基于SRS服务器实现Android-Web端视频通话（3）：Android端向SRS服务器推送WebRTC流

冬季穿短裤

已于 2022-09-16 22:07:25 修改

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分类专栏： WebRTC 文章标签： android webrtc srs

于 2021-09-18 17:52:38 首次发布

本文链接：https://blog.csdn.net/csdn_shen0221/article/details/120331004

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基于SRS服务器实现Android-Web端视频通话（3）：Android端向SRS服务器推送WebRTC流

实现效果
引库
推流流程
代码实现
Github传送门

基于SRS服务器实现Android-Web端视频通话（1）：SRS服务器启用HTTPS
基于SRS服务器实现Android-Web端视频通话（2）：Android端从SRS服务器拉取WebRTC流
 基于SRS服务器实现Android-Web端视频通话（3）：Android端向SRS服务器推送WebRTC流

实现效果

在这里插入图片描述

引库

implementation 'org.webrtc:google-webrtc:1.0.32006'

其他版本，详见

推流流程

createPeerConnectionFactory -> createPeerConnection(addTransceiver) -> createOffer -> setLocalDescription(OFFER) -> get remote sdp(network requset) -> setRemoteDescription(ANSWER)

代码实现

初始化

//加载并初始化 WebRTC，在创建 PeerConnectionFactory 之前必须至少调用一次
PeerConnectionFactory.initialize(
    PeerConnectionFactory.InitializationOptions
        .builder(applicationContext).createInitializationOptions()
)

private val eglBaseContext = EglBase.create().eglBaseContext

createPeerConnectionFactory

private val peerConnectionFactory: PeerConnectionFactory = createPeerConnectionFactory()

private fun createPeerConnectionFactory(): PeerConnectionFactory {
	//先做默认配置，后面可能会遇到坑
    val options = PeerConnectionFactory.Options()
    val encoderFactory = DefaultVideoEncoderFactory(eglBaseContext, true, true)
    val decoderFactory = DefaultVideoDecoderFactory(eglBaseContext)
    return PeerConnectionFactory.builder()
        .setOptions(options)
        .setVideoEncoderFactory(encoderFactory)
        .setVideoDecoderFactory(decoderFactory)
        .createPeerConnectionFactory()
}

createPeerConnection(addTransceiver)

private fun initPeerConnection() {
    val createAudioSource = peerConnectionFactory.createAudioSource(createAudioConstraints())
    val audioTrack =
        peerConnectionFactory.createAudioTrack("local_audio_track", createAudioSource)

    cameraVideoCapturer = createVideoCapture(this)
    cameraVideoCapturer?.let { capture ->
        val videoSource = peerConnectionFactory.createVideoSource(capture.isScreencast)
        videoTrack =
            peerConnectionFactory.createVideoTrack("local_video_track", videoSource).apply {
            	//显示到本地画面上
                addSink(mBinding.svr)
            }
        surfaceTextureHelper =
            SurfaceTextureHelper.create("surface_texture_thread", eglBaseContext)
        capture.initialize(surfaceTextureHelper, this, videoSource.capturerObserver)
        //开始捕帧，宽、高、帧率。
        capture.startCapture(640, 480, 20)
    }

    val rtcConfig = PeerConnection.RTCConfiguration(emptyList())
    /*
 	<p>For users who wish to send multiple audio/video streams and need to stay interoperable with legacy WebRTC implementations, specify PLAN_B.
 	<p>For users who wish to send multiple audio/video streams and/or wish to use the new RtpTransceiver API, specify UNIFIED_PLAN.
 	*/
	//使用PeerConnection.SdpSemantics.UNIFIED_PLAN
    rtcConfig.sdpSemantics = PeerConnection.SdpSemantics.UNIFIED_PLAN

    peerConnection = peerConnectionFactory.createPeerConnection(
        rtcConfig,
        PeerConnectionObserver()
    )?.apply {
    	//不可使用addStream()的方式，否则会报错
    	//addTransceiver()调用顺序会影响到生成sdp信息中video、audio信息顺序，对应关系，下面会提到；
        videoTrack?.let {
        	//添加视频轨，设置仅发送即可
            addTransceiver(
                it,
                RtpTransceiver.RtpTransceiverInit(RtpTransceiver.RtpTransceiverDirection.SEND_ONLY)
            )
        }
        //添加音频轨，设置仅发送即可
        addTransceiver(
            audioTrack,
            RtpTransceiver.RtpTransceiverInit(RtpTransceiver.RtpTransceiverDirection.SEND_ONLY)
        )
    }
}

private fun createAudioConstraints(): MediaConstraints {
    val audioConstraints = MediaConstraints()
    //回声消除
    audioConstraints.mandatory.add(
        MediaConstraints.KeyValuePair(
            "googEchoCancellation",
            "true"
        )
    )
    //自动增益
    audioConstraints.mandatory.add(MediaConstraints.KeyValuePair("googAutoGainControl", "true"))
    //高音过滤
    audioConstraints.mandatory.add(MediaConstraints.KeyValuePair("googHighpassFilter", "true"))
    //噪音处理
    audioConstraints.mandatory.add(
        MediaConstraints.KeyValuePair(
            "googNoiseSuppression",
            "true"
        )
    )
    return audioConstraints
}

private fun createVideoCapture(context: Context): CameraVideoCapturer? {
    val enumerator: CameraEnumerator = if (Camera2Enumerator.isSupported(context)) {
        Camera2Enumerator(context)
    } else {
        Camera1Enumerator()
    }
    for (name in enumerator.deviceNames) {
        if (enumerator.isFrontFacing(name)) {
            return enumerator.createCapturer(name, null)
        }
    }
    for (name in enumerator.deviceNames) {
        if (enumerator.isBackFacing(name)) {
            return enumerator.createCapturer(name, null)
        }
    }
    return null
}

createOffer && setLocalDescription

peerConnection.createOffer(object : SdpAdapter("createOffer") {
	override fun onCreateSuccess(description: SessionDescription?) {
		description?.let {
			if (it.type == SessionDescription.Type.OFFER) {
				connection.setLocalDescription(SdpAdapter("setLocalDescription"), it)
				//这个offerSdp将用于向SRS服务进行网络请求
				val offerSdp = it.description
				getRemoteSdp(offerSdp)
			}
		}
	}
}, MediaConstraints())

get remote sdp(netword requset)

基本配置，根据自己实际情况进行调整

object Constant {
    /**
     * SRS服务器IP
     */
    const val SRS_SERVER_IP = "192.168.2.91"

    /**
     * SRS服务http请求端口，默认1985
     */
    const val SRS_SERVER_HTTP_PORT = "1985"

    /**
     * SRS服务https请求端口，默认1990
     */
    const val SRS_SERVER_HTTPS_PORT = "1990"

    const val SRS_SERVER_HTTP = "$SRS_SERVER_IP:$SRS_SERVER_HTTP_PORT"

    const val SRS_SERVER_HTTPS = "$SRS_SERVER_IP:$SRS_SERVER_HTTPS_PORT"
}

Request Body (application/json)

data class SrsRequestBean(
    /**
     * [PeerConnection.createOffer]返回的sdp
     */
    @Json(name = "sdp")
    val sdp: String?,
    /**
     * 拉取的WebRTC流地址
     */
    @Json(name = "streamurl")
    val streamUrl: String?
)

Response Body (application/json)

data class SrsResponseBean(
    /**
     * 0：成功
     */
    @Json(name = "code")
    val code: Int,
    /**
     * 用于设置[PeerConnection.setRemoteDescription]
     */
    @Json(name = "sdp") val sdp: String?,
    @Json(name = "server")
    val server: String?,
    @Json(name = "sessionid")
    val sessionId: String?
)

网络请求地址
http请求：http://ip:port/rtc/v1/publish/
https请求：https://ip:port/rtc/v1/publish/
Method：POST

在Android P(28)系统的设备上，禁止应用使用的是非加密的明文流量的HTTP 网络请求。

retrofit事例

interface ApiService {

    @POST("/rtc/v1/publish/")
    suspend fun publish(@Body body: SrsRequestBean): SrsResponseBean
}

getRemoteSdp

private fun getRemoteSdp(offerSdp: String){
	//webrtc流地址
	val webrtcUrl="webrtc://${Constant.SRS_SERVER_IP}/live/camera"
    val srsBean = SrsRequestBean(offerSdp, webrtcUrl)
    lifecycleScope.launch {
        val result = try {
            withContext(Dispatchers.IO) {
                retrofitClient.apiService.publish(srsBean)
            }
        } catch (e: Exception) {
            println("网络请求出错：${e.printStackTrace()}")
            toastError("网络请求出错：${e.printStackTrace()}")
            null
        }

        result?.let { bean ->
            if (bean.code == 0) {
                println("网络请求成功，code：${bean.code}")
				setRemoteDescription(bean.sdp)
            } else {
                println("网络请求失败，code：${bean.code}")
            }
        }
    }
}

注意：这一步接口可能会返回code:400的情况，如果没有可跳过。

接口返回400 原因可能是：

1、推流地址被占用，解决方案换个推流地址即可；
2、服务器中打印create session : create session : add publisher : publish negotiate : no found valid H.264 payload type，原因：就是接口请求里offer sdp中m=video无H.264相关信息，即WebRTC在createOffer时，返回的sdp没有H.264相关信息；SRS中使用WebRTC，视频编码仅支持H.264。

针对第二种情况，解决方案可参考我的另外一篇博客。
这样createPeerConnectionFactory就要改一下：
引库WebRTCExtension：

private val peerConnectionFactory: PeerConnectionFactory = createPeerConnectionFactory()

private fun createPeerConnectionFactory(): PeerConnectionFactory {
    val options = PeerConnectionFactory.Options()
    //调整VideoEncoderFactory
	val encoderFactory = createCustomVideoEncoderFactory(eglBaseContext, true, true,
            object : VideoEncoderSupportedCallback {
                override fun isSupportedH264(info: MediaCodecInfo): Boolean {
                	//这里自行添加支持H.264编码的MediaCodecInfo
                    return true
                }

            })
    val decoderFactory = DefaultVideoDecoderFactory(eglBaseContext)
    return PeerConnectionFactory.builder()
        .setOptions(options)
        .setVideoEncoderFactory(encoderFactory)
        .setVideoDecoderFactory(decoderFactory)
        .createPeerConnectionFactory()
}

setRemoteDescription

private fun setRemoteDescription(answerSdp: String){
	val remoteSdp = SessionDescription(SessionDescription.Type.ANSWER, /*关键点*/answerSdp)
	//注意这一步，可能会报错：Failed to set remote answer sdp: The order of m-lines in answer doesn't match order in offer. Rejecting answer.
	peerConnection.setRemoteDescription(SdpAdapter("setRemoteDescription"), remoteSdp)
}

如果你遇到这个错误：
Failed to set remote answer sdp: The order of m-lines in answer doesn’t match order in offer. Rejecting answer.

可以看下我的另外一篇博客。
具体原因就是offer中sdp的video、audio和answer中sdp的video、audio顺序不一致，详见#3179。
解决方案目前大致有三种：
1、升级SRS服务器版本，截止2022-09-16，升级到v4.0.265及以上，即可解决，云服务器这个版本做了兼容处理；
2、如果无法升级SRS，可以调整上面提到的调整**addTransceiver()**调用顺序，确保顺序与SRS接口返回的sdp中video、audio顺序一致；

ps: 上面两种方案解决方式还要非常感谢SRS-杨成立指点，十分感谢！

3、使用博客中第一种原因的解决方式，我们需要手动调换下位置（当然这个方法也可以作为最后的保护手段）：

/**
 * 转换AnswerSdp
 * @param offerSdp offerSdp：创建offer时生成的sdp
 * @param answerSdp answerSdp：网络请求srs服务器返回的sdp
 * @return 转换后的AnswerSdp
 */
private fun convertAnswerSdp(offerSdp: String, answerSdp: String?): String {
	if (answerSdp.isNullOrBlank()){
		return ""
    }
    val indexOfOfferVideo = offerSdp.indexOf("m=video")
    val indexOfOfferAudio = offerSdp.indexOf("m=audio")
    if (indexOfOfferVideo == -1 || indexOfOfferAudio == -1) {
        return answerSdp
    }
    val indexOfAnswerVideo = answerSdp.indexOf("m=video")
    val indexOfAnswerAudio = answerSdp.indexOf("m=audio")
    if (indexOfAnswerVideo == -1 || indexOfAnswerAudio == -1) {
        return answerSdp
    }

    val isFirstOfferVideo = indexOfOfferVideo < indexOfOfferAudio
    val isFirstAnswerVideo = indexOfAnswerVideo < indexOfAnswerAudio
    return if (isFirstOfferVideo == isFirstAnswerVideo) {
        //顺序一致
        answerSdp
    } else {
        //需要调换顺序
        buildString {
            append(answerSdp.substring(0, indexOfAnswerVideo.coerceAtMost(indexOfAnswerAudio)))
            append(
                answerSdp.substring(
                    indexOfAnswerVideo.coerceAtLeast(indexOfOfferVideo),
                    answerSdp.length
                )
            )
            append(
                answerSdp.substring(
                    indexOfAnswerVideo.coerceAtMost(indexOfAnswerAudio),
                    indexOfAnswerVideo.coerceAtLeast(indexOfOfferVideo)
                )
            )
        }
    }
}

修改方法：

private fun setRemoteDescription(offerSdp: String, answerSdp: String){
	val remoteSdp = SessionDescription(SessionDescription.Type.ANSWER, /*关键点*/convertAnswerSdp(offerSdp, answerSdp))
	peerConnection.setRemoteDescription(SdpAdapter("setRemoteDescription"), remoteSdp)
}

关闭

释放资源，避免内存泄漏

mBinding.svr.release()
cameraVideoCapturer?.dispose()
surfaceTextureHelper?.dispose()
videoTrack?.dispose()
peerConnection?.dispose()
peerConnectionFactory.dispose()

至此，推流流程结束。如有错误欢迎指正。

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基于SRS服务器实现Android-Web端视频通话（3）：Android端向SRS服务器推送WebRTC流

基于SRS服务器实现Android-Web端视频通话（3）：Android端向SRS服务器推送WebRTC流
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