temp_file.py

"""
recorder based on streamlit-webrtc 
streamlit run st_recorder.py --server.port 8606
"""
from streamlit_webrtc import (
    webrtc_streamer,
    WebRtcMode,
    WebRtcStreamerContext,
)
from aiortc.contrib.media import MediaRecorder
import soundfile as sf
import numpy as np
import matplotlib.pyplot as plt
import streamlit as st
import queue
from pathlib import Path
import time
import pydub

# from streamlit_lottie import st_lottie
import json

# file_ = '16581-audio.json'
# with open(file_, 'r', encoding='utf-8') as f:
#     lottie_json = json.load(f)

TMP_DIR = Path('temp')
if not TMP_DIR.exists():
    TMP_DIR.mkdir(exist_ok=True, parents=True)

MEDIA_STREAM_CONSTRAINTS = {
    "video": False,
    "audio": {
        # these setting doesn't work
        # "sampleRate": 48000,
        # "sampleSize": 16,
        # "channelCount": 1,
        "echoCancellation": False,  # don't turn on else it would reduce wav quality
        "noiseSuppression": True,
        "autoGainControl": True,
    },
}


def aiortc_audio_recorder(wavpath):
    def recorder_factory():
        return MediaRecorder(wavpath)

    webrtc_ctx: WebRtcStreamerContext = webrtc_streamer(
        key="sendonly-audio",
        # mode=WebRtcMode.SENDONLY,
        mode=WebRtcMode.SENDRECV,
        in_recorder_factory=recorder_factory,
        media_stream_constraints=MEDIA_STREAM_CONSTRAINTS,
    )


def save_frames_from_audio_receiver(wavpath):
    webrtc_ctx = webrtc_streamer(
        key="sendonly-audio",
        mode=WebRtcMode.SENDONLY,
        media_stream_constraints=MEDIA_STREAM_CONSTRAINTS,
    )

    if "audio_buffer" not in st.session_state:
        st.session_state["audio_buffer"] = pydub.AudioSegment.empty()

    status_indicator = st.empty()
    lottie = False
    while True:
        if webrtc_ctx.audio_receiver:
            try:
                audio_frames = webrtc_ctx.audio_receiver.get_frames(timeout=1)
            except queue.Empty:
                status_indicator.info("No frame arrived.")
                continue

            # if not lottie:  # voice gif
            #     st_lottie(lottie_json, height=80)
            #     lottie = True

            for i, audio_frame in enumerate(audio_frames):
                sound = pydub.AudioSegment(
                    data=audio_frame.to_ndarray().tobytes(),
                    sample_width=audio_frame.format.bytes,
                    frame_rate=audio_frame.sample_rate,
                    channels=len(audio_frame.layout.channels),
                )
                # st.markdown(f'{len(audio_frame.layout.channels)}, {audio_frame.format.bytes}, {audio_frame.sample_rate}')
                # 2, 2, 48000
                st.session_state["audio_buffer"] += sound
        else:
            lottie = True
            break

    audio_buffer = st.session_state["audio_buffer"]

    if not webrtc_ctx.state.playing and len(audio_buffer) > 0:
        audio_buffer.export(wavpath, format="wav")
        st.session_state["audio_buffer"] = pydub.AudioSegment.empty()


def display_wavfile(wavpath):
    audio_bytes = open(wavpath, 'rb').read()
    file_type = Path(wavpath).suffix
    st.audio(audio_bytes, format=f'audio/{file_type}', start_time=0)


def plot_wav(wavpath):
    audio, sr = sf.read(str(wavpath))
    fig = plt.figure()
    plt.plot(audio)
    plt.xticks(
        np.arange(0, audio.shape[0], sr / 2), np.arange(0, audio.shape[0] / sr, 0.5)
    )
    plt.xlabel('time')
    st.pyplot(fig)


def record_page():
    st.markdown('# recorder')
    if "wavpath" not in st.session_state:
        cur_time = time.strftime("%Y-%m-%d_%H:%M:%S", time.localtime())
        tmp_wavpath = TMP_DIR / f'{cur_time}.wav'
        st.session_state["wavpath"] = str(tmp_wavpath)

    wavpath = st.session_state["wavpath"]

    aiortc_audio_recorder(wavpath)  # first way
    # save_frames_from_audio_receiver(wavpath)  # second way

    if Path(wavpath).exists():
        st.markdown(wavpath)
        display_wavfile(wavpath)
        plot_wav(wavpath)


if __name__ == "__main__":
    record_page()