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craft-math-function-scripts
31
Commits
| Author | SHA1 | Message | Date | |
|---|---|---|---|---|
 Dylan M. Taylorandmpratsaki
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2b35d24324 |
craft-audio: an AAC program config element shaped like 3.0–5.1 gets those speakers (#964)
* craft-audio: a program config element in the shape of a channel configuration gets its speakers An AAC stream with channel configuration 0 whose program config element lists a centre, a front pair and then exactly the elements of configuration 3 to 6 decoded as C L R Ls Rs LFE and was reported as an unordered layout, so a caller reading it in WAV order sent the centre to the left speaker. Such a stream now has the configuration's layout and WAV order. Any other element list keeps the decoded order, as before. Ported from storytold/filmcraft#4280 (aac_stream_order in filmcraft-codecs). * craft-audio: the ADTS program config oracle expects 5.1 in WAV order ffmpeg 9 writes its 5.1 program config element centre first, so the stream now has 5.1's speakers: each decoded channel must match ffmpeg's channel of the same index. An element list in another order keeps the old check. --------- Co-authored-by: mpratsaki <110930793+mpratsaki@users.noreply.github.com> |
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Dylan M. Taylor
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119f6e252e |
craft-audio: integer PCM of WAV and AIFF files bit for bit, and minimal files around it (#917)
wav_pcm and aiff_pcm (feature decode-native) return a file's integer PCM as stored (RawPcm: little-endian interleaved bytes, 8-bit unsigned, with channels, rate and bits), built on the existing WAV and AIFF chunk parsers. pcm_wav_file, pcm_wav_header, pcm_aiff_file and pcm_aiff_header (feature pcm-encode) write the 44-byte canonical WAVE header and the plain AIFF header around such bytes. For essence embedded in AAF and OMF files, which is moved bit for bit and whose descriptors hold a bare header as a summary. The writers are FilmCraft's (filmcraft-interchange wav.rs); the readers replace its second WAV/AIFF parser with this crate's. |
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Dylan M. Taylor
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0ef4c07c4d | craft-audio: bound the work of MP4 audio timestamp lookups, not the stts entry count (#910) | ||
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Dylan M. Taylor
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1ebc34b0cf |
craft-audio: MP3 (Layer III) encoder from FilmCraft (feature mp3-encode) (#869)
Moves filmcraft-mp3enc (storytold/filmcraft@4bec22b) into craft_audio::mp3, on craft-media's BitWriter. `encode` turns the feature on, mp3::Error converts into AudioError::Encode, and a test covers every sample rate and bit rate with extreme signals. |
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f19f131247 |
craft-audio: a damaged FLAC keeps its timing; AAC uses BitReader::peek (#846)
* craft-audio: AAC Huffman lookahead uses BitReader::peek `BitReader::peek` reads from one 8-byte load since the bit reader was made faster, which is what the AAC decoder's private `peek32` did. The private copy goes; the lookahead is the same zero-padded 32 bits. Ported from storytold/filmcraft#4232. * craft-audio: a damaged FLAC keeps its length and timing symphonia's FLAC demuxer drops a frame whose checksum fails and resyncs on the next one, so the decoder never saw an error: the file decoded shorter and all later audio came out one block early, with `damaged` at 0. The jump in the packet timestamps is the audio that was lost. It is returned as silence, in blocks of at most 65 536 frames, and counted once in `AudioInfo::damaged`. It never passes the length the file declares; with no declared length a jump of more than 2^20 frames is not believed. A seek into the lost frames lands on silence at the right position. Ported from storytold/soundcraft#556, onto the streaming decoder. --------- Co-authored-by: vangel76 <vangel1976@googlemail.com> Co-authored-by: Gert Pellin <gert@pellin.be> |
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Dylan M. Taylor
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c814fdfe7c | craft-audio: bound MP4 audio stts entries and frames x channels per packet (port of soundcraft #1332) (#825) | ||
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Dylan M. Taylor
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0e60ef3b0a |
craft-audio: extension and media-type tables, linear WAV chunk scan, SoundCraft's later tests (#824)
DECODE_EXTENSIONS lists the file extensions of the formats decode reads. EncodeFormat gains ALL, extension() and mime_type(). SoundCraft kept these tables in its own FileFormat type; they are format knowledge, so they live here. A WAV with many empty data chunks parsed in quadratic time: whether an empty data chunk is followed by chunks alone was decided for every one of them, by walking the rest of the file. Only the first data chunk is used, so it is decided once. Tests SoundCraft added to audio-io after craft-audio was cut from it: Matroska element sizes beyond the file, an AIFF channel count with the sign bit set, probe and decode agreeing on the length of AAC in MP4, and one small file per decodable extension (tests/fixtures). |
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Dylan M. Taylor
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44b4a23137 |
craft-audio, craft-media: the changes FilmCraft and SoundCraft made to their copies since extraction (#807)
* craft-audio: FilmCraft's FLAC encoder and MP1/MP2 decoder speed-ups FLAC: per-order predictor kernels, merged Rice partition sums, the LPC window kept per block length, and batches consumed by offset (filmcraft 5eaff794, d9dad4a5). The output is unchanged: 64 golden cases hash the same before and after. MP1/MP2: the faster synthesis filterbank (filmcraft 51c755e5), with its bit-exact test. * craft-audio: FilmCraft's AAC and Opus decoder speed-ups and refactors AAC: table-driven Huffman decoding, per-stage FFT twiddles and fused stages, reused per-channel work buffers, and the encoder's memoised per-band work (filmcraft f9627280, d587ceed, 6121635c, dc124ab7, 7a532eca). Opus: radix-2/3/4 FFT kernels in the CELT IMDCT, the PVQ table built once, one de-emphasis loop for both channels, the SILK resampler without per-sample divisions and with eight partial sums, concealment and FEC capped at 120 ms (filmcraft bf7ac420, 12e32026, c4dc64d1, eb352ec8, 87b8e22d, 014b5e56, 78d77b72). The per-frame mid-only flag this crate fixed in the SILK decoder is kept in the refactored decode. * craft-media: Ogg CRC-32 by slice-by-16 FilmCraft's 6763f9bf: sixteen independent lookups per 16 input bytes, so opening a file is several times faster. Checked against a bit-at-a-time reference on every short length and alignment, and across every split. * craft-audio: SoundCraft's CAF cookie-size and FLAC vendor-length checks A CAF magic cookie size above the signed range is refused instead of passed to symphonia (soundcraft 3bd3965), and a Vorbis comment block whose vendor string claims more than the block holds is refused (soundcraft ddfe991). Each with its regression test. * craft-audio: AC-3 reads bits through the shared BitReader FilmCraft's 1c9c2e82: the AC-3 decoder's private bit-at-a-time reader is replaced by craft-media's BitReader, with its two tests: the decoded output of 2,000 synthetic and 12 real syncframes (and their damaged variants) hashes the same as before, and the reader agrees with the old one on random reads, skips and overruns. The three AC-3 features now turn on craft-media's bitstream module. * craft-audio: the quantiser rounds without a libm call SoundCraft's d293336, the audio-io part: round-half-away by integer arithmetic in the bounce's inner loop, with its test against f64::round over the quantiser's range. |
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Dylan M. Taylorandwhatever-industries
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2765c84bda |
craft-audio: AC-3 and E-AC-3 packet encoders (#806)
* feat(audio): add AC-3 and E-AC-3 packet encoders * craft-audio: AC-3 encoder errors convert to AudioError::Encode --------- Co-authored-by: whatever-industries <77seventy77@users.noreply.github.com> |
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Aleksander HelgakerandDylan M. Taylor
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7e40e12989 |
craft-audio: ALAC decoder and encoder (#731)
* craft-audio: ALAC decoder and encoder, ported from Apple's reference Add `craft_audio::alac`, a Rust port of Apple's ALAC reference implementation (macosforge/alac@c38887c, Apache-2.0): - `alac::Decoder`: packets to planar i32, bit depths 1-32, 1-8 channels, cookies in MP4/Matroska (24-byte config), CAF and old QuickTime form. Hostile input returns errors; arithmetic wraps where C would overflow. - `alac::Encoder`: 16/20/24/32 bits, 1-8 channels, fast mode, any frame length; same packets as the reference for the same input. - `SpecificConfig` (cookie parse/write), `wav_order`/`alac_order`, `channel_mask`. - Features `alac-decode` (in `decode-native`) and `alac-encode` (in `encode`). `TrackDecoder` decodes ALAC natively, so ALAC in MP4 and Matroska no longer goes through symphonia (CAF still does). Apple's reference writes the cookie's channel layout tag in host byte order; this writes it big-endian as its cookie description specifies and reads either. Tests: unit tests for each part, an MP4 decode-and-seek test without symphonia, and ffmpeg oracles both ways (bit-exact, 1-8 channels). NOTICE and each file's header record the Apache-2.0 origin. * craft-audio: ALAC review fixes (#731) - Decode the full-width residual escape 0xffffffff as i32::MIN; the reference's `nd + 1` wraps it to 0. - Require a whole element tag after the last channel (the end tag, after any fill or data stream elements), so a packet that lost its last byte is an error instead of decoding without damage. Other elements there end the packet, as in the reference. - Refuse an ALAC track with no sample rate in its cookie or container. - Conceal a damaged packet with the length its header gives (`alac::Decoder::packet_frames`), falling back to the last packet's. - Matroska: place ALAC packet starts at multiples of the frame length instead of the rounded block timestamps, so seeks land on the exact frame. MP4 keeps its exact sample-table timing. * craft-audio: exact Matroska ALAC seeks after a short packet (#731) Packet starts were multiples of the frame length, so a short packet before the end moved every later packet and seeks after it returned the wrong frames. When a packet's timestamp is further from that multiple than its rounding allows, the starts are now counted from the earlier packets' headers (read once and kept). Other codecs with a fixed packet length fall back to the timestamp, as before. --------- Co-authored-by: Dylan M. Taylor <dylan@dylanmtaylor.com> |
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ca53e90b18 |
craft-audio: MP1/MP2 decoder (#448)
FilmCraft's MPEG audio Layer I/II decoder as `craft_audio::mpa` behind `mpa-decode`, kept over symphonia's: the same quality where both decode, and it also decodes the MP2 files ffmpeg writes by default (mono at 384 kbit/s), which symphonia rejects. `TrackDecoder` and the native read path take MP1/MP2 in MP4, Matroska and raw streams; symphonia keeps MP3 only, so its `mp1` and `mp2` features are dropped. Co-authored-by: Claude Opus 5.5 <noreply@anthropic.com> Co-authored-by: Dylan M. Taylor <dylan@dylanmtaylor.com> |
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b1c64766e6 |
craft-audio: AC-3 and E-AC-3 decoder (9/10) (#447)
* craft-audio: AC-3 and E-AC-3 decoder FilmCraft's AC-3 decoder as `craft_audio::ac3` behind `ac3-decode`: every AC-3 channel mode and frame size, and E-AC-3 independent substream 0 with spectral extension, in WAV order without downmix. `TrackDecoder` and the native read path take AC-3 and E-AC-3 in MP4, Matroska and raw streams. Not bit-exact: zero-bit mantissas are filled with decoder-specific dither (A/52 §7.3.4), so tests compare at 65-94 dB SNR on tones and 27-48 dB on white noise. 7.1 E-AC-3 decodes as its 5.1 core. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * craft-audio: review fixes, round 1 (#447) An AC-3 / E-AC-3 packet is as long as all the primary syncframes in it. Before the first decode, TrackDecoder::frames_per_packet gave 1536 for any AC-3 track, and conceal used the first syncframe's header only. An E-AC-3 stream of one-, two- or three-block syncframes then got the wrong MP4 length and seek positions and the wrong Matroska positions. A packet that groups several such syncframes concealed at a fraction of its length. Seeks in AC-3 and E-AC-3 now have to match the whole-file decode bit for bit, as every other codec's do. The dither is seeded from each syncframe, so the old 25 dB bound was looser than the decoder needs. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * craft-audio: review fixes, round 2 (#447) - AC-3 / E-AC-3 packets that decode to different lengths (in the packets read ahead, or any decoded since) have no frames_per_packet, so a Matroska seek finds its packet by timestamp rather than by index times the first packet's length. - E-AC-3 syncframes of fewer than six blocks may reuse exponents from any syncframe before them: once one is seen, a seek decodes from the start of the stream instead of 1536 frames before the target. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * craft-audio: review fixes, round 3 (#447) - Matroska E-AC-3: the first seek reads every packet's syncframe headers (only the headers, header stripping undone) and places packets by the frames their primary syncframes hold. Packets that started to differ in length after the eight read ahead were placed as the first packet's length times their index. - A packet whose later syncframe fails no longer sets the length lost packets are concealed at, nor marks the packets' lengths as differing: it decoded only in part. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * Merge main; E-AC-3 in Matroska: read packet headers only up to the seek target, and count a packet of dependent syncframes as zero frames --------- Co-authored-by: Claude Opus 5.5 <noreply@anthropic.com> Co-authored-by: Dylan M. Taylor <dylan@dylanmtaylor.com> |
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861c5e172d |
craft-audio: Opus decoder (7/10) (#445)
* craft-audio: Opus decoder FilmCraft's `filmcraft-opus` decoder as `craft_audio::opus` behind `opus-decode`: SILK, hybrid and CELT, every frame size, PLC and FEC, multistream (`Decoder`) and single stream (`StreamDecoder`), 8-48 kHz output. - `TrackDecoder` decodes Opus; mono and stereo without an `OpusHead` (RTP, MPEG-TS) get a default one. Lost packets use Opus's own concealment. - The native read path takes Ogg Opus (pre-skip and end trimming per RFC 7845), and Opus in Matroska/WebM and MP4, where packet starts after a seek come from the packets' own lengths. The RFC 8251 conformance test runs all 36 vectors. `scripts/fetch-opus-vectors.sh` downloads them (SHA-256 checked) and the CI test job caches them. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * craft-audio: review fixes, round 1 (#445) - CELT concealment keeps its pitch in its own field. It reused postfilter_period_old, which the next good frame's post-filter cross-fades from. - Matroska Opus seeks: a failed packet read no longer caches a wrong start; a one-byte packet is read as one byte. - A malformed OpusHead in Ogg is reported as malformed instead of falling back. - The libopus oracle tests skip through the testkit, so CRAFT_MEDIA_REQUIRE_ORACLES fails them. - Decoder::reset documents that it re-arms pre-skip trimming. - fetch-opus-vectors.sh retries its size probe. - CI: the vectors are fetched after ffmpeg is installed, and apt's timeout runs as root so a stalled apt-get is killed and frees the dpkg lock. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * craft-audio: review fixes, round 2 (#445) - CELT: the noise-based concealment sets skip_plc, as libopus does, so pitch-based concealment waits for two good frames in a row instead of extending history that holds noise. - SILK: decode_only_middle is per call, as in libopus. A lost or mono frame no longer reuses the mid-only flag from an earlier frame. - A damaged Opus packet is concealed at its own length (from its TOC and framing) rather than the previous packet's. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> --------- Co-authored-by: Claude Opus 5.5 <noreply@anthropic.com> Co-authored-by: Dylan M. Taylor <dylan@dylanmtaylor.com> |
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Aleksander HelgakerandClaude Opus 5.5
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ca99f91a18 |
craft-audio: AAC encoder (6/10) (#444)
* craft-audio: AAC decoder
FilmCraft's `filmcraft-aac` decoder as `craft_audio::aac` behind
`aac-decode`, kept over symphonia's: the same quality (about 135 dB
SNR against ffmpeg), and it decodes 5.1, which symphonia refuses.
HE-AAC decodes as its AAC-LC core, upsampled 2x to the stream's rate.
- `TrackDecoder` decodes AAC, including ADTS-framed packets without
extradata (MPEG-TS, raw `.aac`).
- The native read path takes AAC in MP4 and Matroska and raw ADTS
streams (frame sync over the file, with resync after damage).
- AAC errors convert into `AudioError`.
The AAC module's shared parts (Huffman, MDCT, TNS, ICS) arrive whole
as ported; their encoder half is unused until the encoder PR.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 1 (#443)
Raw ADTS and TrackDecoder:
- Channel configuration 0 (a PCE in the stream, as ffmpeg's -aac_pce and
AudioToolbox's 7.x write) decodes; it was refused.
- Frames with several raw data blocks are refused; only their first block
was decoded, silently losing the rest.
- A frame that changes the sample rate, profile or channel configuration is
concealed; it played at the wrong speed.
- A source that fails during the frame scan is AudioError::Io; it looked
like the end of the file.
- A stream after an ID3 tag longer than the file head is found.
- conceal counts all the ADTS frames of a packet, and a packet of more than
1024 frames is refused before it is decoded into memory.
AAC decoder:
- Indexed channel configurations place elements by arrival order, not tag,
as ffmpeg does; tagged differently, channels were swapped.
- A PCE in an access unit that fails no longer changes the layout, which
left every later packet failing.
- Non-finite samples are zeroed before the output and the overlap.
MP4: probe and decode agree: the samples' durations end the track when no
single edit does, and an edit cannot outlast the media.
The routing oracle fails on a fixture ffmpeg cannot write when oracles are
required. The README's Limits names HE-AAC v2 as mono, multi-block ADTS,
iTunSMPB, Matroska's tick precision and AAC in CAF.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 2 (#443)
Corrects round 1:
- A multi-block ADTS frame after the stream has started is concealed at its
whole length; round 1's filter skipped it, shifting the audio after it.
- Sample durations only trim an MP4's last packet: wrong ones (some count a
HE-AAC packet's core frames) were cutting audio. Durations of 0 with an edit
use the edit.
Raw ADTS packets now keep their headers, so TrackDecoder alone handles format
changes, multi-block frames and concealment, for apps as for files, and nested
ADTS payloads are not split again. Headers come from the frame scan's reads,
not one read per frame. The stream's format is the one most of its first frames
share, so one damaged header does not silence the file. The ID3 walk and a
packet's ADTS split are bounded.
AAC:
- HE-AAC is delayed by the SBR filterbanks' 481 core samples; it played 19.5 ms
early against ffmpeg in every container.
- A failed access unit leaves the decoder's state as it was, so the next packet
decodes as if it had not been there.
- reset restarts the noise generator, so a seek into PNS repeats.
- A signalled rate that is not twice the core's plays the core; it reported the
signalled rate with the core's samples.
The decoder oracle's tolerance tightens from 1e-3 to 1e-5. Tests cover HE-AAC.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 3 (#443)
Corrects round 2:
- ADTS headers are compared with the stream's format only when the first header
gave it. With extradata the format comes from there, so ADTS in MP4 whose
headers say something else decodes again; every packet had been concealed.
- A stream's only program config element is found in its first frames even when
that frame's header is damaged.
- The test of wrong sample durations patched the second run's count, not the
delta, so "durations of one" changed nothing. It now patches the delta, and
checks a file was changed and that seeks land on the same audio.
MP4 seeks place a packet by its index when every packet decodes to the same
length, as decoding does; they used the durations round 2 had stopped trusting
for the length, and returned the wrong audio. A last delta of 0 is overridden too.
Docs: what TrackInfo says for raw ADTS, that opening one reads the whole file, and
which codecs symphonia reads.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 4 (#443)
- TrackDecoder takes an ADTS stream's format from the one most of the lookahead
shares, as the file path did, so one damaged first header does not fail every
packet of an MPEG-TS stream. open_adts now just asks it.
- MP4 samples of several ADTS frames keep the sample table's timing; a packet
count of one frame each truncated them silently.
- Sample durations past what the packets decode to no longer over-report the
length.
- split_adts_upto is not behind aac-decode, as split_adts is not: an aac-encode
build in the next PR did not compile.
Removes an unreachable ID_END check and open_adts's unused core rate.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 5 (#443)
No change in decoding. An aac-encode build in the next PR needs the
split_adts_upto re-export behind aac-decode, and ID_END only in tests.
Docs say the lookahead's headers give an ADTS stream's configuration, that AAC in
CAF still goes through symphonia, and that ProgramConfig is not #[non_exhaustive].
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 6 (#443)
Two tests from round 1 failed on ffmpeg before 9, including CI's 8.0. Neither was
a decoding fault:
- ffmpeg 9 ends an MP4 with no edit list at its sample durations and earlier
versions present the whole last packet, so the length is checked against the
file's known duration, not ffmpeg's decode.
- The order ffmpeg gives channels with -aac_pce differs by version, so each decoded
channel must match a different one of ffmpeg's, whatever the order.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 7 (#443)
Reading:
- an element of another type or a tag a program config does not list is
refused, not written to another channel; only an LFE coded as an SCE
is still allowed, as ffmpeg allows it
- an explicit sampling rate of 0 is refused
- a packet's later ADTS frames are searched for SBR
- a multi-frame ADTS packet that fails leaves the decoder as it was, and
one whose frames change the channel count is an error
- the whole ADTS frames before a damaged end of a packet decode
- a failed read of a raw stream's first frames is an I/O error
- an MP4 sample of several ADTS frames past the lookahead keeps its length
- an Ogg file with leading bytes keeps its track order and mask comments
Writing:
- ProgramConfig::write, AudioSpecificConfig::to_bytes and
AdtsHeader::write refuse what their fields cannot hold
Tests:
- the routing test checks one alignment for the ffmpeg version under test
- a default decode past an Ogg stream symphonia has no track for
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 8 (#443)
From wanikhawar's review:
- AudioSpecificConfig::parse reads backward-compatible SBR and PS signalling
(a 0x2b7 sync extension after the config), which craft-media's Matroska
demuxer writes for legacy HE-AAC IDs. Without lookahead those streams decoded
as their core.
- An MP4 media edit after one empty edit is applied: its priming skip and its
end. The empty edit's delay is not added as silence.
- A packet lost whole is concealed as long as the last one decoded, so a stream
of multi-frame ADTS packets keeps its timing.
- ADTS-framed MP4 samples keep their sample table's timing, instead of every
sample being read at open to look for several frames in one.
- AdtsHeader::write refuses an object type, sampling frequency index or channel
configuration its fields cannot hold; it masked them into other values.
- AudioSpecificConfig::lc no longer panics for a sampling frequency index with no
rate, and the decoder refuses such a config.
- AudioSpecificConfig::to_bytes refuses channel configuration 0 without a program
config element, which would not parse back.
- Upsample2x starts each channel's history afresh when the channel count
changes.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 9 (#443)
From CodeRabbit's review of round 8:
- AudioSpecificConfig::parse consumes extensionFlag3 after the config when
extensionFlag is set, so the 0x2b7 sync extension after it is found.
- AudioSpecificConfig::to_bytes refuses a sampling rate of 0, which would not
parse back.
- ADTS-framed HE-AAC in MP4 whose sample durations count core frames is not cut
to the durations' half length: durations short of one frame per sample no
longer end an ADTS-framed track, and its length is unknown until decoded.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 10 (#443)
ADTS-framed AAC in MP4: durations short of the frames in the samples read
ahead no longer end the track. With HE-AAC samples of two ADTS frames each,
timed as two core frames, the durations matched one output frame per sample,
so the undercount check passed and the track stopped at half its length,
which probe reported as known. The frames each read-ahead sample holds are
now counted, and durations short of them leave the length unknown until
decoded, as durations short of one frame per sample already did. Correctly
timed AAC-LC samples of several frames keep their end trim.
Reported by FredApps.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: review fixes, round 11 (#443)
Corrects round 10. A short ADTS-framed MP4 track, whose last sample is in
the read-ahead, lost that sample's trim when it cut more than one frame:
the trim counted as durations falling short, so the length was left to
decoding and its padding was presented. The track's last sample is now left
out of the read-ahead comparison.
Reported by CodeRabbit.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: document the ADTS-in-MP4 seek and edit limits (#443)
AAC in MP4 stored as ADTS frames, with sample durations that count fewer
frames than the samples hold, decodes in full from the start, but seeks
place samples by those durations and land later than asked, and an edit
list measured in them can end the track early. The README's Limits section
now says so, and the seek example points to it.
Reported by FredApps.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
* craft-audio: say why the ADTS-in-MP4 seek limit is kept (#443)
The Limits entry from
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alehelandClaude Opus 5.5
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ac8c5aae5d |
craft-audio: review fixes, round 8 (#443)
From wanikhawar's review: - AudioSpecificConfig::parse reads backward-compatible SBR and PS signalling (a 0x2b7 sync extension after the config), which craft-media's Matroska demuxer writes for legacy HE-AAC IDs. Without lookahead those streams decoded as their core. - An MP4 media edit after one empty edit is applied: its priming skip and its end. The empty edit's delay is not added as silence. - A packet lost whole is concealed as long as the last one decoded, so a stream of multi-frame ADTS packets keeps its timing. - ADTS-framed MP4 samples keep their sample table's timing, instead of every sample being read at open to look for several frames in one. - AdtsHeader::write refuses an object type, sampling frequency index or channel configuration its fields cannot hold; it masked them into other values. - AudioSpecificConfig::lc no longer panics for a sampling frequency index with no rate, and the decoder refuses such a config. - AudioSpecificConfig::to_bytes refuses channel configuration 0 without a program config element, which would not parse back. - Upsample2x starts each channel's history afresh when the channel count changes. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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542ca365e2 |
craft-audio: review fixes, round 7 (#443)
Reading: - an element of another type or a tag a program config does not list is refused, not written to another channel; only an LFE coded as an SCE is still allowed, as ffmpeg allows it - an explicit sampling rate of 0 is refused - a packet's later ADTS frames are searched for SBR - a multi-frame ADTS packet that fails leaves the decoder as it was, and one whose frames change the channel count is an error - the whole ADTS frames before a damaged end of a packet decode - a failed read of a raw stream's first frames is an I/O error - an MP4 sample of several ADTS frames past the lookahead keeps its length - an Ogg file with leading bytes keeps its track order and mask comments Writing: - ProgramConfig::write, AudioSpecificConfig::to_bytes and AdtsHeader::write refuse what their fields cannot hold Tests: - the routing test checks one alignment for the ffmpeg version under test - a default decode past an Ogg stream symphonia has no track for Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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d51518952e |
craft-audio: review fixes, round 6 (#443)
Two tests from round 1 failed on ffmpeg before 9, including CI's 8.0. Neither was a decoding fault: - ffmpeg 9 ends an MP4 with no edit list at its sample durations and earlier versions present the whole last packet, so the length is checked against the file's known duration, not ffmpeg's decode. - The order ffmpeg gives channels with -aac_pce differs by version, so each decoded channel must match a different one of ffmpeg's, whatever the order. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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f318d0157f |
craft-audio: review fixes, round 4 (#443)
- TrackDecoder takes an ADTS stream's format from the one most of the lookahead shares, as the file path did, so one damaged first header does not fail every packet of an MPEG-TS stream. open_adts now just asks it. - MP4 samples of several ADTS frames keep the sample table's timing; a packet count of one frame each truncated them silently. - Sample durations past what the packets decode to no longer over-report the length. - split_adts_upto is not behind aac-decode, as split_adts is not: an aac-encode build in the next PR did not compile. Removes an unreachable ID_END check and open_adts's unused core rate. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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ec549d4c89 |
craft-audio: review fixes, round 3 (#443)
Corrects round 2: - ADTS headers are compared with the stream's format only when the first header gave it. With extradata the format comes from there, so ADTS in MP4 whose headers say something else decodes again; every packet had been concealed. - A stream's only program config element is found in its first frames even when that frame's header is damaged. - The test of wrong sample durations patched the second run's count, not the delta, so "durations of one" changed nothing. It now patches the delta, and checks a file was changed and that seeks land on the same audio. MP4 seeks place a packet by its index when every packet decodes to the same length, as decoding does; they used the durations round 2 had stopped trusting for the length, and returned the wrong audio. A last delta of 0 is overridden too. Docs: what TrackInfo says for raw ADTS, that opening one reads the whole file, and which codecs symphonia reads. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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555be7c20c |
craft-audio: review fixes, round 2 (#443)
Corrects round 1: - A multi-block ADTS frame after the stream has started is concealed at its whole length; round 1's filter skipped it, shifting the audio after it. - Sample durations only trim an MP4's last packet: wrong ones (some count a HE-AAC packet's core frames) were cutting audio. Durations of 0 with an edit use the edit. Raw ADTS packets now keep their headers, so TrackDecoder alone handles format changes, multi-block frames and concealment, for apps as for files, and nested ADTS payloads are not split again. Headers come from the frame scan's reads, not one read per frame. The stream's format is the one most of its first frames share, so one damaged header does not silence the file. The ID3 walk and a packet's ADTS split are bounded. AAC: - HE-AAC is delayed by the SBR filterbanks' 481 core samples; it played 19.5 ms early against ffmpeg in every container. - A failed access unit leaves the decoder's state as it was, so the next packet decodes as if it had not been there. - reset restarts the noise generator, so a seek into PNS repeats. - A signalled rate that is not twice the core's plays the core; it reported the signalled rate with the core's samples. The decoder oracle's tolerance tightens from 1e-3 to 1e-5. Tests cover HE-AAC. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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d7a40bc110 |
craft-audio: review fixes, round 1 (#443)
Raw ADTS and TrackDecoder: - Channel configuration 0 (a PCE in the stream, as ffmpeg's -aac_pce and AudioToolbox's 7.x write) decodes; it was refused. - Frames with several raw data blocks are refused; only their first block was decoded, silently losing the rest. - A frame that changes the sample rate, profile or channel configuration is concealed; it played at the wrong speed. - A source that fails during the frame scan is AudioError::Io; it looked like the end of the file. - A stream after an ID3 tag longer than the file head is found. - conceal counts all the ADTS frames of a packet, and a packet of more than 1024 frames is refused before it is decoded into memory. AAC decoder: - Indexed channel configurations place elements by arrival order, not tag, as ffmpeg does; tagged differently, channels were swapped. - A PCE in an access unit that fails no longer changes the layout, which left every later packet failing. - Non-finite samples are zeroed before the output and the overlap. MP4: probe and decode agree: the samples' durations end the track when no single edit does, and an edit cannot outlast the media. The routing oracle fails on a fixture ffmpeg cannot write when oracles are required. The README's Limits names HE-AAC v2 as mono, multi-block ADTS, iTunSMPB, Matroska's tick precision and AAC in CAF. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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7d27e2d0a6 |
craft-audio: AAC decoder
FilmCraft's `filmcraft-aac` decoder as `craft_audio::aac` behind `aac-decode`, kept over symphonia's: the same quality (about 135 dB SNR against ffmpeg), and it decodes 5.1, which symphonia refuses. HE-AAC decodes as its AAC-LC core, upsampled 2x to the stream's rate. - `TrackDecoder` decodes AAC, including ADTS-framed packets without extradata (MPEG-TS, raw `.aac`). - The native read path takes AAC in MP4 and Matroska and raw ADTS streams (frame sync over the file, with resync after damage). - AAC errors convert into `AudioError`. The AAC module's shared parts (Huffman, MDCT, TNS, ICS) arrive whole as ported; their encoder half is unused until the encoder PR. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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57f5d063fe |
craft-audio: review fixes, round 12 (#442)
- FLAC files read their layout from the channel mask comment, or else the one decoders assume for the count, which is what the writer omits the comment for - an unbuilt format is Unsupported before the layout is checked - FLAC encode no longer reserves half the PCM size up front - flac-encode's lib tests stream a file and compare it with encode - the seek test compares samples after the seek Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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afdc045e94 |
craft-audio: review fixes, round 10 (#442)
Tests only; each was confirmed to fail on a mutated encoder. - Every sample rate code in the frame header table. - Signals that reach the Rice parameter and LPC shift limits, where a value past the limit gives a file decoders refuse. - ENCODE_EXTENSIONS lists flac. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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82f63f3a4f |
craft-audio: review fixes, round 8 (#442)
- Code wasted bits (RFC 9639 §9.2.2): low bits that are zero in every sample of a subframe were stored anyway, so 16-bit audio exported at 24 or 32 bits grew by the padding. - Allow residuals up to the 32-bit limit (§9.2.7.3). The limit was half that, so at 32 bits a transient forced its whole block to verbatim. - README: 32-bit FLAC codes stereo channels independently. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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430ad15683 |
craft-audio: review fixes, round 6 (#442)
Tests only, plus docs; the encoder does not change. Each new check was confirmed to fail on a mutated encoder. - Tests: frame numbers count up, LPC and Rice partitions and stereo decorrelation save the space they should, FileEncoder records a FLAC layout, the smallest frame anywhere reaches STREAMINFO, Writer::write applies the frame limit, finish leaves the sink at the end, the exact 655350 Hz cap, the sample size codes. - README limits: a FLAC layout decoders would not assume depends on its mask comment, and ffmpeg and flac ignore a zero mask. - The encode example's comment named no referent. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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8ba7778f8c |
craft-audio: review fixes, round 4 (#442)
The PICTURE metadata test inserts its block without the last-block flag: the encoder's VORBIS_COMMENT follows it, and the flagged version was a broken metadata chain that only decoded by resyncing. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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76df23e86e |
craft-audio: review fixes, round 2 (#442)
- check_room docs: FLAC holds at most 2^36 - 1 frames. - The FLAC round trip covers every channel count from 1 to 8. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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7c88c201a5 |
craft-audio: review fixes, round 1 (#442)
- FileEncoder/encode for FLAC hand the encoder one 4096-frame block at a time: a whole-buffer encode drained the pending samples quadratically (480 s of stereo took 19 s, now 5 s) and copied the whole input twice. - Frame headers name sample rates outside the table (codes 12-14), so streams at rates like 11025 or 37800 Hz stay in the streamable subset. - The decode doctest needs symphonia to read FLAC at this point. - README: FLAC output is verified against ffmpeg, not the flac tool. - Tests: keep the AIFF force-RF64 check; exact 2^36 frame boundary. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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alehelandClaude Opus 5.5
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550732201c |
craft-audio: FLAC encoder
FilmCraft's FLAC encoder behind `flac-encode`, kept over SoundCraft's: linear prediction, stereo decorrelation and the STREAMINFO MD5 give 0.1-37% smaller files on 12 test signals, and the output is bit-exact through `flac` and ffmpeg. It is 3-5x slower than SoundCraft's, still about 56x real time. `lossless-encode` is now `pcm-encode` plus `flac-encode`, and `EncodeFormat::Flac` writes it through `encode` and `FileEncoder`. With only one of the two features, the other's formats are refused as `Unsupported`. The tests that use FLAC files as decode fixtures (truncation, corruption, metadata bounds, zero-frame files, container preflight) come with it, as do the FLAC oracle tests. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> |
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Aleksander HelgakerandClaude Opus 5.5
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38740b43a9 |
craft-audio: core types, WAV/AIFF read and write, peaks, symphonia (#441)
* craft-audio: core types, WAV/AIFF read and write, peaks, symphonia The new crate without any codec module: what every later codec plugs into. - Shared types: `AudioBuffer`, `AudioInfo`, `ChannelLayout`/`Speaker`, `AudioError`, `EncodeOptions` (built with `new` and `with_*`), `EncodeFormat`, `BitDepth`, `BwfInfo`, `SampleFormat`. - WAV/BWF/RF64 and AIFF/AIFF-C reading and writing from SoundCraft's `soundcraft-audio-io`, kept over EffectCraft's and FilmCraft's smaller writers (only it has BWF, dither, any speaker mask and streaming without a known length). `encode` and `FileEncoder` behind `pcm-encode`; `lossless-encode` and `encode` turn it on. - `decode`, `probe`, `tracks` and the streaming, seekable `Decoder` (feature `decode-native`). The native path reads WAV, AIFF, and PCM in MP4/MOV and Matroska through craft-media's demuxers, packet by packet, with edit lists and `CodecDelay`/`DiscardPadding` applied. `TrackDecoder` decodes PCM; each codec PR adds its codec to it. - symphonia 0.6 (MPL-2.0, used unmodified) behind the `symphonia` feature, part of `decode`: the fallback for every format with no native decoder yet. Its size fields are checked before it reads, and its panics are caught where they unwind. - `peaks`: the min/max/RMS pyramid for waveform drawing. Workspace: craft-media is a workspace dependency without default features, so an encode-only craft-audio does not build the MP4 demuxer, and `symphonia-bundle-flac` builds without overflow checks in dev profiles (its 32-bit LPC restore wraps on valid input). CI: `audio.yml`, with clippy per feature, rustdoc, WASM, MSRV, and the tests with ffmpeg oracles. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * craft-audio: review fixes (CodeRabbit on #441) - Reading: AIFF that declares frames without an SSND chunk is Malformed; a Matroska PCM track's length counts from its first frame; seek targets saturate; an empty FLAC has one track; a failed header read is an Io error; deinterleave allocates fallibly; concealed silence is capped like a decoded packet, and symphonia's uses the damaged packet's own duration when known. - Peaks sum squares in f64, so loud float samples give a finite RMS. - Writing: .aifc maps to AIFF; BWF text is cut at a character boundary; a WAV byte rate past 32 bits is an error; the FileEncoder docs say nothing may follow the sink's position. - README: the Limits list says which codecs go through symphonia and which MP4 edit lists apply. - Tests for each, alias and dither coverage, and a FLAC case in the routing oracle. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * craft-audio: review fixes, second round (CodeRabbit on #441) - Every header read is checked: `Native::open`, `native::tracks` and the no-symphonia error read it with `head_checked`. craft-media demuxer errors that are I/O failures (not a file ending early) are returned as `AudioError::Io` instead of sending the file to the other path. - A Matroska track's presented length runs from its first frame plus `CodecDelay` (unclamped, in timestamp ticks with the delay rounded to one) to its end. - symphonia's concealment allocates fallibly, and a damaged first packet is concealed when the demuxer gives its duration. `deinterleave` reserves its channel list fallibly too. - `TrackDecoder::conceal` documents its cap at the decode limit. - Tests: an I/O failure past the first 64 KiB, codec delay with the first block at -20, 0 and 50 ms (checked with `probe`), a seek back after an overshoot, and the loud-sample RMS value. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * craft-audio: TrackDecoder::conceal returns a Result Concealment allocates fallibly and fails with TooLarge when the silence would pass the 2^30-sample decode limit, instead of returning less than the packet's length. The native path, the README, the packets example and the doc example pass the error on. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * craft-audio: review fixes, third round (CodeRabbit on #441) - Concealed silence is held to the decode limit inside pcm::silence, so symphonia's path checks frames × channels too, and symphonia's decoded audio is copied into fallibly allocated planes. - TrackDecoder::reset forgets the last packet's length. - Without symphonia, a damaged MP4, Matroska or Ogg file (truncated, invalid, no moov) is Malformed instead of Unsupported; a feature the demuxer does not read still names the build. - Tests: an MP4 with no moov in a native-only build, conceal after a reset and past the limit. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * craft-audio: review fixes, fourth round (CodeRabbit on #441) - A file the native demuxers find damaged is Malformed with symphonia too, when symphonia calls it unsupported; before, an ftyp-only MP4 was Unsupported in symphonia builds. - symphonia's decoded blocks are held to the 2^30-sample decode limit before allocating. - Symph::seek forgets the last packet's length, so a lost packet after a seek is not concealed with the length of one before it. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * craft-audio: review fixes, fifth round (CodeRabbit on #441) - The damaged-file test covers Matroska as well as MP4, and fails when either branch of native::damage is removed. - native::damage leaves out Ogg: ogg::open reports no damage (it skips damaged pages and calls a file with none it can read not Ogg), and symphonia reports damaged Ogg as malformed itself. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * craft-audio: run the ffmpeg oracle tests on ubuntu-26.04 ubuntu-latest's apt ships ffmpeg 6.1.1, which writes MP2 and E-AC-3 in Matroska with ends a few milliseconds early, so the routing tests on the later codec PRs cannot pass on it. ubuntu-26.04 ships ffmpeg 8.0. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> --------- Co-authored-by: Claude Opus 5.5 <noreply@anthropic.com> |