codec: carry codec-frame overflow across audio chunk boundaries

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.

crates/oak-render/examples/render_audio_wav.rs

@@ -0,0 +1,92 @@
+// Temporary diagnostic: render a media file's audio exactly like the
+// playback prefetch does (one sequence frame per chunk through
+// `eval::render_audio_samples`) and write the result as a PCM s16 WAV,
+// so render-side artifacts can be analyzed offline.
+//
+// Usage: render_audio_wav <media> <stream_index> <start_sec> <end_sec> <out.wav> [fps]
+
+use oak_core::{Rational, TimeRange};
+use oak_render::eval::render_audio_samples;
+use oak_render::ticket::{AudioTicketParams, MontageClip, TicketPayload};
+
+fn main() {
+	let args: Vec<String> = std::env::args().collect();
+	if args.len() < 6 {
+		eprintln!("usage: render_audio_wav <media> <stream_index> <start_sec> <end_sec> <out.wav> [fps]");
+		std::process::exit(64);
+	}
+	let media = &args[1];
+	let stream_index: i32 = args[2].parse().unwrap();
+	let start_sec: f64 = args[3].parse().unwrap();
+	let end_sec: f64 = args[4].parse().unwrap();
+	let out_path = &args[5];
+	let fps: i64 = if args.len() > 6 { args[6].parse().unwrap() } else { 25 };
+
+	let sample_rate = 48000i32;
+	let channel_layout = 0x3u64; // stereo
+	let channels = 2usize;
+
+	let start_frame = (start_sec * fps as f64).round() as i64;
+	let end_frame = (end_sec * fps as f64).round() as i64;
+
+	let montage = vec![MontageClip {
+		filename: media.clone(),
+		stream_index,
+		in_time: Rational::new(start_frame, fps),
+		out_time: Rational::new(end_frame, fps),
+		media_in: Rational::new(0, 1),
+		gain: 1.0,
+		effects: Vec::new(),
+	}];
+
+	let mut pcm: Vec<i16> = Vec::new();
+	for f in start_frame..end_frame {
+		let range = TimeRange::new(Rational::new(f, fps), Rational::new(f + 1, fps));
+		let params = AudioTicketParams {
+			viewer: 1,
+			range,
+			sample_rate,
+			channel_layout,
+			montage: montage.clone(),
+		};
+		match render_audio_samples(&params) {
+			Ok(TicketPayload::Audio(samples)) => {
+				for s in &samples.samples {
+					pcm.push((s.clamp(-1.0, 1.0) * 32767.0) as i16);
+				}
+			}
+			other => {
+				eprintln!("frame {f}: render failed: {:?}", other.is_err());
+				let frames = ((sample_rate as f64) / fps as f64).round() as usize;
+				pcm.extend(std::iter::repeat(0i16).take(frames * channels));
+			}
+		}
+	}
+
+	// Minimal 16-bit PCM WAV.
+	let data_len = (pcm.len() * 2) as u32;
+	let mut w = Vec::with_capacity(44 + data_len as usize);
+	w.extend_from_slice(b"RIFF");
+	w.extend_from_slice(&(36 + data_len).to_le_bytes());
+	w.extend_from_slice(b"WAVEfmt ");
+	w.extend_from_slice(&16u32.to_le_bytes());
+	w.extend_from_slice(&1u16.to_le_bytes()); // PCM
+	w.extend_from_slice(&(channels as u16).to_le_bytes());
+	w.extend_from_slice(&(sample_rate as u32).to_le_bytes());
+	w.extend_from_slice(&(sample_rate as u32 * channels as u32 * 2).to_le_bytes());
+	w.extend_from_slice(&(channels as u16 * 2).to_le_bytes());
+	w.extend_from_slice(&16u16.to_le_bytes());
+	w.extend_from_slice(b"data");
+	w.extend_from_slice(&data_len.to_le_bytes());
+	for s in &pcm {
+		w.extend_from_slice(&s.to_le_bytes());
+	}
+	std::fs::write(out_path, &w).expect("write wav");
+	println!(
+		"wrote {} samples ({} ch @ {} Hz) to {}",
+		pcm.len() / channels,
+		channels,
+		sample_rate,
+		out_path
+	);
+}