可能在 android 中使用（不是 mediaPlayer，而是其他更底层的东西），但是当媒体生态系统的其他部分不这样做时，你真的想要追求 RTSP/RTP 吗？？
IMO - 在 HTML5/WebRTC 的保护伞下有更好的媒体/流方法。比如看看'Ondello' 对流做了什么。
也就是说，这里是一些使用“netty”和“efflux”的 android/RTSP/SDP/RTP 的旧项目代码。它将在 SDP 文件提供者上协商“会话”的某些部分。不记得它是否真的会播放Youtube/RTSP stuff 的音频部分，但那是我当时的目标。 （我认为它使用 AMR-NB 编解码器可以工作，但是有很多问题，我像一个坏习惯一样在 android 上放弃了 RTSP！）
on Git....
@Override
public void mediaDescriptor(Client client, String descriptor)
{
// searches for control: session and media arguments.
final String target = "control:";
Log.d(TAG, "Session Descriptor\n" + descriptor);
int position = -1;
while((position = descriptor.indexOf(target)) > -1)
{
descriptor = descriptor.substring(position + target.length());
resourceList.add(descriptor.substring(0, descriptor.indexOf('\r')));
}
}
private int nextPort()
{
return (port += 2) - 2;
}
private void getRTPStream(TransportHeader transport){
String[] words;
// only want 2000 part of 'client_port=2000-2001' in the Transport header in the response
words = transport.getParameter("client_port").substring(transport.getParameter("client_port").indexOf("=") +1).split("-");
port_lc = Integer.parseInt(words[0]);
words = transport.getParameter("server_port").substring(transport.getParameter("server_port").indexOf("=") +1).split("-");
port_rm = Integer.parseInt(words[0]);
source = transport.getParameter("source").substring(transport.getParameter("source").indexOf("=") +1);
ssrc = transport.getParameter("ssrc").substring(transport.getParameter("ssrc").indexOf("=") +1);
// assume dynamic Packet type = RTP , 99
getRTPStream(session, source, port_lc, port_rm, 99);
//getRTPStream("sessiona", source, port_lc, port_rm, 99);
Log.d(TAG, "raw parms " +port_lc +" " +port_rm +" " +source );
// String[] words = session.split(";");
Log.d(TAG, "session: " +session);
Log.d(TAG, "transport: " +transport.getParameter("client_port")
+" " +transport.getParameter("server_port") +" " +transport.getParameter("source")
+" " +transport.getParameter("ssrc"));
}
private void getRTPStream(String session, String source, int portl, int portr, int payloadFormat ){
// what do u do with ssrc?
InetAddress addr;
try {
addr = InetAddress.getLocalHost();
// Get IP Address
// LAN_IP_ADDR = addr.getHostAddress();
LAN_IP_ADDR = "192.168.1.125";
Log.d(TAG, "using client IP addr " +LAN_IP_ADDR);
} catch (UnknownHostException e1) {
// TODO Auto-generated catch block
e1.printStackTrace();
}
final CountDownLatch latch = new CountDownLatch(2);
RtpParticipant local1 = RtpParticipant.createReceiver(new RtpParticipantInfo(1), LAN_IP_ADDR, portl, portl+=1);
// RtpParticipant local1 = RtpParticipant.createReceiver(new RtpParticipantInfo(1), "127.0.0.1", portl, portl+=1);
RtpParticipant remote1 = RtpParticipant.createReceiver(new RtpParticipantInfo(2), source, portr, portr+=1);
remote1.getInfo().setSsrc( Long.parseLong(ssrc, 16));
session1 = new SingleParticipantSession(session, payloadFormat, local1, remote1);
Log.d(TAG, "remote ssrc " +session1.getRemoteParticipant().getInfo().getSsrc());
session1.init();
session1.addDataListener(new RtpSessionDataListener() {
@Override
public void dataPacketReceived(RtpSession session, RtpParticipantInfo participant, DataPacket packet) {
// System.err.println("Session 1 received packet: " + packet + "(session: " + session.getId() + ")");
//TODO close the file, flush the buffer
// if (_sink != null) _sink.getPackByte(packet);
getPackByte(packet);
// System.err.println("Ssn 1 packet seqn: typ: datasz " +packet.getSequenceNumber() + " " +packet.getPayloadType() +" " +packet.getDataSize());
// System.err.println("Ssn 1 packet sessn: typ: datasz " + session.getId() + " " +packet.getPayloadType() +" " +packet.getDataSize());
// latch.countDown();
}
});
// DataPacket packet = new DataPacket();
// packet.setData(new byte[]{0x45, 0x45, 0x45, 0x45});
// packet.setSequenceNumber(1);
// session1.sendDataPacket(packet);
// try {
// latch.await(2000, TimeUnit.MILLISECONDS);
// } catch (Exception e) {
// fail("Exception caught: " + e.getClass().getSimpleName() + " - " + e.getMessage());
// }
}
//TODO below should collaborate with the audioTrack object and should write to the AT buffr
// audioTrack write was blocking forever
public void getPackByte(DataPacket packet) {
//TODO this is getting called but not sure why only one time
// or whether it is stalling in mid-exec??
//TODO on firstPacket write bytes and start audioTrack
// AMR-nb frames at 12.2 KB or format type 7 frames are handled .
// after the normal header, the getDataArray contains extra 10 bytes of dynamic header that are bypassed by 'limit'
// real value for the frame separator comes in the input stream at position 1 in the data array
// returned by
// int newFrameSep = 0x3c;
// bytes avail = packet.getDataSize() - limit;
// byte[] lbuf = new byte[packet.getDataSize()];
// if ( packet.getDataSize() > 0)
// lbuf = packet.getDataAsArray();
//first frame includes the 1 byte frame header whose value should be used
// to write subsequent frame separators
Log.d(TAG, "getPackByt start and play");
if(!started){
Log.d(TAG, " PLAY audioTrak");
track.play();
started = true;
}
// track.write(packet.getDataAsArray(), limit, (packet.getDataSize() - limit));
track.write(packet.getDataAsArray(), 0, packet.getDataSize() );
Log.d(TAG, "getPackByt aft write");
// if(!started && nBytesRead > minBufferSize){
// Log.d(TAG, " PLAY audioTrak");
// track.play();
// started = true;}
nBytesRead += packet.getDataSize();
if (nBytesRead % 500 < 375) Log.d(TAG, " getPackByte plus 5K received");
}
}