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std::fs::write(out_path, &w).expect("write wav");
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println!(
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"wrote {} samples ({} ch @ {} Hz) to {}",
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pcm.len() / channels,
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channels,
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sample_rate,
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out_path
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);
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}
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