Android 推送手机音频数据到RTMP服务器 AACtoRTMP

Android 推送手机音频数据到RTMP服务器 AACtoRTMP
Android平台上使用AudioRecord采集音视频数据,通过MediaCodec对音视频实施硬编码,回调aac数据流,然后通过librtmp推送到服务器,案例紧写了RTMP的,RTSP的照搬即可,以下是部分代码,具体代码可以看源码连接:https://github.com/printlybyte/AndroidPCMtoAAC_RTMP_RTSP

推流器:DefaultRtmpPublisher

public class DefaultRtmpPublisher implements RtmpPublisher {
   

    private RtmpConnection rtmpConnection;

    public DefaultRtmpPublisher(ConnectCheckerRtmp connectCheckerRtmp) {
        rtmpConnection = new RtmpConnection(connectCheckerRtmp);
    }

    @Override
    public boolean connect(String url) {
        return rtmpConnection.connect(url);
    }

    @Override
    public boolean publish(String publishType) {
        return rtmpConnection.publish(publishType);
    }

    @Override
    public void close() {
        rtmpConnection.close();
    }

    @Override
    public void publishVideoData(byte[] data, int size, int dts) {
        rtmpConnection.publishVideoData(data, size, dts);
    }

    @Override
    public void publishAudioData(byte[] data, int size, int dts) {
        rtmpConnection.publishAudioData(data, size, dts);
    }

    @Override
    public void setVideoResolution(int width, int height) {
        rtmpConnection.setVideoResolution(width, height);
    }

    @Override
    public void setAuthorization(String user, String password) {
        rtmpConnection.setAuthorization(user, password);
    }
}

连接器:RtmpConnection

public class RtmpConnection implements RtmpPublisher {
   

  private static final String TAG = "RtmpConnection";
  private static final Pattern rtmpUrlPattern =
      Pattern.compile("^rtmp://([^/:]+)(:(\\d+))*/([^/]+)(/(.*))*$");
  private static final Pattern rtmpsUrlPattern =
      Pattern.compile("^rtmps://([^/:]+)(:(\\d+))*/([^/]+)(/(.*))*$");

  private int port;
  private String host;
  private String appName;
  private String streamName;
  private String publishType;
  private String swfUrl;
  private String tcUrl;
  private String pageUrl;
  private Socket socket;
  private String socketExceptionCause = "";
  private RtmpSessionInfo rtmpSessionInfo;
  private RtmpDecoder rtmpDecoder;
  private BufferedInputStream inputStream;
  private BufferedOutputStream outputStream;
  private Thread rxPacketHandler;
  private volatile boolean connected = false;
  private volatile boolean publishPermitted = false;
  private final Object connectingLock = new Object();
  private final Object publishLock = new Object();
  private int currentStreamId = 0;
  private int transactionIdCounter = 0;
  private int videoWidth;
  private int videoHeight;
  private ConnectCheckerRtmp connectCheckerRtmp;
  //for secure transport
  private boolean tlsEnabled;
  //for auth
  private String user = null;
  private String password = null;
  private String salt = null;
  private String challenge = null;
  private String opaque = null;
  private boolean onAuth = false;

  public RtmpConnection(ConnectCheckerRtmp connectCheckerRtmp) {
    this.connectCheckerRtmp = connectCheckerRtmp;
  }

  private void handshake(InputStream in, OutputStream out) throws IOException {
    Handshake handshake = new Handshake();
    handshake.writeC0(out);
    handshake.writeC1(out); // Write C1 without waiting for S0
    out.flush();
    handshake.readS0(in);
    handshake.readS1(in);
    handshake.writeC2(out);
    out.flush();
    handshake.readS2(in);
  }

  @Override
  public boolean connect(String url) {
    Matcher rtmpMatcher = rtmpUrlPattern.matcher(url);
    Matcher rtmpsMatcher = rtmpsUrlPattern.matcher(url);
    Matcher matcher;
    if (rtmpMatcher.matches()) {
      matcher = rtmpMatcher;
      tlsEnabled = false;
    } else if (rtmpsMatcher.matches()) {
      matcher = rtmpsMatcher;
      tlsEnabled = true;
    } else {
      connectCheckerRtmp.onConnectionFailedRtmp(
          "Endpoint malformed, should be: rtmp://ip:port/appname/streamname");
      return false;
    }

    tcUrl = url.substring(0, url.lastIndexOf('/'));
    swfUrl = "";
    pageUrl = "";
    host = matcher.group(1);
    String portStr = matcher.group(3);
    port = portStr != null ? Integer.parseInt(portStr) : 1935;
    appName = matcher.group(4);
    streamName = matcher.group(6);

    // socket connection
    Log.d(TAG, "connect() called. Host: "
        + host
        + ", port: "
        + port
        + ", appName: "
        + appName
        + ", publishPath: "
        + streamName);
    rtmpSessionInfo = new RtmpSessionInfo();
    rtmpDecoder = new RtmpDecoder(rtmpSessionInfo);
    try {
      if (!tlsEnabled) {
        socket = new Socket();
        SocketAddress socketAddress = new InetSocketAddress(host, port);
        socket.connect(socketAddress, 3000);
      } else {
        socket = CreateSSLSocket.createSSlSocket(host, port);
        if (socket == null) throw new IOException("Socket creation failed");
      }
      inputStream = new BufferedInputStream(socket.getInputStream());
      outputStream = new BufferedOutputStream(socket.getOutputStream());
      Log.d(TAG, "connect(): socket connection established, doing handhake...");
      handshake(inputStream, outputStream);
      Log.d(TAG, "connect(): handshake done");
    } catch (IOException e) {
      Log.e(TAG, "Error", e);
      connectCheckerRtmp.onConnectionFailedRtmp("Connect error, " + e.getMessage());
      return false;
    }

    // Start the "main" handling thread
    rxPacketHandler = new Thread(new Runnable() {

      @Override
      public void run() {
        try {
          Log.d(TAG, "starting main rx handler loop");
          handleRxPacketLoop();
        } 
  • 0
    点赞
  • 6
    收藏
    觉得还不错? 一键收藏
  • 1
    评论
Android项目中使用FFmpeg来拉取RTSP流并RTMP服务器,可以按照以下步骤进行操作: 步骤1:添加FFmpeg库 首先,将FFmpeg库添加到Android项目中。可以使用开源的FFmpeg库,也可以使用第三方的FFmpeg库。在项目的build.gradle文件中添加FFmpeg库的依赖。 步骤2:初始化FFmpeg 在代码中初始化FFmpeg,可以在Application的onCreate()方法中进行初始化操作。 步骤3:拉取RTSP流 使用FFmpeg命令行或者FFmpeg库提供的API来拉取RTSP流。可以通过设置输入源和输出格式等参数,实现从RTSP服务器拉取流数据。 步骤4:RTMP服务器 使用FFmpeg命令行或者FFmpeg库提供的API来数据RTMP服务器。可以通过设置输出源和RTMP服务器地址等参数,实现将拉取到的流数据RTMP服务器。 步骤5:对过程进行监听和处理 通过FFmpeg库提供的回调函数,可以对过程进行监听和处理。例如,可以监听流传输的状态、的进度等。 步骤6:处理异常情况 在使用FFmpeg的过程中,可能会遇到各种异常情况,比如网络异常、编解码错误等。需要在代码中进行异常处理,保证项目的稳定性和可靠性。 需要注意的是,使用FFmpeg进行RTSP流拉取和RTMP需要一定的编程技巧和对FFmpeg库的理解。并且,由于FFmpeg库在不同平台和版本之间可能存在差异,因此在集成和使用FFmpeg库时,需要确保库的版本和平台的兼容性。

“相关推荐”对你有帮助么?

  • 非常没帮助
  • 没帮助
  • 一般
  • 有帮助
  • 非常有帮助
提交
评论 1
添加红包

请填写红包祝福语或标题

红包个数最小为10个

红包金额最低5元

当前余额3.43前往充值 >
需支付:10.00
成就一亿技术人!
领取后你会自动成为博主和红包主的粉丝 规则
hope_wisdom
发出的红包
实付
使用余额支付
点击重新获取
扫码支付
钱包余额 0

抵扣说明:

1.余额是钱包充值的虚拟货币,按照1:1的比例进行支付金额的抵扣。
2.余额无法直接购买下载,可以购买VIP、付费专栏及课程。

余额充值