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from typing import Any, List, Tuple, Union, Optional | |
import numpy as np | |
import soundfile | |
import io | |
import asyncio | |
from simuleval.agents.pipeline import TreeAgentPipeline | |
from simuleval.agents.states import AgentStates | |
from simuleval.data.segments import Segment, EmptySegment, SpeechSegment | |
import threading | |
import math | |
import logging | |
import sys | |
from pathlib import Path | |
import time | |
from g2p_en import G2p | |
import torch | |
import traceback | |
import time | |
import random | |
import colorlog | |
MODEL_SAMPLE_RATE = 16_000 | |
logger = logging.getLogger(__name__) | |
logger.propagate = False | |
handler = colorlog.StreamHandler(stream=sys.stdout) | |
formatter = colorlog.ColoredFormatter( | |
"%(log_color)s[%(asctime)s][%(levelname)s][%(module)s]:%(reset)s %(message)s", | |
reset=True, | |
log_colors={ | |
"DEBUG": "cyan", | |
"INFO": "green", | |
"WARNING": "yellow", | |
"ERROR": "red", | |
"CRITICAL": "red,bg_white", | |
}, | |
) | |
handler.setFormatter(formatter) | |
logger.addHandler(handler) | |
logger.setLevel(logging.DEBUG) | |
class SpeechAndTextOutput: | |
def __init__( | |
self, | |
text: str = None, | |
speech_samples: list = None, | |
speech_sample_rate: float = None, | |
final: bool = False, | |
): | |
self.text = text | |
self.speech_samples = speech_samples | |
self.speech_sample_rate = speech_sample_rate | |
self.final = final | |
class OutputSegments: | |
def __init__(self, segments: Union[List[Segment], Segment]): | |
if isinstance(segments, Segment): | |
segments = [segments] | |
self.segments: List[Segment] = [s for s in segments] | |
def is_empty(self): | |
return all(segment.is_empty for segment in self.segments) | |
def finished(self): | |
return all(segment.finished for segment in self.segments) | |
def compute_length(self, g2p): | |
lengths = [] | |
for segment in self.segments: | |
if segment.data_type == "text": | |
lengths.append(len([x for x in g2p(segment.content) if x != " "])) | |
elif segment.data_type == "speech": | |
lengths.append(len(segment.content) / MODEL_SAMPLE_RATE) | |
elif isinstance(segment, EmptySegment): | |
continue | |
else: | |
logger.warning( | |
f"Unexpected data_type: {segment.data_type} not in 'speech', 'text'" | |
) | |
return max(lengths) | |
def join_output_buffer( | |
cls, buffer: List[List[Segment]], output: SpeechAndTextOutput | |
): | |
num_segments = len(buffer[0]) | |
for i in range(num_segments): | |
segment_list = [ | |
buffer[j][i] | |
for j in range(len(buffer)) | |
if buffer[j][i].data_type is not None | |
] | |
if len(segment_list) == 0: | |
continue | |
if len(set(segment.data_type for segment in segment_list)) != 1: | |
logger.warning( | |
f"Data type mismatch at {i}: {set(segment.data_type for segment in segment_list)}" | |
) | |
continue | |
data_type = segment_list[0].data_type | |
if data_type == "text": | |
if output.text is not None: | |
logger.warning("Multiple text outputs, overwriting!") | |
output.text = " ".join([segment.content for segment in segment_list]) | |
elif data_type == "speech": | |
if output.speech_samples is not None: | |
logger.warning("Multiple speech outputs, overwriting!") | |
speech_out = [] | |
for segment in segment_list: | |
speech_out += segment.content | |
output.speech_samples = speech_out | |
output.speech_sample_rate = segment.sample_rate | |
elif isinstance(segment_list[0], EmptySegment): | |
continue | |
else: | |
logger.warning( | |
f"Invalid output buffer data type: {data_type}, expected 'speech' or 'text" | |
) | |
return output | |
def __repr__(self) -> str: | |
repr_str = str(self.segments) | |
return f"{self.__class__.__name__}(\n\t{repr_str}\n)" | |
def convert_waveform( | |
waveform: Union[np.ndarray, torch.Tensor], | |
sample_rate: int, | |
normalize_volume: bool = False, | |
to_mono: bool = False, | |
to_sample_rate: Optional[int] = None, | |
) -> Tuple[Union[np.ndarray, torch.Tensor], int]: | |
"""convert a waveform: | |
- to a target sample rate | |
- from multi-channel to mono channel | |
- volume normalization | |
Args: | |
waveform (numpy.ndarray or torch.Tensor): 2D original waveform | |
(channels x length) | |
sample_rate (int): original sample rate | |
normalize_volume (bool): perform volume normalization | |
to_mono (bool): convert to mono channel if having multiple channels | |
to_sample_rate (Optional[int]): target sample rate | |
Returns: | |
waveform (numpy.ndarray): converted 2D waveform (channels x length) | |
sample_rate (float): target sample rate | |
""" | |
try: | |
import torchaudio.sox_effects as ta_sox | |
except ImportError: | |
raise ImportError("Please install torchaudio: pip install torchaudio") | |
effects = [] | |
if normalize_volume: | |
effects.append(["gain", "-n"]) | |
if to_sample_rate is not None and to_sample_rate != sample_rate: | |
effects.append(["rate", f"{to_sample_rate}"]) | |
if to_mono and waveform.shape[0] > 1: | |
effects.append(["channels", "1"]) | |
if len(effects) > 0: | |
is_np_input = isinstance(waveform, np.ndarray) | |
_waveform = torch.from_numpy(waveform) if is_np_input else waveform | |
converted, converted_sample_rate = ta_sox.apply_effects_tensor( | |
_waveform, sample_rate, effects | |
) | |
if is_np_input: | |
converted = converted.numpy() | |
return converted, converted_sample_rate | |
return waveform, sample_rate | |
class SimulevalTranscoder: | |
def __init__(self, agent, sample_rate, debug, buffer_limit): | |
# agent is stateless | |
self.agent = agent | |
self.input_queue = asyncio.Queue() | |
self.output_queue = asyncio.Queue() | |
self.states = self.agent.build_states() | |
if debug: | |
self.get_states_root().debug = True | |
self.incoming_sample_rate = sample_rate | |
self.close = False | |
self.g2p = G2p() | |
# buffer all outgoing translations within this amount of time | |
self.output_buffer_idle_ms = 5000 | |
self.output_buffer_size_limit = ( | |
buffer_limit # phonemes for text, seconds for speech | |
) | |
self.output_buffer_cur_size = 0 | |
self.output_buffer: List[List[Segment]] = [] | |
self.speech_output_sample_rate = None | |
self.last_output_ts = time.time() * 1000 | |
self.timeout_ms = ( | |
30000 # close the transcoder thread after this amount of silence | |
) | |
self.first_input_ts = None | |
self.first_output_ts = None | |
self.debug = debug | |
self.debug_ts = f"{time.time()}_{random.randint(1000, 9999)}" | |
if self.debug: | |
debug_folder = Path(__file__).resolve().parent.parent / "debug" | |
self.test_incoming_wav = soundfile.SoundFile( | |
debug_folder / f"{self.debug_ts}_test_incoming.wav", | |
mode="w+", | |
format="WAV", | |
subtype="PCM_16", | |
samplerate=self.incoming_sample_rate, | |
channels=1, | |
) | |
self.get_states_root().test_input_segments_wav = soundfile.SoundFile( | |
debug_folder / f"{self.debug_ts}_test_input_segments.wav", | |
mode="w+", | |
format="WAV", | |
samplerate=MODEL_SAMPLE_RATE, | |
channels=1, | |
) | |
def get_states_root(self) -> AgentStates: | |
if isinstance(self.agent, TreeAgentPipeline): | |
# self.states is a dict | |
return self.states[self.agent.source_module] | |
else: | |
# self.states is a list | |
return self.states[0] | |
def reset_states(self): | |
if isinstance(self.agent, TreeAgentPipeline): | |
states_iter = self.states.values() | |
else: | |
states_iter = self.states | |
for state in states_iter: | |
state.reset() | |
def debug_log(self, *args): | |
if self.debug: | |
logger.info(*args) | |
def process_incoming_bytes(self, incoming_bytes, target_language, sample_rate): | |
# TODO: currently just taking sample rate here, refactor sample rate | |
# bytes is 16bit signed int | |
self.incoming_sample_rate = sample_rate | |
segment, sr = self._preprocess_wav(incoming_bytes) | |
segment = SpeechSegment( | |
content=segment, sample_rate=sr, tgt_lang=target_language | |
) | |
# # segment is array([0, 0, 0, ..., 0, 0, 0], dtype=int16) | |
self.input_queue.put_nowait(segment) | |
print("process_incoming: put input_queue") | |
def get_input_segment(self): | |
if self.input_queue.empty(): | |
return None | |
chunk = self.input_queue.get_nowait() | |
self.input_queue.task_done() | |
return chunk | |
def _preprocess_wav(self, data: Any) -> Tuple[np.ndarray, int]: | |
segment, sample_rate = soundfile.read( | |
io.BytesIO(data), | |
dtype="float32", | |
always_2d=True, | |
frames=-1, | |
start=0, | |
format="RAW", | |
subtype="PCM_16", | |
samplerate=self.incoming_sample_rate, | |
channels=1, | |
) | |
if self.debug: | |
self.test_incoming_wav.seek(0, soundfile.SEEK_END) | |
self.test_incoming_wav.write(segment) | |
segment = segment.T | |
segment, new_sample_rate = convert_waveform( | |
segment, | |
sample_rate, | |
normalize_volume=False, | |
to_mono=True, | |
to_sample_rate=MODEL_SAMPLE_RATE, | |
) | |
assert MODEL_SAMPLE_RATE == new_sample_rate | |
segment = segment.squeeze(axis=0) | |
return segment, new_sample_rate | |
def process_pipeline_impl(self, input_segment): | |
try: | |
with torch.no_grad(): | |
output_segment = OutputSegments( | |
self.agent.pushpop(input_segment, self.states) | |
) | |
if ( | |
self.get_states_root().first_input_ts is not None | |
and self.first_input_ts is None | |
): | |
# TODO: this is hacky | |
self.first_input_ts = self.get_states_root().first_input_ts | |
if not output_segment.is_empty: | |
print("PUT IN OUTPUT QUEUE") | |
self.output_queue.put_nowait(output_segment) | |
if output_segment.finished: | |
print("OUTPUT SEGMENT IS FINISHED. Resetting states.") | |
self.reset_states() | |
if self.debug: | |
# when we rebuild states, this value is reset to whatever | |
# is in the system dir config, which defaults debug=False. | |
self.get_states_root().debug = True | |
except Exception as e: | |
logger.error(f"Got exception while processing pipeline: {e}") | |
traceback.print_exc() | |
return input_segment | |
def process_pipeline_loop(self): | |
if self.close: | |
print("transcoder closed") | |
return # closes the thread | |
print("processing_pipeline") | |
while not self.close: | |
input_segment = self.get_input_segment() | |
if input_segment is None: | |
if self.get_states_root().is_fresh_state: # TODO: this is hacky | |
time.sleep(0.3) | |
print("loop: input_queue empty") | |
else: | |
time.sleep(0.03) | |
continue | |
print("loop: got input_segment") | |
self.process_pipeline_impl(input_segment) | |
print("finished processing_pipeline") | |
def process_pipeline_once(self): | |
if self.close: | |
return | |
self.debug_log("processing pipeline once") | |
input_segment = self.get_input_segment() | |
if input_segment is None: | |
return | |
self.process_pipeline_impl(input_segment) | |
self.debug_log("finished processing_pipeline_once") | |
def get_output_segment(self): | |
if self.output_queue.empty(): | |
return None | |
output_chunk = self.output_queue.get_nowait() | |
self.output_queue.task_done() | |
return output_chunk | |
def start(self): | |
print("starting transcoder in a thread") | |
threading.Thread(target=self.process_pipeline_loop).start() | |
def first_translation_time(self): | |
return round((self.first_output_ts - self.first_input_ts) / 1000, 2) | |
def get_buffered_output(self) -> SpeechAndTextOutput: | |
now = time.time() * 1000 | |
print(f"get_buffered_output queue size: {self.output_queue.qsize()}") | |
while not self.output_queue.empty(): | |
tmp_out = self.get_output_segment() | |
if tmp_out and tmp_out.compute_length(self.g2p) > 0: | |
if len(self.output_buffer) == 0: | |
self.last_output_ts = now | |
self._populate_output_buffer(tmp_out) | |
self._increment_output_buffer_size(tmp_out) | |
if tmp_out.finished: | |
self.debug_log("tmp_out.finished") | |
res = self._gather_output_buffer_data(final=True) | |
self.debug_log(f"gathered output data: {res}") | |
self.output_buffer = [] | |
self.increment_output_buffer_size = 0 | |
self.last_output_ts = now | |
self.first_output_ts = now | |
return res | |
else: | |
self.debug_log("tmp_out.compute_length is not > 0") | |
if len(self.output_buffer) > 0 and ( | |
now - self.last_output_ts >= self.output_buffer_idle_ms | |
or self.output_buffer_cur_size >= self.output_buffer_size_limit | |
): | |
self.debug_log( | |
"[get_buffered_output] output_buffer is not empty. getting res to return." | |
) | |
self.last_output_ts = now | |
res = self._gather_output_buffer_data(final=False) | |
self.debug_log(f"gathered output data: {res}") | |
self.output_buffer = [] | |
self.output_buffer_phoneme_count = 0 | |
self.first_output_ts = now | |
return res | |
else: | |
self.debug_log("[get_buffered_output] output_buffer is empty...") | |
return None | |
def _gather_output_buffer_data(self, final): | |
output = SpeechAndTextOutput() | |
output.final = final | |
output = OutputSegments.join_output_buffer(self.output_buffer, output) | |
return output | |
def _increment_output_buffer_size(self, segment: OutputSegments): | |
self.output_buffer_cur_size += segment.compute_length(self.g2p) | |
def _populate_output_buffer(self, segment: OutputSegments): | |
self.output_buffer.append(segment.segments) | |
def _compute_phoneme_count(self, string: str) -> int: | |
return len([x for x in self.g2p(string) if x != " "]) | |