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。