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 Android教程網 >> Android技術 >> 關於Android編程 >> Android本地視頻播放器開發--ffmpeg解碼視頻文件中的音頻

Android本地視頻播放器開發--ffmpeg解碼視頻文件中的音頻

編輯：關於Android編程

[cpp]
#include <stdio.h>  
#include <stdlib.h>  
#include <string.h>  
 
#include <assert.h>  
#include <android/log.h>  
 
// for native audio  
#include <SLES/OpenSLES.h>  
#include <SLES/OpenSLES_Android.h>  
 
#include "VideoPlayerDecode.h"  
#include "../ffmpeg/libavutil/avutil.h"  
#include "../ffmpeg/libavcodec/avcodec.h"  
#include "../ffmpeg/libavformat/avformat.h"  
 
#define LOGI(...) ((void)__android_log_print(ANDROID_LOG_INFO, "graduation", __VA_ARGS__))  
 
AVFormatContext *pFormatCtx = NULL; 
int             audioStream, delay_time, videoFlag = 0; 
AVCodecContext  *aCodecCtx; 
AVCodec         *aCodec; 
AVFrame         *aFrame; 
AVPacket        packet; 
int  frameFinished = 0; 
 
// engine interfaces  
static SLObjectItf engineObject = NULL; 
static SLEngineItf engineEngine; 
 
// output mix interfaces  
static SLObjectItf outputMixObject = NULL; 
static SLEnvironmentalReverbItf outputMixEnvironmentalReverb = NULL; 
 
// buffer queue player interfaces  
static SLObjectItf bqPlayerObject = NULL; 
static SLPlayItf bqPlayerPlay; 
static SLAndroidSimpleBufferQueueItf bqPlayerBufferQueue; 
static SLEffectSendItf bqPlayerEffectSend; 
static SLMuteSoloItf bqPlayerMuteSolo; 
static SLVolumeItf bqPlayerVolume; 
 
// aux effect on the output mix, used by the buffer queue player  
static const SLEnvironmentalReverbSettings reverbSettings = 
    SL_I3DL2_ENVIRONMENT_PRESET_STONECORRIDOR; 
 
// file descriptor player interfaces  
static SLObjectItf fdPlayerObject = NULL; 
static SLPlayItf fdPlayerPlay; 
static SLSeekItf fdPlayerSeek; 
static SLMuteSoloItf fdPlayerMuteSolo; 
static SLVolumeItf fdPlayerVolume; 
 
// pointer and size of the next player buffer to enqueue, and number of remaining buffers  
static short *nextBuffer; 
static unsigned nextSize; 
static int nextCount; 
 
// this callback handler is called every time a buffer finishes playing  
void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void *context) 
{ 
    assert(bq == bqPlayerBufferQueue); 
    assert(NULL == context); 
    // for streaming playback, replace this test by logic to find and fill the next buffer  
    if (--nextCount > 0 && NULL != nextBuffer && 0 != nextSize) { 
        SLresult result; 
        // enqueue another buffer  
        result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, nextBuffer, nextSize); 
        // the most likely other result is SL_RESULT_BUFFER_INSUFFICIENT,  
        // which for this code example would indicate a programming error  
        assert(SL_RESULT_SUCCESS == result); 
    } 
} 
 
 
void createEngine(JNIEnv* env, jclass clazz) 
{ 
    SLresult result; 
 
    // create engine  
    result = slCreateEngine(&engineObject, 0, NULL, 0, NULL, NULL); 
    assert(SL_RESULT_SUCCESS == result); 
 
    // realize the engine  
    result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE); 
    assert(SL_RESULT_SUCCESS == result); 
 
    // get the engine interface, which is needed in order to create other objects  
    result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine); 
    assert(SL_RESULT_SUCCESS == result); 
 
    // create output mix, with environmental reverb specified as a non-required interface  
    const SLInterfaceID ids[1] = {SL_IID_ENVIRONMENTALREVERB}; 
    const SLboolean req[1] = {SL_BOOLEAN_FALSE}; 
    result = (*engineEngine)->CreateOutputMix(engineEngine, &outputMixObject, 1, ids, req); 
    assert(SL_RESULT_SUCCESS == result); 
 
    // realize the output mix  
    result = (*outputMixObject)->Realize(outputMixObject, SL_BOOLEAN_FALSE); 
    assert(SL_RESULT_SUCCESS == result); 
 
    // get the environmental reverb interface  
    // this could fail if the environmental reverb effect is not available,  
    // either because the feature is not present, excessive CPU load, or  
    // the required MODIFY_AUDIO_SETTINGS permission was not requested and granted  
    result = (*outputMixObject)->GetInterface(outputMixObject, SL_IID_ENVIRONMENTALREVERB, 
            &outputMixEnvironmentalReverb); 
    if (SL_RESULT_SUCCESS == result) { 
        result = (*outputMixEnvironmentalReverb)->SetEnvironmentalReverbProperties( 
                outputMixEnvironmentalReverb, &reverbSettings); 
    } 
    // ignore unsuccessful result codes for environmental reverb, as it is optional for this example  
} 
 
void createBufferQueueAudioPlayer(JNIEnv* env, jclass clazz, int rate, int channel,int bitsPerSample) 
{ 
    SLresult result; 
 
    // configure audio source  
    SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 2}; 
//    SLDataFormat_PCM format_pcm = {SL_DATAFORMAT_PCM, 2, SL_SAMPLINGRATE_16,  
//        SL_PCMSAMPLEFORMAT_FIXED_16, SL_PCMSAMPLEFORMAT_FIXED_16,  
//        SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT, SL_BYTEORDER_LITTLEENDIAN};  
   SLDataFormat_PCM format_pcm; 
   format_pcm.formatType = SL_DATAFORMAT_PCM; 
format_pcm.numChannels = channel; 
format_pcm.samplesPerSec = rate * 1000; 
 format_pcm.bitsPerSample = bitsPerSample; 
 format_pcm.containerSize = 16; 
if(channel == 2) 
format_pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; 
else 
format_pcm.channelMask = SL_SPEAKER_FRONT_CENTER; 
format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; 
    SLDataSource audioSrc = {&loc_bufq, &format_pcm}; 
 
    // configure audio sink  
    SLDataLocator_OutputMix loc_outmix = {SL_DATALOCATOR_OUTPUTMIX, outputMixObject}; 
    SLDataSink audioSnk = {&loc_outmix, NULL}; 
 
    // create audio player  
    const SLInterfaceID ids[3] = {SL_IID_BUFFERQUEUE, SL_IID_EFFECTSEND, 
            /*SL_IID_MUTESOLO,*/ SL_IID_VOLUME}; 
    const SLboolean req[3] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, 
            /*SL_BOOLEAN_TRUE,*/ SL_BOOLEAN_TRUE}; 
    result = (*engineEngine)->CreateAudioPlayer(engineEngine, &bqPlayerObject, &audioSrc, &audioSnk, 
            3, ids, req); 
    assert(SL_RESULT_SUCCESS == result); 
// realize the player  
    result = (*bqPlayerObject)->Realize(bqPlayerObject, SL_BOOLEAN_FALSE); 
    assert(SL_RESULT_SUCCESS == result); 
 
    // get the play interface  
    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_PLAY, &bqPlayerPlay); 
    assert(SL_RESULT_SUCCESS == result); 
 
    // get the buffer queue interface  
    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_BUFFERQUEUE, 
            &bqPlayerBufferQueue); 
    assert(SL_RESULT_SUCCESS == result); 
 
    // register callback on the buffer queue  
    result = (*bqPlayerBufferQueue)->RegisterCallback(bqPlayerBufferQueue, bqPlayerCallback, NULL); 
    assert(SL_RESULT_SUCCESS == result); 
 
    // get the effect send interface  
    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_EFFECTSEND, 
            &bqPlayerEffectSend); 
    assert(SL_RESULT_SUCCESS == result); 
 
#if 0   // mute/solo is not supported for sources that are known to be mono, as this is  
    // get the mute/solo interface  
    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_MUTESOLO, &bqPlayerMuteSolo); 
    assert(SL_RESULT_SUCCESS == result); 
#endif  
 
    // get the volume interface  
    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_VOLUME, &bqPlayerVolume); 
    assert(SL_RESULT_SUCCESS == result); 
 
// set the player's state to playing  
    result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING); 
    assert(SL_RESULT_SUCCESS == result); 
 
} 
 
void AudioWrite(const void*buffer, int size) 
{ 
    (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, buffer, size); 
} 
 
    JNIEXPORT jint JNICALL Java_com_zhangjie_graduation_videopalyer_jni_VideoPlayerDecode_VideoPlayer 
(JNIEnv *env, jclass clz, jstring fileName) 
{ 
    const char* local_title = (*env)->GetStringUTFChars(env, fileName, NULL); 
    av_register_all();//注冊所有支持的文件格式以及編解碼器  
    /*
     *只讀取文件頭，並不會填充流信息
     */ 
    if(avformat_open_input(&pFormatCtx, local_title, NULL, NULL) != 0) 
        return -1; 
    /*
     *獲取文件中的流信息，此函數會讀取packet，並確定文件中所有流信息，
     *設置pFormatCtx->streams指向文件中的流，但此函數並不會改變文件指針，
     *讀取的packet會給後面的解碼進行處理。
     */ 
    if(avformat_find_stream_info(pFormatCtx, NULL) < 0) 
        return -1; 
    /*
     *輸出文件的信息，也就是我們在使用ffmpeg時能夠看到的文件詳細信息，
     *第二個參數指定輸出哪條流的信息，-1代表ffmpeg自己選擇。最後一個參數用於
     *指定dump的是不是輸出文件，我們的dump是輸入文件，因此一定要為0
     */ 
    av_dump_format(pFormatCtx, -1, local_title, 0); 
    int i = 0; 
    for(i=0; i< pFormatCtx->nb_streams; i++) 
    { 
        if(pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO){ 
            audioStream = i; 
            break; 
        } 
    } 
 
    if(audioStream < 0)return -1; 
    aCodecCtx = pFormatCtx->streams[audioStream]->codec; 
    aCodec = avcodec_find_decoder(aCodecCtx->codec_id); 
    if(avcodec_open2(aCodecCtx, aCodec, NULL) < 0)return -1; 
    aFrame = avcodec_alloc_frame(); 
    if(aFrame == NULL)return -1; 
    int ret; 
    createEngine(env, clz); 
    int flag_start = 0; 
    while(videoFlag != -1) 
    { 
        if(av_read_frame(pFormatCtx, &packet) < 0)break; 
        if(packet.stream_index == audioStream) 
        { 
            ret = avcodec_decode_audio4(aCodecCtx, aFrame, &frameFinished, &packet); 
            if(ret > 0 && frameFinished) 
            { 
                if(flag_start == 0) 
                { 
                    flag_start = 1; 
                    createBufferQueueAudioPlayer(env, clz, aCodecCtx->sample_rate, aCodecCtx->channels, SL_PCMSAMPLEFORMAT_FIXED_16); 
                } 
                int data_size = av_samples_get_buffer_size( 
                        aFrame->linesize,aCodecCtx->channels, 
                        aFrame->nb_samples,aCodecCtx->sample_fmt, 1); 
                LOGI("audioDecodec  :%d : %d, :%d    :%d",data_size,aCodecCtx->channels,aFrame->nb_samples,aCodecCtx->sample_rate); 
                (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, aFrame->data[0], data_size); 
            } 
 
        } 
        usleep(5000); 
        while(videoFlag != 0) 
        { 
            if(videoFlag == 1)//暫停  
            { 
                sleep(1); 
            }else if(videoFlag == -1) //停止  
            { 
                break; 
            } 
        } 
        av_free_packet(&packet); 
    } 
    av_free(aFrame); 
    avcodec_close(aCodecCtx); 
    avformat_close_input(&pFormatCtx); 
    (*env)->ReleaseStringUTFChars(env, fileName, local_title); 
} 
 
JNIEXPORT jint JNICALL Java_com_zhangjie_graduation_videopalyer_jni_VideoPlayerDecode_VideoPlayerPauseOrPlay 
  (JNIEnv *env, jclass clz) 
{ 
        if(videoFlag == 1) 
        { 
                videoFlag = 0; 
        }else if(videoFlag == 0){ 
                videoFlag = 1; 
        } 
        return videoFlag; 
} 
 
JNIEXPORT jint JNICALL Java_com_zhangjie_graduation_videopalyer_jni_VideoPlayerDecode_VideoPlayerStop 
  (JNIEnv *env, jclass clz) 
{ 
        videoFlag = -1; 
} 

#include <stdio.h>
#include <stdlib.h>
#include <string.h>

#include <assert.h>
#include <android/log.h>

// for native audio
#include <SLES/OpenSLES.h>
#include <SLES/OpenSLES_Android.h>

#include "VideoPlayerDecode.h"
#include "../ffmpeg/libavutil/avutil.h"
#include "../ffmpeg/libavcodec/avcodec.h"
#include "../ffmpeg/libavformat/avformat.h"

#define LOGI(...) ((void)__android_log_print(ANDROID_LOG_INFO, "graduation", __VA_ARGS__))

AVFormatContext *pFormatCtx = NULL;
int             audioStream, delay_time, videoFlag = 0;
AVCodecContext  *aCodecCtx;
AVCodec         *aCodec;
AVFrame         *aFrame;
AVPacket        packet;
int  frameFinished = 0;

// engine interfaces
static SLObjectItf engineObject = NULL;
static SLEngineItf engineEngine;

// output mix interfaces
static SLObjectItf outputMixObject = NULL;
static SLEnvironmentalReverbItf outputMixEnvironmentalReverb = NULL;

// buffer queue player interfaces
static SLObjectItf bqPlayerObject = NULL;
static SLPlayItf bqPlayerPlay;
static SLAndroidSimpleBufferQueueItf bqPlayerBufferQueue;
static SLEffectSendItf bqPlayerEffectSend;
static SLMuteSoloItf bqPlayerMuteSolo;
static SLVolumeItf bqPlayerVolume;

// aux effect on the output mix, used by the buffer queue player
static const SLEnvironmentalReverbSettings reverbSettings =
    SL_I3DL2_ENVIRONMENT_PRESET_STONECORRIDOR;

// file descriptor player interfaces
static SLObjectItf fdPlayerObject = NULL;
static SLPlayItf fdPlayerPlay;
static SLSeekItf fdPlayerSeek;
static SLMuteSoloItf fdPlayerMuteSolo;
static SLVolumeItf fdPlayerVolume;

// pointer and size of the next player buffer to enqueue, and number of remaining buffers
static short *nextBuffer;
static unsigned nextSize;
static int nextCount;

// this callback handler is called every time a buffer finishes playing
void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
{
    assert(bq == bqPlayerBufferQueue);
    assert(NULL == context);
    // for streaming playback, replace this test by logic to find and fill the next buffer
    if (--nextCount > 0 && NULL != nextBuffer && 0 != nextSize) {
        SLresult result;
        // enqueue another buffer
        result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, nextBuffer, nextSize);
        // the most likely other result is SL_RESULT_BUFFER_INSUFFICIENT,
        // which for this code example would indicate a programming error
        assert(SL_RESULT_SUCCESS == result);
    }
}


void createEngine(JNIEnv* env, jclass clazz)
{
 SLresult result;

    // create engine
    result = slCreateEngine(&engineObject, 0, NULL, 0, NULL, NULL);
    assert(SL_RESULT_SUCCESS == result);

    // realize the engine
    result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE);
    assert(SL_RESULT_SUCCESS == result);

    // get the engine interface, which is needed in order to create other objects
    result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine);
    assert(SL_RESULT_SUCCESS == result);

    // create output mix, with environmental reverb specified as a non-required interface
    const SLInterfaceID ids[1] = {SL_IID_ENVIRONMENTALREVERB};
    const SLboolean req[1] = {SL_BOOLEAN_FALSE};
    result = (*engineEngine)->CreateOutputMix(engineEngine, &outputMixObject, 1, ids, req);
    assert(SL_RESULT_SUCCESS == result);

    // realize the output mix
    result = (*outputMixObject)->Realize(outputMixObject, SL_BOOLEAN_FALSE);
    assert(SL_RESULT_SUCCESS == result);

    // get the environmental reverb interface
    // this could fail if the environmental reverb effect is not available,
    // either because the feature is not present, excessive CPU load, or
    // the required MODIFY_AUDIO_SETTINGS permission was not requested and granted
    result = (*outputMixObject)->GetInterface(outputMixObject, SL_IID_ENVIRONMENTALREVERB,
            &outputMixEnvironmentalReverb);
    if (SL_RESULT_SUCCESS == result) {
        result = (*outputMixEnvironmentalReverb)->SetEnvironmentalReverbProperties(
                outputMixEnvironmentalReverb, &reverbSettings);
    }
    // ignore unsuccessful result codes for environmental reverb, as it is optional for this example
}

void createBufferQueueAudioPlayer(JNIEnv* env, jclass clazz, int rate, int channel,int bitsPerSample)
{
 SLresult result;

    // configure audio source
    SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 2};
//    SLDataFormat_PCM format_pcm = {SL_DATAFORMAT_PCM, 2, SL_SAMPLINGRATE_16,
//        SL_PCMSAMPLEFORMAT_FIXED_16, SL_PCMSAMPLEFORMAT_FIXED_16,
//        SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT, SL_BYTEORDER_LITTLEENDIAN};
   SLDataFormat_PCM format_pcm;
   format_pcm.formatType = SL_DATAFORMAT_PCM;
format_pcm.numChannels = channel;
format_pcm.samplesPerSec = rate * 1000;
 format_pcm.bitsPerSample = bitsPerSample;
 format_pcm.containerSize = 16;
if(channel == 2)
format_pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
else
format_pcm.channelMask = SL_SPEAKER_FRONT_CENTER;
format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
    SLDataSource audioSrc = {&loc_bufq, &format_pcm};

    // configure audio sink
    SLDataLocator_OutputMix loc_outmix = {SL_DATALOCATOR_OUTPUTMIX, outputMixObject};
    SLDataSink audioSnk = {&loc_outmix, NULL};

    // create audio player
    const SLInterfaceID ids[3] = {SL_IID_BUFFERQUEUE, SL_IID_EFFECTSEND,
            /*SL_IID_MUTESOLO,*/ SL_IID_VOLUME};
    const SLboolean req[3] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE,
            /*SL_BOOLEAN_TRUE,*/ SL_BOOLEAN_TRUE};
    result = (*engineEngine)->CreateAudioPlayer(engineEngine, &bqPlayerObject, &audioSrc, &audioSnk,
            3, ids, req);
    assert(SL_RESULT_SUCCESS == result);
// realize the player
    result = (*bqPlayerObject)->Realize(bqPlayerObject, SL_BOOLEAN_FALSE);
    assert(SL_RESULT_SUCCESS == result);

    // get the play interface
    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_PLAY, &bqPlayerPlay);
    assert(SL_RESULT_SUCCESS == result);

    // get the buffer queue interface
    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_BUFFERQUEUE,
            &bqPlayerBufferQueue);
    assert(SL_RESULT_SUCCESS == result);

    // register callback on the buffer queue
    result = (*bqPlayerBufferQueue)->RegisterCallback(bqPlayerBufferQueue, bqPlayerCallback, NULL);
    assert(SL_RESULT_SUCCESS == result);

    // get the effect send interface
    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_EFFECTSEND,
            &bqPlayerEffectSend);
    assert(SL_RESULT_SUCCESS == result);

#if 0   // mute/solo is not supported for sources that are known to be mono, as this is
    // get the mute/solo interface
    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_MUTESOLO, &bqPlayerMuteSolo);
    assert(SL_RESULT_SUCCESS == result);
#endif

    // get the volume interface
    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_VOLUME, &bqPlayerVolume);
    assert(SL_RESULT_SUCCESS == result);

// set the player's state to playing
    result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING);
    assert(SL_RESULT_SUCCESS == result);

}

void AudioWrite(const void*buffer, int size)
{
 (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, buffer, size);
}

 JNIEXPORT jint JNICALL Java_com_zhangjie_graduation_videopalyer_jni_VideoPlayerDecode_VideoPlayer
(JNIEnv *env, jclass clz, jstring fileName)
{
 const char* local_title = (*env)->GetStringUTFChars(env, fileName, NULL);
 av_register_all();//注冊所有支持的文件格式以及編解碼器
 /*
  *只讀取文件頭，並不會填充流信息
  */
 if(avformat_open_input(&pFormatCtx, local_title, NULL, NULL) != 0)
  return -1;
 /*
  *獲取文件中的流信息，此函數會讀取packet，並確定文件中所有流信息，
  *設置pFormatCtx->streams指向文件中的流，但此函數並不會改變文件指針，
  *讀取的packet會給後面的解碼進行處理。
  */
 if(avformat_find_stream_info(pFormatCtx, NULL) < 0)
  return -1;
 /*
  *輸出文件的信息，也就是我們在使用ffmpeg時能夠看到的文件詳細信息，
  *第二個參數指定輸出哪條流的信息，-1代表ffmpeg自己選擇。最後一個參數用於
  *指定dump的是不是輸出文件，我們的dump是輸入文件，因此一定要為0
  */
 av_dump_format(pFormatCtx, -1, local_title, 0);
 int i = 0;
 for(i=0; i< pFormatCtx->nb_streams; i++)
 {
  if(pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO){
   audioStream = i;
   break;
  }
 }

 if(audioStream < 0)return -1;
 aCodecCtx = pFormatCtx->streams[audioStream]->codec;
 aCodec = avcodec_find_decoder(aCodecCtx->codec_id);
 if(avcodec_open2(aCodecCtx, aCodec, NULL) < 0)return -1;
 aFrame = avcodec_alloc_frame();
 if(aFrame == NULL)return -1;
 int ret;
 createEngine(env, clz);
 int flag_start = 0;
 while(videoFlag != -1)
 {
  if(av_read_frame(pFormatCtx, &packet) < 0)break;
  if(packet.stream_index == audioStream)
  {
   ret = avcodec_decode_audio4(aCodecCtx, aFrame, &frameFinished, &packet);
   if(ret > 0 && frameFinished)
   {
    if(flag_start == 0)
    {
     flag_start = 1;
     createBufferQueueAudioPlayer(env, clz, aCodecCtx->sample_rate, aCodecCtx->channels, SL_PCMSAMPLEFORMAT_FIXED_16);
    }
    int data_size = av_samples_get_buffer_size(
      aFrame->linesize,aCodecCtx->channels,
      aFrame->nb_samples,aCodecCtx->sample_fmt, 1);
    LOGI("audioDecodec  :%d : %d, :%d    :%d",data_size,aCodecCtx->channels,aFrame->nb_samples,aCodecCtx->sample_rate);
    (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, aFrame->data[0], data_size);
   }

  }
  usleep(5000);
  while(videoFlag != 0)
  {
   if(videoFlag == 1)//暫停
   {
    sleep(1);
   }else if(videoFlag == -1) //停止
   {
    break;
   }
  }
  av_free_packet(&packet);
 }
 av_free(aFrame);
 avcodec_close(aCodecCtx);
 avformat_close_input(&pFormatCtx);
 (*env)->ReleaseStringUTFChars(env, fileName, local_title);
}

JNIEXPORT jint JNICALL Java_com_zhangjie_graduation_videopalyer_jni_VideoPlayerDecode_VideoPlayerPauseOrPlay
  (JNIEnv *env, jclass clz)
{
        if(videoFlag == 1)
        {
                videoFlag = 0;
        }else if(videoFlag == 0){
                videoFlag = 1;
        }
        return videoFlag;
}

JNIEXPORT jint JNICALL Java_com_zhangjie_graduation_videopalyer_jni_VideoPlayerDecode_VideoPlayerStop
  (JNIEnv *env, jclass clz)
{
        videoFlag = -1;
}
然後就是需要在Android.mk中添加OpenSL ES的庫支持，代碼如下:


[cpp]
LOCAL_PATH := $(call my-dir) 
#######################################################  
##########      ffmpeg-prebuilt     #######  
#######################################################  
#declare the prebuilt library  
include $(CLEAR_VARS) 
LOCAL_MODULE := ffmpeg-prebuilt 
LOCAL_SRC_FILES := ffmpeg/android/armv7-a/libffmpeg-neon.so 
LOCAL_EXPORT_C_INCLUDES := ffmpeg/android/armv7-a/include 
LOCAL_EXPORT_LDLIBS := ffmpeg/android/armv7-a/libffmpeg-neon.so 
LOCAL_PRELINK_MODULE := true 
include $(PREBUILT_SHARED_LIBRARY) 
 
########################################################  
##      ffmpeg-test-neno.so     ########  
########################################################  
include $(CLEAR_VARS) 
TARGET_ARCH_ABI=armeabi-v7a 
LOCAL_ARM_MODE=arm 
LOCAL_ARM_NEON=true 
LOCAL_ALLOW_UNDEFINED_SYMBOLS=false 
LOCAL_MODULE := ffmpeg-test-neon 
#LOCAL_SRC_FILES := jniffmpeg/VideoPlayerDecode.c  
LOCAL_SRC_FILES := jniffmpeg/Decodec_Audio.c     
 
LOCAL_C_INCLUDES := $(LOCAL_PATH)/ffmpeg/android/armv7-a/include \ 
            $(LOCAL_PATH)/ffmpeg \ 
            $(LOCAL_PATH)/ffmpeg/libavutil \ 
            $(LOCAL_PATH)/ffmpeg/libavcodec \ 
            $(LOCAL_PATH)/ffmpeg/libavformat \ 
            $(LOCAL_PATH)/ffmpeg/libavcodec \ 
            $(LOCAL_PATH)/ffmpeg/libswscale \ 
            $(LOCAL_PATH)/jniffmpeg \ 
            $(LOCAL_PATH) 
LOCAL_SHARED_LIBRARY := ffmpeg-prebuilt 
LOCAL_LDLIBS    := -llog -lGLESv2 -ljnigraphics -lz -lm $(LOCAL_PATH)/ffmpeg/android/armv7-a/libffmpeg-neon.so 
LOCAL_LDLIBS    += -lOpenSLES  
include $(BUILD_SHARED_LIBRARY) 

LOCAL_PATH := $(call my-dir)
#######################################################
##########  ffmpeg-prebuilt  #######
#######################################################
#declare the prebuilt library
include $(CLEAR_VARS)
LOCAL_MODULE := ffmpeg-prebuilt
LOCAL_SRC_FILES := ffmpeg/android/armv7-a/libffmpeg-neon.so
LOCAL_EXPORT_C_INCLUDES := ffmpeg/android/armv7-a/include
LOCAL_EXPORT_LDLIBS := ffmpeg/android/armv7-a/libffmpeg-neon.so
LOCAL_PRELINK_MODULE := true
include $(PREBUILT_SHARED_LIBRARY)

########################################################
##  ffmpeg-test-neno.so  ########
########################################################
include $(CLEAR_VARS)
TARGET_ARCH_ABI=armeabi-v7a
LOCAL_ARM_MODE=arm
LOCAL_ARM_NEON=true
LOCAL_ALLOW_UNDEFINED_SYMBOLS=false
LOCAL_MODULE := ffmpeg-test-neon
#LOCAL_SRC_FILES := jniffmpeg/VideoPlayerDecode.c
LOCAL_SRC_FILES := jniffmpeg/Decodec_Audio.c 

LOCAL_C_INCLUDES := $(LOCAL_PATH)/ffmpeg/android/armv7-a/include \
      $(LOCAL_PATH)/ffmpeg \
      $(LOCAL_PATH)/ffmpeg/libavutil \
      $(LOCAL_PATH)/ffmpeg/libavcodec \
      $(LOCAL_PATH)/ffmpeg/libavformat \
      $(LOCAL_PATH)/ffmpeg/libavcodec \
      $(LOCAL_PATH)/ffmpeg/libswscale \
      $(LOCAL_PATH)/jniffmpeg \
      $(LOCAL_PATH)
LOCAL_SHARED_LIBRARY := ffmpeg-prebuilt
LOCAL_LDLIBS    := -llog -lGLESv2 -ljnigraphics -lz -lm $(LOCAL_PATH)/ffmpeg/android/armv7-a/libffmpeg-neon.so
LOCAL_LDLIBS    += -lOpenSLES
include $(BUILD_SHARED_LIBRARY)
由於OpenSLES最低版本需要9所以要在Application.mk中添加平台


[cpp]
# The ARMv7 is significanly faster due to the use of the hardware FPU  
APP_ABI := armeabi  
APP_PLATFORM := android-9 
APP_STL := stlport_static 
APP_CPPFLAGS += -fno-rtti 
#APP_ABI := armeabi 

# The ARMv7 is significanly faster due to the use of the hardware FPU
APP_ABI := armeabi
APP_PLATFORM := android-9
APP_STL := stlport_static
APP_CPPFLAGS += -fno-rtti
#APP_ABI := armeabi
最後在終端運行ndk-build,就會將代碼添加到[cpp] view plaincopyprint?ffmpeg-test-neon.so這個庫中 

ffmpeg-test-neon.so這個庫中
最後在Android端調用

[cpp]
VideoPlayer這個函數就會自動播放視頻的聲音，測試發現雖然聲音正常但是有雜音，可能采樣率設置的不對，獲取其他的配置有問題，下一章著重解決這個問題，同時使用隊列的方式來從視頻中取音頻包，然後從音頻包隊列中取出，然後解碼播放。 

VideoPlayer這個函數就會自動播放視頻的聲音，測試發現雖然聲音正常但是有雜音，可能采樣率設置的不對，獲取其他的配置有問題，下一章著重解決這個問題，同時使用隊列的方式來從視頻中取音頻包，然後從音頻包隊列中取出，然後解碼播放。

 

 

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