FFmpeg总结（十一）用ffmpeg进行转格式，Android下播放网络音频流

请尊重分享成果，转载请注明出处，本文来自逆流的鱼yuiop，原文链接： http://blog.csdn.net/hejjunlin/article/details/71308337

思路： 1、mp3转成pcm(音频数据)，ffmpeg做的事 2、OpenSL ES引擎创建AudioPlayer，实际调用了AudioTrack

遇到的错误： Error #include nested too deeply 原因：c文件互相引用 解决方案：

• 1、将两个头文件共用的那一部分抽出来单独建一个头文件。
• 2、加预处理#ifndef.. #define…#endif

x86平台没有编译出来so，怀疑存在版本不兼容，编译别的相关so，x86下没有异常。有空这里再更新下原因

studio写ndk相当方便：

工程结构：

Java代码：

package com.hejunlin.ffmpegaudio; import android.support.v7.app.AppCompatActivity; import android.os.Bundle; import android.view.View; import android.widget.EditText; import android.widget.TextView; public class MainActivity extends AppCompatActivity { private EditText mInput; @Override protected void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState);
        setContentView(R.layout.activity_main);
        mInput = (EditText) findViewById(R.id.et_input);
     mInput.setText("http://qzone.60dj.com/huiyuan/201704/19/201704190533197825_35285.mp3");
        findViewById(R.id.bt_play).setOnClickListener(new View.OnClickListener() { @Override public void onClick(View view) {
                NativePlayer.play(mInput.getText().toString().trim());
            }
        });
        findViewById(R.id.bt_pause).setOnClickListener(new View.OnClickListener() { @Override public void onClick(View view) {
                NativePlayer.stop();
            }
        });
    }
}

NativePlayer:

package com.hejunlin.ffmpegaudio; /**
 * Created by hejunlin on 17/5/6.
 */ public class NativePlayer { static {
        System.loadLibrary("NativePlayer");
    } public static native void play(String url); public static native void stop();
}

布局文件：

 <RelativeLayout  xmlns:android="http://schemas.android.com/apk/res/android" xmlns:tools="http://schemas.android.com/tools" android:id="@+id/activity_main" android:layout_width="match_parent" android:layout_height="match_parent" tools:context="com.hejunlin.ffmpegaudio.MainActivity"> <TextView  android:id="@+id/tv_input" android:layout_width="wrap_content" android:layout_height="wrap_content" android:padding="10dp" android:layout_marginTop="30dp" android:text="播放链接:" android:textSize="20sp"/> <EditText  android:id="@+id/et_input" android:layout_width="match_parent" android:layout_height="wrap_content" android:layout_toRightOf="@id/tv_input" android:padding="10dp"/> <LinearLayout  android:layout_width="match_parent" android:layout_height="wrap_content" android:layout_below="@id/et_input" android:orientation="horizontal"> <Button  android:id="@+id/bt_play" android:layout_width="wrap_content" android:layout_height="wrap_content" android:layout_marginTop="10dp" android:layout_marginLeft="60dp" android:background="@drawable/button_shape" android:textColor="@color/white" android:text="播放" /> <Button  android:id="@+id/bt_pause" android:layout_width="wrap_content" android:layout_height="wrap_content" android:layout_marginTop="10dp" android:background="@drawable/button_shape" android:textColor="@color/white" android:layout_marginLeft="80dp" android:text="暂停" /> LinearLayout> RelativeLayout> 

jni相关代码： OpenSL_ES_Core.c

// // Created by hejunlin on 17/5/6. // #include "OpenSL_ES_Core.h" #include "FFmpegCore.h" #include  #include  #include  #include  #include  // for native asset manager #include  #include  #include  #include "log.h" // 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; static void *buffer; static size_t bufferSize; // this callback handler is called every time a buffer finishes playing void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
{
    LOGD(">> buffere queue callback");
    assert(bq == bqPlayerBufferQueue);
    bufferSize = 0; //assert(NULL == context); getPCM(&buffer, &bufferSize); // for streaming playback, replace this test by logic to find and fill the next buffer if (NULL != buffer && 0 != bufferSize) {
        SLresult result; // enqueue another buffer result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, buffer,
                                                 bufferSize); // 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)result;
    }
} void initOpenSLES()
{
    LOGD(">> initOpenSLES...");
    SLresult result; // 1、create engine result = slCreateEngine(&engineObject, 0, NULL, 0, NULL, NULL);
    LOGD(">> initOpenSLES... step 1, result = %d", result); // 2、realize the engine result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE);
    LOGD(">> initOpenSLES...step 2, result = %d", result); // 3、get the engine interface, which is needed in order to create other objects result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine);
    LOGD(">> initOpenSLES...step 3, result = %d", result); // 4、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, 0, 0, 0);
    LOGD(">> initOpenSLES...step 4, result = %d", result); // 5、realize the output mix result = (*outputMixObject)->Realize(outputMixObject, SL_BOOLEAN_FALSE);
    LOGD(">> initOpenSLES...step 5, result = %d", result); // 6、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);
        LOGD(">> initOpenSLES...step 6, result = %d", result);
    }

} // init buffer queue void initBufferQueue(int rate, int channel, int bitsPerSample)
{
    LOGD(">> initBufferQueue");
    SLresult result; // configure audio source SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, 2};
    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);
    (void)result; // realize the player result = (*bqPlayerObject)->Realize(bqPlayerObject, SL_BOOLEAN_FALSE);
    assert(SL_RESULT_SUCCESS == result);
    (void)result; // get the play interface result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_PLAY, &bqPlayerPlay);
    assert(SL_RESULT_SUCCESS == result);
    (void)result; // get the buffer queue interface result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_BUFFERQUEUE,
                                             &bqPlayerBufferQueue);
    assert(SL_RESULT_SUCCESS == result);
    (void)result; // register callback on the buffer queue result = (*bqPlayerBufferQueue)->RegisterCallback(bqPlayerBufferQueue, bqPlayerCallback, NULL);
    assert(SL_RESULT_SUCCESS == result);
    (void)result; // get the effect send interface result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_EFFECTSEND,
                                             &bqPlayerEffectSend);
    assert(SL_RESULT_SUCCESS == result);
    (void)result; // get the volume interface result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_VOLUME, &bqPlayerVolume);
    assert(SL_RESULT_SUCCESS == result);
    (void)result; // set the player's state to playing result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING);
    assert(SL_RESULT_SUCCESS == result);
    (void)result;
} // stop the native audio system void stop()
{ // destroy buffer queue audio player object, and invalidate all associated interfaces if (bqPlayerObject != NULL) {
        (*bqPlayerObject)->Destroy(bqPlayerObject);
        bqPlayerObject = NULL;
        bqPlayerPlay = NULL;
        bqPlayerBufferQueue = NULL;
        bqPlayerEffectSend = NULL;
        bqPlayerMuteSolo = NULL;
        bqPlayerVolume = NULL;
    } // destroy output mix object, and invalidate all associated interfaces if (outputMixObject != NULL) {
        (*outputMixObject)->Destroy(outputMixObject);
        outputMixObject = NULL;
        outputMixEnvironmentalReverb = NULL;
    } // destroy engine object, and invalidate all associated interfaces if (engineObject != NULL) {
        (*engineObject)->Destroy(engineObject);
        engineObject = NULL;
        engineEngine = NULL;
    } // 释放FFmpeg解码器 releaseFFmpeg();
} void play(char *url)
{ int rate, channel;
    LOGD("...get url=%s", url); // 1、初始化FFmpeg解码器 initFFmpeg(&rate, &channel, url); // 2、初始化OpenSLES initOpenSLES(); // 3、初始化BufferQueue initBufferQueue(rate, channel, SL_PCMSAMPLEFORMAT_FIXED_16); // 4、启动音频播放 bqPlayerCallback(bqPlayerBufferQueue, NULL);
}

FFmpegCore.c

#include "log.h" #include "FFmpegCore.h" #include "libavcodec/avcodec.h" #include "libavformat/avformat.h" #include "libswscale/swscale.h" #include "libswresample/swresample.h" #include "libavutil/samplefmt.h" #include  #include  uint8_t *outputBuffer;
size_t outputBufferSize;

AVPacket packet; int audioStream;
AVFrame *aFrame;
SwrContext *swr;
AVFormatContext *aFormatCtx;
AVCodecContext *aCodecCtx; int initFFmpeg(int *rate, int *channel, char *url) {

    av_register_all();
    aFormatCtx = avformat_alloc_context();
    LOGD("ffmpeg get url=:%s", url); // 网络音频流 char *file_name = url; // Open audio file if (avformat_open_input(&aFormatCtx, file_name, NULL, NULL) != 0) {
        LOGE("Couldn't open file:%s\n", file_name); return -1; // Couldn't open file } // Retrieve stream information if (avformat_find_stream_info(aFormatCtx, NULL) < 0) {
        LOGE("Couldn't find stream information."); return -1;
    } // Find the first audio stream int i;
    audioStream = -1; for (i = 0; i < aFormatCtx->nb_streams; i++) { if (aFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
            audioStream < 0) {
            audioStream = i;
        }
    } if (audioStream == -1) {
        LOGE("Couldn't find audio stream!"); return -1;
    } // Get a pointer to the codec context for the video stream aCodecCtx = aFormatCtx->streams[audioStream]->codec; // Find the decoder for the audio stream AVCodec *aCodec = avcodec_find_decoder(aCodecCtx->codec_id); if (!aCodec) {
        fprintf(stderr, "Unsupported codec!\n"); return -1;
    } if (avcodec_open2(aCodecCtx, aCodec, NULL) < 0) {
        LOGE("Could not open codec."); return -1; // Could not open codec }

    aFrame = av_frame_alloc(); // 设置格式转换 swr = swr_alloc();
    av_opt_set_int(swr, "in_channel_layout",  aCodecCtx->channel_layout, 0);
    av_opt_set_int(swr, "out_channel_layout", aCodecCtx->channel_layout, 0);
    av_opt_set_int(swr, "in_sample_rate",     aCodecCtx->sample_rate, 0);
    av_opt_set_int(swr, "out_sample_rate",    aCodecCtx->sample_rate, 0);
    av_opt_set_sample_fmt(swr, "in_sample_fmt",  aCodecCtx->sample_fmt, 0);
    av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
    swr_init(swr); // 分配PCM数据缓存 outputBufferSize = 8196;
    outputBuffer = (uint8_t *) malloc(sizeof(uint8_t) * outputBufferSize); // 返回sample rate和channels *rate = aCodecCtx->sample_rate;
    *channel = aCodecCtx->channels; return 0;
} // 获取PCM数据, 自动回调获取 int getPCM(void **pcm, size_t *pcmSize) {
    LOGD(">> getPcm"); while (av_read_frame(aFormatCtx, &packet) >= 0) { int frameFinished = 0; // Is this a packet from the audio stream? if (packet.stream_index == audioStream) {
            avcodec_decode_audio4(aCodecCtx, aFrame, &frameFinished, &packet); if (frameFinished) { // data_size为音频数据所占的字节数 int data_size = av_samples_get_buffer_size(
                        aFrame->linesize, aCodecCtx->channels,
                        aFrame->nb_samples, aCodecCtx->sample_fmt, 1);
                LOGD(">> getPcm data_size=%d", data_size); // 这里内存再分配可能存在问题 if (data_size > outputBufferSize) {
                    outputBufferSize = data_size;
                    outputBuffer = (uint8_t *) realloc(outputBuffer, sizeof(uint8_t) * outputBufferSize);
                } // 音频格式转换 swr_convert(swr, &outputBuffer, aFrame->nb_samples,
                            (uint8_t const **) (aFrame->extended_data),
                        aFrame->nb_samples); // 返回pcm数据 *pcm = outputBuffer;
                *pcmSize = data_size; return 0;
            }
        }
    } return -1;
} // 释放相关资源 int releaseFFmpeg()
{
    av_packet_unref(&packet);
    av_free(outputBuffer);
    av_free(aFrame);
    avcodec_close(aCodecCtx);
    avformat_close_input(&aFormatCtx); return 0;

NativePlayer.c

// // Created by hejunlin on 17/5/6. // #include "log.h" #include "com_hejunlin_ffmpegaudio_NativePlayer.h" #include "OpenSL_ES_Core.h" JNIEXPORT void JNICALL
Java_com_hejunlin_ffmpegaudio_NativePlayer_play(JNIEnv *env, jclass type, jstring url_) {
    const char *url = (*env)->GetStringUTFChars(env, url_, 0);
    LOGD("start playaudio... url=%s", url);

    play(url);
    (*env)->ReleaseStringUTFChars(env, url_, url);
}

JNIEXPORT void JNICALL
Java_com_hejunlin_ffmpegaudio_NativePlayer_stop(JNIEnv *env, jclass type) {

    LOGD("stop");
    stop();
}

通过cmake，或ndk-build都可以编译，会生成一个NativePlayer.so

效果图：

log输出如下：

05-07 10:14:04.476 6001-6097/com.hejunlin.ffmpegaudio D/Surface: Surface::setBuffersDimensions(this=0xf45b6300,w=1080,h=1920) 05-07 10:14:04.491 6001-6097/com.hejunlin.ffmpegaudio D/Surface: Surface::setBuffersDimensions(this=0xf45b6300,w=1080,h=1920) 05-07 10:14:04.516 6001-6097/com.hejunlin.ffmpegaudio D/Surface: Surface::setBuffersDimensions(this=0xf45b6300,w=1080,h=1920) 05-07 10:14:04.525 6001-6097/com.hejunlin.ffmpegaudio D/Surface: Surface::setBuffersDimensions(this=0xf45b6300,w=1080,h=1920) 05-07 10:14:04.542 6001-6097/com.hejunlin.ffmpegaudio D/Surface: Surface::setBuffersDimensions(this=0xf45b6300,w=1080,h=1920) 05-07 10:14:04.556 6001-6097/com.hejunlin.ffmpegaudio D/Surface: Surface::setBuffersDimensions(this=0xf45b6300,w=1080,h=1920) 05-07 10:14:04.573 6001-6097/com.hejunlin.ffmpegaudio D/Surface: Surface::setBuffersDimensions(this=0xf45b6300,w=1080,h=1920) 05-07 10:14:04.577 6001-6001/com.hejunlin.ffmpegaudio W/linker: libNativePlayer.so: unused DT entry: type 0x6ffffffe arg 0x1414 05-07 10:14:04.577 6001-6001/com.hejunlin.ffmpegaudio W/linker: libNativePlayer.so: unused DT entry: type 0x6fffffff arg 0x4 05-07 10:14:04.578 6001-6001/com.hejunlin.ffmpegaudio W/linker: libavcodec-57.so: unused DT entry: type 0x6ffffffe arg 0x5da4 05-07 10:14:04.578 6001-6001/com.hejunlin.ffmpegaudio W/linker: libavcodec-57.so: unused DT entry: type 0x6fffffff arg 0x2 05-07 10:14:04.578 6001-6001/com.hejunlin.ffmpegaudio W/linker: libavformat-57.so: unused DT entry: type 0x6ffffffe arg 0x6408 05-07 10:14:04.578 6001-6001/com.hejunlin.ffmpegaudio W/linker: libavformat-57.so: unused DT entry: type 0x6fffffff arg 0x2 05-07 10:14:04.578 6001-6001/com.hejunlin.ffmpegaudio W/linker: libswresample-2.so: unused DT entry: type 0x6ffffffe arg 0xcd4 05-07 10:14:04.578 6001-6001/com.hejunlin.ffmpegaudio W/linker: libswresample-2.so: unused DT entry: type 0x6fffffff arg 0x1 05-07 10:14:04.578 6001-6001/com.hejunlin.ffmpegaudio W/linker: libswscale-4.so: unused DT entry: type 0x6ffffffe arg 0xd70 05-07 10:14:04.578 6001-6001/com.hejunlin.ffmpegaudio W/linker: libswscale-4.so: unused DT entry: type 0x6fffffff arg 0x1 05-07 10:14:04.589 6001-6001/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/NativePlayer.c: Java_com_hejunlin_ffmpegaudio_NativePlayer_play:start playaudio... url=http://qzone.60dj.com/huiyuan/201704/19/201704190533197825_35285.mp3 05-07 10:14:04.589 6001-6001/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/OpenSL_ES_Core.c: play:...get url=http://qzone.60dj.com/huiyuan/201704/19/201704190533197825_35285.mp3 05-07 10:14:04.590 6001-6001/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/FFmpegCore.c: initFFmpeg:ffmpeg get url=:http://qzone.60dj.com/huiyuan/201704/19/201704190533197825_35285.mp3 05-07 10:14:04.696 6001-6001/com.hejunlin.ffmpegaudio D/libc-netbsd: getaddrinfo: qzone.60dj.com get result from proxy >> 05-07 10:14:04.949 6001-6001/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/OpenSL_ES_Core.c: initOpenSLES:>> initOpenSLES... 05-07 10:14:04.950 6001-6001/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/OpenSL_ES_Core.c: initOpenSLES:>> initOpenSLES... step 1, result = 0 05-07 10:14:04.950 6001-6001/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/OpenSL_ES_Core.c: initOpenSLES:>> initOpenSLES...step 2, result = 0 05-07 10:14:04.950 6001-6001/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/OpenSL_ES_Core.c: initOpenSLES:>> initOpenSLES...step 3, result = 0 05-07 10:14:04.950 6001-6001/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/OpenSL_ES_Core.c: initOpenSLES:>> initOpenSLES...step 4, result = 0 05-07 10:14:04.950 6001-6001/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/OpenSL_ES_Core.c: initOpenSLES:>> initOpenSLES...step 5, result = 0 05-07 10:14:04.950 6001-6001/com.hejunlin.ffmpegaudio W/libOpenSLES: Leaving Object::GetInterface (SL_RESULT_FEATURE_UNSUPPORTED) 05-07 10:14:04.950 6001-6001/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/OpenSL_ES_Core.c: initBufferQueue:>> initBufferQueue 05-07 10:14:04.951 6001-6001/com.hejunlin.ffmpegaudio V/AudioTrack: set(): streamType 3, sampleRate 44100, format 0x1, channelMask 0x3, frameCount 0, flags #0, notificationFrames 0, sessionId 774, transferType 0 05-07 10:14:04.951 6001-6001/com.hejunlin.ffmpegaudio V/AudioTrack: set() streamType 3 frameCount 0 flags 0000 05-07 10:14:04.951 6001-6001/com.hejunlin.ffmpegaudio D/AudioTrack: audiotrack 0xf459cd80 set Type 3, rate 44100, fmt 1, chn 3, fcnt 0, flags 0000 05-07 10:14:04.951 6001-6001/com.hejunlin.ffmpegaudio D/AudioTrack: mChannelMask 0x3 05-07 10:14:04.953 6001-6001/com.hejunlin.ffmpegaudio V/AudioTrack: createTrack_l() output 2 afLatency 21 05-07 10:14:04.953 6001-6001/com.hejunlin.ffmpegaudio V/AudioTrack: afFrameCount=1024, minBufCount=1, afSampleRate=48000, afLatency=21 05-07 10:14:04.953 6001-6001/com.hejunlin.ffmpegaudio V/AudioTrack: minFrameCount: 2822, afFrameCount=1024, minBufCount=3, sampleRate=44100, afSampleRate=48000, afLatency=21 05-07 10:14:04.954 6001-6001/com.hejunlin.ffmpegaudio D/AudioTrackCenter: addTrack, trackId:0xdaf0c000, frameCount:2822, sampleRate:44100, trackPtr:0xf459cd80 05-07 10:14:04.954 6001-6001/com.hejunlin.ffmpegaudio D/AudioTrackCenter: mAfSampleRate 48000, sampleRate 44100, AfFrameCount 1024 , mAfSampleRate 48000, frameCount 2822 05-07 10:14:04.979 6001-6001/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/OpenSL_ES_Core.c: bqPlayerCallback:>> buffere queue callback 05-07 10:14:04.979 6001-6001/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/FFmpegCore.c: getPCM:>> getPcm 05-07 10:14:04.979 6001-6681/com.hejunlin.ffmpegaudio D/AudioTrackShared: front(0x0), mIsOut 1, avail 2822, mFrameCount 2822, filled 0 05-07 10:14:04.979 6001-6681/com.hejunlin.ffmpegaudio V/AudioTrack: obtainBuffer(940) returned 2822 = 940 + 1882 err 0 05-07 10:14:04.979 6001-6681/com.hejunlin.ffmpegaudio D/AudioTrackShared: front(0x0), mIsOut 1, interrupt() FUTEX_WAKE 05-07 10:14:04.979 6001-6681/com.hejunlin.ffmpegaudio D/AudioTrack: audiotrack 0xf459cd80 stop done 05-07 10:14:04.980 6001-6001/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/FFmpegCore.c: getPCM:>> getPcm data_size=188 05-07 10:14:04.980 6001-6681/com.hejunlin.ffmpegaudio D/AudioTrackShared: front(0x0), mIsOut 1, avail 2822, mFrameCount 2822, filled 0 05-07 10:14:04.980 6001-6681/com.hejunlin.ffmpegaudio V/AudioTrack: obtainBuffer(940) returned 2822 = 940 + 1882 err 0 05-07 10:14:04.980 6001-6681/com.hejunlin.ffmpegaudio D/AudioTrackShared: front(0x0), mIsOut 1, avail 2775, mFrameCount 2822, filled 47 05-07 10:14:04.980 6001-6681/com.hejunlin.ffmpegaudio V/AudioTrack: obtainBuffer(893) returned 2775 = 893 + 1882 err 0 05-07 10:14:04.980 6001-6681/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/OpenSL_ES_Core.c: bqPlayerCallback:>> buffere queue callback 05-07 10:14:04.980 6001-6681/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/FFmpegCore.c: getPCM:>> getPcm 05-07 10:14:04.980 6001-6681/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/FFmpegCore.c: getPCM:>> getPcm data_size=4608 05-07 10:14:04.980 6001-6681/com.hejunlin.ffmpegaudio D/AudioTrackShared: front(0x0), mIsOut 1, avail 1882, mFrameCount 2822, filled 940 05-07 10:14:04.980 6001-6681/com.hejunlin.ffmpegaudio V/AudioTrack: obtainBuffer(940) returned 1882 = 940 + 942 err 0 05-07 10:14:04.980 6001-6681/com.hejunlin.ffmpegaudio D/AudioTrackShared: front(0x0), mIsOut 1, avail 1623, mFrameCount 2822, filled 1199 05-07 10:14:04.980 6001-6681/com.hejunlin.ffmpegaudio V/AudioTrack: obtainBuffer(681) returned 1623 = 681 + 942 err 0 05-07 10:14:04.980 6001-6681/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/OpenSL_ES_Core.c: bqPlayerCallback:>> buffere queue callback 05-07 10:14:04.980 6001-6681/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/FFmpegCore.c: getPCM:>> getPcm 05-07 10:14:04.981 6001-6681/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/FFmpegCore.c: getPCM:>> getPcm data_size=4608 05-07 10:14:04.981 6001-6681/com.hejunlin.ffmpegaudio D/AudioTrackShared: front(0x0), mIsOut 1, avail 942, mFrameCount 2822, filled 1880 05-07 10:14:04.981 6001-6681/com.hejunlin.ffmpegaudio V/AudioTrack: obtainBuffer(940) returned 942 = 940 + 2 err 0 05-07 10:14:04.981 6001-6681/com.hejunlin.ffmpegaudio D/AudioTrackShared: front(0x0), mIsOut 1, avail 471, mFrameCount 2822, filled 2351 05-07 10:14:04.981 6001-6681/com.hejunlin.ffmpegaudio V/AudioTrack: obtainBuffer(469) returned 471 = 469 + 2 err 0 05-07 10:14:04.981 6001-6681/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/OpenSL_ES_Core.c: bqPlayerCallback:>> buffere queue callback 05-07 10:14:04.981 6001-6681/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/FFmpegCore.c: getPCM:>> getPcm 05-07 10:14:04.981 6001-6681/com.hejunlin.ffmpegaudio D//Users/hejunlin/AndroidStudioProjects/FFmpegAudio/app/src/main/jni/FFmpegCore.c: getPCM:>> getPcm data_size=4608

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