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。