text-to-speech
Skill72 starsChanged 46 days agoApache-2.0View on GitHub
Write correct Sarvam Bulbul TTS code — bulbul:v4-flash persona speaker IDs (voice_language_style), REST, HTTP stream, WebSocket, pronunciation dictionaries, and v3/v4 traps (short names rejected on v4, param ranges). Use this skill when generating speech in an app with Python or JS/TS. For live TTS in chat via MCP, use sarvam-mcp instead.
- Reads credentials
21 other projects use a version of this.
Adopted by
Most recently taken up by
- OpenMontageInc/OpenMontagetheir own version, first seen Nov 8, 2026
- calesthio/OpenMontagetheir own version, first seen Nov 8, 2026
- dmccreary/ibook-skillstheir own version, first seen Nov 3, 2026
- MoizIbnYousaf/marketing-clitheir own version, first seen Nov 1, 2026
- eins78/agent-skillstheir own version, first seen Oct 30, 2026
- deepgram/skillstheir own version, first seen Oct 25, 2026
Used with
Projects with it also use voice-changer, speech-engine and 9 more.
Other files people use alongside this one.
Files found in the same public projects as a version of this one, more often than they turn up anywhere else. Each project counts once, large collections count for less, and this file's own repository is left out.
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Remove background noise and isolate vocals/speech from audio using ElevenLabs Voice Isolator (audio isolation) API. Use when cleaning up noisy recordings, removing music or background ambience from dialogue, isolating speech from field recordings, preparing audio for transcription, extracting vocals, or any "denoise / clean up / isolate voice" task.
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LinHao-citySkillsetup-api-key
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MoizIbnYousafSkillvisual-style
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- +5 more from LinHao-city
Other versions21 projects use a version of this.
Word for word · 0
None found.
Edited · 0
None found.
Wrote their own version · 21
- calesthio/OpenMontage★ 78ksame name
- OpenMontageInc/OpenMontage★ 78ksame name
- wlzh/skills★ 694same name
- elevenlabs/skills★ 539same name
- tadaspetra/loop★ 296same name
- dmccreary/claude-skills★ 110same name
- dmccreary/ibook-skills★ 110same name
- awesome-genmedia/skills★ 36same name
- MoizIbnYousaf/marketing-cli★ 33same name
- deepgram/skills★ 26same name
11 more not listed
How matches are found. Word for word: the exact same text. Edited: shares passages with this file (“99% same” means 99% of its five-word phrases also appear here; 10% or more counts). Their own version: similar meaning, or, weakest, the same name. We can’t tell which project had it first. Projects are counted among the 200 most-starred candidates, and mentions are links to the repository we found, up to 50 per site, so both counts are a floor.
In this repository7 other files it gives its agents.
- chatSkill
- doc-aiSkill
- dubbingSkill
- sarvam-mcpSkill
- speech-to-textSkill
- vibe-codingSkill
- voice-agentsSkill
The file
We split this skill into the rules it gives an agent
1 rule. 0 are common, 0 are distinctive, and 1 is very specific to this project.
0
Common
Most of the 18 edited versions we read keep these.
0
Distinctive
This author's own choices. Worth a look.
1
Very specific
Tied to their project's names, files or tools. Probably skip.
- Common
- Distinctive
- Very specific
- Very specific: line 44
Do not copy speaker="ishita" from the Voices “generate your own sample” snippet — that is a v3 short name and fails on v4-flash.
“Common” is measured: kept by at least 6 in 10 of the 18 edited versions of this file we could read. “Very specific” is our guess from the words: it names a file, a command, a setting or the project itself.
Files
Text-to-Speech — Bulbul
Live in-chat TTS → sarvam-mcp (
sarvam_tools_tts_*). This skill = SDK code.
[!IMPORTANT] Auth:
api-subscription-keyheader — NOTAuthorization: Bearer. Base URL:https://api.sarvam.ai(NOT/v1— that prefix is only for the OpenAI-compatible chat endpoint) SDK floor for the code below: Pythonsarvamai>=0.1.29, JSsarvamai@>=1.1.8. On older versions the TTS param istarget_language_code— see Gotchas.
Model
bulbul:v4-flash — latest, use this for new integrations. Same REST / HTTP stream / WebSocket contract as v3. 200+ persona voices (222 IDs).
bulbul:v3 — what the API uses when model is omitted. Short names (shubh, priya). Still the right model for Malayalam and Odia, which have no v4 personas yet.
Speakers (v4 Flash)
IDs are lowercase voice_language_style: aparna_hi_customer, simran_en_customer, shubh_enhi_ads. Omit speaker and the default is shubh_enhi_ads. v2/v3 short names (shubh, priya, ishita, anushka) return 400.
The middle token is a language tag (hi, en, bn, gu, kn, mr, pa, ta, te) or enhi for English–Hindi code-mix. Use an _enhi_ persona when one utterance is genuinely mixed; a few English loanwords in Hindi script do not need it.
language_code still takes the usual BCP-47 values (hi-IN, en-IN, … including ml-IN and od-IN). There are no ml or od persona IDs — for those languages set model="bulbul:v3" and a v3 short name.
| Use case | Hindi | English |
| Customer care | aparna_hi_customer | aparna_en_companion |
| Sales | aparna_hi_customer, shubh_hi_customer | simran_en_sales |
| Collections | simran_hi_recovery | shubh_en_recovery |
| Hinglish (one utterance) | simran_enhi_customer | — |
Do not copy speaker="ishita" from the Voices “generate your own sample” snippet — that is a v3 short name and fails on v4-flash.
Coverage by language (live-verified 2026-10-06; one working persona each — pick others from the live list):
| Language | Example persona | Language | Example persona |
en-IN | simran_en_customer, sunny_en_social | mr-IN | ishita_mr_conversational |
hi-IN | aparna_hi_customer, shubh_hi_customer | gu-IN | bhavik_gu_conversation |
hi-IN (Hinglish) | shubh_enhi_ads (the default) | kn-IN | chaitra_kn_conversation |
bn-IN | roopa_bn_conversational, arnab_bn_conversation | pa-IN | anand_pa_conversation |
ta-IN | gokul_ta_narration, aravind_ta_ads | te-IN | kavitha_te_conversation |
The roster changes. Sarvam renamed voices on 2026-10-06 (bappa_bn_conversation → arnab_bn_conversation, vetri_ta_ads/vetri_ta_suspense → aravind_ta_*), and the old IDs now 400. Do not hardcode IDs from memory or old blog posts. An unknown speaker returns 400 with the full live list, so the list is one bad request away. The API also lists 2 Assamese (_as_) personas that the docs catalog does not.
Quick Start (Python)
from sarvamai import SarvamAI
from sarvamai.play import save
client = SarvamAI()
response = client.text_to_speech.convert(
text="नमस्ते, आप कैसे हैं?",
language_code="hi-IN",
model="bulbul:v4-flash",
speaker="aparna_hi_customer",
enable_preprocessing=True,
)
save(response, "output.wav")
# HTTP Stream (lower latency, binary audio)
chunks = []
for chunk in client.text_to_speech.convert_stream(
text="Hello from Sarvam AI",
language_code="en-IN",
model="bulbul:v4-flash",
speaker="simran_en_customer",
output_audio_codec="wav",
):
chunks.append(chunk)
audio = b"".join(chunks)Quick Start (JavaScript/TypeScript)
import { SarvamAIClient } from "sarvamai";
import { writeFile } from "fs/promises";
const client = new SarvamAIClient({ apiSubscriptionKey: "YOUR_SARVAM_API_KEY" });
// REST
const response = await client.textToSpeech.convert({
text: "नमस्ते, आप कैसे हैं?",
language_code: "hi-IN",
model: "bulbul:v4-flash",
speaker: "aparna_hi_customer",
enable_preprocessing: true,
});
// HTTP Stream (lower latency, returns BinaryResponse)
const streamResponse = await client.textToSpeech.convertStream({
text: "Hello from Sarvam AI",
language_code: "en-IN",
model: "bulbul:v4-flash",
speaker: "simran_en_customer",
output_audio_codec: "wav",
});
const bytes = await streamResponse.bytes();
await writeFile("output.wav", bytes);WebSocket Streaming
import asyncio
from sarvamai import AsyncSarvamAI
async def tts_stream():
client = AsyncSarvamAI()
async with client.text_to_speech_streaming.connect(model="bulbul:v4-flash") as ws:
await ws.configure(
target_language_code="hi-IN",
speaker="aparna_hi_customer",
output_audio_codec="wav",
)
await ws.convert("Your text here")
await ws.flush()
async for message in ws:
pass # base64 audio chunks
asyncio.run(tts_stream())Character Limits
| Method | Max Text |
REST (convert) | 2,500 chars |
HTTP Stream (convert_stream) | 3,500 chars |
Legacy inputs array (raw REST) | 500 chars per item — send text instead to get the 2,500 limit |
| WebSocket | 2,500 chars/msg (keep <500 for lowest latency; send many messages per connection) |
Gotchas
| Gotcha | Detail |
Set model or you get v3 | Omitting model selects bulbul:v3. v3 short names then work; v4 persona IDs do not. New code sets model="bulbul:v4-flash". |
| v3/v2 short names on v4 | shubh, priya, ishita, anushka, etc. are rejected. Use voice_language_style. Unknown IDs 400 with the live speaker list. |
language_code is version-gated | Python >=0.1.29 and JS >=1.1.8 (both 2026-08-03) take language_code; earlier versions take target_language_code. Hard rename — no alias, no deprecation shim, and the JSON body key changed too, so the wrong name raises TypeError before any request goes out. Check with pip show sarvamai / npm ls sarvamai if you hit that. |
| WebSocket was not renamed | Python ws.configure() still takes target_language_code (mapped to language_code on the wire). Docs that pass language_code= into ws.configure() raise TypeError. JS configureConnection() takes language_code. |
| JS method name | client.textToSpeech.convert({...}) and .convertStream({...}) — camelCase. Stream returns BinaryResponse with .stream(), .bytes(), .blob(). |
pitch / loudness ranges | pitch −0.5–0.5 (default 0), loudness 0.1–2.5 (default 1), pace 0.5–2.0. Out-of-range values return 400. Bulbul v3 used to reject pitch / loudness; as of 2026-10-06 it accepts them too (live-verified). v2-only ranges in the model-page limits table do not apply. |
| Streaming codec default | HTTP stream and WebSocket do not default to WAV. Set output_audio_codec (wav or mp3; also flac / aac on HTTP stream). opus can appear in the schema and currently errors on /text-to-speech. |
| Sample rate | speech_sample_rate 8000, 16000, 24000 and 48000 verified on v4-flash REST (the returned WAV matches the requested rate). |
| Sample rate >24kHz | 32kHz, 44.1kHz, 48kHz only via REST, not streaming. Omitted → 22050 Hz. |
| REST response | Base64-encoded audio in response.audios[0]. Use sarvamai.play.save() or base64.b64decode(). Writing the raw base64 string to disk is not playable audio. |
| Speaker must match model | The mismatch fails both ways: a v3 short name on bulbul:v4-flash and a v4 persona on bulbul:v3 both return 400. Pair model and speaker explicitly. |
| No SSML | SSML markup is NOT supported. Use pace (and on v4, pitch / loudness) plus sentence-boundary chunking. |
| Use native script | Romanized Indic input ("Aapka order confirm ho gaya hai") degrades quality. Write Indic words in native script; keep English loanwords in Latin ("आपका order confirm हो गया है"). With enable_preprocessing=true, do not hand-expand numbers, dates, or currency. |
ml-IN / od-IN have no v4 personas | Codes are accepted. The catalog has no Malayalam or Odia IDs. Use bulbul:v3 for those languages. |
| Pronunciation dictionary | Pass dict_id on v4-flash the same way as v3 (10 dicts/user, 100 words/dict). Create via Python client.pronunciation_dictionary.create(file=f). JS SDK upload is broken (missing multipart Content-Type) — use raw fetch + FormData with an explicit Blob type. The pronunciation-page limits row that says “v3 only” is stale; the examples on that page send dict_id with bulbul:v4-flash. |
Full Docs
Fetch the voice catalog, ranked picks, streaming protocol, and codec options from:
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