Android本地视频播放器开发--ffmpeg解码视频文件中的音频(2)
在Android本地视频播放器开发--ffmpeg解码视频文件中的音频(1)中我们从视频文件中解码出音频,这一章中将使用OpenSL ES来播放解码的音频数据,首先关于OpenSL ES这里暂不介绍,可以查看官网以及NDK中samples下面的native-audio里面的文件,这里我也是扣取了其中的代码,我们播放音频的部分在上一章的基础上进行添加的,代码如下:
#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();//注册所有支持的文件格式以及编解码器 /* *只读取文件头,并不会填充流信息 */ if(avformat_open_input(&pFormatCtx, local_title, NULL, NULL) != 0) return -1; /* *获取文件中的流信息,此函数会读取packet,并确定文件中所有流信息, *设置pFormatCtx->streams指向文件中的流,但此函数并不会改变文件指针, *读取的packet会给后面的解码进行处理。 */ if(avformat_find_stream_info(pFormatCtx, NULL) < 0) return -1; /* *输出文件的信息,也就是我们在使用ffmpeg时能够看到的文件详细信息, *第二个参数指定输出哪条流的信息,-1代表ffmpeg自己选择。最后一个参数用于 *指定dump的是不是输出文件,我们的dump是输入文件,因此一定要为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)//暂停 { 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; }
然后就是需要在Android.mk中添加OpenSL ES的库支持,代码如下:
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)
由于OpenSLES最低版本需要9所以要在Application.mk中添加平台
# 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
最后在终端运行ndk-build,就会将代码添加到
ffmpeg-test-neon.so这个库中
最后在Android端调用
VideoPlayer这个函数就会自动播放视频的声音,测试发现虽然声音正常但是有杂音,可能采样率设置的不对,获取其他的配置有问题,下一章着重解决这个问题,同时使用队列的方式来从视频中取音频包,然后从音频包队列中取出,然后解码播放。