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简介
观看手游直播时，我们观众端看到的是选手的屏幕上的内容，这是如何实现的呢？这篇博客将手写一个录屏直播Demo，实现类似手游投屏直播的效果
获取屏幕数据很简单，Android 系统有提供对应的服务，难点在于传输数据到直播服务器，我们使用 RtmpDump 来传输 Rtmp 数据，由于 RtmpDump 使用 C 语言实现，我们还需要用到 NDK 开发，单单用 Java 无法实现哈，当年没有使用开发实现的RTMP协议的开源库，所以在RTMP协议包上，使用了RtmpDump C库。
实现效果
实现效果
基本流程
- 获取录屏数据
- 对数据进行 h264 编码
- Rtmp 数据包
- 上传到直播服务器推流地址
RTMP直播实现流程
获取录屏数据
MediaProjection 视频采集 SDK中的接口
A token granting applications the ability to capture screen contents and/or record system audio. The exact capabilities granted depend on the type of MediaProjection.
A screen capture session can be started through
MediaProjectionManager.createScreenCaptureIntent(). This grants the ability to capture screen contents, but not system audio.
@Override
protected void onActivityResult(int requestCode, int resultCode, Intent data) {
super.onActivityResult(requestCode, resultCode, data);
if (requestCode == 100 && resultCode == Activity.RESULT_OK) {
if (editText.getText()!=null&&!TextUtils.isEmpty(editText.getText().toString())) {
url=editText.getText().toString();
Log.i("tuch", "url: "+url);
}
Log.i(TAG, " url:"+url);
mediaProjection = mediaProjectionManager.getMediaProjection(resultCode, data);
}
}
public void startLive(View view) {
this.mediaProjectionManager = (MediaProjectionManager)getSystemService(Context.MEDIA_PROJECTION_SERVICE);
Intent captureIntent = mediaProjectionManager.createScreenCaptureIntent();
startActivityForResult(captureIntent, 100);
}
mediaProjectionManager.createScreenCaptureIntent
获取录屏数据
通过 Intent 获取到 MediaProjectionService，继而获取到 Mediaprojection 的 VirtualCanvas，我们录屏的原始数据就是从中得来的
public VirtualDisplay createVirtualDisplay (String name, int width, int height, int dpi, int flags, Surface surface, VirtualDisplay.Callback callback, Handler handler)
Creates a VirtualDisplay to capture the contents of the screen.
createVirtualDisplay API
mediaProjection--->产生录屏数据
对数据进行 h264 编码
通过 MediaProjection 获取到的 YUV 裸数据，我们先需要对其进行 h264 编码，此时我们使用原生 MediaCodec 进行硬编码,本例子是Demo工程，没有考虑编码兼容性的问题，直接使用 MediaCodec。MediaCodec的使用参考 MediaCodec 介绍
public void startLive(MediaProjection mediaProjection) {
this.mediaProjection = mediaProjection;
MediaFormat format = MediaFormat.createVideoFormat(MediaFormat.MIMETYPE_VIDEO_AVC,
720,
1280);
format.setInteger(MediaFormat.KEY_COLOR_FORMAT,
MediaCodecInfo.CodecCapabilities.COLOR_FormatSurface);
//码率，帧率，分辨率，关键帧间隔
format.setInteger(MediaFormat.KEY_BIT_RATE, 400_000);
format.setInteger(MediaFormat.KEY_FRAME_RATE, 15);
format.setInteger(MediaFormat.KEY_I_FRAME_INTERVAL, 1);
try {
mediaCodec = MediaCodec.createEncoderByType(MediaFormat.MIMETYPE_VIDEO_AVC);//手机
mediaCodec.configure(format, null, null,
MediaCodec.CONFIGURE_FLAG_ENCODE);
Surface surface = mediaCodec.createInputSurface();
virtualDisplay = mediaProjection.createVirtualDisplay(
"screen-codec",
720, 1280, 1,
DisplayManager.VIRTUAL_DISPLAY_FLAG_PUBLIC,
surface, null, null);
} catch (IOException e) {
e.printStackTrace();
}
LiveTaskManager.getInstance().execute(this);
}
Surface surface = mediaCodec.createInputSurface();从编码器创建一个画布, 画布上的图像会被编码器自动编码, 调用createVirtualDisplay创建虚拟显示器VirtualDisplay ，即会将手机屏幕镜像到虚拟显示器上。在createVirtualDisplay时，需要传递一个Surface(画布)。需要获取图像数据即可从这个Surface中读取。
VirtualDisplay-Surface-MediaCodec
配置完成后，从MediaCodec中获取数据
@Override
public void run() {
isLiving = true;
mediaCodec.start();
MediaCodec.BufferInfo bufferInfo = new MediaCodec.BufferInfo();
while (isLiving){
//若时间差大于 2 s，则通知编码器，生成 I 帧
if (System.currentTimeMillis() - timeStamp >= 2000){
// Bundle 通知 Dsp
Bundle msgBundle = new Bundle();
msgBundle.putInt(MediaCodec.PARAMETER_KEY_REQUEST_SYNC_FRAME,0);
mediaCodec.setParameters(msgBundle);
timeStamp = System.currentTimeMillis();
}
// 接下来就是 MediaCodec 常规操作，获取 Buffer 可用索引，这里不需要获取输出索引，内部已经操作了
int outputBufferIndex = mediaCodec.dequeueOutputBuffer(bufferInfo,100_000);
if (outputBufferIndex >=0){
// 获取到了
ByteBuffer byteBuffer = mediaCodec.getOutputBuffer(outputBufferIndex);
byte[] outData = new byte[bufferInfo.size];
byteBuffer.get(outData);
}
}
VideoCodec线程的润方法中，不断的判int outputBufferIndex = mediaCodec.dequeueOutputBuffer(bufferInfo,100_000);是否有编码完成的H264数据，为什么是h264编码的，在配置的时候，MediaFormat.MIMETYPE_VIDEO_AVC参数定义的！
此时可以把好的数据，以byte的方式写入文件查看
image-20210326175405628
可以看到sps pps I帧等关键信息
也可以通过ffplay命令进行播放
ffplay -i codec.h264
此时，我们获得了编码好的 h264 数据，接下来封装 Rtmp数据包。
LIBRTMP
C语言开源RTMP库，封装 Socket 建立TCP通信，并实现了RTMP数据的收发。
RTMPDump
rtmpdump is a toolkit for RTMP streams. All forms of RTMP are supported, including rtmp://, rtmpt://, rtmpe://, rtmpte://, and rtmps://.
License: GPLv2
Copyright (C) 2009 Andrej Stepanchuk
Copyright (C) 2010-2011 Howard Chu
Download the source:
git clone git://git.ffmpeg.org/rtmpdump
The latest release is 2.4 which you can check out from git. Aside from various minor bugfixes since 2.3, RTMPE type 9 handshakes are now supported.
使用第三方库 Rtmpdump 来实现推流到直播服务器，由于 Rtmpdump 的代码量不是很多，我们直接拷贝源代码到 Android 的 cpp 文件
#定义宏 如果代码中定义了 #defind NO_CRYPTO
#就表示不适用ssl，不支持rtmps。我们这里不支持ssl
set(CMAKE_C_FLAGS "${CMAKE_C_FLAGS} -DNO_CRYPTO")
# 把当前目录下所有得文件 变成一个 SOURCE变量表示
aux_source_directory(. SOURCE)
# 编译成librtmp.a 静态库，编译源文件引用${SOURCE}获取
add_library(rtmp STATIC ${SOURCE})
工程cmake引入编译好的rtmp静态库
# 加入子文件夹
add_subdirectory(librtmp)
# 链接引入
target_link_libraries( # Specifies the target library.
native-lib
# Links the target library to the log library
# included in the NDK.
${log-lib}
rtmp)
RtmpDump 的使用
RtmpDump
- 连接服务器
- RTMP_Init(RTMP *r) 初始化
- RTMP_EnableWrite(RTMP *r) 配置开启数据写入
- RTMP_Connect(RTMP *r, RTMPPacket *cp)
- RTMP_ConnectStream(RTMP *r, int seekTime)
- 发送数据
RTMPPacket_Alloc(RTMPPacket *p, int nSize)
RTMP_SendPacket(RTMP *r, RTMPPacket *packet, int queue)
RTMPPacket_Free(RTMPPacket *p)
- Rtmp 协议关键帧协议格式
image
- Rtmp 协议非关键帧协议格式
image-20210326195045243
- SPS PPS数据包
sps-pps
启动配置好的SRS推流服务器
SRS推流服务器
./objs/srs -c conf/rtmp.conf
lsof -i :1935
连接直播服务器
这一步中，需要预先准备直播推流地址，然后实现 native 方法
extern "C"
JNIEXPORT jboolean JNICALL
Java_com_lecture_rtmtscreenlive_ScreenLive_connect(JNIEnv *env, jobject thiz, jstring url_) {
// 首先 Java 的转成 C 的字符串，不然无法使用
const char *url = env->GetStringUTFChars(url_, 0);
int ret;
do {
live = (Live *) malloc(sizeof(Live));
memset(live, 0, sizeof(Live));
live->rtmp = RTMP_Alloc();// Rtmp 申请内存
RTMP_Init(live->rtmp);
live->rtmp->Link.timeout = 10;// 设置 rtmp 初始化参数，比如超时时间、url
LOGI("connect %s", url);
if (!(ret = RTMP_SetupURL(live->rtmp, (char *) url))) break;
RTMP_EnableWrite(live->rtmp);// 开启 Rtmp 写入
LOGI("RTMP_Connect");
if (!(ret = RTMP_Connect(live->rtmp, 0))) break;
LOGI("RTMP_ConnectStream ");
if (!(ret = RTMP_ConnectStream(live->rtmp, 0))) break;
LOGI("connect success");
} while (0);
if (!ret && live) {
free(live);
live = nullptr;
}
env->ReleaseStringUTFChars(url_, url);
return ret;
}
2021-03-26 20:25:33.202 9139-9211/com.lecture.rtmtscreenlive I/DDDDDD: connect rtmp://192.168.10.224/live/livestream
2021-03-26 20:25:33.202 9139-9211/com.lecture.rtmtscreenlive I/DDDDDD: RTMP_Connect
2021-03-26 20:25:33.422 9139-9211/com.lecture.rtmtscreenlive I/DDDDDD: RTMP_ConnectStream
2021-03-26 20:25:33.562 9139-9211/com.lecture.rtmtscreenlive I/DDDDDD: connect success
2021-03-26 20:25:34.152 9139-9222/com.lecture.rtmtscreenlive I/------>dddd<---------: run: -2
2021-03-26 20:25:34.152 9139-9222/com.lecture.rtmtscreenlive I/------>dddd<---------: run: 0
服务器连接成功，现在开始发送RTMP视频数据
- 视频数据封包
RTMPPacket *createVideoPackage(int8_t *buf, int len, const long tms, Live *live) {
// 分隔符被抛弃了 --buf指的是651
buf += 4;
len -= 4;
int body_size = len + 9;
RTMPPacket *packet = (RTMPPacket *) malloc(sizeof(RTMPPacket));
RTMPPacket_Alloc(packet, len + 9);
packet->m_body[0] = 0x27;
if (buf[0] == 0x65) { //关键帧
packet->m_body[0] = 0x17;
LOGI("发送关键帧 data");
}
packet->m_body[1] = 0x01;
packet->m_body[2] = 0x00;
packet->m_body[3] = 0x00;
packet->m_body[4] = 0x00;
//长度
packet->m_body[5] = (len >> 24) & 0xff;
packet->m_body[6] = (len >> 16) & 0xff;
packet->m_body[7] = (len >> 8) & 0xff;
packet->m_body[8] = (len) & 0xff;
//数据
memcpy(&packet->m_body[9], buf, len);
packet->m_packetType = RTMP_PACKET_TYPE_VIDEO;
packet->m_nBodySize = body_size;
packet->m_nChannel = 0x04;
packet->m_nTimeStamp = tms;
packet->m_hasAbsTimestamp = 0;
packet->m_headerType = RTMP_PACKET_SIZE_LARGE;
packet->m_nInfoField2 = live->rtmp->m_stream_id;
return packet;
}
- Sps pps 封包
RTMPPacket *createVideoPackage(Live *live) {
int body_size = 13 + live->sps_len + 3 + live->pps_len;
RTMPPacket *packet = (RTMPPacket *) malloc(sizeof(RTMPPacket));
RTMPPacket_Alloc(packet, body_size);
int i = 0;
//AVC sequence header 与IDR一样
packet->m_body[i++] = 0x17;
//AVC sequence header 设置为0x00
packet->m_body[i++] = 0x00;
//CompositionTime
packet->m_body[i++] = 0x00;
packet->m_body[i++] = 0x00;
packet->m_body[i++] = 0x00;
//AVC sequence header
packet->m_body[i++] = 0x01; //configurationVersion 版本号 1
packet->m_body[i++] = live->sps[1]; //profile 如baseline、main、 high
packet->m_body[i++] = live->sps[2]; //profile_compatibility 兼容性
packet->m_body[i++] = live->sps[3]; //profile level
packet->m_body[i++] = 0xFF; // reserved（111111） + lengthSizeMinusOne（2位 nal 长度） 总是0xff
//sps
packet->m_body[i++] = 0xE1; //reserved（111） + lengthSizeMinusOne（5位 sps 个数） 总是0xe1
//sps length 2字节
packet->m_body[i++] = (live->sps_len >> 8) & 0xff; //第0个字节
packet->m_body[i++] = live->sps_len & 0xff; //第1个字节
memcpy(&packet->m_body[i], live->sps, live->sps_len);
i += live->sps_len;
/*pps*/
packet->m_body[i++] = 0x01; //pps number
//pps length
packet->m_body[i++] = (live->pps_len >> 8) & 0xff;
packet->m_body[i++] = live->pps_len & 0xff;
memcpy(&packet->m_body[i], live->pps, live->pps_len);
packet->m_packetType = RTMP_PACKET_TYPE_VIDEO;
packet->m_nBodySize = body_size;
packet->m_nChannel = 0x04;
packet->m_nTimeStamp = 0;
packet->m_hasAbsTimestamp = 0;
packet->m_headerType = RTMP_PACKET_SIZE_LARGE;
packet->m_nInfoField2 = live->rtmp->m_stream_id;
return packet;
}
int sendPacket(RTMPPacket *packet) {
int r = RTMP_SendPacket(live->rtmp, packet, 1);
RTMPPacket_Free(packet);
free(packet);
return r;
}
void prepareVideo(int8_t *data, int len, Live *live) {
for (int i = 0; i < len; i++) {
//0x00 0x00 0x00 0x01
if (i + 4 < len) {
if (data[i] == 0x00 && data[i + 1] == 0x00
&& data[i + 2] == 0x00
&& data[i + 3] == 0x01) {
//0x00 0x00 0x00 0x01 7 sps 0x00 0x00 0x00 0x01 8 pps
//将sps pps分开
//找到pps
if (data[i + 4] == 0x68) {
//去掉界定符
live->sps_len = i - 4;
live->sps = static_cast(malloc(live->sps_len));
memcpy(live->sps, data + 4, live->sps_len);
live->pps_len = len - (4 + live->sps_len) - 4;
live->pps = static_cast(malloc(live->pps_len));
memcpy(live->pps, data + 4 + live->sps_len + 4, live->pps_len);
LOGI("sps:%d pps:%d", live->sps_len, live->pps_len);
break;
}
}
}
}
}
测试
ffplay rtmp://192.168.10.224/live/livestream
image-20210326211259263
获取音频数据
- AudioRecord 采集
//录音工具类 采样位数 通道数 采样评率 固定了 设备没关系 录音 数据一样的
minBufferSize = AudioRecord.getMinBufferSize(44100,
AudioFormat.CHANNEL_IN_MONO,
AudioFormat.ENCODING_PCM_16BIT);
audioRecord = new AudioRecord(
MediaRecorder.AudioSource.MIC, 44100,
AudioFormat.CHANNEL_IN_MONO,
AudioFormat.ENCODING_PCM_16BIT, minBufferSize);
- 使用MediaCodec对采集的PCM数据编码
MediaFormat format = MediaFormat.createAudioFormat(MediaFormat.MIMETYPE_AUDIO_AAC, 44100, 1);
//录音质量
format.setInteger(MediaFormat.KEY_AAC_PROFILE, MediaCodecInfo.CodecProfileLevel
.AACObjectLC);
//一秒的码率 aac
format.setInteger(MediaFormat.KEY_BIT_RATE, 64_000);
mediaCodec = MediaCodec.createEncoderByType(MediaFormat.MIMETYPE_AUDIO_AAC);
mediaCodec.configure(format, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
mediaCodec.start();
- 麦克风的数据读取出来 pcm
audioRecord.startRecording();
// 容器 固定
byte[] buffer = new byte[minBufferSize];
// 麦克风的数据读取出来 pcm buffer aac
int len = audioRecord.read(buffer, 0, buffer.length);
- 获取经过mediaCodec编码的aac数据
int index = mediaCodec.dequeueInputBuffer(0);
if (index >= 0) {
ByteBuffer inputBuffer = mediaCodec.getInputBuffer(index);
inputBuffer.clear();
inputBuffer.put(buffer, 0, len);
//填充数据后再加入队列
mediaCodec.queueInputBuffer(index, 0, len,
System.nanoTime() / 1000, 0);
}
RTMP 包中封装的音视频数据流，其实和FLV/tag封装音频和视频数据的方式是相同的，所以我们只需要按照FLV格式封装音视频即可。
image-20210326215407401
RTMPPacket *createAudioPacket(int8_t *buf, const int len, const int type, const long tms,
Live *live) {
// 组装音频包 两个字节 是固定的 af 如果是第一次发 你就是 01 如果后面 00 或者是 01 aac
int body_size = len + 2;
RTMPPacket *packet = (RTMPPacket *) malloc(sizeof(RTMPPacket));
RTMPPacket_Alloc(packet, body_size);
// 音频头
packet->m_body[0] = 0xAF;
if (type == 1) {
// 头
packet->m_body[1] = 0x00;
}else{
packet->m_body[1] = 0x01;
}
memcpy(&packet->m_body[2], buf, len);
packet->m_packetType = RTMP_PACKET_TYPE_AUDIO;
packet->m_nChannel = 0x05;
packet->m_nBodySize = body_size;
packet->m_nTimeStamp = tms;
packet->m_hasAbsTimestamp = 0;
packet->m_headerType = RTMP_PACKET_SIZE_LARGE;
packet->m_nInfoField2 = live->rtmp->m_stream_id;
return packet;
}
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