Draft card (test upload)

README.md

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+---
+license: cc-by-4.0
+pretty_name: "PACC-T: Parallel Acoustic Confound Corpus, Telecoms"
+language:
+  - en
+size_categories:
+  - 100K<n<1M
+task_categories:
+  - audio-classification
+tags:
+  - speech
+  - codec
+  - telephony
+  - bona-fide
+  - robustness
+  - anti-spoofing
+  - parallel-corpus
+---
+
+# PACC-T: Parallel Acoustic Confound Corpus, Telecoms
+
+PACC-T is 5,992 bona fide speech clips passed through 34 speech and audio codecs,
+12 tandem codec chains and 5 resample-only controls. Every condition contains
+the same 5,992 clips, so any clip can be compared with itself across all 51 conditions.
+Every clip has a per-clip record of the exact commands that produced it.
+
+It contains no synthetic or spoofed speech. It is a reference for what telecom channels do to real
+speech: a bona fide baseline, a robustness benchmark and a condition-shift test set.
+
+| | |
+|---|---|
+| Conditions | 51 (34 codecs, 12 tandems, 5 controls) |
+| Clips per condition | 5,992 (same clips in every condition) |
+| Files | 305,592 FLAC, plus 5,992 sources in `pacc_base.tar` |
+| Duration | about 7.2 h per condition, 366 h in total |
+| Size | 20.7 GB of audio; one condition 0.18-0.75 GB; sources 0.89 GB |
+| Download | per condition, so any subset can be fetched on its own |
+
+Companion dataset: **PACC-P** (Presentation), the same 5,992 sources under 50 noise, reverberation,
+filtering and pitch/tempo conditions. (DOI pending)
+
+## Before you measure
+
+### Sample rate and bandwidth are confounds, not side effects
+Every narrowband codec decodes at 8 kHz, and many wideband codecs at 16 kHz.
+Any feature measured on a codec condition mixes what the codec did with what
+the lower sample rate and missing bandwidth did. Attribute an effect to a codec
+only after comparing against the resample control at the same output rate.
+
+### Resample controls
+resample_ctrl_8k / 16k / 22k / 44k / 48k hold the same 5,992 sources,
+resampled with no codec. Each codec condition is paired with the control at its
+output rate (see the condition index). An effect that appears in both the codec
+and its control is a rate effect.
+
+### Output sample rate is not content bandwidth
+A file's sample rate is an upper bound on its content, not a measure of it. `evs_swb_48k_*` is
+stored at 48 kHz but, as super-wideband EVS, carries content only up to about 16 kHz. Tandem chains
+are bounded by their narrowest hop. Estimate content bandwidth from the audio if a feature depends
+on it.
+
+### Native bandwidth differs by pool
+AMI sources are native 16 kHz: content stops at 8 kHz. In resample_ctrl_22k /
+44k / 48k and in codecs decoded above 16 kHz, AMI clips are upsampled and carry
+no content above 8 kHz. VCTK sources are 48 kHz. Compare across rates within a
+pool, or account for this when pooling.
+
+### Formant measurement on band-limited conditions
+Narrowband conditions (8 kHz output, or any condition whose content stops near
+3.4-4 kHz) are a known failure case for LPC formant trackers configured for
+wideband speech. With a ceiling above the available content (Praat's default is
+5500 Hz), the tracker fits spurious poles in the empty band. These "ghost
+formants" pull F2 and F3 low, often by hundreds of Hz, with no error flag. Set
+the formant ceiling from content bandwidth or speaker, check readings against
+the matching resample control, or treat F2 and above as unreliable in these
+conditions.
+
+Tool defaults matter. For example, eGeMAPSv02 derives formants after resampling
+to 11 kHz with an 11th-order autocorrelation LPC, and some voice-quality
+measures (such as glottal-flow quotients) change with sample rate on identical
+content. Validate any feature on the resample controls before attributing an
+effect to a codec.
+
+## Sources
+
+| Pool | Clips | Native format | Source corpus |
+|---|---|---|---|
+| AMI | 2,992 (4.45 h) | 16 kHz WAV, headset microphone | AMI Meeting Corpus: 139 meetings, 157 participants |
+| VCTK_mic1 | 1,500 (1.36 h) | 48 kHz FLAC | CSTR VCTK Corpus 0.92, mic1: 109 speakers |
+| VCTK_mic2 | 1,500 (1.36 h) | 48 kHz FLAC | CSTR VCTK Corpus 0.92, mic2: 108 speakers |
+
+VCTK mic pairing: 1,485 utterances appear in both mic pools; 15 in each pool are unpaired. Joining
+mic1 to mic2 on utterance ID gives 1,485 pairs, not 1,500. All sources are mono 16-bit PCM.
+AMI file IDs encode meeting, channel, segment index and start/end time in seconds.
+
+Per-source metadata ships as `sources/metadata.csv` in `pacc_base.tar`: speaker, gender and where
+the label came from, VCTK age and utterance, AMI meeting, channel, participant and segment times.
+
+### Gender balance
+
+| Pool | Female | Male | Label source |
+|---|---|---|---|
+| VCTK_mic1 | 750 | 750 | VCTK speaker metadata |
+| VCTK_mic2 | 750 | 750 | VCTK speaker metadata |
+| AMI | 1,126 (38%) | 1,866 (62%) | AMI participant metadata |
+
+Sources were selected for gender balance. VCTK is balanced exactly. AMI is not: its clips were
+selected using gender inferred from the audio (pitch and apparent vocal-tract length), and those
+labels proved wrong for 17% of clips, mostly men labelled as women (441 clips, against 68 the other
+way). Checked against AMI's own participant metadata, the AMI pool is 62% male. The `gender`
+column gives the metadata label; the inferred label is kept, for transparency only, as
+`gender_inferred_not_recommended`. For TS meetings, which have no entry in AMI's
+`participants.xml`, sex is taken from the M/F prefix of the participant ID, which agrees with
+`participants.xml` for all 189 participants listed there.
+
+8 AMI clips with no usable speech are excluded from all of PACC (`excluded_sources.csv`). Earlier
+internal builds also held RAVDESS and CREMA-D clips; they were removed so that PACC is CC BY only.
+
+## Conditions
+
+Output sample rate is the codec's native decode rate. Compare each condition with the resample
+control at the same rate (last column) before attributing an effect to the codec.
+
+Framing: **block** = frame-based codec; **sample** = sample-by-sample waveform coder with no frame
+structure. Tools that look for frame boundaries have nothing to find in sample-based conditions.
+
+### Codecs
+
+| Condition | Family | Codec | Bitrate | Framing | Output rate | Rate control |
+|---|---|---|---|---|---|---|
+| `aac_32k` | media | AAC-LC (ffmpeg native) | 32 kbps | block | 22.05 kHz | `resample_ctrl_22k` |
+| `aac_64k` | media | AAC-LC (ffmpeg native) | 64 kbps | block | 44.1 kHz | `resample_ctrl_44k` |
+| `amr_nb_122` | mobile | AMR-NB | 12.2 kbps | block | 8 kHz | `resample_ctrl_8k` |
+| `amr_nb_475` | mobile | AMR-NB | 4.75 kbps | block | 8 kHz | `resample_ctrl_8k` |
+| `amr_wb` | mobile | AMR-WB | 23.85 kbps | block | 16 kHz | `resample_ctrl_16k` |
+| `codec2_1300` | low-rate | Codec 2 (1300 mode, see EDGE_CASES) | 1.3 kbps | block | 8 kHz | `resample_ctrl_8k` |
+| `codec2_700` | low-rate | Codec 2 700C | 0.7 kbps | block | 8 kHz | `resample_ctrl_8k` |
+| `evs_24400_dtxadapt` | mobile (VoLTE) | EVS, DTX adaptive CNG | 24.4 kbps | block | 16 kHz | `resample_ctrl_16k` |
+| `evs_24400_dtxfixed` | mobile (VoLTE) | EVS, DTX fixed 8-frame CNG | 24.4 kbps | block | 16 kHz | `resample_ctrl_16k` |
+| `evs_24400_nodtx` | mobile (VoLTE) | EVS, DTX off | 24.4 kbps | block | 16 kHz | `resample_ctrl_16k` |
+| `evs_9600_dtxadapt` | mobile (VoLTE) | EVS, DTX adaptive CNG | 9.6 kbps | block | 16 kHz | `resample_ctrl_16k` |
+| `evs_9600_dtxfixed` | mobile (VoLTE) | EVS, DTX fixed 8-frame CNG | 9.6 kbps | block | 16 kHz | `resample_ctrl_16k` |
+| `evs_9600_nodtx` | mobile (VoLTE) | EVS, DTX off | 9.6 kbps | block | 16 kHz | `resample_ctrl_16k` |
+| `evs_swb_48k_dtxadapt` | mobile (VoLTE) | EVS super-wideband, DTX adaptive CNG | 24.4 kbps | block | 48 kHz | `resample_ctrl_48k` |
+| `evs_swb_48k_dtxfixed` | mobile (VoLTE) | EVS super-wideband, DTX fixed 8-frame CNG | 24.4 kbps | block | 48 kHz | `resample_ctrl_48k` |
+| `evs_swb_48k_nodtx` | mobile (VoLTE) | EVS super-wideband, DTX off | 24.4 kbps | block | 48 kHz | `resample_ctrl_48k` |
+| `g711_alaw` | PSTN | G.711 A-law | 64 kbps | sample | 8 kHz | `resample_ctrl_8k` |
+| `g711_ulaw` | PSTN | G.711 mu-law | 64 kbps | sample | 8 kHz | `resample_ctrl_8k` |
+| `g722` | PSTN wideband | G.722 (sub-band ADPCM) | 64 kbps | sample | 16 kHz | `resample_ctrl_16k` |
+| `g726_16k` | PSTN | G.726 ADPCM | 16 kbps | sample | 8 kHz | `resample_ctrl_8k` |
+| `g726_24k` | PSTN | G.726 ADPCM | 24 kbps | sample | 8 kHz | `resample_ctrl_8k` |
+| `g726_32k` | PSTN | G.726 ADPCM | 32 kbps | sample | 8 kHz | `resample_ctrl_8k` |
+| `gsm` | mobile | GSM 06.10 full rate | 13 kbps | block | 8 kHz | `resample_ctrl_8k` |
+| `ilbc` | VoIP | iLBC | encoder default | block | 8 kHz | `resample_ctrl_8k` |
+| `lc3` | Bluetooth LE Audio | LC3 | encoder default | block | 16 kHz | `resample_ctrl_16k` |
+| `mp3_128k` | media | MP3 (LAME) | 128 kbps | block | 44.1 kHz | `resample_ctrl_44k` |
+| `mp3_32k` | media | MP3 (LAME) | 32 kbps | block | 22.05 kHz | `resample_ctrl_22k` |
+| `opus_16k_auto` | VoIP | Opus, mode chosen by encoder | 16 kbps | block | 48 kHz | `resample_ctrl_48k` |
+| `opus_16k_celt` | VoIP | Opus, CELT forced (lowdelay) | 16 kbps | block | 48 kHz | `resample_ctrl_48k` |
+| `opus_32k_auto` | VoIP | Opus, mode chosen by encoder | 32 kbps | block | 48 kHz | `resample_ctrl_48k` |
+| `opus_32k_celt` | VoIP | Opus, CELT forced (lowdelay) | 32 kbps | block | 48 kHz | `resample_ctrl_48k` |
+| `opus_6k_auto` | VoIP | Opus, mode chosen by encoder | 6 kbps | block | 48 kHz | `resample_ctrl_48k` |
+| `opus_6k_celt` | VoIP | Opus, CELT forced (lowdelay) | 6 kbps | block | 48 kHz | `resample_ctrl_48k` |
+| `speex_8k` | VoIP | Speex narrowband | encoder default | block | 8 kHz | `resample_ctrl_8k` |
+
+### Tandem chains (two codecs in sequence)
+
+The first codec's output is the second codec's input. Effective bandwidth is bounded by the
+narrowest hop, and the output's frame structure and quantisation reflect the last codec: in
+`amr_wb_to_g711_ulaw` the output carries G.711's sample-by-sample companding lattice at 8 kHz,
+while the speech had already been through AMR-WB's frame-based coding.
+
+| Condition | Chain | Framing | Output rate | Rate control |
+|---|---|---|---|---|
+| `amr_wb_to_g711_ulaw` | AMR-WB 23.85 kbps then G.711 mu-law 64 kbps | block then sample | 8 kHz | `resample_ctrl_8k` |
+| `evs_24400_dtxfixed_to_amr_nb_475` | EVS, DTX fixed 8-frame CNG 24.4 kbps then AMR-NB 4.75 kbps | block then block | 8 kHz | `resample_ctrl_8k` |
+| `evs_24400_dtxfixed_to_amr_wb` | EVS, DTX fixed 8-frame CNG 24.4 kbps then AMR-WB 23.85 kbps | block then block | 16 kHz | `resample_ctrl_16k` |
+| `evs_24400_dtxfixed_to_g711_ulaw` | EVS, DTX fixed 8-frame CNG 24.4 kbps then G.711 mu-law 64 kbps | block then sample | 8 kHz | `resample_ctrl_8k` |
+| `evs_24400_nodtx_to_amr_nb_475` | EVS, DTX off 24.4 kbps then AMR-NB 4.75 kbps | block then block | 8 kHz | `resample_ctrl_8k` |
+| `evs_24400_nodtx_to_amr_wb` | EVS, DTX off 24.4 kbps then AMR-WB 23.85 kbps | block then block | 16 kHz | `resample_ctrl_16k` |
+| `evs_24400_nodtx_to_g711_ulaw` | EVS, DTX off 24.4 kbps then G.711 mu-law 64 kbps | block then sample | 8 kHz | `resample_ctrl_8k` |
+| `opus_32k_auto_to_amr_wb` | Opus, mode chosen by encoder 32 kbps then AMR-WB 23.85 kbps | block then block | 16 kHz | `resample_ctrl_16k` |
+| `opus_32k_auto_to_evs_24400_dtxadapt` | Opus, mode chosen by encoder 32 kbps then EVS, DTX adaptive CNG 24.4 kbps | block then block | 16 kHz | `resample_ctrl_16k` |
+| `opus_32k_auto_to_g711_ulaw` | Opus, mode chosen by encoder 32 kbps then G.711 mu-law 64 kbps | block then sample | 8 kHz | `resample_ctrl_8k` |
+| `opus_32k_celt_to_amr_wb` | Opus, CELT forced (lowdelay) 32 kbps then AMR-WB 23.85 kbps | block then block | 16 kHz | `resample_ctrl_16k` |
+| `opus_32k_celt_to_evs_24400_nodtx` | Opus, CELT forced (lowdelay) 32 kbps then EVS, DTX off 24.4 kbps | block then block | 16 kHz | `resample_ctrl_16k` |
+
+### Resample controls (no codec)
+
+Each control is the source resampled with ffmpeg 8.1's default resampler (swresample, via `-ar`)
+and written as 16-bit FLAC. The codec conditions use the same resampler for their own rate
+conversion, so a control differs from its codec condition only by the codec.
+`resample_ctrl_48k` is sample-identical to the VCTK sources (a no-op and determinism check); for
+AMI it is an upsample from 16 kHz.
+
+| Condition | Output rate |
+|---|---|
+| `resample_ctrl_8k` | 8 kHz |
+| `resample_ctrl_16k` | 16 kHz |
+| `resample_ctrl_22k` | 22.05 kHz |
+| `resample_ctrl_44k` | 44.1 kHz |
+| `resample_ctrl_48k` | 48 kHz |
+
+## Files
+
+| Tarball | Contents |
+|---|---|
+| `pacc_base.tar` | the 5,992 source clips (`sources/<pool>/`), identical in PACC-T and PACC-P |
+| `pacc-t_<condition>.tar` | one condition: `pacc-t/<condition>/<pool>/<file_id>.flac`, `params.csv`, `README.txt` |
+| `pacc-t_sample.tar` | 14 fixed sources (5 VCTK utterances on both mics, 4 AMI clips) through every condition |
+
+All audio is 16-bit mono FLAC. Extracting any set of tarballs builds one tree. `SHA256SUMS` and
+`TARBALL_INDEX.csv` list every tarball; `pacc-t_manifest.csv` lists every file with its sha256.
+Tarballs are deterministic: the same inputs rebuild byte-identical files.
+
+## Per-clip records and verification
+
+Each condition's `params.csv` has one row per clip: input file and sha256, output sha256, sample
+rate, frame count, and the full command sequence, with paths as placeholders.
+
+- **Logged** (6 conditions: the EVS DTX-fixed conditions and their tandems): written at encode time,
+  including the EVS encoder's own DTX status line.
+- **Recovered** (45 conditions, encoded before per-clip logging existed): each recorded command was
+  re-executed on 60 clips per condition (20 per pool) and reproduced the stored audio bit for bit,
+  2,700 of 2,700. Rows were then filled from the files.
+
+The column `params_origin` says which applies. Before packaging, contract tests confirmed: every
+condition holds exactly the 5,992 base sources once each; no excluded source ships; manifest,
+params and file hashes agree.
+
+## Tools and versions
+
+- ffmpeg 8.1 (full_build, www.gyan.dev, Windows static), with libopencore-amrnb, libvo-amrwbenc,
+  libgsm, libilbc, libspeex, libopus, libmp3lame (LAME 3.100), liblc3 and libcodec2. The static
+  build does not expose the other libraries' version strings.
+- EVS: 3GPP TS 26.443 floating-point reference C code, banner "Version 12.7.0 / 13.3.0",
+  mirror github.com/wanglihe/3gpp-evs at commit 519236cc07ca209cb3aa2cc32de6ca686269839b.
+- Decoders: only `ilbc` pins its decoder (`-c:a libilbc`); all others use ffmpeg's default decoder
+  for the stream, as recorded in the params. A start-burst scan of every file (first 40 ms peak
+  against the rest of the clip) flagged none.
+
+## Known edge cases
+
+See `EDGE_CASES.txt`. In short: EVS DTX variants coincide on clips without pauses; LC3 passes two
+very quiet clips through unchanged; `codec2_1300` runs at 1300 bps despite a 700 bps argument;
+350 AMI sources carry clipping from the original recordings.
+
+## Motivation
+
+The design of this corpus was motivated by Delgado et al. (ICASSP 2026), who argue that deepfake
+detection must account for how audio is presented through real communication channels, and by Lee
+et al. (ICASSP 2026), whose noise-aware multi-LoRA framework highlights real-world conditions.
+Neither group was involved in producing this dataset.
+
+- H. Delgado, G. Ramondetti, E. Dalmasso, G. Karvitsky, D. Colibro, H. Talib. "On Deepfake Voice
+  Detection - It's All in the Presentation." ICASSP 2026. arXiv:2509.26471.
+- W. Lee, H. Dinh-Xuan, T.-P. Doan, S. Jung. "Dynamic Noise-Aware Multi LoRA Framework Towards
+  Real-World Audio Deepfake Detection." ICASSP 2026.
+
+## Licence and credits
+
+PACC-T is released under **CC BY 4.0** by Moonscape Software. It is derived from:
+
+- **CSTR VCTK Corpus 0.92.** J. Yamagishi, C. Veaux, K. MacDonald. University of Edinburgh, CSTR,
+  2019. doi:10.7488/ds/2645. CC BY 4.0.
+- **AMI Meeting Corpus.** University of Edinburgh et al. https://groups.inf.ed.ac.uk/ami/corpus/.
+  CC BY 4.0. J. Carletta (2006), "Announcing the AMI Meeting Corpus", ELRA Newsletter 11(1).
+
+Please credit these sources alongside PACC-T.
+
+## Citation
+
+```bibtex
+@dataset{moonscape_pacc_t_2026,
+  author    = {{Moonscape Software}},
+  title     = {{PACC-T: Parallel Acoustic Confound Corpus, Telecoms}},
+  year      = {2026},
+  version   = {1.0},
+  publisher = {Zenodo},
+  doi       = {pending}
+}
+```