请帮帮我!我在 https://github.com/pchab/AndroidRTC 中使用了这个例子将视频和音频从安卓设备流式传输到其他安卓设备。在这个例子中，他们使用了 2 个库:libjingle_peerConnection 和 SocketIo 客户端，但我不知道如何将流数据保存为 h.264 格式？
最佳答案
经过对这个项目的大量尝试和努力，我找到了将视频保存为 mp4 的解决方案，没有任何问题。
将此 VideoFileRenderer.java 添加到您的项目中
package org.webrtc;
import android.media.MediaCodec;
import android.media.MediaCodecInfo;
import android.media.MediaFormat;
import android.media.MediaMuxer;
import android.os.Handler;
import android.os.HandlerThread;
import android.util.Log;
import android.view.Surface;
import org.webrtc.audio.AudioDeviceModule;
import org.webrtc.audio.JavaAudioDeviceModule;
import org.webrtc.audio.JavaAudioDeviceModule.SamplesReadyCallback;
import java.io.IOException;
import java.nio.ByteBuffer;
public class VideoFileRenderer implements VideoSink, SamplesReadyCallback {
private static final String TAG = "VideoFileRenderer";
private final HandlerThread renderThread;
private final Handler renderThreadHandler;
private final HandlerThread audioThread;
private final Handler audioThreadHandler;
private int outputFileWidth = -1;
private int outputFileHeight = -1;
private ByteBuffer[] encoderOutputBuffers;
private ByteBuffer[] audioInputBuffers;
private ByteBuffer[] audioOutputBuffers;
private EglBase eglBase;
private EglBase.Context sharedContext;
private VideoFrameDrawer frameDrawer;
// TODO: these ought to be configurable as well
private static final String MIME_TYPE = "video/avc"; // H.264 Advanced Video Coding
private static final int FRAME_RATE = 30; // 30fps
private static final int IFRAME_INTERVAL = 5; // 5 seconds between I-frames
private MediaMuxer mediaMuxer;
private MediaCodec encoder;
private MediaCodec.BufferInfo bufferInfo, audioBufferInfo;
private int trackIndex = -1;
private int audioTrackIndex;
private boolean isRunning = true;
private GlRectDrawer drawer;
private Surface surface;
private MediaCodec audioEncoder;
private AudioDeviceModule audioDeviceModule;
public VideoFileRenderer(String outputFile, final EglBase.Context sharedContext, boolean withAudio) throws IOException {
renderThread = new HandlerThread(TAG + "RenderThread");
renderThread.start();
renderThreadHandler = new Handler(renderThread.getLooper());
if (withAudio) {
audioThread = new HandlerThread(TAG + "AudioThread");
audioThread.start();
audioThreadHandler = new Handler(audioThread.getLooper());
} else {
audioThread = null;
audioThreadHandler = null;
}
bufferInfo = new MediaCodec.BufferInfo();
this.sharedContext = sharedContext;
// Create a MediaMuxer. We can't add the video track and start() the muxer here,
// because our MediaFormat doesn't have the Magic Goodies. These can only be
// obtained from the encoder after it has started processing data.
mediaMuxer = new MediaMuxer(outputFile,
MediaMuxer.OutputFormat.MUXER_OUTPUT_MPEG_4);
audioTrackIndex = withAudio ? -1 : 0;
}
private void initVideoEncoder() {
MediaFormat format = MediaFormat.createVideoFormat(MIME_TYPE, outputFileWidth, outputFileHeight);
// Set some properties. Failing to specify some of these can cause the MediaCodec
// configure() call to throw an unhelpful exception.
format.setInteger(MediaFormat.KEY_COLOR_FORMAT,
MediaCodecInfo.CodecCapabilities.COLOR_FormatSurface);
format.setInteger(MediaFormat.KEY_BIT_RATE, 6000000);
format.setInteger(MediaFormat.KEY_FRAME_RATE, FRAME_RATE);
format.setInteger(MediaFormat.KEY_I_FRAME_INTERVAL, IFRAME_INTERVAL);
// Create a MediaCodec encoder and configure it with our format. Get a Surface
// we can use for input and wrap it with a class that handles the EGL work.
try {
encoder = MediaCodec.createEncoderByType(MIME_TYPE);
encoder.configure(format, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
renderThreadHandler.post(() -> {
eglBase = EglBase.create(sharedContext, EglBase.CONFIG_RECORDABLE);
surface = encoder.createInputSurface();
eglBase.createSurface(surface);
eglBase.makeCurrent();
drawer = new GlRectDrawer();
});
} catch (Exception e) {
Log.wtf(TAG, e);
}
}
@Override
public void onFrame(VideoFrame frame) {
frame.retain();
if (outputFileWidth == -1) {
outputFileWidth = frame.getRotatedWidth();
outputFileHeight = frame.getRotatedHeight();
initVideoEncoder();
}
renderThreadHandler.post(() -> renderFrameOnRenderThread(frame));
}
private void renderFrameOnRenderThread(VideoFrame frame) {
if (frameDrawer == null) {
frameDrawer = new VideoFrameDrawer();
}
frameDrawer.drawFrame(frame, drawer, null, 0, 0, outputFileWidth, outputFileHeight);
frame.release();
drainEncoder();
eglBase.swapBuffers();
}
/**
* Release all resources. All already posted frames will be rendered first.
*/
public void release() {
isRunning = false;
if (audioThreadHandler != null)
audioThreadHandler.post(() -> {
if (audioEncoder != null) {
audioEncoder.stop();
audioEncoder.release();
}
audioThread.quit();
});
renderThreadHandler.post(() -> {
if (encoder != null) {
encoder.stop();
encoder.release();
}
eglBase.release();
mediaMuxer.stop();
mediaMuxer.release();
renderThread.quit();
});
}
private boolean encoderStarted = false;
private volatile boolean muxerStarted = false;
private long videoFrameStart = 0;
private void drainEncoder() {
if (!encoderStarted) {
encoder.start();
encoderOutputBuffers = encoder.getOutputBuffers();
encoderStarted = true;
return;
}
while (true) {
int encoderStatus = encoder.dequeueOutputBuffer(bufferInfo, 10000);
if (encoderStatus == MediaCodec.INFO_TRY_AGAIN_LATER) {
break;
} else if (encoderStatus == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
// not expected for an encoder
encoderOutputBuffers = encoder.getOutputBuffers();
Log.e(TAG, "encoder output buffers changed");
} else if (encoderStatus == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
// not expected for an encoder
MediaFormat newFormat = encoder.getOutputFormat();
Log.e(TAG, "encoder output format changed: " + newFormat);
trackIndex = mediaMuxer.addTrack(newFormat);
if (audioTrackIndex != -1 && !muxerStarted) {
mediaMuxer.start();
muxerStarted = true;
}
if (!muxerStarted)
break;
} else if (encoderStatus < 0) {
Log.e(TAG, "unexpected result fr om encoder.dequeueOutputBuffer: " + encoderStatus);
} else { // encoderStatus >= 0
try {
ByteBuffer encodedData = encoderOutputBuffers[encoderStatus];
if (encodedData == null) {
Log.e(TAG, "encoderOutputBuffer " + encoderStatus + " was null");
break;
}
// It's usually necessary to adjust the ByteBuffer values to match BufferInfo.
encodedData.position(bufferInfo.offset);
encodedData.limit(bufferInfo.offset + bufferInfo.size);
if (videoFrameStart == 0 && bufferInfo.presentationTimeUs != 0) {
videoFrameStart = bufferInfo.presentationTimeUs;
}
bufferInfo.presentationTimeUs -= videoFrameStart;
if (muxerStarted)
mediaMuxer.writeSampleData(trackIndex, encodedData, bufferInfo);
isRunning = isRunning && (bufferInfo.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) == 0;
encoder.releaseOutputBuffer(encoderStatus, false);
if ((bufferInfo.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0) {
break;
}
} catch (Exception e) {
Log.wtf(TAG, e);
break;
}
}
}
}
private long presTime = 0L;
private void drainAudio() {
if (audioBufferInfo == null)
audioBufferInfo = new MediaCodec.BufferInfo();
while (true) {
int encoderStatus = audioEncoder.dequeueOutputBuffer(audioBufferInfo, 10000);
if (encoderStatus == MediaCodec.INFO_TRY_AGAIN_LATER) {
break;
} else if (encoderStatus == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
// not expected for an encoder
audioOutputBuffers = audioEncoder.getOutputBuffers();
Log.w(TAG, "encoder output buffers changed");
} else if (encoderStatus == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
// not expected for an encoder
MediaFormat newFormat = audioEncoder.getOutputFormat();
Log.w(TAG, "encoder output format changed: " + newFormat);
audioTrackIndex = mediaMuxer.addTrack(newFormat);
if (trackIndex != -1 && !muxerStarted) {
mediaMuxer.start();
muxerStarted = true;
}
if (!muxerStarted)
break;
} else if (encoderStatus < 0) {
Log.e(TAG, "unexpected result fr om encoder.dequeueOutputBuffer: " + encoderStatus);
} else { // encoderStatus >= 0
try {
ByteBuffer encodedData = audioOutputBuffers[encoderStatus];
if (encodedData == null) {
Log.e(TAG, "encoderOutputBuffer " + encoderStatus + " was null");
break;
}
// It's usually necessary to adjust the ByteBuffer values to match BufferInfo.
encodedData.position(audioBufferInfo.offset);
encodedData.limit(audioBufferInfo.offset + audioBufferInfo.size);
if (muxerStarted)
mediaMuxer.writeSampleData(audioTrackIndex, encodedData, audioBufferInfo);
isRunning = isRunning && (audioBufferInfo.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) == 0;
audioEncoder.releaseOutputBuffer(encoderStatus, false);
if ((audioBufferInfo.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0) {
break;
}
} catch (Exception e) {
Log.wtf(TAG, e);
break;
}
}
}
}
@Override
public void onWebRtcAudioRecordSamplesReady(JavaAudioDeviceModule.AudioSamples audioSamples) {
if (!isRunning)
return;
audioThreadHandler.post(() -> {
if (audioEncoder == null) try {
audioEncoder = MediaCodec.createEncoderByType("audio/mp4a-latm");
MediaFormat format = new MediaFormat();
format.setString(MediaFormat.KEY_MIME, "audio/mp4a-latm");
format.setInteger(MediaFormat.KEY_CHANNEL_COUNT, audioSamples.getChannelCount());
format.setInteger(MediaFormat.KEY_SAMPLE_RATE, audioSamples.getSampleRate());
format.setInteger(MediaFormat.KEY_BIT_RATE, 64 * 1024);
format.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel.AACObjectLC);
audioEncoder.configure(format, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
audioEncoder.start();
audioInputBuffers = audioEncoder.getInputBuffers();
audioOutputBuffers = audioEncoder.getOutputBuffers();
} catch (IOException exception) {
Log.wtf(TAG, exception);
}
int bufferIndex = audioEncoder.dequeueInputBuffer(0);
if (bufferIndex >= 0) {
ByteBuffer buffer = audioInputBuffers[bufferIndex];
buffer.clear();
byte[] data = audioSamples.getData();
buffer.put(data);
audioEncoder.queueInputBuffer(bufferIndex, 0, data.length, presTime, 0);
presTime += data.length * 125 / 12; // 1000000 microseconds / 48000hz / 2 bytes
}
drainAudio();
});
}
}
然后为这个录音添加这个实现MediaRecorderImpl.java
package com.vedja.hassan.kavandeh_master.utils;
import android.support.annotation.Nullable;
import android.util.Log;
import com.vedja.hassan.kavandeh_master.utils.utils.EglUtils;
import org.webrtc.VideoFileRenderer;
import org.webrtc.VideoTrack;
import org.webrtc.audio.AudioDeviceModule;
import org.webrtc.audio.JavaAudioDeviceModule;
import java.io.File;
public class MediaRecorderImpl {
private final Integer id;
private final VideoTrack videoTrack;
private final AudioSamplesInterceptor audioInterceptor;
private VideoFileRenderer videoFileRenderer;
private boolean isRunning = false;
private File recordFile;
public MediaRecorderImpl(Integer id, @Nullable VideoTrack videoTrack, @Nullable AudioSamplesInterceptor audioInterceptor) {
this.id = id;
this.videoTrack = videoTrack;
this.audioInterceptor = audioInterceptor;
}
public void startRecording(File file) throws Exception {
recordFile = file;
if (isRunning)
return;
isRunning = true;
//noinspection ResultOfMethodCallIgnored
file.getParentFile().mkdirs();
if (videoTrack != null) {
videoFileRenderer = new VideoFileRenderer(
file.getAbsolutePath(),
EglUtils.getRootEglBaseContext(),
audioInterceptor != null
);
videoTrack.addSink(videoFileRenderer);
if (audioInterceptor != null)
audioInterceptor.attachCallback(id, videoFileRenderer);
} else {
Log.e(TAG, "Video track is null");
if (audioInterceptor != null) {
//TODO(rostopira): audio only recording
throw new Exception("Audio-only recording not implemented yet");
}
}
}
public File getRecordFile() { return recordFile; }
public void stopRecording() {
isRunning = false;
if (audioInterceptor != null)
audioInterceptor.detachCallback(id);
if (videoTrack != null && videoFileRenderer != null) {
videoTrack.removeSink(videoFileRenderer);
videoFileRenderer.release();
videoFileRenderer = null;
}
}
private static final String TAG = "MediaRecorderImpl";
}
并将上面的代码与此代码一起使用
final AudioSamplesInterceptor inputSamplesInterceptor = new AudioSamplesInterceptor();
private OutputAudioSamplesInterceptor outputSamplesInterceptor = null;
private final SparseArray<MediaRecorderImpl> mediaRecorders = new SparseArray<>();
void startRecordingToFile(String path, Integer id, @Nullable VideoTrack videoTrack, @Nullable AudioChannel audioChannel) throws Exception {
AudioSamplesInterceptor interceptor = null;
if (audioChannel == AudioChannel.INPUT)
interceptor = inputSamplesInterceptor;
else if (audioChannel == AudioChannel.OUTPUT) {
if (outputSamplesInterceptor == null)
outputSamplesInterceptor = new OutputAudioSamplesInterceptor(audioDeviceModule);
interceptor = outputSamplesInterceptor;
}
mediaRecorder = new MediaRecorderImpl(id, videoTrack, interceptor);
mediaRecorder.startRecording(new File(path));
mediaRecorders.append(id, mediaRecorder);
}
void stopRecording(Integer id) {
MediaRecorderImpl mediaRecorder = mediaRecorders.get(id);
if (mediaRecorder != null) {
mediaRecorder.stopRecording();
mediaRecorders.remove(id);
File file = mediaRecorder.getRecordFile();
if (file != null) {
ContentValues values = new ContentValues(3);
values.put(MediaStore.Video.Media.TITLE, file.getName());
values.put(MediaStore.Video.Media.MIME_TYPE, "video/mp4");
values.put(MediaStore.Video.Media.DATA, file.getAbsolutePath());
getContentResolver().insert(MediaStore.Video.Media.EXTERNAL_CONTENT_URI, values);
}
}
}
终于用上了这个
try {
VedjaSharedPreference sharedPreference = new VedjaSharedPreference(getContext());
final File dir = new File(sharedPreference.getStringParam(StaticParameter.SAVING_URL) + "/audio/");
dir.mkdirs(); //create folders where write files
final File file = new File(dir, "Vedja-".concat(String.valueOf(System.currentTimeMillis())).concat(".mp3"));
VideoTrack videoTrack = null;
MediaStreamTrack track = slaveManagerActivity.remoteStream.videoTracks.get(0);
if (track instanceof VideoTrack)
videoTrack = (VideoTrack) track;
AudioChannel audioChannel = AudioChannel.OUTPUT;
slaveManagerActivity.startRecordingToFile(file.getPath(), 1, videoTrack, audioChannel);
} catch (Exception e) {
e.printStackTrace();
throw new RuntimeException(
"Failed to open video file for output: ", e);
}
可能在复制这段代码后，在您的项目中找不到某些类。你可以在互联网上搜索这个类。
关于android - 如何在 webRTC android 中将视频流数据记录为 mp4？，我们在Stack Overflow上找到一个类似的问题： https://stackoverflow.com/questions/49830268/