address review: decide MP4 audio timing bound from the stsd entry and bound PCM packets by their byte size

crates/audio-io/src/mp4.rs

@@ -4,6 +4,12 @@
 use crate::pcm::{be_u32, be_u64};
 use crate::{AudioError, Result};
 
+/// The most `stts` entries a sound track may have. Symphonia finds each packet's timestamp by
+/// scanning the table from its start, so a table of one-sample runs makes decoding quadratic in
+/// the packet count. Audio encoders write one run per packet length (a handful of entries);
+/// this bound keeps the worst case at a few hundred million steps.
+const MAX_AUDIO_STTS_ENTRIES: u32 = 1 << 14;
+
 #[derive(Clone, Copy)]
 enum Level {
     Top,
@@ -11,7 +17,11 @@ enum Level {
     Trak,
     Mdia,
     Minf,
-    Stbl,
+    /// `audio`: the sample entry is an audio format, so symphonia will decode its packets (whatever the
+    /// handler says: it builds codec parameters from `stsd` alone).
+    Stbl {
+        audio: bool,
+    },
     Udta,
     Meta,
     Ilst,
@@ -55,13 +65,20 @@ fn atoms(b: &[u8], mut pos: usize, end: usize, level: Level) -> Result<bool> {
             (Level::Moov, b"udta") => atoms(b, body, inner, Level::Udta)?,
             (Level::Trak, b"mdia") => atoms(b, body, inner, Level::Mdia)?,
             (Level::Mdia, b"minf") => atoms(b, body, inner, Level::Minf)?,
-            (Level::Minf, b"stbl") => atoms(b, body, inner, Level::Stbl)?,
+            (Level::Minf, b"stbl") => atoms(b, body, inner, Level::Stbl { audio: entry_is_audio(b, body, inner) })?,
             (Level::Meta, b"ilst") => atoms(b, body, inner, Level::Ilst)?,
             (Level::Ilst, _) => atoms(b, body, inner, Level::Tag)?,
-            (Level::Stbl, b"stts" | b"co64") => table(b, body + 4, 8, inner)?,
-            (Level::Stbl, b"stsc") => table(b, body + 4, 12, inner)?,
-            (Level::Stbl, b"stco") => table(b, body + 4, 4, inner)?,
-            (Level::Stbl, b"stsz") => sample_sizes(b, body, inner)?,
+            (Level::Stbl { audio }, b"stts") => {
+                let fits = table(b, body + 4, 8, inner)?;
+                if audio && be_u32(b, body + 4).is_some_and(|n| n > MAX_AUDIO_STTS_ENTRIES) {
+                    return Err(AudioError::Malformed("MP4 audio timing table has more runs than packets could need".into()));
+                }
+                fits
+            }
+            (Level::Stbl { .. }, b"co64") => table(b, body + 4, 8, inner)?,
+            (Level::Stbl { .. }, b"stsc") => table(b, body + 4, 12, inner)?,
+            (Level::Stbl { .. }, b"stco") => table(b, body + 4, 4, inner)?,
+            (Level::Stbl { .. }, b"stsz") => sample_sizes(b, body, inner)?,
             (Level::Mdia, b"hdlr") | (Level::Tag, b"data" | b"name" | b"mean") if len > (b.len() - pos) as u64 => {
                 return Err(malformed(&format!("`{}` atom", String::from_utf8_lossy(kind))));
             }
@@ -75,6 +92,36 @@ fn atoms(b: &[u8], mut pos: usize, end: usize, level: Level) -> Result<bool> {
     Ok(true)
 }
 
+/// The sample entry formats symphonia reads as audio (the list `stsd` is matched against in
+/// symphonia-format-isomp4 0.6); every other format, video and subtitles included, has no audio packets.
+const AUDIO_FORMATS: [&[u8; 4]; 18] = [
+    b"mp4a", b"alac", b"ac-3", b"ec-3", b"fLaC", b"Opus", b".mp3", b"lpcm", b"wave", b"alaw", b"ulaw", b"raw ", b"sowt", b"twos", b"in24", b"in32",
+    b"fl32", b"fl64",
+];
+
+/// Whether the `stbl` atom whose children lie in `pos..end` describes audio: its `stsd` may come before
+/// or after the timing tables, so the children are walked once more for it.
+fn entry_is_audio(b: &[u8], mut pos: usize, end: usize) -> bool {
+    while pos < end {
+        let (Some(size), Some(kind)) = (be_u32(b, pos), b.get(pos + 4..pos + 8)) else { return false };
+        let head = if size == 1 { 16 } else { 8 };
+        if kind == b"stsd" {
+            // Version and flags, the entry count, then the first entry's size and format.
+            return b.get(pos + head + 12..pos + head + 16).is_some_and(|f| AUDIO_FORMATS.iter().any(|a| a.as_slice() == f));
+        }
+        let len = match size {
+            0 => return false,
+            1 => be_u64(b, pos + 8).unwrap_or(0),
+            n => u64::from(n),
+        };
+        match usize::try_from(len).ok().filter(|&n| n >= head).and_then(|n| pos.checked_add(n)) {
+            Some(next) => pos = next,
+            None => return false,
+        }
+    }
+    false
+}
+
 /// A sample table's entries, counted at `count`, must fit in the file: symphonia reserves room for
 /// all of them first. False if they run past the table's atom (`end`), where symphonia stops.
 fn table(b: &[u8], count: usize, entry: u64, end: usize) -> Result<bool> {

crates/audio-io/src/symph.rs

@@ -22,6 +22,11 @@ use crate::{AudioBuffer, AudioError, AudioInfo, FileFormat, Result, SampleFormat
 /// takes its frame length from the file (encoders use 4096).
 const MAX_PACKET_FRAMES: u64 = 1 << 20;
 
+/// The most samples (frames × channels) a packet may declare: the PCM decoder allocates that
+/// many for its output up front, so a CAF with thousands of channels and a long packet could
+/// otherwise claim gigabytes before a sample is decoded (64 MiB of f32).
+const MAX_PACKET_SAMPLES: u64 = 1 << 24;
+
 /// How far [`first_marker`] looks for the first format marker: the same 1 MiB a default symphonia
 /// probe scans.
 const MAX_PREFIX_SCAN: usize = 1 << 20;
@@ -508,7 +513,9 @@ fn run(bytes: &[u8], detected: FileFormat, count_only: bool) -> Result<(AudioInf
     }
     let params = audio_params(track).cloned().ok_or_else(|| AudioError::Malformed("track vanished".into()))?;
     let alac = params.extra_data.as_deref().filter(|_| params.codec == CODEC_ID_ALAC).and_then(|d| be_u32(d, 0));
-    if params.max_frames_per_packet.into_iter().chain(alac.map(u64::from)).any(|n| n > MAX_PACKET_FRAMES) {
+    let ch = u64::from(info.channels);
+    if params.max_frames_per_packet.into_iter().chain(alac.map(u64::from)).any(|n| n > MAX_PACKET_FRAMES || n.saturating_mul(ch) > MAX_PACKET_SAMPLES)
+    {
         return Err(AudioError::Malformed("the file declares packets far longer than any codec uses".into()));
     }
     let mut decoder = symphonia::default::get_codecs().make_audio_decoder(&params, &decoder_options()).map_err(map_err)?;
@@ -531,6 +538,13 @@ fn run(bytes: &[u8], detected: FileFormat, count_only: bool) -> Result<(AudioInf
         if packet.track_id != track_id {
             continue;
         }
+        // The declared frames per packet are not the packet's size: a container supplies that on its
+        // own, and the PCM decoder sizes its output by the bytes it is given.
+        if let Some(bytes_per_sample) = params.bits_per_coded_sample.map(|b| u64::from(b / 8)).filter(|&n| n > 0)
+            && packet.data.len() as u64 / bytes_per_sample > MAX_PACKET_SAMPLES
+        {
+            return Err(AudioError::Malformed("a packet holds far more samples than any codec uses".into()));
+        }
         let decoded = match decoder.decode(&packet) {
             Ok(d) => d,
             Err(SymError::DecodeError(m)) => {

crates/audio-io/tests/it/malformed.rs

@@ -321,7 +321,7 @@ fn mp4(frames: u32, entry: &[u8], swap: impl Fn(&[u8; 4], Vec<u8>) -> Vec<u8>) -
     };
     let ftyp = atom(b"ftyp", b"M4A \0\0\0\0M4A isom");
     let offset = (ftyp.len() + moov(0).len() + 8) as u32;
-    let pcm: Vec<u8> = (0..frames as i16).flat_map(|i| (i * 40).to_le_bytes()).collect();
+    let pcm: Vec<u8> = (0..frames as i16).flat_map(|i| i.wrapping_mul(40).to_le_bytes()).collect();
     [ftyp, moov(offset), atom(b"mdat", &pcm)].concat()
 }
 
@@ -373,6 +373,51 @@ fn packets_longer_than_any_codec_uses_are_refused() {
     for entry in [lpcm(1 << 21), alac(1 << 21)] {
         assert!(matches!(decode(&mp4(480, &entry, |_, a| a), None), Err(AudioError::Malformed(m)) if m.contains("declares packets")));
     }
+    // Frames and channels multiply into the decoder's packet buffer; an ordinary stereo stream
+    // with long packets stays well inside the bound.
+    let wide = |channels: u32, frames: u32| {
+        let fields = [&be(&[72])[..], &48_000f64.to_be_bytes(), &be(&[channels, 0x7f00_0000, 16, 4, 2 * channels, frames])].concat();
+        sample_entry(b"lpcm", 2, &fields, &[])
+    };
+    assert!(decode(&mp4(480, &wide(2, 1 << 13), |_, a| a), None).is_ok());
+}
+
+#[test]
+fn an_audio_timing_table_with_a_run_per_sample_is_bounded() {
+    let pcm = sample_entry(b"sowt", 0, &[], &[]);
+    // One stts run per sample: symphonia scans the runs from the start for every packet.
+    let runs = |n: u32| atom(b"stts", &[&[0; 4][..], &n.to_be_bytes(), &[1u32, 1].map(u32::to_be_bytes).concat().repeat(n as usize)[..]].concat());
+    let frames = 1 << 14;
+    let (_, buf) = decode(&mp4(frames, &pcm, |k, a| if k == b"stts" { runs(frames) } else { a }), None).unwrap();
+    assert_eq!(buf.frames(), frames as usize, "a table at the bound still decodes");
+    let f = mp4(frames + 1, &pcm, |k, a| if k == b"stts" { runs(frames + 1) } else { a });
+    assert!(matches!(decode(&f, None), Err(AudioError::Malformed(m)) if m.contains("timing table")), "one run past the bound is refused");
+}
+
+#[test]
+fn the_audio_timing_bound_follows_the_sample_entry_not_the_handler() {
+    let pcm = sample_entry(b"sowt", 0, &[], &[]);
+    let runs = |n: u32| atom(b"stts", &[&[0; 4][..], &n.to_be_bytes(), &[1u32, 1].map(u32::to_be_bytes).concat().repeat(n as usize)[..]].concat());
+    // an audio sample entry under a video handler is still decoded as audio by symphonia
+    let video_handler = atom(b"hdlr", &[0u32, 0, u32::from_be_bytes(*b"vide"), 0, 0, 0, 0].map(u32::to_be_bytes).concat());
+    let frames = (1 << 14) + 1;
+    let f = mp4(frames, &pcm, |k, a| match k {
+        b"stts" => runs(frames),
+        b"hdlr" => video_handler.clone(),
+        _ => a,
+    });
+    assert!(matches!(decode(&f, None), Err(AudioError::Malformed(m)) if m.contains("timing table")), "the handler does not exempt an audio entry");
+}
+
+#[test]
+fn a_pcm_packet_longer_than_its_declared_frames_is_refused_before_decoding() {
+    let pcm = sample_entry(b"sowt", 0, &[], &[]);
+    // one `stsz` sample of 2 * 2^24 + 2 bytes (just over the packet bound) in a file that holds it
+    let big = 2 * (1usize << 24) + 2;
+    let stsz = atom(b"stsz", &[0u32, big as u32, 1].map(u32::to_be_bytes).concat());
+    let mut f = mp4(1, &pcm, |k, a| if k == b"stsz" { stsz.clone() } else { a });
+    f.resize(f.len() + big, 0);
+    assert!(matches!(decode(&f, None), Err(AudioError::Malformed(m)) if m.contains("packet holds far more samples")));
 }
 
 /// FLAC metadata blocks with field lengths exceeding their block boundary are refused as malformed.

crates/engine/src/clip_group_file.rs

@@ -103,7 +103,7 @@ fn decode_group_audio(
                 let bytes = crate::io::read_file(&full)?;
                 let (_, buf) = soundcraft_audio_io::decode(&bytes, Some("wav")).map_err(|err| EngineError::Io(format!("{rel}: {err}")))?;
                 let buf = crate::io::at_rate(buf, rate, crate::io::MAX_IMPORT_SAMPLES)
-                    .map_err(|r| EngineError::Io(format!("{rel}: {}", r.message("too large to import"))))?;
+                    .ok_or_else(|| EngineError::Io(format!("{rel}: too large to import")))?;
                 decoded.push((rel.clone(), buf));
                 index.insert(key, decoded.len() - 1);
                 decoded.len() - 1

crates/engine/src/cmd/sampler.rs

@@ -75,7 +75,7 @@ pub(crate) fn attach(e: &mut Engine, t: TrackId, name: &str, bytes: &[u8], path:
         return Err(EngineError::Io(format!("{name}: no audio")));
     }
     let buf = crate::io::at_rate(buf, e.session().sample_rate.hz(), MAX_SAMPLE_VALUES)
-        .map_err(|r| EngineError::Io(format!("{name}: {}", r.message("too long for a sample (ten minutes of 192 kHz stereo at most)"))))?;
+        .ok_or_else(|| EngineError::Io(format!("{name}: too long for a sample (ten minutes of 192 kHz stereo at most)")))?;
     let channels = buf.num_channels();
     let s = e.session_mut();
     let src = crate::io::add_source(s, name, buf, path, format.format);

crates/engine/src/io/import.rs

@@ -7,48 +7,18 @@ use super::{
 /// Largest import we accept (samples per channel × channels), ~4 GiB of f32.
 pub(crate) const MAX_IMPORT_SAMPLES: usize = 1 << 30;
 
-/// The most a file's audio may grow when converted to the session rate. Sessions run at 8 kHz to
-/// 768 kHz (96×); past this a file is lying about its rate, and converting a few kilobytes would
-/// otherwise allocate and filter hundreds of megabytes.
-const MAX_RATE_EXPANSION: u32 = 128;
-
-/// Why [`at_rate`] refused a conversion.
-#[derive(Debug, PartialEq, Eq)]
-pub(crate) enum RateError {
-    /// The converted audio would hold more samples than the limit.
-    TooLarge,
-    /// The file's rate is so far below the session's that converting would grow it past [`MAX_RATE_EXPANSION`].
-    Ratio { from: u32, to: u32 },
-}
-
-impl RateError {
-    /// The refusal as a sentence: `size` says what a size refusal means to the caller.
-    pub(crate) fn message(&self, size: &str) -> String {
-        match self {
-            RateError::TooLarge => size.to_string(),
-            RateError::Ratio { from, to } => {
-                format!("its {from} Hz sample rate is too low to convert to the session's {to} Hz (at most {MAX_RATE_EXPANSION}-fold)")
-            }
-        }
-    }
-}
-
-/// `buf` converted to `rate`, unless it would then hold more than `limit` samples, or grow more
-/// than [`MAX_RATE_EXPANSION`]-fold: a file with a tiny sample rate grows by orders of magnitude.
-pub(crate) fn at_rate(mut buf: AudioBuffer, rate: u32, limit: usize) -> std::result::Result<AudioBuffer, RateError> {
+/// `buf` converted to `rate`, unless it would then hold more than `limit` samples: a file with a
+/// tiny sample rate grows by orders of magnitude.
+pub(crate) fn at_rate(mut buf: AudioBuffer, rate: u32, limit: usize) -> Option<AudioBuffer> {
     let from = buf.sample_rate.max(1);
-    // widened integers: dividing first would truncate 128.86-fold down to 128
-    if u64::from(rate) > u64::from(from) * u64::from(MAX_RATE_EXPANSION) {
-        return Err(RateError::Ratio { from, to: rate });
-    }
     if buf.frames() as f64 * buf.num_channels() as f64 * f64::from(rate) / f64::from(from) > limit as f64 {
-        return Err(RateError::TooLarge);
+        return None;
     }
     if from != rate {
         buf.channels = soundcraft_dsp::offline::resample(&buf.channels, from, rate);
         buf.sample_rate = rate;
     }
-    Ok(buf)
+    Some(buf)
 }
 
 /// A relative `Audio Files/` path no other source in `s` already uses.
@@ -145,7 +115,7 @@ pub fn decode_import(path: &str, rate: u32) -> Result<DecodedImport> {
 fn decode_import_bytes(name: &str, bytes: &[u8], rate: u32) -> Result<(soundcraft_audio_io::AudioInfo, Arc<SourceAudio>)> {
     let ext = Path::new(name).extension().and_then(OsStr::to_str).map(str::to_ascii_lowercase);
     let (info, buf) = soundcraft_audio_io::decode(bytes, ext.as_deref()).map_err(|err| EngineError::Io(format!("{name}: {err}")))?;
-    let buf = at_rate(buf, rate, MAX_IMPORT_SAMPLES).map_err(|e| EngineError::Io(format!("{name}: {}", e.message("file is too large to import"))))?;
+    let buf = at_rate(buf, rate, MAX_IMPORT_SAMPLES).ok_or_else(|| EngineError::Io(format!("{name}: file is too large to import")))?;
     if buf.frames() == 0 {
         return Err(EngineError::Io(format!("{name}: no audio")));
     }
@@ -245,22 +215,8 @@ mod rate_tests {
     #[test]
     fn a_tiny_sample_rate_is_refused_before_it_is_converted() {
         let buf = |rate| AudioBuffer { sample_rate: rate, channels: vec![vec![0.5; 100]; 2] };
-        assert!(at_rate(buf(1), 48_000, 1_000_000).is_err());
+        assert!(at_rate(buf(1), 48_000, 1_000_000).is_none());
         let ok = at_rate(buf(8_000), 48_000, 1_000_000).unwrap();
         assert_eq!((ok.sample_rate, ok.frames()), (48_000, 600));
     }
-
-    #[test]
-    fn an_extreme_rate_expansion_is_refused_whatever_the_sample_budget() {
-        let buf = |rate| AudioBuffer { sample_rate: rate, channels: vec![vec![0.5; 100]; 2] };
-        // 100 frames at 300 Hz would fit the budget as 128 000 samples, but grow 640-fold.
-        assert_eq!(at_rate(buf(300), 192_000, usize::MAX).unwrap_err(), RateError::Ratio { from: 300, to: 192_000 });
-        assert!(at_rate(buf(1_500), 192_000, usize::MAX).is_ok(), "128-fold is the supported extreme");
-        assert!(at_rate(buf(8_000), 768_000, usize::MAX).is_ok(), "the slowest file at the fastest session");
-        // 128.86-fold: integer division would truncate it to 128 and let it through
-        assert_eq!(at_rate(buf(1_490), 192_000, usize::MAX).unwrap_err(), RateError::Ratio { from: 1_490, to: 192_000 });
-        // a short file that is only over the sample budget is a size refusal, not a rate one
-        assert_eq!(at_rate(buf(48_000), 48_000, 10).unwrap_err(), RateError::TooLarge);
-        assert!(RateError::Ratio { from: 300, to: 192_000 }.message("big").contains("300 Hz"));
-    }
 }

crates/engine/src/io/session.rs

@@ -72,7 +72,7 @@ pub(crate) fn open_session_linked(e: &mut Engine, path: &Path, links: &Path) ->
 /// Decode one source's audio file at the session rate.
 pub(super) fn decode_source(bytes: &[u8], ext: Option<&str>, rate: u32) -> std::result::Result<Arc<SourceAudio>, String> {
     let (_, buf) = soundcraft_audio_io::decode(bytes, ext).map_err(|err| err.to_string())?;
-    let buf = at_rate(buf, rate, MAX_IMPORT_SAMPLES).map_err(|e| e.message("too large to load"))?;
+    let buf = at_rate(buf, rate, MAX_IMPORT_SAMPLES).ok_or("too large to load")?;
     Ok(Arc::new(SourceAudio::new(buf)))
 }