请教下webtrc实时音频的一些问题，代码如下 #336

lin-jj-engine · 2024-12-24T09:19:03Z

import wave

import numpy as np
from aiortc import RTCPeerConnection, RTCSessionDescription, AudioStreamTrack
from aiohttp import web
import aiohttp_cors
import asyncio

from av import AudioFrame


class AudioFileTrack(AudioStreamTrack):
    """
    从 WAV 文件中读取音频数据的自定义音频轨道。
    """
    kind = "audio"

    def __init__(self, filename: str):
        super().__init__()
        self._filename = filename
        self._wavefile = wave.open(self._filename, "rb")  # 打开WAV文件
        self._frame_rate = self._wavefile.getframerate()  # 获取帧率
        self._frame_size = self._wavefile.getnframes()  # 获取文件的总帧数
        print(self._frame_rate,self._frame_size)
        self._timestamp = 0  # 初始化时间戳

    async def recv(self):
        """
        从 WAV 文件读取音频帧，并返回音频帧。
        """
        print(2)
        # 从 WAV 文件读取音频数据
        frames = self._wavefile.readframes(320)  # 读取 1024 帧数据
        if len(frames) == 0:
            # 如果没有数据了，返回空的音频帧
            return AudioFrame(format="s16", layout="mono", samples=0)

        # 构造音频帧
        frame = AudioFrame(format="s16", layout="mono", samples=320)
        frame.pts = self._timestamp
        print(self._timestamp/16000)
        frame.sample_rate = self._frame_rate
        frame.time_base = fractions.Fraction(1, self._frame_rate)
        print(frame.time_base)
        self._timestamp += 320  # 增加时间戳
        return frame


async def offer(request):
    params = await request.json()
    offer_sdp = params['offer']

    pc = RTCPeerConnection()

    # 创建视频流轨道
    video_track = AudioFileTrack("test.wav")  # 设置视频路径
    pc.addTrack(video_track)
    # 设置远程描述
    offer = RTCSessionDescription(sdp=offer_sdp, type=params['type'])
    await pc.setRemoteDescription(offer)

    # 创建并发送答案
    answer = await pc.createAnswer()
    await pc.setLocalDescription(answer)
    return web.json_response({"sdp": pc.localDescription.sdp, "type": pc.localDescription.type})


async def init():
    app = web.Application()
    cors = aiohttp_cors.setup(app, defaults={
        "*": aiohttp_cors.ResourceOptions(
            allow_credentials=True,
            expose_headers="*",
            allow_headers="*",
            # allow_methods="*",
        )
    })
    resource = cors.add(app.router.add_resource("/offer"))
    cors.add(resource.add_route("POST", offer))
    runner = web.AppRunner(app)
    await runner.setup()
    site = web.TCPSite(runner, '0.0.0.0', 8900)
    await site.start()


print('------- Http server started at localhost:8900 --------')

# def main():
if __name__ == '__main__':
    loop = asyncio.new_event_loop()  # get_event_loop()
    loop.run_until_complete(init())
    loop.run_forever()

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lin-jj-engine · 2024-12-24T09:21:00Z

前端偶尔能播放出音频，也写过视频的，视频也是偶尔会播放，是为什么呢，实在是不太懂，想问大佬请教下，感谢

lin-jj-engine · 2024-12-24T09:26:10Z

<html lang="en">
	<head>
		<meta charset="UTF-8">
		<meta name="viewport" content="width=device-width, initial-scale=1.0">
		<title>WebRTC Streaming</title>
	</head>
	<body>
		<h2>WebRTC Video Streaming</h2>
		<video id="video" autoplay playsinline></video> <!-- 用于显示视频流 -->
		<h2>WebRTC Video Streaming</h2>
		<audio id="remoteAudio" autoplay></audio> <!-- 用于播放音频流 -->
		<button onclick="startAudio()">Play Audio</button>
		<h2>WebRTC Video Streaming</h2>
		<script>
			const videoElement = document.getElementById("video");
			const remoteAudio = document.getElementById('remoteAudio');
			remoteAudio.muted = false;
			let pc = null;

			// 配置 STUN 服务器（Google 提供的公共 STUN 服务器）
			const config = {
				iceServers: [{
						urls: "stun:stun1.l.google.com:19302"
					} // 公共 STUN 服务器
				]
			};

			// 创建 RTCPeerConnection
			async function start() {
				pc = new RTCPeerConnection(config);

				// 接收并显示视频轨道
				pc.addEventListener('track', (event) => {

					if (event.track.kind === 'audio') {
						console.log('Audio track received.');
						remoteAudio.srcObject = event.streams[0]; // 播放音频流
						remoteAudio.play().catch((err) => {
							console.error('Error playing audio:', err);
						});
						console.log('Stream', event.streams[0]);
						console.log('Stream Active:', event.streams[0].active);
						console.log('Stream Tracks:', event.streams[0].getTracks());

					}
				});

				// 创建本地 offer，并发送给服务器
				const offer = await pc.createOffer({
					offerToReceiveVideo: true,
					offerToReceiveAudio: true
				});
				await pc.setLocalDescription(offer);

				// 向后端发送 offer 并接收 answer
				const response = await fetch('http://localhost:8900/offer', {
					method: 'POST',
					headers: {
						'Content-Type': 'application/json'
					},
					body: JSON.stringify({
						offer: pc.localDescription.sdp,
						type: pc.localDescription.type
					})
				});

				const data = await response.json();

				// 设置远程描述（answer）
				const answer = new RTCSessionDescription({
					type: 'answer',
					sdp: data.sdp
				});
				await pc.setRemoteDescription(answer);
			}

			function startAudio() {
				const remoteAudio = document.getElementById('remoteAudio');
				remoteAudio.play().catch(error => {
					console.log('Audio play failed:', error);
				});
			}
			// 启动 WebRTC 流
			start();
		</script>
	</body>
</html>

lipku · 2024-12-24T11:02:06Z

参考一下aiortc里的例子

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请教下webtrc实时音频的一些问题，代码如下 #336

请教下webtrc实时音频的一些问题，代码如下 #336

lin-jj-engine commented Dec 24, 2024 •

edited

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lin-jj-engine commented Dec 24, 2024

lin-jj-engine commented Dec 24, 2024

lipku commented Dec 24, 2024

请教下webtrc实时音频的一些问题，代码如下 #336

请教下webtrc实时音频的一些问题，代码如下 #336

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lin-jj-engine commented Dec 24, 2024 • edited Loading

lin-jj-engine commented Dec 24, 2024

lin-jj-engine commented Dec 24, 2024

lipku commented Dec 24, 2024

lin-jj-engine commented Dec 24, 2024 •

edited

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