以下是使用 Live555 库进行 RTMP 推流的示例代码，需要在项目中引入 Live555 库：
```csharp
using System;
using System.Collections.Generic;
using System.Linq;
using System.Text;
using System.Threading.Tasks;
using System.IO;
using Live;
using Live.LiveMedia;
namespace RtmpPusher
{
class RtmpPusher
{
private RTMPClient rtmpClient;
private H264VideoStreamSource videoSource;
public RtmpPusher()
{
rtspClient = new RTSPClient(Environment.GetCommandLineArgs()[1], 0, null, 0, -1);
videoSource = H264VideoStreamSource.createNew(Envir.instance(), null);
videoSource.setFramerate(30);
videoSource.start();
videoSource.getNextFrame(true);
rtspClient.setStreamSource(videoSource, null, null, null);
rtspClient.sendDescribeCommand(new RTSPClient.DescribeCallback(describeCallback));
}
private void describeCallback(RTSPClient.clientContinueFunc continueFunc)
{
rtspClient.sendSetupCommand(0, new RTSPClient.SetupCallback(setupCallback));
}
private void setupCallback(RTSPClient.clientContinueFunc continueFunc, int resultCode, string resultString, Session session, MediaSubsession subsession)
{
rtspClient.sendPlayCommand(continueFunc, 0.0f, -1.0f, 1.0f);
rtspClient.sendPauseCommand(continueFunc, 0.0f);
rtspClient.sendPauseCommand(continueFunc, 0.0f);
rtspClient.sendPlayCommand(continueFunc, 0.0f, -1.0f, 1.0f);
rtspClient.sendPauseCommand(continueFunc, 0.0f);
rtspClient.sendPauseCommand(continueFunc, 0.0f);
rtspClient.sendPlayCommand(continueFunc, 0.0f, -1.0f, 1.0f);
rtspClient.sendPauseCommand(continueFunc, 0.0f);
rtspClient.sendPauseCommand(continueFunc, 0.0f);
rtspClient.sendPlayCommand(continueFunc, 0.0f, -1.0f, 1.0f);
rtspClient.sendOptionsCommand(continueFunc, 0);
rtspClient.sendAnnounceCommand(continueFunc, null);
rtspClient.sendRecordCommand(continueFunc, null);
rtspClient.sendTeardownCommand(continueFunc, null);
rtspClient.sendOptionsCommand(continueFunc, 1);
rtspClient.sendSetParameterCommand(continueFunc, null);
rtspClient.sendGetParameterCommand(continueFunc, null);
rtspClient.sendPauseCommand(continueFunc, 0.0f);
rtspClient.sendPlayCommand(continueFunc, 0.0f, -1.0f, 1.0f);
rtspClient.sendGetParameterCommand(continueFunc, null);
}
public void pushStream()
{
// 指定 RTMP 服务器地址和应用名称以及流名称
rtspClient.sendAnnounceCommand(new RTSPClient.AnnounceCallback(announceCallback),
"rtmp://server-ip-address/application/stream-name",
"rtmp",
"video/avc",
"H264",
"96",
null);
}
private void announceCallback(RTSPClient.clientContinueFunc continueFunc, int resultCode, string resultString)
{
rtspClient.sendSetParameterCommand(continueFunc, null);
rtspClient.sendOptionsCommand(continueFunc, 1);
rtspClient.sendGetParameterCommand(continueFunc, null);
rtspClient.sendPauseCommand(continueFunc, 0.0f);
rtspClient.sendPlayCommand(continueFunc, 0.0f, -1.0f, 1.0f);
rtspClient.sendGetParameterCommand(continueFunc, null);
}
public void close()
{
rtspClient.sendTeardownCommand(null, null);
}
}
}
```
这个示例代码使用了 Live555 库中的 RTSPClient 和 H264VideoStreamSource 类，通过 RTSPClient 从 RTSP 服务器获取视频流，然后使用 H264VideoStreamSource 将视频流转换为 H.264 编码格式，并将其推送到 RTMP 服务器上。需要注意的是，这只是一个简单的示例，实际应用中还需要根据具体需求进行修改和调整。
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