Android使用FFmpeg播放音频_qt 安卓ffmpeg avcodec模块不能识别

2401_87225473

于 2024-09-19 02:10:09 发布

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文章标签： android ffmpeg 音视频

本文链接：https://blog.csdn.net/2401_87225473/article/details/142347530

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    LOGE("%s","没有找到音频流");
    return;
}
AVCodecParameters\* pCodecParams = pFormatContext->streams[audio_index]->codecpar;
AVCodecID  pCodecId = pCodecParams->codec_id;
if(pCodecId == NULL){
    LOGE("%s","CodecID == null");
    return;
}
AVCodec\* pCodec = avcodec\_find\_decoder(pCodecId);
if(pCodec == NULL){
    LOGE("%s","没有找到解码器");
    return;
}
AVCodecContext\* pCodecContext = avcodec\_alloc\_context3(pCodec);
if(pCodecContext == NULL){
    LOGE("%s","不能为CodecContext分配内存");
    return;
}
if(avcodec\_parameters\_to\_context(pCodecContext,pCodecParams)<0){
    LOGE("%s","创建codecContext失败");
    return;
}
if(avcodec\_open2(pCodecContext,pCodec,NULL) <0 ){
    LOGE("%s","打开解码器失败");
    return;
}
AVPacket \*avp = av\_packet\_alloc();
AVFrame \*avf = av\_frame\_alloc();

//frame->16bit 44100 PCM 统一音频采样格式与采样率
SwrContext\* swr_cxt = swr\_alloc();
//重采样设置选项-----------------------------------------------------------start
//输入的采样格式
enum AVSampleFormat in_sample_fmt = pCodecContext->sample_fmt;
//输出的采样格式
enum AVSampleFormat out_sample_fmt = AV_SAMPLE_FMT_S16;
//输入的采样率
int in_sample_rate = pCodecContext->sample_rate;
printf("sample rate = %d \n" ,in_sample_rate);
//输出的采样率
int out_sample_rate = 44100;
//输入的声道布局
uint64_t in_ch_layout = pCodecContext->channel_layout;
//输出的声道布局
uint64_t out_ch_layout = AV_CH_LAYOUT_MONO;
//SwrContext 设置参数
swr\_alloc\_set\_opts(swr_cxt,out_ch_layout,out_sample_fmt,out_sample_rate,in_ch_layout,in_sample_fmt,in_sample_rate,0,NULL);
//初始化SwrContext
swr\_init(swr_cxt);
//重采样设置选项-----------------------------------------------------------end
//获取输出的声道个数
int out_channel_nb = av\_get\_channel\_layout\_nb\_channels(out_ch_layout);
jclass clazz = env->GetObjectClass(instance);
//调用Java方法MethodID
jmethodID methodId = env->GetMethodID(clazz,"createTrack","(II)V");
jmethodID methodID1 = env->GetMethodID(clazz,"playTrack","([BI)V");
//通过methodId调用Java方法
env->CallVoidMethod(instance,methodId,44100,out_channel_nb);
//存储pcm数据
uint8_t \*out_buf = (uint8_t\*)av\_malloc(2\*44100);
int got_frame, frame_count = 0;
//6.一帧一帧读取压缩的音频数据AVPacket
int ret;
while(av\_read\_frame(pFormatContext,avp) >= 0){
    if(avp->stream_index == audio_index){
        //解码从avpacket到avframe
        ret = avcodec\_decode\_audio4(pCodecContext,avf,&got_frame,avp);
        // =0 表示解码完成
        if(ret < 0){
            av\_log(NULL,AV_LOG_INFO,"解码完成 \n");
        }
        //表示正在解码
        if(got_frame != 0){
            LOGE("正在解码第%d帧 \n",++frame_count);
            swr\_convert(swr_cxt , &out_buf , 2 \* 44100 , (const uint8_t \*\*)avf->data , avf->nb_samples);
            //获取sample的size
            int out_buf_size = av\_samples\_get\_buffer\_size(NULL,out_channel_nb,avf->nb_samples,out_sample_fmt,1);
            jbyteArray audioArray = env->NewByteArray(out_buf_size);
            env->SetByteArrayRegion(audioArray,0,out_buf_size,(const jbyte\*)out_buf);
            //调用Java方法 把解码后的数据（audioArray）给AudioTrack进行播放
            env->CallVoidMethod(instance,methodID1,audioArray,out_buf_size);
            env->DeleteLocalRef(audioArray);
        }
    }
    av\_packet\_unref(avp);
}
av\_frame\_free(&avf);
swr\_free(&swr_cxt);
avcodec\_close(pCodecContext);
avformat\_close\_input(&pFormatContext);
env->ReleaseStringUTFChars(audioPath,path);

}


下面是Java播放器调用JNI的代码：

open class ThirdActivity : AppCompatActivity() {
private val inputFilePath = “/storage/emulated/0/GreenCheng/video/g4.mp4”
private var audioTrack:AudioTrack? = null
private lateinit var musicPlayer: MusicPlayer
override fun onCreate(savedInstanceState: Bundle?) {
super.onCreate(savedInstanceState)
setContentView(R.layout.activity_third)
musicPlayer = MusicPlayer(inputFilePath)
}

fun play(view: View) {
    surface_view.startPlay(inputFilePath)
}

fun playAudio(view: View){
    Thread(Runnable {
        Log.d("ThirdActivity", "------>>调用native方法")
        playAudio(inputFilePath)
    }).start()

// musicPlayer.playAudio()
}

fun stopAudio(view: View){
    musicPlayer.stopAudio()
}

/\*\*

* 这是初始化AudioTrack的方法，也是在C中调用的
*/
fun createTrack(sampleRateInHn:Int,nbChannel:Int){
val channelConfig =when (nbChannel) {
1 -> AudioFormat.CHANNEL_OUT_MONO
2 -> AudioFormat.CHANNEL_OUT_STEREO
else -> AudioFormat.CHANNEL_OUT_MONO
}
val bufferSize = AudioTrack.getMinBufferSize(sampleRateInHn,channelConfig,AudioFormat.ENCODING_PCM_16BIT)
audioTrack = AudioTrack(AudioManager.STREAM_MUSIC,sampleRateInHn,channelConfig,AudioFormat.ENCODING_PCM_16BIT,bufferSize,AudioTrack.MODE_STREAM)
audioTrack?.play()
}

/\*\*

* 这是在C代码中调用的，就是在解码出PCM就会调用这个方法，让AudioTrack进行播放
*/
fun playTrack(buffer:ByteArray,length:Int){
if(audioTrack != null && audioTrack?.playState == AudioTrack.PLAYSTATE_PLAYING){
audioTrack?.write(buffer,0,length)
}
}

private external fun playAudio(path:String)

companion object {
    init {
        System.loadLibrary("native-lib")
    }
}

}


使用OpenSL EL进行音频播放的代码：  
 JNI代码

SLObjectItf engineObject=NULL;//用SLObjectItf声明引擎接口对象
SLEngineItf engineEngine = NULL;//声明具体的引擎对象

SLObjectItf outputMixObject = NULL;//用SLObjectItf创建混音器接口对象
SLEnvironmentalReverbItf outputMixEnvironmentalReverb = NULL;具体的混音器对象实例
SLEnvironmentalReverbSettings settings = SL_I3DL2_ENVIRONMENT_PRESET_DEFAULT;//默认情况

SLObjectItf audioplayer=NULL;//用SLObjectItf声明播放器接口对象
SLPlayItf slPlayItf=NULL;//播放器接口
SLAndroidSimpleBufferQueueItf slBufferQueueItf=NULL;//缓冲区队列接口

size_t buffersize =0;
void *buffer;
//将pcm数据添加到缓冲区中
void getQueueCallBack(SLAndroidSimpleBufferQueueItf slBufferQueueItf, void* context){

buffersize=0;

getPcm(&buffer,&buffersize);
if(buffer!=NULL&&buffersize!=0){
    //将得到的数据加入到队列中
    (\*slBufferQueueItf)->Enqueue(slBufferQueueItf,buffer,buffersize);
}

}

//创建引擎
void createEngine(){
slCreateEngine(&engineObject,0,NULL,0,NULL,NULL);//创建引擎
(*engineObject)->Realize(engineObject,SL_BOOLEAN_FALSE);//实现engineObject接口对象
(*engineObject)->GetInterface(engineObject,SL_IID_ENGINE,&engineEngine);//通过引擎调用接口初始化SLEngineItf
}

//创建混音器
void createMixVolume(){
(*engineEngine)->CreateOutputMix(engineEngine,&outputMixObject,0,0,0);//用引擎对象创建混音器接口对象
(*outputMixObject)->Realize(outputMixObject,SL_BOOLEAN_FALSE);//实现混音器接口对象
SLresult sLresult = (*outputMixObject)->GetInterface(outputMixObject,SL_IID_ENVIRONMENTALREVERB,&outputMixEnvironmentalReverb);//利用混音器实例对象接口初始化具体的混音器对象
//设置
if (SL_RESULT_SUCCESS == sLresult) {
(*outputMixEnvironmentalReverb)->
SetEnvironmentalReverbProperties(outputMixEnvironmentalReverb, &settings);
}
}

//创建播放器
void createPlayer(const char* path){
//初始化ffmpeg
int rate;
int channels;
createFFmpeg(&rate,&channels,path);
LOGE(“RATE %d”,rate);
LOGE(“channels %d”,channels);
/*
* typedef struct SLDataLocator_AndroidBufferQueue_ {
SLuint32 locatorType;//缓冲区队列类型
SLuint32 numBuffers;//buffer位数
} */

SLDataLocator_AndroidBufferQueue android_queue = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,2};
/\*\*

typedef struct SLDataFormat_PCM_ {
SLuint32 formatType; pcm
SLuint32 numChannels; 通道数
SLuint32 samplesPerSec; 采样率
SLuint32 bitsPerSample; 采样位数
SLuint32 containerSize; 包含位数
SLuint32 channelMask; 立体声
SLuint32 endianness; end标志位
} SLDataFormat_PCM;
*/
SLDataFormat_PCM pcm = {SL_DATAFORMAT_PCM,(SLuint32)channels,(SLuint32)rate*1000
,SL_PCMSAMPLEFORMAT_FIXED_16
,SL_PCMSAMPLEFORMAT_FIXED_16
,SL_SPEAKER_FRONT_LEFT|SL_SPEAKER_FRONT_RIGHT,SL_BYTEORDER_LITTLEENDIAN};

/\*

* typedef struct SLDataSource_ {
void *pLocator;//缓冲区队列
void *pFormat;//数据样式,配置信息
} SLDataSource;
* */
SLDataSource dataSource = {&android_queue,&pcm};

SLDataLocator_OutputMix slDataLocator_outputMix={SL_DATALOCATOR_OUTPUTMIX,outputMixObject};


SLDataSink slDataSink = {&slDataLocator_outputMix,NULL};


const SLInterfaceID ids[3]={SL_IID_BUFFERQUEUE,SL_IID_EFFECTSEND,SL_IID_VOLUME};
const SLboolean req[3]={SL_BOOLEAN_FALSE,SL_BOOLEAN_FALSE,SL_BOOLEAN_FALSE};

/\*

* SLresult (*CreateAudioPlayer) (
SLEngineItf self,
SLObjectItf * pPlayer,
SLDataSource *pAudioSrc,//数据设置
SLDataSink *pAudioSnk,//关联混音器
SLuint32 numInterfaces,
const SLInterfaceID * pInterfaceIds,
const SLboolean * pInterfaceRequired
);
* */
LOGE(“执行到此处”)
(*engineEngine)->CreateAudioPlayer(engineEngine,&audioplayer,&dataSource,&slDataSink,3,ids,req);
(*audioplayer)->Realize(audioplayer,SL_BOOLEAN_FALSE);
LOGE(“执行到此处2”)
(*audioplayer)->GetInterface(audioplayer,SL_IID_PLAY,&slPlayItf);//初始化播放器
//注册缓冲区,通过缓冲区里面的数据进行播放
(*audioplayer)->GetInterface(audioplayer,SL_IID_BUFFERQUEUE,&slBufferQueueItf);
//设置回调接口
(*slBufferQueueItf)->RegisterCallback(slBufferQueueItf,getQueueCallBack,NULL);
//播放
(*slPlayItf)->SetPlayState(slPlayItf,SL_PLAYSTATE_PLAYING);

//开始播放
getQueueCallBack(slBufferQueueItf,NULL);

}
//释放资源
void releaseResource(){
if(audioplayer!=NULL){
(*audioplayer)->Destroy(audioplayer);
audioplayer=NULL;
slBufferQueueItf=NULL;
slPlayItf=NULL;
}
if(outputMixObject!=NULL){
(*outputMixObject)->Destroy(outputMixObject);
outputMixObject=NULL;
outputMixEnvironmentalReverb=NULL;
}
if(engineObject!=NULL){
(*engineObject)->Destroy(engineObject);
engineObject=NULL;
engineEngine=NULL;
}
releaseFFmpeg();
}

AVFormatContext *pFormatCtx;
AVCodecContext *pCodecCtx;
AVCodec *pCodex;
AVPacket *packet;
AVFrame *frame;
SwrContext *swrContext;
uint8_t *out_buffer;
int out_channer_nb;
int audio_stream_idx=-1;
//opensl es调用 int * rate,int *channel
int createFFmpeg(int *rate,int *channel,const char* path){
av_register_all();
const char *input = path;
pFormatCtx = avformat_alloc_context();
LOGE(“Lujng %s”,input);
LOGE(“xxx %p”,pFormatCtx);
int error;
char buf[] = “”;
//打开视频地址并获取里面的内容(解封装)
if (error = avformat_open_input(&pFormatCtx, input, NULL, NULL) < 0) {
av_strerror(error, buf, 1024);
// LOGE(“%s” ,inputPath)
LOGE(“Couldn’t open file %s: %d(%s)”, input, error, buf);
// LOGE(“%d”,error)
LOGE(“打开视频失败”);
}
//3.获取视频信息
if(avformat_find_stream_info(pFormatCtx,NULL) < 0){
LOGE(“%s”,“获取视频信息失败”);
return -1;
}

int i=0;
for (int i = 0; i < pFormatCtx->nb_streams; ++i) {
    if (pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
        LOGE(" 找到音频id %d", pFormatCtx->streams[i]->codec->codec_type);
        audio_stream_idx=i;
        break;
    }
}

// mp3的解码器

// 获取音频编解码器
pCodecCtx=pFormatCtx->streams[audio_stream_idx]->codec;
LOGE("获取视频编码器上下文 %p ",pCodecCtx);

pCodex = avcodec\_find\_decoder(pCodecCtx->codec_id);
LOGE("获取视频编码 %p",pCodex);

if (avcodec\_open2(pCodecCtx, pCodex, NULL)<0) {
}
packet = (AVPacket \*)av\_malloc(sizeof(AVPacket));

// av_init_packet(packet);
// 音频数据

frame = av\_frame\_alloc();

// mp3 里面所包含的编码格式转换成 pcm SwcContext
swrContext = swr_alloc();

int length=0;
int got_frame;

// 44100*2
out_buffer = (uint8_t *) av_malloc(44100 * 2);
uint64_t out_ch_layout=AV_CH_LAYOUT_STEREO;
// 输出采样位数 16位
enum AVSampleFormat out_formart=AV_SAMPLE_FMT_S16;
//输出的采样率必须与输入相同
int out_sample_rate = pCodecCtx->sample_rate;

swr\_alloc\_set\_opts(swrContext, out_ch_layout, out_formart, out_sample_rate,
                   pCodecCtx->channel_layout, pCodecCtx->sample_fmt, pCodecCtx->sample_rate, 0,
                   NULL);

swr\_init(swrContext);

// 获取通道数 2
out_channer_nb = av_get_channel_layout_nb_channels(AV_CH_LAYOUT_STEREO);
*rate = pCodecCtx->sample_rate;
*channel = pCodecCtx->channels;
return 0;
}
//
int getPcm(void **pcm,size_t *pcm_size){
int frameCount=0;
int got_frame;
while (av_read_frame(pFormatCtx, packet) >= 0) {
if (packet->stream_index == audio_stream_idx) {
// 解码 mp3 编码格式frame----pcm frame
avcodec_decode_audio4(pCodecCtx, frame, &got_frame, packet);
if (got_frame) {
LOGE(“解码”);
/**
* int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,
const uint8_t **in , int in_count);