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server: use mp3 as default for audio storage

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Closes #223
This commit is contained in:
Mathieu Virbel
2023-09-13 16:18:12 +02:00
committed by Mathieu Virbel
parent fb93c55993
commit 2b9eef6131
4 changed files with 27 additions and 77 deletions
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16
server/reflector/processors/audio_file_writer.py
View File

@@ -1,7 +1,8 @@
from reflector.processors.base import Processor
import av
from pathlib import Path
import av
from reflector.processors.base import Processor
class AudioFileWriterProcessor(Processor):
"""
@@ -15,6 +16,8 @@ class AudioFileWriterProcessor(Processor):
super().__init__()
if isinstance(path, str):
path = Path(path)
if path.suffix not in (".mp3", ".wav"):
raise ValueError("Only mp3 and wav files are supported")
self.path = path
self.out_container = None
self.out_stream = None
@@ -22,10 +25,19 @@ class AudioFileWriterProcessor(Processor):
async def _push(self, data: av.AudioFrame):
if not self.out_container:
self.path.parent.mkdir(parents=True, exist_ok=True)
suffix = self.path.suffix
if suffix == ".mp3":
self.out_container = av.open(self.path.as_posix(), "w", format="mp3")
self.out_stream = self.out_container.add_stream(
"libmp3lame", rate=data.sample_rate
)
elif suffix == ".wav":
self.out_container = av.open(self.path.as_posix(), "w", format="wav")
self.out_stream = self.out_container.add_stream(
"pcm_s16le", rate=data.sample_rate
)
else:
raise ValueError("Only mp3 and wav files are supported")
for packet in self.out_stream.encode(data):
self.out_container.mux(packet)
await self.emit(data)

66
server/reflector/views/transcripts.py
View File

@@ -1,12 +1,10 @@
import json
import shutil
from datetime import datetime
from pathlib import Path
from tempfile import NamedTemporaryFile
from typing import Annotated, Optional
from uuid import uuid4
import av
import reflector.auth as auth
from fastapi import (
APIRouter,
Depends,
@@ -17,13 +15,11 @@ from fastapi import (
)
from fastapi_pagination import Page, paginate
from pydantic import BaseModel, Field
from starlette.concurrency import run_in_threadpool
import reflector.auth as auth
from reflector.db import database, transcripts
from reflector.logger import logger
from reflector.settings import settings
from reflector.utils.audio_waveform import get_audio_waveform
from starlette.concurrency import run_in_threadpool
from ._range_requests_response import range_requests_response
from .rtc_offer import PipelineEvent, RtcOffer, rtc_offer_base
@@ -112,35 +108,12 @@ class Transcript(BaseModel):
def topics_dump(self, mode="json"):
return [topic.model_dump(mode=mode) for topic in self.topics]
def convert_audio_to_mp3(self):
fn = self.audio_mp3_filename
if fn.exists():
return
logger.info(f"Converting audio to mp3: {self.audio_filename}")
inp = av.open(self.audio_filename.as_posix(), "r")
# create temporary file for mp3
with NamedTemporaryFile(suffix=".mp3", delete=False) as tmp:
out = av.open(tmp.name, "w")
stream = out.add_stream("mp3")
for frame in inp.decode(audio=0):
frame.pts = None
for packet in stream.encode(frame):
out.mux(packet)
for packet in stream.encode(None):
out.mux(packet)
out.close()
# move temporary file to final location
shutil.move(tmp.name, fn.as_posix())
def convert_audio_to_waveform(self, segments_count=256):
fn = self.audio_waveform_filename
if fn.exists():
return
waveform = get_audio_waveform(
path=self.audio_filename, segments_count=segments_count
path=self.audio_mp3_filename, segments_count=segments_count
)
try:
with open(fn, "w") as fd:
@@ -158,10 +131,6 @@ class Transcript(BaseModel):
def data_path(self):
return Path(settings.DATA_DIR) / self.id
@property
def audio_filename(self):
return self.data_path / "audio.wav"
@property
def audio_mp3_filename(self):
return self.data_path / "audio.mp3"
@@ -373,27 +342,6 @@ async def transcript_delete(
return DeletionStatus(status="ok")
@router.get("/transcripts/{transcript_id}/audio")
async def transcript_get_audio(
request: Request,
transcript_id: str,
user: Annotated[Optional[auth.UserInfo], Depends(auth.current_user_optional)],
):
user_id = user["sub"] if user else None
transcript = await transcripts_controller.get_by_id(transcript_id, user_id=user_id)
if not transcript:
raise HTTPException(status_code=404, detail="Transcript not found")
if not transcript.audio_filename.exists():
raise HTTPException(status_code=404, detail="Audio not found")
return range_requests_response(
request,
transcript.audio_filename,
content_type="audio/wav",
)
@router.get("/transcripts/{transcript_id}/audio/mp3")
async def transcript_get_audio_mp3(
request: Request,
@@ -405,11 +353,9 @@ async def transcript_get_audio_mp3(
if not transcript:
raise HTTPException(status_code=404, detail="Transcript not found")
if not transcript.audio_filename.exists():
if not transcript.audio_mp3_filename.exists():
raise HTTPException(status_code=404, detail="Audio not found")
await run_in_threadpool(transcript.convert_audio_to_mp3)
return range_requests_response(
request,
transcript.audio_mp3_filename,
@@ -427,7 +373,7 @@ async def transcript_get_audio_waveform(
if not transcript:
raise HTTPException(status_code=404, detail="Transcript not found")
if not transcript.audio_filename.exists():
if not transcript.audio_mp3_filename.exists():
raise HTTPException(status_code=404, detail="Audio not found")
await run_in_threadpool(transcript.convert_audio_to_waveform)
@@ -640,7 +586,7 @@ async def transcript_record_webrtc(
request,
event_callback=handle_rtc_event,
event_callback_args=transcript_id,
audio_filename=transcript.audio_filename,
audio_filename=transcript.audio_mp3_filename,
source_language=transcript.source_language,
target_language=transcript.target_language,
)

9
server/tests/test_transcripts_audio_download.py
View File

@@ -24,8 +24,8 @@ async def fake_transcript(tmpdir):
await transcripts_controller.update(transcript, {"status": "finished"})
# manually copy a file at the expected location
audio_filename = transcript.audio_filename
path = Path(__file__).parent / "records" / "test_mathieu_hello.wav"
audio_filename = transcript.audio_mp3_filename
path = Path(__file__).parent / "records" / "test_mathieu_hello.mp3"
audio_filename.parent.mkdir(parents=True, exist_ok=True)
shutil.copy(path, audio_filename)
yield transcript
@@ -35,7 +35,6 @@ async def fake_transcript(tmpdir):
@pytest.mark.parametrize(
"url_suffix,content_type",
[
["", "audio/wav"],
["/mp3", "audio/mp3"],
],
)
@@ -52,7 +51,6 @@ async def test_transcript_audio_download(fake_transcript, url_suffix, content_ty
@pytest.mark.parametrize(
"url_suffix,content_type",
[
["", "audio/wav"],
["/mp3", "audio/mp3"],
],
)
@@ -76,7 +74,6 @@ async def test_transcript_audio_download_range(
@pytest.mark.parametrize(
"url_suffix,content_type",
[
["", "audio/wav"],
["/mp3", "audio/mp3"],
],
)
@@ -104,4 +101,4 @@ async def test_transcript_audio_download_waveform(fake_transcript):
assert response.status_code == 200
assert response.headers["content-type"] == "application/json"
assert isinstance(response.json()["data"], list)
assert len(response.json()["data"]) == 256
assert len(response.json()["data"]) >= 255

5
server/tests/test_transcripts_rtc_ws.py
View File

@@ -200,11 +200,6 @@ async def test_transcript_rtc_and_websocket(
assert resp.status_code == 200
assert resp.json()["status"] == "ended"
# check that audio is available
resp = await ac.get(f"/transcripts/{tid}/audio")
assert resp.status_code == 200
assert resp.headers["Content-Type"] == "audio/wav"
# check that audio/mp3 is available
resp = await ac.get(f"/transcripts/{tid}/audio/mp3")
assert resp.status_code == 200