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import asyncio
import io
import logging
import traceback
from typing import List
import av
import numpy as np
import streamlit as st
from streamlit_webrtc import WebRtcMode, webrtc_streamer
import pydub
from dotenv import load_dotenv
load_dotenv()
from sample_utils.turn import get_ice_servers
logger = logging.getLogger(__name__)
class StreamingMP3ToFrames:
def __init__(self):
self.append = False
def process_chunk(self, chunk):
audio_frames = []
try:
if self.append:
self.bytes_io.write(chunk)
self.append = False
self.bytes_io.seek(0)
else:
self.bytes_io = io.BytesIO(chunk)
container = av.open(self.bytes_io, 'r', format='mp3')
audio_stream = next(s for s in container.streams if s.type == 'audio')
for frame in container.decode(audio_stream):
# Convert the audio frame to a NumPy array
array = frame.to_ndarray()
# Now you can use av.AudioFrame.from_ndarray
# audio_frame = av.AudioFrame.from_ndarray(array, format='flt', layout='mono')
audio_frame = av.AudioFrame.from_ndarray(array, format='fltp', layout='mono')
audio_frame.sample_rate = 44100
audio_frames.append(audio_frame)
return audio_frames
except Exception as e:
print (e)
self.append = True
self.bytes_io.seek(0, io.SEEK_END)
return audio_frames
def video_frame_callback(
frame: av.VideoFrame,
) -> av.VideoFrame:
return frame
streaming_mp3_to_frames = StreamingMP3ToFrames()
with open("chunks.pkl", "rb") as f:
import pickle
debug_chunks = pickle.load(f)
debug_frames = []
debug_frame_idx = 0
for chunk in debug_chunks:
new_frames = streaming_mp3_to_frames.process_chunk(chunk)
for frame in new_frames:
debug_frames.append(frame)
# print (frame)
def dequeue_frame():
global debug_frame_idx, debug_frames
enqueued_frame = debug_frames[debug_frame_idx]
debug_frame_idx += 1
if debug_frame_idx >= len(debug_frames):
debug_frame_idx = 0
return enqueued_frame
# emptry array of type int16
sample_buffer = np.zeros((0), dtype=np.int16)
def process_frame(old_frame):
try:
output_channels = 2
output_sample_rate = 44100
required_samples = old_frame.samples
global sample_buffer
while sample_buffer.shape[0] < required_samples:
dequeued_frame = dequeue_frame()
if dequeued_frame is None:
break
# convert dequeued_frame to same format as old_frame
float_samples = dequeued_frame.to_ndarray()
max_sample = np.max(np.abs(float_samples))
min_sample = np.min(np.abs(float_samples))
if max_sample > 1.0 or min_sample > 1.0:
print(f"WARNING: max_sample: {max_sample}, min_sample: {min_sample}")
int_samples = np.int16(float_samples * 32767)
sound = pydub.AudioSegment(
data=int_samples.tobytes(),
sample_width=2,
frame_rate=output_sample_rate,
channels=len(dequeued_frame.layout.channels),
)
sound = sound.set_frame_rate(old_frame.sample_rate)
samples = np.array(sound.get_array_of_samples(), dtype=np.int16)
sample_buffer = np.append(sample_buffer, samples)
# handle case where we ran out of frames
if sample_buffer.shape[0] < required_samples:
empty_samples = np.zeros((required_samples - sample_buffer.shape[0]), dtype=np.int16)
sample_buffer = np.append(sample_buffer, empty_samples)
# take the first required_samples samples from the buffer
samples = sample_buffer[:required_samples]
sample_buffer = sample_buffer[required_samples:]
# Duplicate mono channel for stereo
if output_channels == 2:
samples = np.vstack((samples, samples)).reshape((-1,), order='F')
samples = samples.reshape(1, -1)
layout = 'stereo' if output_channels == 2 else 'mono'
new_frame = av.AudioFrame.from_ndarray(samples, format='s16', layout=layout)
new_frame.sample_rate = old_frame.sample_rate
new_frame.pts = old_frame.pts
return new_frame
except Exception as e:
print (e)
traceback.print_exc()
raise(e)
def audio_frame_callback(old_frame: av.AudioFrame) -> av.AudioFrame:
global debug_frame_idx, debug_frames
new_frame = process_frame(old_frame)
# print (f"new_frames: {len(new_frames)}, frames: {len(frames)}")
print (f"frame: {old_frame}, pts: {old_frame.pts}")
print (f"new_frame: {new_frame}, pts: {new_frame.pts}")
return new_frame
# return old_frame
webrtc_streamer(
key="delay",
mode=WebRtcMode.SENDRECV,
rtc_configuration={"iceServers": get_ice_servers()},
video_frame_callback=video_frame_callback,
audio_frame_callback=audio_frame_callback,
)
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