android使用ffmpeg

cygwin上文编译文章.


在ffmpeg/arm添加的文件夹Android.mk 的主要目的是为了宣布动态库libs下一个

LOCAL_PATH:= $(call my-dir)
 
include $(CLEAR_VARS)
LOCAL_MODULE:= libavcodec
LOCAL_SRC_FILES:= lib/libavcodec-55.so
LOCAL_EXPORT_C_INCLUDES := $(LOCAL_PATH)/include
include $(PREBUILT_SHARED_LIBRARY)
 
include $(CLEAR_VARS)
LOCAL_MODULE:= libavformat
LOCAL_SRC_FILES:= lib/libavformat-55.so
LOCAL_EXPORT_C_INCLUDES := $(LOCAL_PATH)/include
include $(PREBUILT_SHARED_LIBRARY)
 
include $(CLEAR_VARS)
LOCAL_MODULE:= libswscale
LOCAL_SRC_FILES:= lib/libswscale-2.so
LOCAL_EXPORT_C_INCLUDES := $(LOCAL_PATH)/include
include $(PREBUILT_SHARED_LIBRARY)
 
include $(CLEAR_VARS)
LOCAL_MODULE:= libavutil
LOCAL_SRC_FILES:= lib/libavutil-52.so
LOCAL_EXPORT_C_INCLUDES := $(LOCAL_PATH)/include
include $(PREBUILT_SHARED_LIBRARY)
 
include $(CLEAR_VARS)
LOCAL_MODULE:= libavfilter
LOCAL_SRC_FILES:= lib/libavfilter-4.so
LOCAL_EXPORT_C_INCLUDES := $(LOCAL_PATH)/include
include $(PREBUILT_SHARED_LIBRARY)
 
include $(CLEAR_VARS)
LOCAL_MODULE:= libwsresample
LOCAL_SRC_FILES:= lib/libswresample-0.so
LOCAL_EXPORT_C_INCLUDES := $(LOCAL_PATH)/include
include $(PREBUILT_SHARED_LIBRARY)


添加Media.h

#pragma once
#include <jni.h>
#include <android/native_window_jni.h>
#include "utils/Lock.h"
#include <pthread.h>

//ffmpeg 须要先定义 __STDC_CONSTANT_MACROS 才干通过 c++ 编译
#define __STDC_CONSTANT_MACROS
#ifndef INT64_C
#define INT64_C(c) (c ## LL)
#define UINT64_C(c) (c ## ULL)
#endif
extern "C" {
	#include <libavcodec/avcodec.h>
	#include <libavformat/avformat.h>
	#include <libavutil/avutil.h>
	#include <libavutil/dict.h>
	#include <libavutil/frame.h>
	#include <libavutil/mem.h>
	#include <libavutil/pixfmt.h>
	#include <libswscale/swscale.h>
	#include <libavutil/time.h>
	#include <libavutil/opt.h>
	#include <libswresample/swresample.h>
}

class Media
{
	public:

		Media();
		~Media();
		void setSurface(JNIEnv *pEnv, jobject pSurface,int pWidth,int pHeight);
		bool initPath(const char * path);
		bool initCodec(int width,int height);
		int getResWidth();
		int getResHeight();
		void play();
		void pause();
		void stop();
		bool isPlaying();
		void decodeAndRenderPic(void *pBuffer,int dwBufsize);
		void decodeAudioAndPlay(void *pBuffer,int dwBufsize);
	private:
		static void* decodeAndRenderAdpt(void *params);
		void decodeAndRender();
	private:
		bool bInit;

		ANativeWindow* 		window;
		char 				*videoFileName;
		AVFormatContext 	*formatCtx;
		int 				videoStream;
		int               audioStream;
		AVCodecContext  	*codecCtx;
		AVCodecContext  	*codecCtxAudio;
		AVFrame         	*decodedFrame;
		AVFrame         	*frameRGBA ;
		jobject				bitmap;
		void*				buffer;
		struct SwsContext   *sws_ctx;
		struct SwrContext   *swr_ctx;
		int 				width;
		int 				height;
		bool               _stop;

		pthread_t decodeThread;

		Mutex mutexSurface;
		Mutex lockWindow;
};


核心代码例如以下:


添加Media.cpp


		if(packet.stream_index==videoStream) {
			// Decode video frame
			avcodec_decode_video2(codecCtx, decodedFrame, &frameFinished,
			   &packet);
			// Did we get a video frame?

			if(frameFinished) {
				// Convert the image from its native format to RGBA
				sws_scale
				(
					sws_ctx,
					(uint8_t const * const *)decodedFrame->data,
					decodedFrame->linesize,
					0,
					codecCtx->height,
					frameRGBA->data,
					frameRGBA->linesize
				);

				if(packet.dts == AV_NOPTS_VALUE
						 && decodedFrame->opaque && *(uint64_t*)decodedFrame->opaque != AV_NOPTS_VALUE)
				{
					pts = *(uint64_t *)decodedFrame->opaque;
					LOGD("pst1: %d",pts);
				}
				else if(packet.dts != AV_NOPTS_VALUE) {
				  pts = packet.dts;
				  LOGD("pst2: %d",pts);
				} else {
				  pts = 0;
				  LOGD("pst3: %d",pts);
				}
				//pts = av_q2d(codecCtx->time_base) * 1000000.0 * i * 2;
				pts *= 1000;
				//LOGD("debug %d,%d,%f",pts, (long)(av_q2d(codecCtx->time_base) * 1000000.0 * i * 2), av_q2d(codecCtx->time_base));
				if(0 == pts || 0 == baseTime)
				{
					baseTime = av_gettime() - pts;
					LOGD("BASETIME: %d",baseTime);
				}else{
					waitTime = (baseTime + pts) - av_gettime();
					LOGD("WAITTIME: %d, %d",waitTime,pts);
				}

				//waitTime = (av_q2d(codecCtx->time_base) * 1000.0 - 0.0) * 1000;
				if(waitTime>0)
					usleep(waitTime);
				if(!_stop)
				{
					synchronized(lockWindow)
					{
						if(!_stop && NULL!=window)
						{
							// lock the window buffer
							if (ANativeWindow_lock(pWin, &windowBuffer, NULL) < 0) {
								LOGE("cannot lock window");
							} else {
								// draw the frame on buffer
								//LOGD("copy buffer %d:%d:%d", width, height, width*height*RGB_SIZE);
								//LOGD("window buffer: %d:%d:%d", windowBuffer.width, windowBuffer.height, windowBuffer.stride);
								//memcpy(windowBuffer.bits, buffer,  width * height * RGB_SIZE);
								if(windowBuffer.width >= windowBuffer.stride){
									//LOGE("1=========windowBuffer: %d,%d,%d,%d", windowBuffer.format,windowBuffer.stride,windowBuffer.width,windowBuffer.height);
									memcpy(windowBuffer.bits, buffer,  width * height * RGB_SIZE);
								}else{
									//LOGE("2=========windowBuffer: %d,%d,%d,%d", windowBuffer.format,windowBuffer.stride,windowBuffer.width,windowBuffer.height);
									//skip stride-width 跳过padding部分内存
									for(int i=0;i<height;++i)
										memcpy(windowBuffer.bits +  windowBuffer.stride * i * RGB_SIZE
												, buffer + width * i * RGB_SIZE
												, width * RGB_SIZE);
								}
								// unlock the window buffer and post it to display
								ANativeWindow_unlockAndPost(pWin);
								// count number of frames
								++i;
							}
						}
					}
				}
			}
		}else if(packet.stream_index==audioStream) {
			int ret = avcodec_decode_audio4(codecCtxAudio,decodedFrame, &frameFinished,
			   &packet);
			// LOGD("avcodec_decode_audio4, %d , ret %d" , frameFinished, ret);
			if(frameFinished)
			{
				// LOGD("read audio play");
	            size_t unpadded_linesize = decodedFrame->nb_samples * av_get_bytes_per_sample((AVSampleFormat)decodedFrame->format);
	            /* Write the raw audio data samples of the first plane. This works
	             * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
	             * most audio decoders output planar audio, which uses a separate
	             * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
	             * In other words, this code will write only the first audio channel
	             * in these cases.
	             * You should use libswresample or libavfilter to convert the frame
	             * to packed data. */
	            if(NULL!=swr_ctx)
	            {
	            	int dst_linesize = 0;
					int dst_nb_samples =av_rescale_rnd(decodedFrame->nb_samples, decodedFrame->sample_rate, codecCtxAudio->sample_rate, AV_ROUND_UP);
					int dst_nb_channels = av_get_channel_layout_nb_channels(codecCtxAudio->channels ==1 ?AV_CH_LAYOUT_MONO:AV_CH_LAYOUT_STEREO);
					av_samples_alloc_array_and_samples(&dst_data,&dst_linesize,dst_nb_channels,dst_nb_samples,codecCtxAudio->sample_fmt == AV_SAMPLE_FMT_U8?
AV_SAMPLE_FMT_U8:AV_SAMPLE_FMT_S16, 0);
	            	int ret = audio_swr_resampling_audio(swr_ctx,decodedFrame,dst_data);
	            	if(ret>0){
	            		writeAudio(dst_data[0],ret);
	            		//fwrite(dst_data[0], 1, ret, stream);
	            	}
	            	if (dst_data)
	            	{
	            		av_freep(&dst_data[0]);
	            	}
	            	av_freep(&dst_data);
	            }else{
					writeAudio(decodedFrame->extended_data[0], unpadded_linesize);
					//fwrite(decodedFrame->extended_data[0], 1, unpadded_linesize, stream);
	            }
	            //fwrite(decodedFrame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
	            LOGD("read audio buffer: %d ,%d", unpadded_linesize, decodedFrame->linesize[0]);
			}else{
				//LOGD("===read audio buffer: %d", packet.size);
				//writeAudio(packet.data, packet.size);
			}
		}else{
			 LOGD("unkown stream index: %d", packet.stream_index);
		}
		// Free the packet that was allocated by av_read_frame
		av_free_packet(&packet);
	}
	//fclose(stream);
	LOGI("total No. of frames decoded and rendered %d", i);
}

void Media::decodeAndRenderPic(void *pBuffer,int dwBufsize) {
	ANativeWindow_Buffer 	windowBuffer;
	AVPacket        		packet;
	int            			frameFinished;
	int 					lineCnt;
	ANativeWindow * pWin;
	pWin=window;
	ARect rect;

	rect.left=0;
	rect.top=0;
	rect.right = width;
	rect.bottom = height;

	memset(&packet,0x00,sizeof(AVPacket));
	packet.data = (uint8_t*)pBuffer;//这里填入一个指向完整H264数据帧的指针
	packet.size = dwBufsize;//这个填入H264数据帧的大小
	// Decode video frame
	avcodec_decode_video2(codecCtx, decodedFrame, &frameFinished,
	   &packet);
	// Did we get a video frame?

	//LOGD("111111111111111111111111");
	if(frameFinished && NULL!=window && bInit) {
		// Convert the image from its native format to RGBA
		sws_scale
		(
			sws_ctx,
			(uint8_t const * const *)decodedFrame->data,
			decodedFrame->linesize,
			0,
			codecCtx->height,
			frameRGBA->data,
			frameRGBA->linesize
		);
		//LOGD("22222222222222222222222222222");
		synchronized(lockWindow)
		{
			if(NULL!=window)
			{
				// lock the window buffer
				if (ANativeWindow_lock(pWin, &windowBuffer, &rect) < 0) {
					LOGE("cannot lock window");
				} else {
					//LOGD("333333333333333333333333333");
					// draw the frame on buffer
					LOGD("copy buffer %d:%d:%d   lineSize:%d", width, height, width*height*RGB_SIZE, frameRGBA->linesize[0]);
					LOGD("RECT : %d,%d,%d,%d",rect.left,rect.top,rect.right,rect.bottom);
					//LOGD("window buffer: %d:%d:%d", windowBuffer.width,windowBuffer.height, windowBuffer.stride);
					if(windowBuffer.width >= windowBuffer.stride){
						//LOGE("1=========windowBuffer: %d,%d,%d,%d", windowBuffer.format,windowBuffer.stride,windowBuffer.width,windowBuffer.height);
						memcpy(windowBuffer.bits, buffer,  width * height * RGB_SIZE);
					}else{
						//LOGE("2=========windowBuffer: %d,%d,%d,%d", windowBuffer.format,windowBuffer.stride,windowBuffer.width,windowBuffer.height);
						//skip stride-width 跳过padding部分内存
						for(int i=0;i<height;++i)
							memcpy(windowBuffer.bits +  windowBuffer.stride * i * RGB_SIZE
									, buffer + width * i * RGB_SIZE
									, width * RGB_SIZE);
					}
					//LOGD("666666666666666666666666666");
					// unlock the window buffer and post it to display
					ANativeWindow_unlockAndPost(pWin);
					// count number of frames
					//SaveFrame(pEnv, bitmap, codecCtx->width, codecCtx->height, i);
					//stop = 1;
				}
			}
		}
	}
	//LOGD("44444444444444444444444");
	// Free the packet that was allocated by av_read_frame
	av_free_packet(&packet);
	//LOGD("5555555555555555555555555");
}

void Media::decodeAudioAndPlay(void *pBuffer,int dwBufsize) {
	AVPacket        		packet;
	int            			frameFinished;

	LOGD("decodeAudioAndPlay start");
	if(NULL == codecCtxAudio)
	{
		LOGD("codecCtxAudio not init!");
		return;
	}

	memset(&packet,0x00,sizeof(AVPacket));
	packet.data = (uint8_t*)pBuffer;//这里填入一个指向完整H264数据帧的指针
	packet.size = dwBufsize;//这个填入H264数据帧的大小
	// Decode audio frame
	int ret = avcodec_decode_audio4(codecCtxAudio,decodedFrame, &frameFinished,
						   &packet);
	LOGD("avcodec_decode_audio4, %d , ret %d" , frameFinished, ret);
	// Did we get a audio frame?

	if(frameFinished && bInit) {
		size_t unpadded_linesize = decodedFrame->nb_samples * av_get_bytes_per_sample((AVSampleFormat)decodedFrame->format);
		writeAudio(decodedFrame->extended_data[0], unpadded_linesize);
		LOGD("writeAudio");
	}else{
		LOGD("writeAudio fail!");
	}
	// Free the packet that was allocated by av_read_frame
	av_free_packet(&packet);
	LOGD("decodeAudioAndPlay end");
}



Audio.h 音频做成了单例模式... 没有怎么封装, 使用java的AudioTrack,使用native的话不同版本号Android的so文件不一样所以不考虑了.

#pragma once

void initAudio(int mhz=44100,bool bMono=false,bool b16Bit=true);
void writeAudio(void * buffer,int size);
void releaseAudio();

Audio.cpp 

2:0x4|0x8,//0x4|0x8,//2,            /*CHANNEL_CONFIGURATION_MONO*/
				b16Bit?2:3,            /*ENCODING_PCM_16BIT*/
				buffer_size,  /*bufferSizeInBytes*/
				1             /*AudioTrack.MODE_STREAM*/
		);
		audio_track = (jobject)pEnv->NewGlobalRef(audio_track);
		//setvolume
		LOGD("setStereoVolume 1");
		jmethodID setStereoVolume =pEnv->GetMethodID(audio_track_cls,"setStereoVolume","(FF)I");
	   pEnv->CallIntMethod(audio_track,setStereoVolume,1.0,1.0);
		LOGD("setStereoVolume 2");
		//play
		jmethodID method_play =pEnv->GetMethodID(audio_track_cls, "play",
				"()V");
	   pEnv->CallVoidMethod(audio_track, method_play);

		//write
		method_write =pEnv->GetMethodID(audio_track_cls,"write","([BII)I");
		//method_write = (jmethodID)pEnv->NewGlobalRef(method_write);

		LOGI("initAudio OK, BufferSize/4:%d",buffer_size/4 );

		static pthread_t thread=NULL;
		if(NULL==thread)
			pthread_create(&thread, NULL, audioThread, NULL);
		init = true;
}

void* audioThread(void *params)
{
	LOGW("create thread : %d Audio.cpp audioThread", syscall(SYS_gettid));
	AttachCurrentThread();
	JNIEnv * env = getEnv();
	while(true)
	{
		 SE e = pop();
		 if(_stop)
			 continue;
		 int size = e->size;
		 int wirteSize = 0;
		 jbyte * buf= e->buf.get();
		 while(size > BUFFER_SIZE)
		 {
			 // LOGD("writeAudio , BufferSize/4:%d",BUFFER_SIZE );
			 env->SetByteArrayRegion(buffer, 0,BUFFER_SIZE, buf + wirteSize);
			 //LOGD("writeAudio , ==========" );
			 env->CallVoidMethod(audio_track,method_write,buffer,0,BUFFER_SIZE);
			 wirteSize += BUFFER_SIZE;
			 size -= BUFFER_SIZE;
		 }
		 if(size>0)
		 {
			 //LOGD("writeAudio , size:%d",size );
			 env->SetByteArrayRegion(buffer, 0,size, buf + wirteSize);
			 env->CallVoidMethod(audio_track,method_write,buffer,0,size);
		 }
		 //LOGD("writeAudio , OK! size:%d",e->size );
	}

	DetachCurrentThread();
	return NULL;
}

void writeAudio(void * buf,int size)
{
	sharedptr<jbyte> b(new jbyte[size]);
    memcpy(b.get(),buf,size);
    ElementBuf *eb =new ElementBuf();
    eb->buf = b;
    eb->size = size;
	SE e(eb);
	push(e);
}

void releaseAudio()
{
	_stop = true;
}


其他相关代码:

bool AttachCurrentThread()
{
	LOGI("AttachCurrentThread ing");
	JNIEnv * env;
	int status = 0;
	env = getEnv();
	if(NULL==env){
		int ret = g_jvm->AttachCurrentThread(&env, NULL);
		LOGI("AttachCurrentThread ok");
		return ret>=0;
	}
	LOGW("AttachCurrentThread fail, thread is attached");
	return false;
}
void DetachCurrentThread()
{
	LOGI("DetachCurrentThread ing");
	if(NULL!=getEnv())
		g_jvm->DetachCurrentThread();
	LOGI("DetachCurrentThread ok");
}
JNIEnv * getEnv()
{
<span style="white-space:pre">	</span>JNIEnv* env;
<span style="white-space:pre">	</span>if (g_jvm->GetEnv((void **)&env, JNI_VERSION_1_6) != JNI_OK) {
<span style="white-space:pre">		</span> return NULL;
<span style="white-space:pre">	</span>}
<span style="white-space:pre">	</span>return env;
}


项目的Android.mk 我这里包括了我使用的libspeex库,没用到的能够不用

LOCAL_PATH := $(call my-dir)

include $(CLEAR_VARS)

LOCAL_MODULE    := cu
#LOCAL_SRC_FILES := cu.cpp

FILE_LIST := $(wildcard $(LOCAL_PATH)/*.cpp) 
LOCAL_SRC_FILES += $(FILE_LIST:$(LOCAL_PATH)/%=%) 

FILE_LIST := $(wildcard $(LOCAL_PATH)/*.c) 
LOCAL_SRC_FILES += $(FILE_LIST:$(LOCAL_PATH)/%=%)

FILE_LIST := $(wildcard $(LOCAL_PATH)/*/*.cpp) 
LOCAL_SRC_FILES += $(FILE_LIST:$(LOCAL_PATH)/%=%)

LOCAL_LDLIBS := -llog -ljnigraphics -lz -landroid
LOCAL_SHARED_LIBRARIES := libavformat libavcodec libswscale libavutil libwsresample libspeex

include $(BUILD_SHARED_LIBRARY)

$(call import-add-path,$(LOCAL_PATH))
$(call import-add-path,$(LOCAL_PATH)/ffmpeg/arm/include)
$(call import-module, ffmpeg/arm)
$(call import-module, speex)
include $(all-subdir-makefiles)

Application.mk: 

APP_ABI := armeabi
#APP_ABI := armeabi-v7a
APP_PLATFORM := android-9
APP_STL := stlport_static
APP_CPPFLAGS += -fexceptions
APP_CFLAGS += -Wno-error=format-security


临时贴这么多代码出来啦! 

音频视频同步展示仅仅做了最简单的依据视屏的pts做同步,我发现pts和网上说的不太一样.
音频使用的是:java 的AudioTrack




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