android 使用ffmpeg音视频播放（二）

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上篇是视频解码播放，这篇讲音频解码播放，解码过程同视频解码，不过取的是音频流，android播放用的audiotracker，因为实现简单。

编写Android代码

因为是audiotracker播放，在代理类里除了编写native方法，还要提供给c调用的返回audiotracker的设定采样率和声道的createAudioTrack（）。

public class YoungPlayer {    public native void render(String input,Surface surface);    public native void sound(String input,String output);    public native void play(String input,Surface surface);    static{        System.loadLibrary("avutil-54");        System.loadLibrary("swresample-1");        System.loadLibrary("avcodec-56");        System.loadLibrary("avformat-56");        System.loadLibrary("swscale-3");        System.loadLibrary("postproc-53");        System.loadLibrary("avfilter-5");        System.loadLibrary("avdevice-56");        System.loadLibrary("yuv");        System.loadLibrary("myffmpeg");    }    public AudioTrack createAudioTrack(int sampleRateInHz, int nb_channels){        //固定格式的音频码流        int audioFormat = AudioFormat.ENCODING_PCM_16BIT;        Log.i("yang", "nb_channels:"+nb_channels);        //声道布局        int channelConfig;        if(nb_channels == 1){            channelConfig = android.media.AudioFormat.CHANNEL_OUT_MONO;        }else if(nb_channels == 2){            channelConfig = android.media.AudioFormat.CHANNEL_OUT_STEREO;        }else{            channelConfig = android.media.AudioFormat.CHANNEL_OUT_STEREO;        }        int bufferSizeInBytes = AudioTrack.getMinBufferSize(sampleRateInHz, channelConfig, audioFormat);        AudioTrack audioTrack = new AudioTrack(                AudioManager.STREAM_MUSIC,                 sampleRateInHz, channelConfig,                 audioFormat,                 bufferSizeInBytes, AudioTrack.MODE_STREAM);        //播放        //audioTrack.play();        //写入PCM        //audioTrack.write(audioData, offsetInBytes, sizeInBytes);        return audioTrack;    }}

调用代码：

    public void sound(View btn){        String video = sp_video.getSelectedItem().toString();        final String input = new File(Environment.getExternalStorageDirectory(),video).getAbsolutePath();        final String output = new File(Environment.getExternalStorageDirectory(),"Output.pcm").getAbsolutePath();        //Surface传入到Native函数中，用于绘制        new Thread(new Runnable() {            public void run() {                player.sound(input,output);            }        }).start();    }

C/C++代码实现音频文件播放

#include "com_yang_ffmpegDemo_YoungPlayer.h"#include <stdlib.h>#include <unistd.h>#include <android/log.h>#define LOGI(FORMAT,...) __android_log_print(ANDROID_LOG_INFO,"jason",FORMAT,##__VA_ARGS__);#define LOGE(FORMAT,...) __android_log_print(ANDROID_LOG_ERROR,"jason",FORMAT,##__VA_ARGS__);#define MAX_AUDIO_FRME_SIZE 48000 * 4//封装格式#include "libavformat/avformat.h"//解码#include "libavcodec/avcodec.h"//缩放#include "libswscale/swscale.h"//重采样#include "libswresample/swresample.h"JNIEXPORT void JNICALL Java_com_yang_ffmpegDemo_YoungPlayer_sound(JNIEnv *env, jobject jthiz, jstring input_jstr, jstring output_jstr){    const char* input_cstr = (*env)->GetStringUTFChars(env,input_jstr,NULL);    const char* output_cstr = (*env)->GetStringUTFChars(env,output_jstr,NULL);    LOGI("%s","sound");    //注册组件    av_register_all();    AVFormatContext *pFormatCtx = avformat_alloc_context();    //打开音频文件    if(avformat_open_input(&pFormatCtx,input_cstr,NULL,NULL) != 0){        LOGI("%s","无法打开音频文件");        return;    }    //获取输入文件信息    if(avformat_find_stream_info(pFormatCtx,NULL) < 0){        LOGI("%s","无法获取输入文件信息");        return;    }    //获取音频流索引位置    int i = 0, audio_stream_idx = -1;    for(; i < pFormatCtx->nb_streams;i++){        if(pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO){            audio_stream_idx = i;            break;        }    }    //获取解码器    AVCodecContext *codecCtx = pFormatCtx->streams[audio_stream_idx]->codec;    AVCodec *codec = avcodec_find_decoder(codecCtx->codec_id);    if(codec == NULL){        LOGI("%s","无法获取解码器");        return;    }    //打开解码器    if(avcodec_open2(codecCtx,codec,NULL) < 0){        LOGI("%s","无法打开解码器");        return;    }    //压缩数据    AVPacket *packet = (AVPacket *)av_malloc(sizeof(AVPacket));    //解压缩数据    AVFrame *frame = av_frame_alloc();    //frame->16bit 44100 PCM 统一音频采样格式与采样率    SwrContext *swrCtx = swr_alloc();    //重采样设置参数-------------start    //输入的采样格式    enum AVSampleFormat in_sample_fmt = codecCtx->sample_fmt;    //输出采样格式16bit PCM    enum AVSampleFormat out_sample_fmt = AV_SAMPLE_FMT_S16;    //输入采样率    int in_sample_rate = codecCtx->sample_rate;    //输出采样率    int out_sample_rate = in_sample_rate;    //获取输入的声道布局    //根据声道个数获取默认的声道布局（2个声道，默认立体声stereo）    //av_get_default_channel_layout(codecCtx->channels);    uint64_t in_ch_layout = codecCtx->channel_layout;    //输出的声道布局（立体声）    uint64_t out_ch_layout = AV_CH_LAYOUT_STEREO;    swr_alloc_set_opts(swrCtx,            out_ch_layout,out_sample_fmt,out_sample_rate,            in_ch_layout,in_sample_fmt,in_sample_rate,            0, NULL);    swr_init(swrCtx);    //输出的声道个数    int out_channel_nb = av_get_channel_layout_nb_channels(out_ch_layout);    //重采样设置参数-------------end    //JNI begin------------------    //JasonPlayer    jclass player_class = (*env)->GetObjectClass(env,jthiz);    //AudioTrack对象    jmethodID create_audio_track_mid = (*env)->GetMethodID(env,player_class,"createAudioTrack","(II)Landroid/media/AudioTrack;");    jobject audio_track = (*env)->CallObjectMethod(env,jthiz,create_audio_track_mid,out_sample_rate,out_channel_nb);    //调用AudioTrack.play方法    jclass audio_track_class = (*env)->GetObjectClass(env,audio_track);    jmethodID audio_track_play_mid = (*env)->GetMethodID(env,audio_track_class,"play","()V");    (*env)->CallVoidMethod(env,audio_track,audio_track_play_mid);    //AudioTrack.write    jmethodID audio_track_write_mid = (*env)->GetMethodID(env,audio_track_class,"write","([BII)I");    //JNI end------------------    FILE *fp_pcm = fopen(output_cstr,"wb");    //16bit 44100 PCM 数据    uint8_t *out_buffer = (uint8_t *)av_malloc(MAX_AUDIO_FRME_SIZE);    int got_frame = 0,index = 0, ret;    //不断读取压缩数据    while(av_read_frame(pFormatCtx,packet) >= 0){        //解码音频类型的Packet        if(packet->stream_index == audio_stream_idx){            //解码            ret = avcodec_decode_audio4(codecCtx,frame,&got_frame,packet);            if(ret < 0){                LOGI("%s","解码完成");            }            //解码一帧成功            if(got_frame > 0){                LOGI("解码：%d",index++);                swr_convert(swrCtx, &out_buffer, MAX_AUDIO_FRME_SIZE,(const uint8_t **)frame->data,frame->nb_samples);                //获取sample的size                int out_buffer_size = av_samples_get_buffer_size(NULL, out_channel_nb,                        frame->nb_samples, out_sample_fmt, 1);                fwrite(out_buffer,1,out_buffer_size,fp_pcm);                //out_buffer缓冲区数据，转成byte数组                jbyteArray audio_sample_array = (*env)->NewByteArray(env,out_buffer_size);                jbyte* sample_bytep = (*env)->GetByteArrayElements(env,audio_sample_array,NULL);                //out_buffer的数据复制到sampe_bytep                memcpy(sample_bytep,out_buffer,out_buffer_size);                //同步                (*env)->ReleaseByteArrayElements(env,audio_sample_array,sample_bytep,0);                //AudioTrack.write PCM数据                (*env)->CallIntMethod(env,audio_track,audio_track_write_mid,                        audio_sample_array,0,out_buffer_size);                //释放局部引用                (*env)->DeleteLocalRef(env,audio_sample_array);                usleep(1000 * 16);            }        }        av_free_packet(packet);    }    av_frame_free(&frame);    av_free(out_buffer);    swr_free(&swrCtx);    avcodec_close(codecCtx);    avformat_close_input(&pFormatCtx);    (*env)->ReleaseStringUTFChars(env,input_jstr,input_cstr);    (*env)->ReleaseStringUTFChars(env,output_jstr,output_cstr);}

以上通过C调用Java类audiotracker的播放功能实现，仅仅几行代码就实现了。
PCM 数据播放在开发中也经常使用，例如自己编写播放器，解码之后的音频PCM数据，就可以通过OpenSL 播放，比用Java层的AudioTrack更快，延迟更低。如果要达到更好的效果可以参考Android+FFmpeg+OpenSL ES音频解码播放。

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