NDK38_RTMP：音频编码与推流

NDK开发汇总

一 集成faac

1 下载编译faac生成静态库和.h头文件

2 项目添加libfaac.a和.h头文件，CmakeList：

cmake_minimum_required(VERSION 3.4.1)

# 引入指定目录下的CMakeLists.txt
add_subdirectory(src/main/cpp/librtmp)


add_library(
             native-lib

             SHARED

             src/main/cpp/native-lib.cpp
             src/main/cpp/VideoChannel.cpp
             src/main/cpp/AudioChannel.cpp)

include_directories(src/main/cpp/include)
set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -L${CMAKE_SOURCE_DIR}/src/main/cpp/libs/${ANDROID_ABI}")

target_link_libraries(
                       native-lib
                       rtmp
                       x264
                       faac
            log)

二 获取音频

AudioChannel

public class AudioChannel {

    private int inputSamples;
    private ExecutorService executor;
    private AudioRecord audioRecord;
    private LivePusher mLivePusher;
    private int channels = 1;
    private boolean isLiving;

    public AudioChannel(LivePusher livePusher) {
        mLivePusher = livePusher;
        executor = Executors.newSingleThreadExecutor();
        //准备录音机 采集pcm 数据
        int channelConfig;
        if (channels == 2) {
            channelConfig = AudioFormat.CHANNEL_IN_STEREO;
        } else {
            channelConfig = AudioFormat.CHANNEL_IN_MONO;
        }


        mLivePusher.native_setAudioEncInfo(44100, channels);
        //16 位 2个字节
        inputSamples = mLivePusher.getInputSamples() * 2;

        //最小需要的缓冲区
        int minBufferSize = AudioRecord.getMinBufferSize(44100, channelConfig, AudioFormat.ENCODING_PCM_16BIT) * 2;
        //1、麦克风 2、采样率 3、声道数 4、采样位
        audioRecord = new AudioRecord(MediaRecorder.AudioSource.MIC, 44100, channelConfig, AudioFormat.ENCODING_PCM_16BIT, minBufferSize > inputSamples ? minBufferSize : inputSamples);
    }


    public void startLive() {
        isLiving = true;
        executor.submit(new AudioTeask());
    }

    public void stopLive() {
        isLiving = false;
    }


    public void release() {
        audioRecord.release();
    }


    class AudioTeask implements Runnable {

        @Override
        public void run() {
            //启动录音机
            audioRecord.startRecording();
            byte[] bytes = new byte[inputSamples];
            while (isLiving) {
                int len = audioRecord.read(bytes, 0, bytes.length);
                if (len > 0) {
                    //送去编码
                    mLivePusher.native_pushAudio(bytes);
                }
            }
            //停止录音机
            audioRecord.stop();
        }
    }
}

LivePusher

public class LivePusher {



    private AudioChannel audioChannel;
    private VideoChannel videoChannel;

    public LivePusher(Activity activity, int width, int height, int bitrate,
                      int fps, int cameraId) {

        native_init();
        videoChannel = new VideoChannel(this,activity, width, height, bitrate, fps, cameraId);
        audioChannel = new AudioChannel(this);
    }

    public void setPreviewDisplay(SurfaceHolder surfaceHolder) {
        videoChannel.setPreviewDisplay(surfaceHolder);
    }

    public void switchCamera() {
        videoChannel.switchCamera();
    }

    public void startLive(String path) {
        native_start(path);
        videoChannel.startLive();
        audioChannel.startLive();
    }

    public void stopLive(){
        videoChannel.stopLive();
        audioChannel.stopLive();
        native_stop();
    }

    public void release(){
        videoChannel.release();
        audioChannel.release();
        native_release();
    }


    public native void native_init();

    public native void native_start(String path);

    public native void native_setVideoEncInfo(int width, int height, int fps, int bitrate);

    public native void native_setAudioEncInfo(int sampleRateInHz, int channelConfig);

    public native void native_pushVideo(byte[] data);

    public native void native_pushAudio(byte[] data);

    public native int getInputSamples();

    public native void native_stop();

    public native void native_release();
}

三 音频解码与推送

native-lib

#include <jni.h>
#include <string>
#include "safe_queue.h"
#include "librtmp/rtmp.h"
#include "VideoChannel.h"
#include "AudioChannel.h"
#include "macro.h"

SafeQueue<RTMPPacket *> packets;
VideoChannel *videoChannel = 0;
int isStart = 0;
pthread_t pid;

int readyPushing = 0;
uint32_t start_time;


AudioChannel *audioChannel = 0;

void releasePackets(RTMPPacket *&packet) {
    if (packet) {
        RTMPPacket_Free(packet);
        delete packet;
        packet = 0;
    }
}

void callback(RTMPPacket *packet) {
    if (packet) {
        //设置时间戳
        packet->m_nTimeStamp = RTMP_GetTime() - start_time;
        packets.push(packet);
    }
}

extern "C" JNIEXPORT void JNICALL
Java_com_cn_ray_rtmpdump_LivePusher_native_1init(JNIEnv *env, jobject instance) {

    //准备一个Video编码器的工具类 ：进行编码
    videoChannel = new VideoChannel;
    videoChannel->setVideoCallback(callback);

    audioChannel = new AudioChannel;
    audioChannel->setAudioCallback(callback);

    //准备一个队列,打包好的数据 放入队列，在线程中统一的取出数据再发送给服务器
    packets.setReleaseCallback(releasePackets);

}

extern "C" JNIEXPORT void JNICALL
Java_com_cn_ray_rtmpdump_LivePusher_native_1setVideoEncInfo(JNIEnv *env, jobject instance, jint width, jint height, jint fps,
                                                            jint bitrate) {
    if (videoChannel) {
        videoChannel->setVideoEncInfo(width, height, fps, bitrate);
    }


}

void *start(void *args) {
    char *url = static_cast<char *>(args);
    RTMP *rtmp = 0;
    do {
        rtmp = RTMP_Alloc();
        if (!rtmp) {
            LOGE("alloc rtmp失败");
            break;
        }
        RTMP_Init(rtmp);
        int ret = RTMP_SetupURL(rtmp, url);
        if (!ret) {
            LOGE("设置地址失败:%s", url);
            break;
        }
        //5s超时时间
        rtmp->Link.timeout = 5;
        RTMP_EnableWrite(rtmp);
        ret = RTMP_Connect(rtmp, 0);
        if (!ret) {
            LOGE("连接服务器:%s", url);
            break;
        }
        ret = RTMP_ConnectStream(rtmp, 0);
        if (!ret) {
            LOGE("连接流:%s", url);
            break;
        }
        //记录一个开始时间
        start_time = RTMP_GetTime();
        //表示可以开始推流了
        readyPushing = 1;
        packets.setWork(1);
        //保证第一个数据是 aac解码数据包
        callback(audioChannel->getAudioTag());
        RTMPPacket *packet = 0;
        while (readyPushing) {
            packets.pop(packet);
            if (!readyPushing) {
                break;
            }
            if (!packet) {
                continue;
            }
            packet->m_nInfoField2 = rtmp->m_stream_id;
            //发送rtmp包 1：队列
            // 意外断网？发送失败，rtmpdump 内部会调用RTMP_Close
            // RTMP_Close 又会调用 RTMP_SendPacket
            // RTMP_SendPacket  又会调用 RTMP_Close
            // 将rtmp.c 里面WriteN方法的 Rtmp_Close注释掉
            ret = RTMP_SendPacket(rtmp, packet, 1);
            releasePackets(packet);
            if (!ret) {
                LOGE("发送失败");
                break;
            }
        }
        releasePackets(packet);
    } while (0);
    //
    isStart = 0;
    readyPushing = 0;
    packets.setWork(0);
    packets.clear();
    if (rtmp) {
        RTMP_Close(rtmp);
        RTMP_Free(rtmp);
    }
    delete (url);
    return 0;
}


extern "C"
JNIEXPORT void JNICALL
Java_com_cn_ray_rtmpdump_LivePusher_native_1start(JNIEnv *env, jobject instance, jstring path_) {
    if (isStart) {
        return;
    }
    isStart = 1;
    const char *path = env->GetStringUTFChars(path_, 0);
    char *url = new char[strlen(path) + 1];
    strcpy(url, path);
    pthread_create(&pid, 0, start, url);
    env->ReleaseStringUTFChars(path_, path);
}

extern "C"
JNIEXPORT void JNICALL
Java_com_cn_ray_rtmpdump_LivePusher_native_1pushVideo(JNIEnv *env, jobject instance,
                                                      jbyteArray data_) {
    if (!videoChannel || !readyPushing) {
        return;
    }
    jbyte *data = env->GetByteArrayElements(data_, NULL);
    videoChannel->encodeData(data);
    env->ReleaseByteArrayElements(data_, data, 0);
}

extern "C"
JNIEXPORT void JNICALL
Java_com_cn_ray_rtmpdump_LivePusher_native_1stop(JNIEnv *env, jobject instance) {

    readyPushing = 0;
    //关闭队列工作
    packets.setWork(0);
    pthread_join(pid, 0);

}

extern "C"
JNIEXPORT void JNICALL
Java_com_cn_ray_rtmpdump_LivePusher_native_1release(JNIEnv *env, jobject instance) {
    DELETE(videoChannel);
    DELETE(audioChannel);
}

extern "C"
JNIEXPORT void JNICALL
Java_com_cn_ray_rtmpdump_LivePusher_native_1setAudioEncInfo(JNIEnv *env, jobject instance,
                                                            jint sampleRateInHz,
                                                            jint channelConfig) {

    if(audioChannel){
        audioChannel->setAudioEncInfo(sampleRateInHz,channelConfig);
    }

}extern "C"
JNIEXPORT jint JNICALL
Java_com_cn_ray_rtmpdump_LivePusher_getInputSamples(JNIEnv *env, jobject instance) {
    if(audioChannel){
        audioChannel->getInputSamples();
    }
    return -1;
}

extern "C"
JNIEXPORT void JNICALL
Java_com_cn_ray_rtmpdump_LivePusher_native_1pushAudio(JNIEnv *env, jobject instance,
                                                      jbyteArray data_) {
    if (!audioChannel || !readyPushing) {
        return;
    }
    jbyte *data = env->GetByteArrayElements(data_, NULL);
    audioChannel->encodeData(data);
    env->ReleaseByteArrayElements(data_, data, 0);
}

AudioChannel

//


#include <cstring>
#include "AudioChannel.h"
#include "macro.h"

AudioChannel::AudioChannel() {

}

AudioChannel::~AudioChannel() {
    DELETE(buffer);
    //释放编码器
    if (audioCodec) {
        faacEncClose(audioCodec);
        audioCodec = 0;
    }
}

void AudioChannel::setAudioCallback(AudioCallback audioCallback) {
    this->audioCallback = audioCallback;
}

void AudioChannel::setAudioEncInfo(int samplesInHZ, int channels) {
    //打开编码器
    mChannels = channels;
    //3、一次最大能输入编码器的样本数量 也编码的数据的个数 (一个样本是16位 2字节)
    //4、最大可能的输出数据  编码后的最大字节数
    audioCodec = faacEncOpen(samplesInHZ, channels, &inputSamples, &maxOutputBytes);

    //设置编码器参数
    faacEncConfigurationPtr config = faacEncGetCurrentConfiguration(audioCodec);
    //指定为 mpeg4 标准
    config->mpegVersion = MPEG4;
    //lc 标准
    config->aacObjectType = LOW;
    //16位
    config->inputFormat = FAAC_INPUT_16BIT;
    // 编码出原始数据 既不是adts也不是adif
    config->outputFormat = 0;
    faacEncSetConfiguration(audioCodec, config);

    //输出缓冲区 编码后的数据 用这个缓冲区来保存
    buffer = new u_char[maxOutputBytes];
}

int AudioChannel::getInputSamples() {
    return inputSamples;
}

RTMPPacket *AudioChannel::getAudioTag() {
    u_char *buf;
    u_long len;
    faacEncGetDecoderSpecificInfo(audioCodec, &buf, &len);
    int bodySize = 2 + len;
    RTMPPacket *packet = new RTMPPacket;
    RTMPPacket_Alloc(packet, bodySize);
    //双声道
    packet->m_body[0] = 0xAF;
    if (mChannels == 1) {
        packet->m_body[0] = 0xAE;
    }
    packet->m_body[1] = 0x00;
    //图片数据
    memcpy(&packet->m_body[2], buf, len);

    packet->m_hasAbsTimestamp = 0;
    packet->m_nBodySize = bodySize;
    packet->m_packetType = RTMP_PACKET_TYPE_AUDIO;
    packet->m_nChannel = 0x11;
    packet->m_headerType = RTMP_PACKET_SIZE_LARGE;
    return packet;
}

void AudioChannel::encodeData(int8_t *data) {
    //返回编码后数据字节的长度
    int bytelen = faacEncEncode(audioCodec, reinterpret_cast<int32_t *>(data), inputSamples, buffer,
                                maxOutputBytes);
    if (bytelen > 0) {
        //看表
        int bodySize = 2 + bytelen;
        RTMPPacket *packet = new RTMPPacket;
        RTMPPacket_Alloc(packet, bodySize);
        //双声道
        packet->m_body[0] = 0xAF;
        if (mChannels == 1) {
            packet->m_body[0] = 0xAE;
        }
        //编码出的声音 都是 0x01
        packet->m_body[1] = 0x01;
        //图片数据
        memcpy(&packet->m_body[2], buffer, bytelen);

        packet->m_hasAbsTimestamp = 0;
        packet->m_nBodySize = bodySize;
        packet->m_packetType = RTMP_PACKET_TYPE_AUDIO;
        packet->m_nChannel = 0x11;
        packet->m_headerType = RTMP_PACKET_SIZE_LARGE;
        audioCallback(packet);
    }
}

四 Demo

RTMPDump