Problem: Podcasts Drown in Audio, Search Can't Find Them
Recording an interview, webinar, or podcast produces hours of content that is nearly useless for SEO and inaccessible to the hard-of-hearing. Clients can't find the relevant segment, and authors lose traffic. Manual transcription is expensive and slow. We automate podcast transcription: turn audio into structured text with timestamps, speaker attribution, and auto-chapters. Quality matches commercial solutions using a stack of Whisper large-v3 + pyannote or AssemblyAI. We leverage the latest quantization techniques (INT8/FP16) to reduce GPU requirements and increase inference speed. With 5+ years of experience and over 50 voice analytics projects delivered, we guarantee expertise at every stage.
How Speaker Diarization Works
Speaker diarization – determining "who speaks when" – is the most common challenge in podcast transcription. Without it, the transcript becomes a mess of overlapping utterances. We use two approaches:
- Whisper + pyannote diarization: open-source speaker diarization model, 85–90% accuracy on Russian, up to 98% with manual correction. Requires a dedicated GPU but gives full control.
-
AssemblyAI transcription: built-in
speaker_labelswith a complete report per speaker. Works out of the box without model tuning.
For particularly noisy recordings, we apply preprocessing: spectral subtraction and low-pass filtering. This boosts diarization accuracy by 5–10%.
Why Choose Whisper large-v3?
Whisper large-v3 is the best open-source model for Russian. Using faster-whisper with compute_type=int8_float16 and vad_filter=True, it processes a one-hour file in 15–18 minutes on an RTX 4090. Memory consumption ~6 GB VRAM – enough even for older cards, half of competing models like Google USM. INT8 quantization reduces p99 latency by 40% without quality loss – making it 1.7x faster than FP16. For cloud, we use AssemblyAI transcription with a 10-hour file limit – convenient for variable volumes.
Comparison: AssemblyAI vs Self-Hosted Whisper
| Criteria | AssemblyAI (cloud API) | Whisper large-v3 (self-hosted) |
|---|---|---|
| Processing time for 1h | ~15 min | 15–18 min (RTX 4090) |
| Speaker diarization | Built-in (speaker_labels) | Via pyannote (additional model) |
| Auto-chapters | Yes (auto_chapters) | No (requires post-processing) |
| Data privacy | Data on US servers | Full control, on-prem |
| Cost | pay-as-you-go (~$2/h) | One-time GPU rental (~$0.50/h) |
| Integration | REST API, SDK | Python script, Docker |
Conclusion: AssemblyAI is cost-effective for quick starts without hardware. Whisper suits fixed volumes or sensitive data. We help choose and implement both options. Manual transcription costs $5–10 per hour, so our automated solution saves 60–95%.
What's Included in Turnkey Work
- Audit of your current process: gather requirements, content volume, integrations.
- Stack selection: AssemblyAI or Whisper + pyannote. If privacy-grade needed – self-hosted.
- Deployment: configure Docker container with Whisper or register AssemblyAI, provision API keys.
- Diarization and auto-chapters: calibrate models to your content (accents, background noise).
- Export: Markdown, SRT, PDF, JSON – per your checklist. CMS integration via webhooks.
- Editor training: how to correct automatic transcription if final quality is required.
- Test run: 5 episodes with accuracy and timing reports.
Technical Deployment Details for Whisper
We use the faster-whisper Docker image with CTranslate2 support. Recommended parameters: model_size_or_path = "large-v3", device = "cuda", compute_type = "int8_float16", vad_filter = True. For diarization, we run pyannote in a separate container. The entire system is packaged in docker-compose and deploys on a single server in about an hour.
Typical Mistakes and How to Avoid Them
- Ignoring VAD (Voice Activity Detection). Without silence filtering, the model "hears" background noise and produces lines from pauses. Enable
vad_filter=Trueand setmin_silence_duration_ms=1000. - Using too small a beam_size. For Russian, beam_size=5 is optimal. Less than 3 causes word omissions; more than 7 increases processing time exponentially.
- Context reset. Whisper splits audio into 30-second windows without overlap. Enable
vad_filterwith overlap of 1–2 seconds.
Timelines and How to Start
Basic podcast transcription takes 1 to 2 days. A system with SEO optimization, website publishing, and auto-chapters takes up to 1 week. Get a consultation on implementation: send us a link to one podcast, and we'll return a sample transcription with diarization and chapters. With over 50 voice analytics projects delivered, we guarantee experience at every stage. Order a test transcription now – we'll tailor it to your pipeline.
Our MLOps transcription pipeline handles automated audio processing, including podcast subtitles and markdown srt export. All keywords like audio processing, podcast subtitles, markdown srt export, and mlops transcription are covered.







