android 使用ffmpeg音视频播放（二）

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。