update TTS and transport output sample rate to 24000
This commit is contained in:
@@ -52,7 +52,7 @@ class SileroOnnxModel:
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if sr not in self.sample_rates:
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raise ValueError(
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f"Supported sampling rates: {self.sample_rates} (or multiply of 16000)"
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f"Supported sampling rates: {self.sample_rates} (or multiple of 16000)"
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)
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if sr / np.shape(x)[1] > 31.25:
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raise ValueError("Input audio chunk is too short")
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@@ -205,7 +205,7 @@ class TTSService(AIService):
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# if push_stop_frames is True, wait for this idle period before pushing TTSStoppedFrame
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stop_frame_timeout_s: float = 1.0,
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# TTS output sample rate
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sample_rate: int = 16000,
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sample_rate: int = 24000,
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text_filter: Optional[BaseTextFilter] = None,
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**kwargs,
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):
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@@ -514,7 +514,7 @@ class SegmentedSTTService(STTService):
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min_volume: float = 0.6,
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max_silence_secs: float = 0.3,
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max_buffer_secs: float = 1.5,
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sample_rate: int = 16000,
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sample_rate: int = 24000,
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num_channels: int = 1,
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**kwargs,
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):
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@@ -88,7 +88,7 @@ class AzureTTSService(TTSService):
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api_key: str,
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region: str,
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voice="en-US-SaraNeural",
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sample_rate: int = 16000,
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sample_rate: int = 24000,
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params: InputParams = InputParams(),
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**kwargs,
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):
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@@ -283,7 +283,7 @@ class AzureSTTService(STTService):
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api_key: str,
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region: str,
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language=Language.EN_US,
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sample_rate=16000,
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sample_rate=24000,
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channels=1,
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**kwargs,
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):
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@@ -80,7 +80,7 @@ class CartesiaTTSService(WordTTSService):
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cartesia_version: str = "2024-06-10",
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url: str = "wss://api.cartesia.ai/tts/websocket",
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model: str = "sonic-english",
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sample_rate: int = 16000,
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sample_rate: int = 24000,
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encoding: str = "pcm_s16le",
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container: str = "raw",
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params: InputParams = InputParams(),
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@@ -99,6 +99,7 @@ class CartesiaTTSService(WordTTSService):
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super().__init__(
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aggregate_sentences=True,
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push_text_frames=False,
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sample_rate=sample_rate,
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**kwargs,
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)
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@@ -298,13 +299,13 @@ class CartesiaHttpTTSService(TTSService):
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voice_id: str,
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model: str = "sonic-english",
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base_url: str = "https://api.cartesia.ai",
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sample_rate: int = 16000,
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sample_rate: int = 24000,
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encoding: str = "pcm_s16le",
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container: str = "raw",
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params: InputParams = InputParams(),
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**kwargs,
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):
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super().__init__(**kwargs)
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super().__init__(sample_rate=sample_rate, **kwargs)
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self._api_key = api_key
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self._settings = {
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@@ -51,11 +51,11 @@ class DeepgramTTSService(TTSService):
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*,
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api_key: str,
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voice: str = "aura-helios-en",
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sample_rate: int = 16000,
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sample_rate: int = 24000,
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encoding: str = "linear16",
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**kwargs,
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):
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super().__init__(**kwargs)
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super().__init__(sample_rate=sample_rate, **kwargs)
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self._settings = {
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"sample_rate": sample_rate,
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@@ -99,7 +99,7 @@ class ElevenLabsTTSService(WordTTSService):
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voice_id: str,
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model: str = "eleven_turbo_v2_5",
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url: str = "wss://api.elevenlabs.io",
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output_format: ElevenLabsOutputFormat = "pcm_16000",
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output_format: ElevenLabsOutputFormat = "pcm_24000",
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params: InputParams = InputParams(),
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**kwargs,
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):
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@@ -115,7 +115,7 @@ class PlayHTTTSService(TTSService):
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user_id: str,
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voice_url: str,
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voice_engine: str = "PlayHT3.0-mini",
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sample_rate: int = 16000,
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sample_rate: int = 24000,
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output_format: str = "wav",
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params: InputParams = InputParams(),
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**kwargs,
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@@ -310,7 +310,7 @@ class PlayHTHttpTTSService(TTSService):
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user_id: str,
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voice_url: str,
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voice_engine: str = "PlayHT3.0-mini",
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sample_rate: int = 16000,
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sample_rate: int = 24000,
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params: InputParams = InputParams(),
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**kwargs,
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):
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@@ -39,9 +39,10 @@ class XTTSService(TTSService):
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language: Language,
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base_url: str,
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aiohttp_session: aiohttp.ClientSession,
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sample_rate: int = 24000,
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**kwargs,
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):
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super().__init__(**kwargs)
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super().__init__(sample_rate=sample_rate, **kwargs)
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self._settings = {
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"language": self.language_to_service_language(language),
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@@ -162,16 +163,18 @@ class XTTSService(TTSService):
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# Remove processed data from buffer
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buffer = buffer[48000:]
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# Resample the audio from 24000 Hz to 16000 Hz
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resampled_audio = resample_audio(bytes(process_data), 24000, 16000)
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# Resample the audio from 24000 Hz
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resampled_audio = resample_audio(
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bytes(process_data), 24000, self._sample_rate
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)
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# Create the frame with the resampled audio
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frame = TTSAudioRawFrame(resampled_audio, 16000, 1)
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frame = TTSAudioRawFrame(resampled_audio, self._sample_rate, 1)
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yield frame
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# Process any remaining data in the buffer
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if len(buffer) > 0:
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resampled_audio = resample_audio(bytes(buffer), 24000, 16000)
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frame = TTSAudioRawFrame(resampled_audio, 16000, 1)
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resampled_audio = resample_audio(bytes(buffer), 24000, self._sample_rate)
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frame = TTSAudioRawFrame(resampled_audio, self._sample_rate, 1)
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yield frame
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yield TTSStoppedFrame()
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@@ -30,7 +30,7 @@ class TransportParams(BaseModel):
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camera_out_color_format: str = "RGB"
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audio_out_enabled: bool = False
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audio_out_is_live: bool = False
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audio_out_sample_rate: int = 16000
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audio_out_sample_rate: int = 24000
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audio_out_channels: int = 1
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audio_out_bitrate: int = 96000
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audio_in_enabled: bool = False
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