我正在尝试将音频从Raspberry Pi流式传输到VM。
Raspberry Pi插入了一个麦克风，其管道就像 因此（已删除IP /主机名信息）：
gst-launch-1.0 -ev alsasrc device=plughw:1,0 ! audioconvert ! rtpL24pay ! udpsink host=xxxxx port=xxxx
VM正在运行以下管道：
gst-launch-1.0 -ev udpsrc port=xxxx caps="application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L24, encoding-params=(string)2, channels=(int)2, payload=(int)96, ssrc=(uint)636287891, timestamp-offset=(uint)692362821, seqnum-offset=(uint)11479" ! rtpL24depay ! decodebin ! audioconvert ! wavenc ! filesink location=test.wav
当我用Ctrl + C结束时，通过命令行运行它就可以了 （与-e开关结合使用），并且文件可读。我想做的事 但是，通过命令行保持管道在Raspberry pi上运行， 但是将Java应用程序用于VM的管道。该Java应用程序是 挂钩到REST端点“ / start”和“ / stop”。 “ / start”启动管道 和“ / stop” 应该停止管道并写入文件，但是当命中“ / stop”端点时，文件的大小为零，并且不可读。我的代码在这篇文章的底部。关于改进管道或如何使文件可读的任何想法都将是很棒的。我最初的想法是，这与我发送EOS消息的方式有关，但并不太确定。谢谢！
编辑：还忘了提及我在Docker容器中运行此JAR文件，因此它可能与端口有关，但再次-不确定。
package service.rest.controllers;
import org.apache.logging.log4j.LogManager;
import org.apache.logging.log4j.Logger;
import org.freedesktop.gstreamer.*;
import org.freedesktop.gstreamer.event.EOSEvent;
import org.freedesktop.gstreamer.message.EOSMessage;
import org.freedesktop.gstreamer.message.Message;
import org.springframework.beans.factory.annotation.Autowired;
import org.springframework.web.bind.annotation.RequestMapping;
import org.springframework.web.bind.annotation.RestController;
import service.config.ClientSettings;
import service.config.FileSettings;
import service.postgres.Database;
import service.postgres.ExecuteDatabase;
import java.text.SimpleDateFormat;
import java.util.Date;
//pipeline running on pi@raspberrypi:
//gst-launch-1.0 -ev alsasrc device=plughw:1,0 ! audioconvert ! rtpL24pay ! udpsink host=xxxxx port=xxxx
@RestController
public class AudioCaptureController {
@Autowired
public Database database;
@Autowired
ExecuteDatabase db_executor;
@Autowired
ClientSettings clientSettings;
@Autowired
FileSettings fileSettings;
private static final Logger LOGGER = LogManager.getLogger(AudioCaptureController.class.getName());
private static final String startTemplate = "Pipeline started at %s.";
private static final String stopTemplate = "File recorded for time window %s to %s.";
private static final SimpleDateFormat ft = new SimpleDateFormat("yyyy-MM-dd HH:mm:ss");
private Pipeline pipe;
private Date startTime;
private int port;
private int defaultLength;
private int defaultRecordingDuration;
private String defaultDirectory;
public AudioCaptureController() {
}
/**
* Initializes GStreamer pipeline.
* udpsrc ! rtpL24depay ! decodebin ! audioconvert ! wavenc ! filesink
*/
public void init() {
port = clientSettings.getUdp_port();
defaultLength = fileSettings.getDefault_length();
defaultRecordingDuration = fileSettings.getDefault_recording_duration();
defaultDirectory = fileSettings.getDefault_directory();
Gst.init("Receiver");
//CREATE ELEMENTS
Element source = ElementFactory.make("udpsrc", "source");
Element depayloader = ElementFactory.make("rtpL24depay", "depayloader");
Element decoder = ElementFactory.make("decodebin", "decoder");
Element converter = ElementFactory.make("audioconvert", "converter");
Element encoder = ElementFactory.make("wavenc", "encoder");
Element sink = ElementFactory.make("filesink", "sink");
//CONFIGURE ELEMENTS
Caps caps = Caps.fromString("application/x-rtp, " +
"media=(string)audio, " +
"clock-rate=(int)44100, " +
"encoding-name=(string)L24, " +
"encoding-params=(string)2, " +
"channels=(int)2, " +
"payload=(int)96, " +
"ssrc=(uint)636287891, " +
"timestamp-offset=(uint)692362821, " +
"seqnum-offset=(uint)11479");
source.set("port", port);
source.setCaps(caps);
//GENERATE WAV FILE - **Currently generating only one file**
//todo: need a way to save specific file names. probably have to pause and restart the stream each time.
//consider splitting the file post-processing
//can't use multifilesink or splitmuxsink b/c no native support for wav
//https://stackoverflow.com/questions/25662392/gstreamer-multifilesink-wav-files-splitting
sink.set("location", defaultDirectory + "test.wav");
// sink.set("location", "test.wav");
//SET UP PIPELINE
pipe = new Pipeline();
pipe.addMany(source, depayloader, decoder, converter, encoder, sink);
//LINK PADS
source.link(depayloader);
depayloader.link(decoder);
decoder.link(converter);
converter.link(encoder);
encoder.link(sink);
//HANDLE EOS/ERROR/WARNING ON THE BUS
Bus bus = pipe.getBus();
bus.connect((Bus.EOS) gstObject -> System.out.println("EOS " + gstObject));
bus.connect((Bus.ERROR) (gstObject, i, s) -> System.out.println("ERROR " + i + " " + s + " " + gstObject));
bus.connect((Bus.WARNING) (gstObject, i, s) -> System.out.println("WARN " + i + " " + s + " " + gstObject));
bus.connect((Bus.EOS) obj -> {
pipe.stop();
Gst.deinit();
Gst.quit();
});
}
/**
* Starts the GStreamer pipeline.
*/
@RequestMapping("/start")
public String startRecording() {
//START PIPELINE
pipe.play();
startTime = new Date(System.currentTimeMillis());
LOGGER.info(String.format(startTemplate, ft.format(startTime)));
return String.format(startTemplate, ft.format(startTime));
}
/**
* Stops the GStreamer pipeline and pushes the file to database.
*/
@RequestMapping("/stop")
public String stopRecording() {
// if (pipe.isPlaying()) { //might have to comment this out
// pipe.stop();
pipe.getBus().post(new EOSMessage(pipe.getS));
// Gst.quit();
Date endTime = new Date(System.currentTimeMillis());
String filePath = defaultDirectory + "test.wav";
db_executor.insertRecord(database.getConnection(), ft.format(startTime), ft.format(endTime), filePath);
LOGGER.info(String.format(stopTemplate, ft.format(startTime), ft.format(endTime)));
return String.format(stopTemplate, ft.format(startTime), ft.format(endTime));
// } else {
// LOGGER.info("Pipeline is already at state " + pipe.getState());
// return "Pipeline is already at state " + pipe.getState();
// }
}
}
答案 0 :(得分：0)
Decodebin具有动态源填充，您需要在它们显示时链接它们（它们在流开始之前就不存在，因为Decodebin无法知道它将要处理的内容以及需要多少个填充）
对于您的情况，您可能不需要它，因为您不需要解码，只需使用rtp depayloader。如果要保留它，请确保注册一个pad-added回调，一旦创建便笺簿，便会得到。在回调中，您应该将其链接到管道的其余部分（如果需要，您甚至可以在此时创建其余部分）。