[Test] Add e2e tests for Qwen3-TTS speech endpoint (#1206)

Signed-off-by: linyueqian <linyueqian@outlook.com>
Signed-off-by: Yueqian Lin <70319226+linyueqian@users.noreply.github.com>
This commit is contained in:
Yueqian Lin authored and GitHub committed 2026-02-10 16:33:39 +08:00
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@@ -228,6 +228,26 @@ steps:
volumes:
- "/fsx/hf_cache:/fsx/hf_cache"
- label: "Qwen3-TTS E2E Test"
timeout_in_minutes: 10
depends_on: image-build
commands:
- export VLLM_LOGGING_LEVEL=DEBUG
- export VLLM_WORKER_MULTIPROC_METHOD=spawn
- pytest -s -v tests/e2e/online_serving/test_qwen3_tts.py
agents:
queue: "gpu_4_queue"
plugins:
- docker#v5.2.0:
image: public.ecr.aws/q9t5s3a7/vllm-ci-test-repo:$BUILDKITE_COMMIT
always-pull: true
propagate-environment: true
shm-size: "8gb"
environment:
- "HF_HOME=/fsx/hf_cache"
volumes:
- "/fsx/hf_cache:/fsx/hf_cache"
# - label: "Omni Model Test with H100"
# timeout_in_minutes: 30
# depends_on: image-build
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@@ -8,7 +8,7 @@ COPY . .
# Install system dependencies
RUN apt-get update && \
apt-get install -y ffmpeg && \
apt-get install -y ffmpeg sox libsox-fmt-all && \
apt-get clean && \
rm -rf /var/lib/apt/lists/*
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@@ -8,6 +8,7 @@ nav:
- OpenAI-Compatible API:
- Image Generation: serving/image_generation_api.md
- Image Edit: serving/image_edit_api.md
- Text to Speech: serving/speech_api.md
- Examples:
- examples/README.md
- Offline Inference:
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@@ -0,0 +1,267 @@
# Speech API
vLLM-Omni provides an OpenAI-compatible API for text-to-speech (TTS) generation using Qwen3-TTS models.
Each server instance runs a single model (specified at startup via `vllm serve <model> --omni`).
## Quick Start
### Start the Server
```bash
# CustomVoice model (predefined speakers)
vllm serve Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice \
--stage-configs-path vllm_omni/model_executor/stage_configs/qwen3_tts.yaml \
--omni \
--port 8091 \
--trust-remote-code \
--enforce-eager
```
### Generate Speech
**Using curl:**
```bash
curl -X POST http://localhost:8091/v1/audio/speech \
-H "Content-Type: application/json" \
-d '{
"input": "Hello, how are you?",
"voice": "vivian",
"language": "English"
}' --output output.wav
```
**Using Python:**
```python
import httpx
response = httpx.post(
"http://localhost:8091/v1/audio/speech",
json={
"input": "Hello, how are you?",
"voice": "vivian",
"language": "English",
},
timeout=300.0,
)
with open("output.wav", "wb") as f:
f.write(response.content)
```
**Using OpenAI SDK:**
```python
from openai import OpenAI
client = OpenAI(base_url="http://localhost:8091/v1", api_key="none")
response = client.audio.speech.create(
model="Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice",
voice="vivian",
input="Hello, how are you?",
)
response.stream_to_file("output.wav")
```
## API Reference
### Endpoint
```
POST /v1/audio/speech
Content-Type: application/json
```
### Request Parameters
#### OpenAI Standard Parameters
| Parameter | Type | Default | Description |
|-----------|------|---------|-------------|
| `input` | string | **required** | The text to synthesize into speech |
| `model` | string | server's model | Model to use (optional, should match server if specified) |
| `voice` | string | "vivian" | Speaker name (e.g., vivian, ryan, aiden) |
| `response_format` | string | "wav" | Audio format: wav, mp3, flac, pcm, aac, opus |
| `speed` | float | 1.0 | Playback speed (0.25-4.0) |
#### vLLM-Omni Extension Parameters
| Parameter | Type | Default | Description |
|-----------|------|---------|-------------|
| `task_type` | string | "CustomVoice" | TTS task type: CustomVoice, VoiceDesign, or Base |
| `language` | string | "Auto" | Language (see supported languages below) |
| `instructions` | string | "" | Voice style/emotion instructions |
| `max_new_tokens` | integer | 2048 | Maximum tokens to generate |
**Supported languages:** Auto, Chinese, English, Japanese, Korean, German, French, Russian, Portuguese, Spanish, Italian
#### Voice Clone Parameters (Base task)
| Parameter | Type | Default | Description |
|-----------|------|---------|-------------|
| `ref_audio` | string | null | Reference audio (URL or base64 data URL) |
| `ref_text` | string | null | Transcript of reference audio |
| `x_vector_only_mode` | bool | null | Use speaker embedding only (no ICL) |
### Response Format
Returns binary audio data with appropriate `Content-Type` header (e.g., `audio/wav`).
### Voices Endpoint
```
GET /v1/audio/voices
```
Lists available voices for the loaded model.
```json
{
"voices": ["aiden", "dylan", "eric", "ono_anna", "ryan", "serena", "sohee", "uncle_fu", "vivian"]
}
```
## Examples
### CustomVoice with Style Instruction
```bash
curl -X POST http://localhost:8091/v1/audio/speech \
-H "Content-Type: application/json" \
-d '{
"input": "I am so excited!",
"voice": "vivian",
"instructions": "Speak with great enthusiasm"
}' --output excited.wav
```
### VoiceDesign (Natural Language Voice Description)
```bash
# Start server with VoiceDesign model first
vllm serve Qwen/Qwen3-TTS-12Hz-1.7B-VoiceDesign \
--stage-configs-path vllm_omni/model_executor/stage_configs/qwen3_tts.yaml \
--omni \
--port 8091 \
--trust-remote-code \
--enforce-eager
```
```bash
curl -X POST http://localhost:8091/v1/audio/speech \
-H "Content-Type: application/json" \
-d '{
"input": "Hello world",
"task_type": "VoiceDesign",
"instructions": "A warm, friendly female voice with a gentle tone"
}' --output designed.wav
```
### Base (Voice Cloning)
```bash
# Start server with Base model first
vllm serve Qwen/Qwen3-TTS-12Hz-1.7B-Base \
--stage-configs-path vllm_omni/model_executor/stage_configs/qwen3_tts.yaml \
--omni \
--port 8091 \
--trust-remote-code \
--enforce-eager
```
```bash
curl -X POST http://localhost:8091/v1/audio/speech \
-H "Content-Type: application/json" \
-d '{
"input": "Hello, this is a cloned voice",
"task_type": "Base",
"ref_audio": "https://example.com/reference.wav",
"ref_text": "Original transcript of the reference audio"
}' --output cloned.wav
```
## Supported Models
| Model | Task Type | Description |
|-------|-----------|-------------|
| `Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice` | CustomVoice | Predefined speaker voices with optional style control |
| `Qwen/Qwen3-TTS-12Hz-1.7B-VoiceDesign` | VoiceDesign | Natural language voice style description |
| `Qwen/Qwen3-TTS-12Hz-1.7B-Base` | Base | Voice cloning from reference audio |
| `Qwen/Qwen3-TTS-12Hz-0.6B-CustomVoice` | CustomVoice | Smaller/faster variant |
| `Qwen/Qwen3-TTS-12Hz-0.6B-Base` | Base | Smaller/faster variant for voice cloning |
## Error Responses
### 400 Bad Request
Invalid parameters:
```json
{
"error": {
"message": "Input text cannot be empty",
"type": "BadRequestError",
"param": null,
"code": 400
}
}
```
### 404 Not Found
Model not found:
```json
{
"error": {
"message": "The model `xxx` does not exist.",
"type": "NotFoundError",
"param": "model",
"code": 404
}
}
```
## Troubleshooting
### "TTS model did not produce audio output"
Ensure you're using the correct model variant for your task type:
- CustomVoice task → CustomVoice model
- VoiceDesign task → VoiceDesign model
- Base task → Base model
### Server Not Running
```bash
# Check if server is responding
curl http://localhost:8091/v1/audio/voices
```
### Out of Memory
If you encounter OOM errors:
1. Use smaller model variant: `Qwen/Qwen3-TTS-12Hz-0.6B-CustomVoice`
2. Reduce `--gpu-memory-utilization`
### Unsupported Speaker
Use `/v1/audio/voices` to list available voices for the loaded model.
## Development
Enable debug logging:
```bash
vllm serve Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice \
--stage-configs-path vllm_omni/model_executor/stage_configs/qwen3_tts.yaml \
--omni \
--port 8091 \
--trust-remote-code \
--enforce-eager \
--uvicorn-log-level debug
```
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@@ -192,6 +192,6 @@ extend-ignore-identifiers-re = [
".*nothink.*",
".*NOTHINK.*",
".*nin.*",
"Ono_Anna",
".*[Oo]no_[Aa]nna.*",
".*cann.*",
]
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@@ -0,0 +1,226 @@
# SPDX-License-Identifier: Apache-2.0
# SPDX-FileCopyrightText: Copyright contributors to the vLLM project
"""
E2E Online tests for Qwen3-TTS model with text input and audio output.
These tests verify the /v1/audio/speech endpoint works correctly with
actual model inference, not mocks.
"""
import os
os.environ["VLLM_WORKER_MULTIPROC_METHOD"] = "spawn"
os.environ["VLLM_TEST_CLEAN_GPU_MEMORY"] = "0"
from pathlib import Path
import httpx
import pytest
from tests.conftest import OmniServer
from tests.utils import hardware_test
MODEL = "Qwen/Qwen3-TTS-12Hz-0.6B-CustomVoice"
def get_stage_config():
"""Get the stage config path for Qwen3-TTS."""
return str(
Path(__file__).parent.parent.parent.parent / "vllm_omni" / "model_executor" / "stage_configs" / "qwen3_tts.yaml"
)
@pytest.fixture(scope="module")
def omni_server():
"""Start vLLM-Omni server with CustomVoice model."""
stage_config_path = get_stage_config()
print(f"Starting OmniServer with model: {MODEL}")
with OmniServer(
MODEL,
[
"--stage-configs-path",
stage_config_path,
"--stage-init-timeout",
"120",
"--trust-remote-code",
"--enforce-eager",
],
) as server:
print("OmniServer started successfully")
yield server
print("OmniServer stopping...")
print("OmniServer stopped")
def make_speech_request(
host: str,
port: int,
text: str,
voice: str = "vivian",
language: str = "English",
task_type: str | None = None,
instructions: str | None = None,
timeout: float = 120.0,
) -> httpx.Response:
"""Make a request to the /v1/audio/speech endpoint."""
url = f"http://{host}:{port}/v1/audio/speech"
payload = {
"input": text,
"voice": voice,
"language": language,
}
if task_type:
payload["task_type"] = task_type
if instructions:
payload["instructions"] = instructions
with httpx.Client(timeout=timeout) as client:
return client.post(url, json=payload)
def verify_wav_audio(content: bytes) -> bool:
"""Verify that content is valid WAV audio data."""
# WAV files start with "RIFF" header
if len(content) < 44: # Minimum WAV header size
return False
return content[:4] == b"RIFF" and content[8:12] == b"WAVE"
# Minimum expected audio size for a short sentence (~1 second of 24kHz 16-bit mono WAV)
MIN_AUDIO_BYTES = 10000
class TestQwen3TTSCustomVoice:
"""E2E tests for Qwen3-TTS CustomVoice model."""
@pytest.mark.core_model
@pytest.mark.omni
@hardware_test(res={"cuda": "L4"}, num_cards=4)
def test_speech_english_basic(self, omni_server) -> None:
"""Test basic English TTS generation."""
response = make_speech_request(
host=omni_server.host,
port=omni_server.port,
text="Hello, how are you?",
voice="vivian",
language="English",
)
assert response.status_code == 200, f"Request failed: {response.text}"
assert response.headers.get("content-type") == "audio/wav"
assert verify_wav_audio(response.content), "Response is not valid WAV audio"
assert len(response.content) > MIN_AUDIO_BYTES, (
f"Audio content too small ({len(response.content)} bytes), expected at least {MIN_AUDIO_BYTES} bytes"
)
@pytest.mark.core_model
@pytest.mark.omni
@hardware_test(res={"cuda": "L4"}, num_cards=4)
def test_speech_chinese_basic(self, omni_server) -> None:
"""Test basic Chinese TTS generation."""
response = make_speech_request(
host=omni_server.host,
port=omni_server.port,
text="你好,我是通义千问",
voice="vivian",
language="Chinese",
)
assert response.status_code == 200, f"Request failed: {response.text}"
assert response.headers.get("content-type") == "audio/wav"
assert verify_wav_audio(response.content), "Response is not valid WAV audio"
assert len(response.content) > MIN_AUDIO_BYTES, (
f"Audio content too small ({len(response.content)} bytes), expected at least {MIN_AUDIO_BYTES} bytes"
)
@pytest.mark.core_model
@pytest.mark.omni
@hardware_test(res={"cuda": "L4"}, num_cards=4)
def test_speech_different_voices(self, omni_server) -> None:
"""Test TTS with different voice options."""
voices = ["vivian", "ryan"]
for voice in voices:
response = make_speech_request(
host=omni_server.host,
port=omni_server.port,
text="Testing voice selection.",
voice=voice,
language="English",
)
assert response.status_code == 200, f"Request failed for voice {voice}: {response.text}"
assert verify_wav_audio(response.content), f"Invalid WAV for voice {voice}"
@pytest.mark.core_model
@pytest.mark.omni
@hardware_test(res={"cuda": "L4"}, num_cards=4)
def test_speech_binary_response_not_utf8_error(self, omni_server) -> None:
"""
Regression test: Verify binary audio is returned, not UTF-8 error.
This test ensures the multimodal_output property correctly retrieves
audio from completion outputs, preventing the "TTS model did not
produce audio output" error.
"""
response = make_speech_request(
host=omni_server.host,
port=omni_server.port,
text="This should return binary audio, not a JSON error.",
voice="vivian",
language="English",
)
# Should NOT be a JSON error response
assert response.status_code == 200, f"Request failed: {response.text}"
# Verify it's binary audio, not JSON
try:
# If this succeeds and starts with {"error", it's a bug
text = response.content.decode("utf-8")
assert not text.startswith('{"error"'), f"Got error response instead of audio: {text}"
except UnicodeDecodeError:
# This is expected - binary audio can't be decoded as UTF-8
pass
assert verify_wav_audio(response.content), "Response is not valid WAV audio"
class TestQwen3TTSAPIEndpoints:
"""Test API endpoint functionality."""
@pytest.mark.core_model
@pytest.mark.omni
@hardware_test(res={"cuda": "L4"}, num_cards=4)
def test_list_voices_endpoint(self, omni_server) -> None:
"""Test the /v1/audio/voices endpoint returns available voices."""
url = f"http://{omni_server.host}:{omni_server.port}/v1/audio/voices"
with httpx.Client(timeout=30.0) as client:
response = client.get(url)
assert response.status_code == 200
data = response.json()
assert "voices" in data
assert isinstance(data["voices"], list)
assert len(data["voices"]) > 0
# Check some expected voices are present
voices_lower = [v.lower() for v in data["voices"]]
assert "vivian" in voices_lower or "ryan" in voices_lower
@pytest.mark.core_model
@pytest.mark.omni
@hardware_test(res={"cuda": "L4"}, num_cards=4)
def test_models_endpoint(self, omni_server) -> None:
"""Test the /v1/models endpoint returns loaded model."""
url = f"http://{omni_server.host}:{omni_server.port}/v1/models"
with httpx.Client(timeout=30.0) as client:
response = client.get(url)
assert response.status_code == 200
data = response.json()
assert "data" in data
assert len(data["data"]) > 0
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@@ -221,24 +221,33 @@ class OrchestratorAggregator:
stage_id: int,
request_id: str,
) -> None:
if (
output_to_yield.final_output_type == "audio"
and finished
and (multimodal_output := output_to_yield.multimodal_output.get("audio")) is not None
):
nframes = int(multimodal_output[-1].shape[0])
stage_events_for_req = self.stage_events.get(request_id, [])
if stage_events_for_req:
for stage_event in stage_events_for_req:
if stage_event.stage_id == stage_id:
stage_event.audio_generated_frames += nframes
break
else:
logger.warning(
"Failed to record audio generated frames for request %s at stage %s: no stage event found",
request_id,
stage_id,
)
try:
if (
output_to_yield.final_output_type == "audio"
and finished
and (multimodal_output := output_to_yield.multimodal_output.get("audio")) is not None
and len(multimodal_output) > 0
):
last = multimodal_output[-1]
nframes = int(last.shape[0]) if last.ndim > 0 else 1
stage_events_for_req = self.stage_events.get(request_id, [])
if stage_events_for_req:
for stage_event in stage_events_for_req:
if stage_event.stage_id == stage_id:
stage_event.audio_generated_frames += nframes
break
else:
logger.warning(
"Failed to record audio generated frames for request %s at stage %s: no stage event found",
request_id,
stage_id,
)
except Exception:
logger.debug(
"Failed to record audio frames for request %s",
request_id,
exc_info=True,
)
def _as_stage_request_stats(
self,