[FFMPEG-2]最想实现的第一点一个功能-从mp3中提取pcm数据，重采样和加wav头

忙了几天终于实现了转码，但是只是转码还是不够的，还想要进行重采样。同样的还是要求助于万能的google，找了一大堆代码，再加上参考了官方的demo。折腾了两天，终于搞定了。

大部分的代码和转码的代码差不多，多的地方是需要设置输出的格式比如采样率，声道，bit_rate等信息。从输入文件解码得到数据之后要进行重采样的操作，然后保存重采样之后的数据。一开始找了一个讲解很详细的博客，但是他是用的老版本，很多函数都废弃了。后来又想起来官方的example，然后结合了一下终于搞出来了！因为是基于别人的代码修改的，所以代码风格和上一篇不怎么也一样。

#include    <stdlib.h>#include    <stdio.h>#include    <string.h>#include    <math.h>#include     <stdint.h>#include     <libavutil/avutil.h>#include    <libavutil/attributes.h>#include    <libavutil/opt.h>#include    <libavutil/mathematics.h>#include    <libavutil/imgutils.h>#include    <libavutil/samplefmt.h>#include    <libavutil/timestamp.h>#include    <libavformat/avformat.h>#include    <libavcodec/avcodec.h>#include    <libswscale/swscale.h>#include    <libavutil/mathematics.h>#include    <libswresample/swresample.h>#include    <libavutil/channel_layout.h>#include    <libavutil/common.h>#include    <libavformat/avio.h>#include    <libavutil/file.h>#include    <libswresample/swresample.h>FILE *fin, *fout;#define SWR_CH_MAX 1024int ffmpeg_audio_decode(const char * inFile, const char * outFile,int channel_num, int bit_rate, int sample_rate) {av_register_all();AVFrame* frame = av_frame_alloc();if (!frame) {fprintf(stderr, "Error allocating the frame");return 1;}AVFormatContext* formatContext = NULL;if (avformat_open_input(&formatContext, inFile, NULL, NULL) != 0) {av_free(frame);fprintf(stderr, "Error opening the file");return 1;}if (avformat_find_stream_info(formatContext, NULL) < 0) {av_free(frame);avformat_close_input(&formatContext);fprintf(stderr, "Error finding the stream info");return 1;}AVStream* audioStream = NULL;unsigned int i;// Find the audio stream (some container files can have multiple streams in them)for (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);avformat_close_input(&formatContext);fprintf(stderr, "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);avformat_close_input(&formatContext);fprintf(stderr, "Couldn't find a proper decoder");return 1;} else if (avcodec_open2(codecContext, codecContext->codec, NULL) != 0) {av_free(frame);avformat_close_input(&formatContext);fprintf(stderr, "Couldn't open the context with the decoder");return 1;}/*START*************************START***********************START*//*设置输出格式，为了未来jni调用的简单，所以都传的int，反正是自己用的，直观一些*/int64_t outChannelLayout;if (channel_num == 1) {outChannelLayout = AV_CH_LAYOUT_MONO;} else {outChannelLayout = AV_CH_LAYOUT_STEREO;}enum AVSampleFormat outSampleFormat;if (bit_rate == 32) {outSampleFormat = AV_SAMPLE_FMT_S32;} else {outSampleFormat = AV_SAMPLE_FMT_S16;}int outSampleRate = sample_rate; //44100;SwrContext* swrContext = swr_alloc_set_opts(NULL, outChannelLayout,outSampleFormat, outSampleRate,av_get_default_channel_layout(codecContext->channels),codecContext->sample_fmt, codecContext->sample_rate, 0, NULL);if (swrContext == NULL) {av_free(frame);avcodec_close(codecContext);avformat_close_input(&formatContext);fprintf(stderr, "Couldn't create the SwrContext");return 1;}if (swr_init(swrContext) != 0) {av_free(frame);avcodec_close(codecContext);avformat_close_input(&formatContext);swr_free(&swrContext);fprintf(stderr, "Couldn't initialize the SwrContext");return 1;}/*END**************************END*************************END*/fout = fopen(outFile, "wb+");AVPacket packet;av_init_packet(&packet);// Read the packets in a loopwhile (av_read_frame(formatContext, &packet) == 0) {if (packet.stream_index == audioStream->index) {AVPacket decodingPacket = packet;while (decodingPacket.size > 0) {// Try to decode the packet into a frameint frameFinished = 0;int result = avcodec_decode_audio4(codecContext, frame,&frameFinished, &decodingPacket);if (result < 0 || frameFinished == 0) {break;}unsigned char buffer[100000];unsigned char* pointers[SWR_CH_MAX];pointers[0] = &buffer[0];//这个地方还不是很清楚，看官方的example感觉好复杂。尤其是参数。int numSamplesOut = swr_convert(swrContext, pointers,outSampleRate,(const unsigned char**) frame->extended_data,frame->nb_samples);fwrite((short *) buffer, sizeof(short),(size_t) numSamplesOut * channel_num, fout);decodingPacket.size -= result;decodingPacket.data += result;}}av_free_packet(&packet);}// Clean up!av_free(frame);avcodec_close(codecContext);avformat_close_input(&formatContext);fclose(fout);return 0;}int main() {char *inFile = "ping.mp3";char *outFile = "ping.wav";fprintf(stderr, "res:%d\n",ffmpeg_audio_decode(inFile, outFile, 2, 16, 44100));return 0;}

代码未经充分测试，总感觉又一些问题。忘了加头了，现在把头加上。感觉自己写的程序越来越不靠谱了！但是基本呢功能还是能实现的。

#include    <stdlib.h>#include    <stdio.h>#include    <string.h>#include    <math.h>#include <stdint.h>#include  <libavutil/avutil.h>#include  <libavutil/attributes.h>#include    <libavutil/opt.h>#include    <libavutil/mathematics.h>#include    <libavutil/imgutils.h>#include    <libavutil/samplefmt.h>#include    <libavutil/timestamp.h>#include    <libavformat/avformat.h>#include    <libavcodec/avcodec.h>#include    <libswscale/swscale.h>#include    <libavutil/mathematics.h>#include    <libswresample/swresample.h>#include    <libavutil/channel_layout.h>#include    <libavutil/common.h>#include    <libavformat/avio.h>#include    <libavutil/file.h>#include    <libswresample/swresample.h>FILE *fin, *fout;#define SWR_CH_MAX 1024typedef struct {unsigned long samplerate;unsigned int bits_per_sample;unsigned int channels;unsigned long total_samples;} WavHeader;int write_wav_header(unsigned char header[], WavHeader *wavHeader) {//unsigned char header[44];unsigned char* p = header;unsigned int bytes = (wavHeader->bits_per_sample + 7) / 8;float data_size = (float) bytes * wavHeader->total_samples;unsigned long word32;*p++ = 'R';*p++ = 'I';*p++ = 'F';*p++ = 'F'; //RIFF 4word32 = (unsigned long) data_size + (44 - 8);*p++ = (unsigned char) (word32 >> 0);*p++ = (unsigned char) (word32 >> 8);*p++ = (unsigned char) (word32 >> 16);*p++ = (unsigned char) (word32 >> 24); //数据长度 4+4*p++ = 'W';*p++ = 'A';*p++ = 'V';*p++ = 'E'; //WAVE 4+4+4*p++ = 'f';*p++ = 'm';*p++ = 't';*p++ = ' '; //fmt 4+4+4+4*p++ = 0x10;*p++ = 0x00;*p++ = 0x00;*p++ = 0x00; //16  4+4+4+4+4=20*p++ = 0x01;*p++ = 0x00; //1   4+4+4+4+4+2=22*p++ = (unsigned char) (wavHeader->channels >> 0);*p++ = (unsigned char) (wavHeader->channels >> 8); //声道数   4+4+4+4+4+2+2=24word32 = (unsigned long) (wavHeader->samplerate + 0.5);*p++ = (unsigned char) (word32 >> 0);*p++ = (unsigned char) (word32 >> 8);*p++ = (unsigned char) (word32 >> 16);*p++ = (unsigned char) (word32 >> 24); //采样率   4+4+4+4+4+2+2+4=28word32 = wavHeader->samplerate * bytes * wavHeader->channels;*p++ = (unsigned char) (word32 >> 0);*p++ = (unsigned char) (word32 >> 8);*p++ = (unsigned char) (word32 >> 16);*p++ = (unsigned char) (word32 >> 24); //每秒所需字节数  4+4+4+4+4+2+2+4=32word32 = bytes * wavHeader->channels;*p++ = (unsigned char) (word32 >> 0);*p++ = (unsigned char) (word32 >> 8); // blockAlign（2个字节） 4+4+4+4+4+2+2+4+2=34//每个采样需要的字节数，计算公式：声道数 * 每个采样需要的bit  / 8*p++ = (unsigned char) (wavHeader->bits_per_sample >> 0);*p++ = (unsigned char) (wavHeader->bits_per_sample >> 8); //bitPerSample（2个字节）4+4+4+4+4+2+2+4+2+2=36//每个采样需要的bit数，一般为8或16*p++ = 'd';*p++ = 'a';*p++ = 't';*p++ = 'a'; //data 4+4+4+4+4+2+2+4+2+2+4=40word32 = (unsigned long) data_size;*p++ = (unsigned char) (word32 >> 0);*p++ = (unsigned char) (word32 >> 8);*p++ = (unsigned char) (word32 >> 16);*p++ = (unsigned char) (word32 >> 24);//size2（4个字节）   4+4+4+4+4+2+2+4+2+2+4+4=44// 录音数据的长度，不包括头部长度return 0;}int ffmpeg_audio_decode(const char * inFile, const char * outFile,int channel_num, int bit_rate, int sample_rate) {av_register_all();AVFrame* frame = av_frame_alloc();if (!frame) {fprintf(stderr, "Error allocating the frame");return 1;}AVFormatContext* formatContext = NULL;if (avformat_open_input(&formatContext, inFile, NULL, NULL) != 0) {av_free(frame);fprintf(stderr, "Error opening the file");return 1;}if (avformat_find_stream_info(formatContext, NULL) < 0) {av_free(frame);avformat_close_input(&formatContext);fprintf(stderr, "Error finding the stream info");return 1;}AVStream* audioStream = NULL;unsigned int i;// Find the audio stream (some container files can have multiple streams in them)for (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);avformat_close_input(&formatContext);fprintf(stderr, "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);avformat_close_input(&formatContext);fprintf(stderr, "Couldn't find a proper decoder");return 1;} else if (avcodec_open2(codecContext, codecContext->codec, NULL) != 0) {av_free(frame);avformat_close_input(&formatContext);fprintf(stderr, "Couldn't open the context with the decoder");return 1;}/*START*************************START***********************START*//*设置输出格式，为了未来jni调用的简单，所以都传的int，反正是自己用的，直观一些*/int64_t outChannelLayout;if (channel_num == 1) {outChannelLayout = AV_CH_LAYOUT_MONO;} else {outChannelLayout = AV_CH_LAYOUT_STEREO;}enum AVSampleFormat outSampleFormat;if (bit_rate == 32) {outSampleFormat = AV_SAMPLE_FMT_S32;} else {outSampleFormat = AV_SAMPLE_FMT_S16;}int outSampleRate = sample_rate; //44100;SwrContext* swrContext = swr_alloc_set_opts(NULL, outChannelLayout,outSampleFormat, outSampleRate,av_get_default_channel_layout(codecContext->channels),codecContext->sample_fmt, codecContext->sample_rate, 0, NULL);if (swrContext == NULL) {av_free(frame);avcodec_close(codecContext);avformat_close_input(&formatContext);fprintf(stderr, "Couldn't create the SwrContext");return 1;}if (swr_init(swrContext) != 0) {av_free(frame);avcodec_close(codecContext);avformat_close_input(&formatContext);swr_free(&swrContext);fprintf(stderr, "Couldn't initialize the SwrContext");return 1;}/*END**************************END*************************END*/fout = fopen(outFile, "wb+");AVPacket packet;av_init_packet(&packet);double duration = (double) formatContext->duration / 1000.0 / 1000.0;fprintf(stderr, "duration=%f\n", duration);int total_sample = (int) duration * channel_num * sample_rate;fprintf(stderr, "total_sample=%d\n", total_sample);int count = 0;fseek(fout,44, SEEK_SET);while (av_read_frame(formatContext, &packet) == 0) {if (packet.stream_index == audioStream->index) {AVPacket decodingPacket = packet;while (decodingPacket.size > 0) {// Try to decode the packet into a frameint frameFinished = 0;int result = avcodec_decode_audio4(codecContext, frame,&frameFinished, &decodingPacket);if (result < 0 || frameFinished == 0) {break;}unsigned char buffer[100000];unsigned char* pointers[SWR_CH_MAX];pointers[0] = &buffer[0];//这个地方还不是很清楚，看官方的example感觉好复杂。尤其是参数。int numSamplesOut = swr_convert(swrContext, pointers,outSampleRate,(const unsigned char**) frame->extended_data,frame->nb_samples);fwrite((short *) buffer, sizeof(short),(size_t) numSamplesOut * channel_num, fout);count += numSamplesOut* channel_num;decodingPacket.size -= result;decodingPacket.data += result;}}av_free_packet(&packet);}fprintf(stderr, "count=%d\n", count);unsigned char header[44];WavHeader h;h.bits_per_sample = bit_rate;h.channels = channel_num;h.samplerate = sample_rate;h.total_samples = count;write_wav_header(header, &h);fseek(fout,0, SEEK_SET);fwrite(&header, 1, 44, fout);// Clean up!av_free(frame);avcodec_close(codecContext);avformat_close_input(&formatContext);fclose(fout);return 0;}int main() {char *inFile = "ping.mp3";char *outFile = "ping.wav";fprintf(stderr, "res:%d\n",ffmpeg_audio_decode(inFile, outFile, 2, 16, 44100));return 0;}