from queue import Queue from transformers.generation.streamers import BaseStreamer from typing import Optional class ParlerTTSStreamer(BaseStreamer): def __init__( self, model: ParlerTTSForConditionalGeneration, device: Optional[str] = None, play_steps: Optional[int] = 10, stride: Optional[int] = None, timeout: Optional[float] = None, ): """ Streamer that stores playback-ready audio in a queue, to be used by a downstream application as an iterator. This is useful for applications that benefit from accessing the generated audio in a non-blocking way (e.g. in an interactive Gradio demo). Parameters: model (`ParlerTTSForConditionalGeneration`): The Parler-TTS model used to generate the audio waveform. device (`str`, *optional*): The torch device on which to run the computation. If `None`, will default to the device of the model. play_steps (`int`, *optional*, defaults to 10): The number of generation steps with which to return the generated audio array. Using fewer steps will mean the first chunk is ready faster, but will require more codec decoding steps overall. This value should be tuned to your device and latency requirements. stride (`int`, *optional*): The window (stride) between adjacent audio samples. Using a stride between adjacent audio samples reduces the hard boundary between them, giving smoother playback. If `None`, will default to a value equivalent to play_steps // 6 in the audio space. timeout (`int`, *optional*): The timeout for the audio queue. If `None`, the queue will block indefinitely. Useful to handle exceptions in `.generate()`, when it is called in a separate thread. """ self.decoder = model.decoder self.audio_encoder = model.audio_encoder self.generation_config = model.generation_config self.device = device if device is not None else model.device # variables used in the streaming process self.play_steps = play_steps if stride is not None: self.stride = stride else: hop_length = math.floor(self.audio_encoder.config.sampling_rate / self.audio_encoder.config.frame_rate) self.stride = hop_length * (play_steps - self.decoder.num_codebooks) // 6 self.token_cache = None self.to_yield = 0 # varibles used in the thread process self.audio_queue = Queue() self.stop_signal = None self.timeout = timeout def apply_delay_pattern_mask(self, input_ids): # build the delay pattern mask for offsetting each codebook prediction by 1 (this behaviour is specific to Parler) _, delay_pattern_mask = self.decoder.build_delay_pattern_mask( input_ids[:, :1], bos_token_id=self.generation_config.bos_token_id, pad_token_id=self.generation_config.decoder_start_token_id, max_length=input_ids.shape[-1], ) # apply the pattern mask to the input ids input_ids = self.decoder.apply_delay_pattern_mask(input_ids, delay_pattern_mask) # revert the pattern delay mask by filtering the pad token id mask = (delay_pattern_mask != self.generation_config.bos_token_id) & (delay_pattern_mask != self.generation_config.pad_token_id) input_ids = input_ids[mask].reshape(1, self.decoder.num_codebooks, -1) # append the frame dimension back to the audio codes input_ids = input_ids[None, ...] # send the input_ids to the correct device input_ids = input_ids.to(self.audio_encoder.device) decode_sequentially = ( self.generation_config.bos_token_id in input_ids or self.generation_config.pad_token_id in input_ids or self.generation_config.eos_token_id in input_ids ) if not decode_sequentially: output_values = self.audio_encoder.decode( input_ids, audio_scales=[None], ) else: sample = input_ids[:, 0] sample_mask = (sample >= self.audio_encoder.config.codebook_size).sum(dim=(0, 1)) == 0 sample = sample[:, :, sample_mask] output_values = self.audio_encoder.decode(sample[None, ...], [None]) audio_values = output_values.audio_values[0, 0] return audio_values.cpu().float().numpy() def put(self, value): batch_size = value.shape[0] // self.decoder.num_codebooks if batch_size > 1: raise ValueError("ParlerTTSStreamer only supports batch size 1") if self.token_cache is None: self.token_cache = value else: self.token_cache = torch.concatenate([self.token_cache, value[:, None]], dim=-1) if self.token_cache.shape[-1] % self.play_steps == 0: audio_values = self.apply_delay_pattern_mask(self.token_cache) self.on_finalized_audio(audio_values[self.to_yield : -self.stride]) self.to_yield += len(audio_values) - self.to_yield - self.stride def end(self): """Flushes any remaining cache and appends the stop symbol.""" if self.token_cache is not None: audio_values = self.apply_delay_pattern_mask(self.token_cache) else: audio_values = np.zeros(self.to_yield) self.on_finalized_audio(audio_values[self.to_yield :], stream_end=True) def on_finalized_audio(self, audio: np.ndarray, stream_end: bool = False): """Put the new audio in the queue. If the stream is ending, also put a stop signal in the queue.""" self.audio_queue.put(audio, timeout=self.timeout) if stream_end: self.audio_queue.put(self.stop_signal, timeout=self.timeout) def __iter__(self): return self def __next__(self): value = self.audio_queue.get(timeout=self.timeout) if not isinstance(value, np.ndarray) and value == self.stop_signal: raise StopIteration() else: return value