Merge pull request #664 from pipecat-ai/aleix/fix-aws-stuttering
services(aws): read stream and resample in a thread
This commit is contained in:
@@ -9,6 +9,9 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
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### Added
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- Added `TavusVideoService`. This is an integration for Tavus digital twins.
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(see https://www.tavus.io/)
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- Added `DailyTransport.update_subscriptions()`. This allows you to have fine
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grained control of what media subscriptions you want for each participant in a
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room.
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@@ -19,6 +22,10 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
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output to 24000 and also the default output transport sample rate. This
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improves audio quality at the cost of some extra bandwidth.
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### Fixed
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- Fixed an issue that was causing stuttering with AWS TTS service.
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## [0.0.47] - 2024-10-22
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### Added
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@@ -4,6 +4,8 @@
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# SPDX-License-Identifier: BSD 2-Clause License
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#
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import asyncio
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from typing import AsyncGenerator, Optional
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from loguru import logger
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@@ -165,6 +167,14 @@ class AWSTTSService(TTSService):
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return ssml
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async def run_tts(self, text: str) -> AsyncGenerator[Frame, None]:
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def read_audio_data(**args):
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response = self._polly_client.synthesize_speech(**args)
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if "AudioStream" in response:
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audio_data = response["AudioStream"].read()
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resampled = resample_audio(audio_data, 16000, self._settings["sample_rate"])
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return resampled
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return None
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logger.debug(f"Generating TTS: [{text}]")
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try:
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@@ -186,23 +196,24 @@ class AWSTTSService(TTSService):
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# Filter out None values
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filtered_params = {k: v for k, v in params.items() if v is not None}
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response = self._polly_client.synthesize_speech(**filtered_params)
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audio_data = await asyncio.to_thread(read_audio_data, **filtered_params)
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if not audio_data:
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logger.error(f"{self} No audio data returned")
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yield None
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return
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await self.start_tts_usage_metrics(text)
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yield TTSStartedFrame()
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if "AudioStream" in response:
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with response["AudioStream"] as stream:
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audio_data = stream.read()
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chunk_size = 8192
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for i in range(0, len(audio_data), chunk_size):
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chunk = audio_data[i : i + chunk_size]
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if len(chunk) > 0:
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await self.stop_ttfb_metrics()
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resampled = resample_audio(chunk, 16000, self._settings["sample_rate"])
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frame = TTSAudioRawFrame(resampled, self._settings["sample_rate"], 1)
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yield frame
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chunk_size = 8192
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for i in range(0, len(audio_data), chunk_size):
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chunk = audio_data[i : i + chunk_size]
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if len(chunk) > 0:
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await self.stop_ttfb_metrics()
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frame = TTSAudioRawFrame(chunk, self._settings["sample_rate"], 1)
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yield frame
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yield TTSStoppedFrame()
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