ffmpeg 解码amr音频 到pcm文件有噪音

		char *outfilename="D:/cvsroot/ffmpeg/ffmpegDecodeAndEncode/Debug/3.pcm";



		FILE* pcm;		

		pcm = fopen(outfilename,"wb");



		av_register_all();



		AVFrame* frame = avcodec_alloc_frame();

		if (!frame)

		{

			printf( "Error allocating the frame" );

			return 1;

		}



		AVFormatContext* formatContext = NULL;

		//if (avformat_open_input(&formatContext, "D:/cvsroot/ffmpeg/ffmpegDecodeAndEncode/Debug/1.mp3", NULL, NULL) != 0)

		if (avformat_open_input(&formatContext, "D:/cvsroot/ffmpeg/ffmpegDecodeAndEncode/Debug/1.amr", NULL, NULL) != 0)

		{

			av_free(frame);

			printf( "Error opening the file" );

			return 1;

		}



		if (avformat_find_stream_info(formatContext, NULL) < 0)

		{

			av_free(frame);

			av_close_input_file(formatContext);

			printf( "Error finding the stream info" );

			return 1;

		}



		AVStream* audioStream = NULL;

		for (unsigned int i = 0; i < formatContext->nb_streams; ++i)

		{

			if (formatContext->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO)

			{

				audioStream = formatContext->streams[i];

				break;

			}

		}



		if (audioStream == NULL)

		{

			av_free(frame);

			av_close_input_file(formatContext);

			printf( "Could not find any audio stream in the file" );

			return 1;

		}



		AVCodecContext* codecContext = audioStream->codec;



		codecContext->codec = avcodec_find_decoder(codecContext->codec_id);

		if (codecContext->codec == NULL)

		{

			av_free(frame);

			av_close_input_file(formatContext);

			printf( "Couldn't find a proper decoder" );

			return 1;

		}

		else if (avcodec_open2(codecContext, codecContext->codec, NULL) != 0)

		{

			av_free(frame);

			av_close_input_file(formatContext);

			printf( "Couldn't open the context with the decoder" );

			return 1;

		}



		SwrContext* swrContext = NULL;



		swrContext = swr_alloc_set_opts(swrContext,

			codecContext->channel_layout, // out channel layout

			codecContext->sample_fmt, // out sample format

			codecContext->sample_rate, // out sample rate

			codecContext->channel_layout, // in channel layout

			codecContext->sample_fmt, // in sample format

			codecContext->sample_rate, // in sample rate

			0, // log offset

			NULL); // log context

		swr_init(swrContext);









		if(codecContext->sample_fmt==AV_SAMPLE_FMT_S16)

		{

			swrContext = swr_alloc_set_opts(swrContext,

				codecContext->channel_layout,

				AV_SAMPLE_FMT_S16P,

				codecContext->sample_rate,

				codecContext->channel_layout,

				AV_SAMPLE_FMT_S16,

				codecContext->sample_rate,

				0,

				0);

			if(swr_init(swrContext)<0)

			{

				printf("swr_init() for AV_SAMPLE_FMT_S16P fail");

				return 0;

			}

		}

		else if(codecContext->sample_fmt==AV_SAMPLE_FMT_FLT)

		{

			swrContext = swr_alloc_set_opts(swrContext,

				codecContext->channel_layout,

				AV_SAMPLE_FMT_S16P,

				codecContext->sample_rate,

				codecContext->channel_layout,

				AV_SAMPLE_FMT_FLT,

				codecContext->sample_rate,

				0,

				0);

			if(swr_init(swrContext)<0)

			{

				printf("swr_init() for AV_SAMPLE_FMT_S16P fail");

				return 0;

			}

		}		



		

		if (!swrContext)

		{

			av_free(frame);

			avcodec_close(codecContext);

			av_close_input_file(formatContext);

			printf( "Couldn't allocate and set the resampling context" );

			return 1;

		}



		printf(" bit_rate = %d \r\n", codecContext->bit_rate);

		printf(" sample_rate = %d \r\n", codecContext->sample_rate);

		printf(" channels = %d \r\n", codecContext->channels);

		printf(" code_name = %s \r\n", codecContext->codec->name);



		int bufSize = av_samples_get_buffer_size(NULL, codecContext->channels, codecContext->sample_rate, codecContext->sample_fmt, 0);

		uint8_t* buf = new uint8_t[bufSize];



		AVPacket packet;

		av_init_packet(&packet);



		int packetCount = 0;

		int decodedFrameCount = 0;		



		static uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2]; 

		uint8_t *out[]={audio_buf};  

uint8_t *pktdata;

        int pktsize,flush_complete=0;

		while (av_read_frame(formatContext, &packet) == 0)

		{

			++packetCount;

			if (packet.stream_index == audioStream->index)

			{

                             pktdata = packet.data;

                             pktsize = packet.size;

				int frameFinished = 0;

				avcodec_decode_audio4(codecContext, frame, &frameFinished, &packet);



				if (frameFinished)

				{							

                                        pktsize -= len;

					pktdata += len;

					int data_size = av_samples_get_buffer_size(NULL,codecContext->channels,frame->nb_samples,codecContext->sample_fmt, 1);

					

					/*****************************************************

					以下代码使用swr_convert函数进行转换，但是转换后的文件连mp3到pcm文件都不能播放了，所以注释了

					const uint8_t *in[] = {frame->data[0]};



					int len=swr_convert(swrContext,out,sizeof(audio_buf)/codecContext->channels/av_get_bytes_per_sample(AV_SAMPLE_FMT_S16P),  

						in,frame->linesize[0]/codecContext->channels/av_get_bytes_per_sample(codecContext->sample_fmt));					



					len=len*codecContext->channels*av_get_bytes_per_sample(AV_SAMPLE_FMT_S16P);  



					if (len < 0) {

						fprintf(stderr, "audio_resample() failed\n");

						break;

					}

					if (len == sizeof(audio_buf) / codecContext->channels / av_get_bytes_per_sample(AV_SAMPLE_FMT_S16P)) {

						fprintf(stderr, "warning: audio buffer is probably too small\n");

						swr_init(swrContext);

					}

					*****************************************************/

					

					char *data = (char *)malloc(data_size);

					short *sample_buffer = (short *)frame->data[0];

					for (int i = 0; i < data_size/2; i++)

					{

						data[i*2] = (char)(sample_buffer[i/2] & 0xFF);

						data[i*2+1] = (char)((sample_buffer[i/2] >>8) & 0xFF);		



					}

					fwrite(data, data_size, 1, pcm);

					fflush(pcm);



					

				}

			}

			

			av_free_packet(&packet);

		}



		delete [] buf;

		swr_free(&swrContext);

		av_free(frame);

		avcodec_close(codecContext);

		av_close_input_file(formatContext);

		fclose(pcm);