250 hours of Punjabi Telephony Speech

Speech captured over a real telephony path — a genuine call through the network, not studio audio downsampled afterwards to imitate one.

Scenario-driven calls between a caller and an agent or IVR flow, each leg recorded separately, across a deliberate spread of handsets and network conditions.

Yield at this style: 50–60% of recorded time is accepted. Dropped calls, network artefacts beyond tolerance and unusable legs are discarded. Real network conditions are the point of the style and also its main cost.

• Call scenarios drawn from real contact-centre intents: balance enquiry, complaint, booking change, escalation
• IVR flows scripted with deliberate mis-entry and barge-in paths, since those are where deployed systems fail
• Handset spread specified up front — low-end Android, feature phone, landline — because handset variance is a real acoustic axis
• Background conditions varied on purpose: street, vehicle, indoor, since real callers are rarely in quiet rooms
• Built against Punjabi specifically: Punjabi is tonal: high, low and level tones distinguish words, and tone is not marked in Gurmukhi orthography
• Code-mixing handled explicitly rather than edited out — Punjabi speech mixes Hindi and English freely, with strong diaspora influence in urban registers.

Six to eight weeks. Beyond three cities the bottleneck moves from recruitment to transcription throughput, which is why the transcriber count rises faster than the engineer count.

Four to five batches on a fortnightly cadence, sequenced so the dialects you care about most land first.

Cover all three historic regions (Majha, Malwa, Doaba); tone realisation differs measurably between them.

At 250 hours quotas are enforced per dialect and reconciled fortnightly. Aggregate demographics are reported per batch so drift is visible while there is still time to correct it.

Recruitment must be stratified by handset and carrier as well as by dialect, which adds a screening axis the other styles do not have.

• Punjabi is tonal: high, low and level tones distinguish words, and tone is not marked in Gurmukhi orthography
• Punjabi speech mixes Hindi and English freely, with strong diaspora influence in urban registers.
• Tonal variation is essentially unmodelled in public Punjabi data, and Malwai/Doabi rural speech is scarce.

• Codec and sample rate verified per file, rejecting any studio audio that has been downsampled to fake a telephony path
• Echo and double-talk checked on both legs
• DTMF events verified against the call log
• Level normalisation applied per leg, since handset output levels vary far more than studio microphones do
• Tone is unrepresented in text, so pronunciation lexicons must be built from audio, not from spelling
• Shahmukhi vs Gurmukhi script decisions must be fixed per project

100% technical QA, 15% content QA with stratified sampling across city, dialect and transcriber, so no single team's output goes unchecked.

• Studio audio downsampled to 8 kHz and passed off as telephony, which has none of the codec or packet-loss characteristics that matter
• Echo and double-talk that make the agent leg unusable
• Carrier and handset monoculture, producing a corpus that only represents one acoustic path
• Over-clean recordings from participants who move somewhere quiet to take the call, defeating the purpose

Cost at this band is driven by: Transcription depth — verbatim with disfluencies costs materially more than clean-read transcription; Number of distinct dialect quotas rather than raw hours.

• Transcription becomes the critical path and stays there for the rest of the volume bands
• Coordinator turnover mid-build causes quota drift that is only visible in the final demographic report unless it is audited weekly
• Punjabi carries 5 recognised varieties across Punjab, Haryana, Delhi, so the quota matrix is wider than the headline volume suggests
• Tonal variation is essentially unmodelled in public Punjabi data, and Malwai/Doabi rural speech is scarce.

• WAV audio to your naming convention, with the per-file manifest
• Verbatim Gurmukhi transcripts with utterance-level timestamps
• Call metadata: duration, codec, handset class, carrier, packet-loss events
• Intent label per call and per turn
• DTMF and hold, transfer and barge-in events
• Agent and caller leg identifiers
• Per-speaker metadata: age band, gender, region, dialect, education band
• Consent records mapped to speaker IDs
• QA report with pass rates, rejection reasons and agreement statistics
• Speaker-disjoint train / dev / test splits on request

• Contact-centre and IVR ASR
• Voice bots operating over the phone network
• Intent classification on narrowband audio
• Robustness to codec and packet loss

Narrowband telephony audio is the wrong input for TTS or any wideband model — the frequency content simply is not there. Use it for models that will be deployed on a phone line and nothing else.