ffmpeg音频转码，采用swr_convert重取样

1、概述：

和上一篇音频转码流程一样，只是把filter重取样改为了swr_convert重取样，理解本例子对AVFrame中data如何存储音频数据有更好的理解。

2、代码：

 /*
 *最简单的音频转码器（只处理音频）
 *缪国凯 Mickel
 *[email protected]
 *本程序实现从一个视频格式转码到另一个视频格式，只处理音频，视频忽略，若有多个音频流，只处理第一个,采用swr_convert重取样
 *2015-5-8
 */


 #include "stdafx.h"

 #ifdef __cplusplus
 extern"C"
 {
 #endif
 #include
 #include "libavcodec/avcodec.h"
 #include "libavfilter/avfiltergraph.h"
 #include "libavfilter/buffersink.h"
 #include "libavfilter/buffersrc.h"
 #include "libavutil/avutil.h"
 #include "libavutil/opt.h"
 #include "libavutil/pixdesc.h"
 #include "libswresample\swresample.h"
 #include "libavutil\fifo.h"


 #pragma comment(lib, "avcodec.lib")
 #pragma comment(lib, "avformat.lib")
 #pragma comment(lib, "avutil.lib")
 //#pragma comment(lib, "avdevice.lib")
 #pragma comment(lib, "avfilter.lib")
 //#pragma comment(lib, "postproc.lib")
 #pragma comment(lib, "swresample.lib")
 //#pragma comment(lib, "swscale.lib")
 #ifdef __cplusplus
 };
 #endif

 #define BUF_SIZE_20K 2048000
 #define BUF_SIZE_1K 1024000

 SwrContext* pSwrCtx = NULL;
 AVStream *out_stream = NULL;
 AVFormatContext *in_fmt_ctx = NULL, *out_fmt_ctx = NULL;
 int audio_index = -1;

 void initSwr()
 {
     if (out_fmt_ctx->streams[0]->codec->channels != in_fmt_ctx->streams[audio_index]->codec->channels
         || out_fmt_ctx->streams[0]->codec->sample_rate != in_fmt_ctx->streams[audio_index]->codec->sample_rate
         || out_fmt_ctx->streams[0]->codec->sample_fmt != in_fmt_ctx->streams[audio_index]->codec->sample_fmt)
     {
         if ( NULL == pSwrCtx )
         {
             pSwrCtx = swr_alloc();
         }
 #if LIBSWRESAMPLE_VERSION_MINOR >= 17    // 根据版本不同，选用适当函数
         av_opt_set_int(pSwrCtx, "ich", in_fmt_ctx->streams[audio_index]->codec->channels, 0);
         av_opt_set_int(pSwrCtx, "och", out_fmt_ctx->streams[0]->codec->channels, 0);
         av_opt_set_int(pSwrCtx, "in_sample_rate",  in_fmt_ctx->streams[audio_index]->codec->sample_rate, 0);
         av_opt_set_int(pSwrCtx, "out_sample_rate",  out_fmt_ctx->streams[0]->codec->sample_rate, 0);
         av_opt_set_sample_fmt(pSwrCtx, "in_sample_fmt", in_fmt_ctx->streams[audio_index]->codec->sample_fmt, 0);
         av_opt_set_sample_fmt(pSwrCtx, "out_sample_fmt", out_fmt_ctx->streams[0]->codec->sample_fmt, 0);

 #else
         pSwrCtx = swr_alloc_set_opts(pSwrCtx,
             in_fmt_ctx->streams[audio_index]->codec->channel_layout,
             AV_SAMPLE_FMT_S16,
             in_fmt_ctx->streams[audio_index]->codec->sample_rate,
             in_fmt_ctx->streams[audio_index]->codec->channel_layout,
             in_fmt_ctx->streams[audio_index]->codec->sample_fmt,
             in_fmt_ctx->streams[audio_index]->codec->sample_rate,
             0, NULL);
 #endif
         swr_init(pSwrCtx);
     }
 }

 //setup_array函数摘自ffmpeg例程
 static void setup_array(uint8_t* out[SWR_CH_MAX], AVFrame* in_frame, int format, int samples)
 {
     if (av_sample_fmt_is_planar((AVSampleFormat)format))
     {
         int i;
         int plane_size = av_get_bytes_per_sample((AVSampleFormat)(format & 0xFF)) * samples;
         format &= 0xFF;
         //从decoder出来的frame中的data数据不是连续分布的，所以不能这样写：in_frame->data[0]+i*plane_size;
         for (i = 0; i < in_frame->channels; i++)
         {
             out[i] = in_frame->data[i];
         }
     }
     else
     {
         out[0] = in_frame->data[0];
     }
 }

 int TransSample(AVFrame *in_frame, AVFrame *out_frame)
 {
     int ret;


     int max_dst_nb_samples = 4096;
     //int64_t dst_nb_samples;
     int64_t src_nb_samples = in_frame->nb_samples;
     out_frame->pts = in_frame->pts;
     uint8_t* paudiobuf;
     int decode_size, input_size, len;
     if (pSwrCtx != NULL)
     {
         out_frame->nb_samples = av_rescale_rnd(swr_get_delay(pSwrCtx, out_fmt_ctx->streams[0]->codec->sample_rate) + src_nb_samples,
             out_fmt_ctx->streams[0]->codec->sample_rate, out_fmt_ctx->streams[0]->codec->sample_rate, AV_ROUND_UP);

         ret = av_samples_alloc(out_frame->data, &out_frame->linesize[0],
             out_fmt_ctx->streams[0]->codec->channels, out_frame->nb_samples, out_fmt_ctx->streams[0]->codec->sample_fmt, 0);
         if (ret < 0)
         {
             av_log(NULL, AV_LOG_WARNING, "[%s.%d %s() Could not allocate samples Buffer\n", __FILE__, __LINE__, __FUNCTION__);
             return -1;
         }
         max_dst_nb_samples = out_frame->nb_samples;

         //输入也可能是分平面的，所以要做如下处理
         uint8_t* m_ain[SWR_CH_MAX];
         setup_array(m_ain, in_frame, in_fmt_ctx->streams[audio_index]->codec->sample_fmt, src_nb_samples);

         //注意这里，out_count和in_count是samples单位，不是byte
         //所以这样av_get_bytes_per_sample(in_fmt_ctx->streams[audio_index]->codec->sample_fmt) * src_nb_samples是错的
         len = swr_convert(pSwrCtx, out_frame->data, out_frame->nb_samples,
             (const uint8_t**)m_ain, src_nb_samples);

         if (len < 0)
         {
             char errmsg[BUF_SIZE_1K];
             av_strerror(len, errmsg, sizeof(errmsg));
             av_log(NULL, AV_LOG_WARNING, "[%s:%d] swr_convert!(%d)(%s)", __FILE__, __LINE__, len, errmsg);
             return -1;
         }
     }
     else
     {
         printf("pSwrCtx with out init!\n");
         return -1;
     }
     return 0;
 }

 int flush_encoder(AVFormatContext *fmt_ctx, unsigned int stream_index)
 {
     int ret;
     int got_frame;
     AVPacket enc_pkt;
     if (!(fmt_ctx->streams[stream_index]->codec->codec->capabilities &
         CODEC_CAP_DELAY))
     {
         return 0;
     }
     int i = 0;
     while(1)
     {
         enc_pkt.data = NULL;
         enc_pkt.size = 0;
         av_init_packet(&enc_pkt);
         ret = avcodec_encode_audio2(out_fmt_ctx->streams[stream_index]->codec, &enc_pkt,
             NULL, &got_frame);

         if (ret < 0)
             break;
         if (!got_frame)
             break;
         /* prepare packet for muxing */
         enc_pkt.stream_index = stream_index;
         enc_pkt.dts = av_rescale_q_rnd(enc_pkt.dts,
             out_fmt_ctx->streams[stream_index]->codec->time_base,
             out_fmt_ctx->streams[stream_index]->time_base,
             (AVRounding)(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
         enc_pkt.pts = av_rescale_q_rnd(enc_pkt.pts,
             out_fmt_ctx->streams[stream_index]->codec->time_base,
             out_fmt_ctx->streams[stream_index]->time_base,
             (AVRounding)(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
         enc_pkt.duration = av_rescale_q(enc_pkt.duration,
             out_fmt_ctx->streams[stream_index]->codec->time_base,
             out_fmt_ctx->streams[stream_index]->time_base);

         /* mux encoded frame */
         ret = av_interleaved_write_frame(out_fmt_ctx, &enc_pkt);
         av_free_packet(&enc_pkt);
         if (ret < 0)
         {
             break;
         }
         i++;
     }
     printf("flusher write %d frame", i);
     return ret;
 }

 int _tmain(int argc, _TCHAR* argv[])
 {
     AVFrame *frame, *frame_out;
     AVPacket pkt_in, pkt_out;

     int ret;

     if (argc < 3)
     {
         printf("error in input param");
         getchar();
         return -1;
     }

     av_register_all();
     avfilter_register_all();
     //input
     if (avformat_open_input(&in_fmt_ctx, argv[1], NULL, NULL) < 0)
     {
         printf("can not open input file context");
         goto end;
     }
     if (avformat_find_stream_info(in_fmt_ctx, NULL) < 0)
     {
         printf("can not find input stream info!\n");
         goto end;
     }

     //output
     avformat_alloc_output_context2(&out_fmt_ctx, NULL, NULL, argv[2]);
     if (!out_fmt_ctx)
     {
         printf("can not alloc output context!\n");
         goto end;
     }
     //open decoder & new out stream & open encoder
     for (int i = 0; i < in_fmt_ctx->nb_streams; i++)
     {
         if (in_fmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO)
         {
             //open decoder
             if(0 > avcodec_open2(in_fmt_ctx->streams[i]->codec, avcodec_find_decoder(in_fmt_ctx->streams[i]->codec->codec_id), NULL))
             {
                 printf("can not find or open decoder!\n");
                 goto end;
             }

             audio_index = i;

             //new stream
             out_stream = avformat_new_stream(out_fmt_ctx, NULL);
             if (!out_stream)
             {
                 printf("can not new stream for output!\n");
                 goto end;
             }

             //set codec context param
             //use default audio encoder
             out_stream->codec->codec = avcodec_find_encoder(out_fmt_ctx->oformat->audio_codec);
             //use the input audio encoder
             //out_stream->codec->codec = avcodec_find_encoder(ifmt_ctx->streams[i]->codec->codec_id);

             out_stream->codec->sample_rate = in_fmt_ctx->streams[i]->codec->sample_rate;
             out_stream->codec->channel_layout = in_fmt_ctx->streams[i]->codec->channel_layout;
             out_stream->codec->channels = av_get_channel_layout_nb_channels(out_stream->codec->channel_layout);
             // take first format from list of supported formats
             out_stream->codec->sample_fmt = out_stream->codec->codec->sample_fmts[0];
             AVRational time_base={1, out_stream->codec->sample_rate};
             out_stream->codec->time_base = time_base;

             //open encoder
             if (!out_stream->codec->codec)
             {
                 printf("can not find the encoder!\n");
                 goto end;
             }
             if ((avcodec_open2(out_stream->codec, out_stream->codec->codec, NULL)) < 0)
             {
                 printf("can not open the encoder\n");
                 goto end;
             }

             if (out_fmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
                 out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;

             break;
         }
     }

     //dump input info
     av_dump_format(in_fmt_ctx, 0, argv[1], 0);
     //dump output info
     av_dump_format(out_fmt_ctx, 0, argv[2], 1);

     if (-1 == audio_index)
     {
         printf("found no audio stream in input file!\n");
         goto end;
     }

     if (!(out_fmt_ctx->oformat->flags & AVFMT_NOFILE))
     {
         if(avio_open(&out_fmt_ctx->pb, argv[2], AVIO_FLAG_WRITE) < 0)
         {
             printf("can not open output file handle!\n");
             goto end;
         }
     }

     if(avformat_write_header(out_fmt_ctx, NULL) < 0)
     {
         printf("can not write the header of the output file!\n");
         goto end;
     }

     //
     initSwr();

     int got_frame, got_picture;
     int frame_index = 0;
     int i = 0;
     for(int i = 0; ;i++)
     {
         pkt_in.data = NULL;
         pkt_in.size = 0;

         got_frame = -1;
         got_picture = -1;

         if (av_read_frame(in_fmt_ctx, &pkt_in) < 0)
         {
             break;
         }
         if (pkt_in.stream_index != audio_index)
         {
             continue;
         }
         frame = av_frame_alloc();
         if ((ret = avcodec_decode_audio4(in_fmt_ctx->streams[audio_index]->codec, frame, &got_frame, &pkt_in)) < 0)
         {
             av_frame_free(&frame);
             printf("can not decoder a frame");
             break;
         }
         av_free_packet(&pkt_in);

         if (got_frame)
         {
             frame->pts = av_frame_get_best_effort_timestamp(frame);

             frame_out = av_frame_alloc();

             if (0 != TransSample(frame, frame_out))
             {
                 printf("can not swr the audio data!\n");
                 break;
             }

             av_init_packet(&pkt_out);
             int ret = avcodec_encode_audio2(out_fmt_ctx->streams[0]->codec, &pkt_out, frame_out, &got_picture);
             av_free(frame_out->data[0]);
             av_frame_free(&frame_out);
             if (got_picture )
             {
                 pkt_out.stream_index = out_stream->index;
                 pkt_out.dts = av_rescale_q_rnd(pkt_out.dts,
                     out_fmt_ctx->streams[out_stream->index]->codec->time_base,
                     out_fmt_ctx->streams[out_stream->index]->time_base,
                     (AVRounding)(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));

                 pkt_out.pts = av_rescale_q_rnd(pkt_out.pts,
                     out_fmt_ctx->streams[out_stream->index]->codec->time_base,
                     out_fmt_ctx->streams[out_stream->index]->time_base,
                     (AVRounding)(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));

                 pkt_out.duration = av_rescale_q(pkt_out.duration,
                     out_fmt_ctx->streams[out_stream->index]->codec->time_base,
                     out_fmt_ctx->streams[out_stream->index]->time_base);

                 av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
                 /* mux encoded frame */
                 ret = av_interleaved_write_frame(out_fmt_ctx, &pkt_out);
                 av_free_packet(&pkt_out);
                 if (ret < 0)
                 {
                     printf("write a frame failed!\n");
                     break;
                 }
                 printf("success write a frame:index %d\n", frame_index++);
             }

         }
     }
     ret = flush_encoder(out_fmt_ctx, out_stream->index);

     if (ret < 0)
     {
         printf("Flushing encoder failed");
         return -1;
     }

     //write file trailer
     av_write_trailer(out_fmt_ctx);

     //clean
     avcodec_close(out_stream->codec);
     avcodec_close(in_fmt_ctx->streams[audio_index]->codec);

 end:
     avformat_close_input(&in_fmt_ctx);

     if (out_fmt_ctx && !(out_fmt_ctx->oformat->flags & AVFMT_NOFILE))
     {
         avio_close(out_fmt_ctx->pb);
     }
     avformat_free_context(out_fmt_ctx);
     getchar();
     return 0;
 }

3、工程下载地址：

http://download.csdn.net/detail/dancing_night/8685669

目录(?)[+]

1、概述：

和上一篇音频转码流程一样，只是把filter重取样改为了swr_convert重取样，理解本例子对AVFrame中data如何存储音频数据有更好的理解。

2、代码：

 /*
 *最简单的音频转码器（只处理音频）
 *缪国凯 Mickel
 *[email protected]
 *本程序实现从一个视频格式转码到另一个视频格式，只处理音频，视频忽略，若有多个音频流，只处理第一个,采用swr_convert重取样
 *2015-5-8
 */


 #include "stdafx.h"

 #ifdef __cplusplus
 extern"C"
 {
 #endif
 #include
 #include "libavcodec/avcodec.h"
 #include "libavfilter/avfiltergraph.h"
 #include "libavfilter/buffersink.h"
 #include "libavfilter/buffersrc.h"
 #include "libavutil/avutil.h"
 #include "libavutil/opt.h"
 #include "libavutil/pixdesc.h"
 #include "libswresample\swresample.h"
 #include "libavutil\fifo.h"


 #pragma comment(lib, "avcodec.lib")
 #pragma comment(lib, "avformat.lib")
 #pragma comment(lib, "avutil.lib")
 //#pragma comment(lib, "avdevice.lib")
 #pragma comment(lib, "avfilter.lib")
 //#pragma comment(lib, "postproc.lib")
 #pragma comment(lib, "swresample.lib")
 //#pragma comment(lib, "swscale.lib")
 #ifdef __cplusplus
 };
 #endif

 #define BUF_SIZE_20K 2048000
 #define BUF_SIZE_1K 1024000

 SwrContext* pSwrCtx = NULL;
 AVStream *out_stream = NULL;
 AVFormatContext *in_fmt_ctx = NULL, *out_fmt_ctx = NULL;
 int audio_index = -1;

 void initSwr()
 {
     if (out_fmt_ctx->streams[0]->codec->channels != in_fmt_ctx->streams[audio_index]->codec->channels
         || out_fmt_ctx->streams[0]->codec->sample_rate != in_fmt_ctx->streams[audio_index]->codec->sample_rate
         || out_fmt_ctx->streams[0]->codec->sample_fmt != in_fmt_ctx->streams[audio_index]->codec->sample_fmt)
     {
         if ( NULL == pSwrCtx )
         {
             pSwrCtx = swr_alloc();
         }
 #if LIBSWRESAMPLE_VERSION_MINOR >= 17    // 根据版本不同，选用适当函数
         av_opt_set_int(pSwrCtx, "ich", in_fmt_ctx->streams[audio_index]->codec->channels, 0);
         av_opt_set_int(pSwrCtx, "och", out_fmt_ctx->streams[0]->codec->channels, 0);
         av_opt_set_int(pSwrCtx, "in_sample_rate",  in_fmt_ctx->streams[audio_index]->codec->sample_rate, 0);
         av_opt_set_int(pSwrCtx, "out_sample_rate",  out_fmt_ctx->streams[0]->codec->sample_rate, 0);
         av_opt_set_sample_fmt(pSwrCtx, "in_sample_fmt", in_fmt_ctx->streams[audio_index]->codec->sample_fmt, 0);
         av_opt_set_sample_fmt(pSwrCtx, "out_sample_fmt", out_fmt_ctx->streams[0]->codec->sample_fmt, 0);

 #else
         pSwrCtx = swr_alloc_set_opts(pSwrCtx,
             in_fmt_ctx->streams[audio_index]->codec->channel_layout,
             AV_SAMPLE_FMT_S16,
             in_fmt_ctx->streams[audio_index]->codec->sample_rate,
             in_fmt_ctx->streams[audio_index]->codec->channel_layout,
             in_fmt_ctx->streams[audio_index]->codec->sample_fmt,
             in_fmt_ctx->streams[audio_index]->codec->sample_rate,
             0, NULL);
 #endif
         swr_init(pSwrCtx);
     }
 }

 //setup_array函数摘自ffmpeg例程
 static void setup_array(uint8_t* out[SWR_CH_MAX], AVFrame* in_frame, int format, int samples)
 {
     if (av_sample_fmt_is_planar((AVSampleFormat)format))
     {
         int i;
         int plane_size = av_get_bytes_per_sample((AVSampleFormat)(format & 0xFF)) * samples;
         format &= 0xFF;
         //从decoder出来的frame中的data数据不是连续分布的，所以不能这样写：in_frame->data[0]+i*plane_size;
         for (i = 0; i < in_frame->channels; i++)
         {
             out[i] = in_frame->data[i];
         }
     }
     else
     {
         out[0] = in_frame->data[0];
     }
 }

 int TransSample(AVFrame *in_frame, AVFrame *out_frame)
 {
     int ret;


     int max_dst_nb_samples = 4096;
     //int64_t dst_nb_samples;
     int64_t src_nb_samples = in_frame->nb_samples;
     out_frame->pts = in_frame->pts;
     uint8_t* paudiobuf;
     int decode_size, input_size, len;
     if (pSwrCtx != NULL)
     {
         out_frame->nb_samples = av_rescale_rnd(swr_get_delay(pSwrCtx, out_fmt_ctx->streams[0]->codec->sample_rate) + src_nb_samples,
             out_fmt_ctx->streams[0]->codec->sample_rate, out_fmt_ctx->streams[0]->codec->sample_rate, AV_ROUND_UP);

         ret = av_samples_alloc(out_frame->data, &out_frame->linesize[0],
             out_fmt_ctx->streams[0]->codec->channels, out_frame->nb_samples, out_fmt_ctx->streams[0]->codec->sample_fmt, 0);
         if (ret < 0)
         {
             av_log(NULL, AV_LOG_WARNING, "[%s.%d %s() Could not allocate samples Buffer\n", __FILE__, __LINE__, __FUNCTION__);
             return -1;
         }
         max_dst_nb_samples = out_frame->nb_samples;

         //输入也可能是分平面的，所以要做如下处理
         uint8_t* m_ain[SWR_CH_MAX];
         setup_array(m_ain, in_frame, in_fmt_ctx->streams[audio_index]->codec->sample_fmt, src_nb_samples);

         //注意这里，out_count和in_count是samples单位，不是byte
         //所以这样av_get_bytes_per_sample(in_fmt_ctx->streams[audio_index]->codec->sample_fmt) * src_nb_samples是错的
         len = swr_convert(pSwrCtx, out_frame->data, out_frame->nb_samples,
             (const uint8_t**)m_ain, src_nb_samples);

         if (len < 0)
         {
             char errmsg[BUF_SIZE_1K];
             av_strerror(len, errmsg, sizeof(errmsg));
             av_log(NULL, AV_LOG_WARNING, "[%s:%d] swr_convert!(%d)(%s)", __FILE__, __LINE__, len, errmsg);
             return -1;
         }
     }
     else
     {
         printf("pSwrCtx with out init!\n");
         return -1;
     }
     return 0;
 }

 int flush_encoder(AVFormatContext *fmt_ctx, unsigned int stream_index)
 {
     int ret;
     int got_frame;
     AVPacket enc_pkt;
     if (!(fmt_ctx->streams[stream_index]->codec->codec->capabilities &
         CODEC_CAP_DELAY))
     {
         return 0;
     }
     int i = 0;
     while(1)
     {
         enc_pkt.data = NULL;
         enc_pkt.size = 0;
         av_init_packet(&enc_pkt);
         ret = avcodec_encode_audio2(out_fmt_ctx->streams[stream_index]->codec, &enc_pkt,
             NULL, &got_frame);

         if (ret < 0)
             break;
         if (!got_frame)
             break;
         /* prepare packet for muxing */
         enc_pkt.stream_index = stream_index;
         enc_pkt.dts = av_rescale_q_rnd(enc_pkt.dts,
             out_fmt_ctx->streams[stream_index]->codec->time_base,
             out_fmt_ctx->streams[stream_index]->time_base,
             (AVRounding)(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
         enc_pkt.pts = av_rescale_q_rnd(enc_pkt.pts,
             out_fmt_ctx->streams[stream_index]->codec->time_base,
             out_fmt_ctx->streams[stream_index]->time_base,
             (AVRounding)(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
         enc_pkt.duration = av_rescale_q(enc_pkt.duration,
             out_fmt_ctx->streams[stream_index]->codec->time_base,
             out_fmt_ctx->streams[stream_index]->time_base);

         /* mux encoded frame */
         ret = av_interleaved_write_frame(out_fmt_ctx, &enc_pkt);
         av_free_packet(&enc_pkt);
         if (ret < 0)
         {
             break;
         }
         i++;
     }
     printf("flusher write %d frame", i);
     return ret;
 }

 int _tmain(int argc, _TCHAR* argv[])
 {
     AVFrame *frame, *frame_out;
     AVPacket pkt_in, pkt_out;

     int ret;

     if (argc < 3)
     {
         printf("error in input param");
         getchar();
         return -1;
     }

     av_register_all();
     avfilter_register_all();
     //input
     if (avformat_open_input(&in_fmt_ctx, argv[1], NULL, NULL) < 0)
     {
         printf("can not open input file context");
         goto end;
     }
     if (avformat_find_stream_info(in_fmt_ctx, NULL) < 0)
     {
         printf("can not find input stream info!\n");
         goto end;
     }

     //output
     avformat_alloc_output_context2(&out_fmt_ctx, NULL, NULL, argv[2]);
     if (!out_fmt_ctx)
     {
         printf("can not alloc output context!\n");
         goto end;
     }
     //open decoder & new out stream & open encoder
     for (int i = 0; i < in_fmt_ctx->nb_streams; i++)
     {
         if (in_fmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO)
         {
             //open decoder
             if(0 > avcodec_open2(in_fmt_ctx->streams[i]->codec, avcodec_find_decoder(in_fmt_ctx->streams[i]->codec->codec_id), NULL))
             {
                 printf("can not find or open decoder!\n");
                 goto end;
             }

             audio_index = i;

             //new stream
             out_stream = avformat_new_stream(out_fmt_ctx, NULL);
             if (!out_stream)
             {
                 printf("can not new stream for output!\n");
                 goto end;
             }

             //set codec context param
             //use default audio encoder
             out_stream->codec->codec = avcodec_find_encoder(out_fmt_ctx->oformat->audio_codec);
             //use the input audio encoder
             //out_stream->codec->codec = avcodec_find_encoder(ifmt_ctx->streams[i]->codec->codec_id);

             out_stream->codec->sample_rate = in_fmt_ctx->streams[i]->codec->sample_rate;
             out_stream->codec->channel_layout = in_fmt_ctx->streams[i]->codec->channel_layout;
             out_stream->codec->channels = av_get_channel_layout_nb_channels(out_stream->codec->channel_layout);
             // take first format from list of supported formats
             out_stream->codec->sample_fmt = out_stream->codec->codec->sample_fmts[0];
             AVRational time_base={1, out_stream->codec->sample_rate};
             out_stream->codec->time_base = time_base;

             //open encoder
             if (!out_stream->codec->codec)
             {
                 printf("can not find the encoder!\n");
                 goto end;
             }
             if ((avcodec_open2(out_stream->codec, out_stream->codec->codec, NULL)) < 0)
             {
                 printf("can not open the encoder\n");
                 goto end;
             }

             if (out_fmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
                 out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;

             break;
         }
     }

     //dump input info
     av_dump_format(in_fmt_ctx, 0, argv[1], 0);
     //dump output info
     av_dump_format(out_fmt_ctx, 0, argv[2], 1);

     if (-1 == audio_index)
     {
         printf("found no audio stream in input file!\n");
         goto end;
     }

     if (!(out_fmt_ctx->oformat->flags & AVFMT_NOFILE))
     {
         if(avio_open(&out_fmt_ctx->pb, argv[2], AVIO_FLAG_WRITE) < 0)
         {
             printf("can not open output file handle!\n");
             goto end;
         }
     }

     if(avformat_write_header(out_fmt_ctx, NULL) < 0)
     {
         printf("can not write the header of the output file!\n");
         goto end;
     }

     //
     initSwr();

     int got_frame, got_picture;
     int frame_index = 0;
     int i = 0;
     for(int i = 0; ;i++)
     {
         pkt_in.data = NULL;
         pkt_in.size = 0;

         got_frame = -1;
         got_picture = -1;

         if (av_read_frame(in_fmt_ctx, &pkt_in) < 0)
         {
             break;
         }
         if (pkt_in.stream_index != audio_index)
         {
             continue;
         }
         frame = av_frame_alloc();
         if ((ret = avcodec_decode_audio4(in_fmt_ctx->streams[audio_index]->codec, frame, &got_frame, &pkt_in)) < 0)
         {
             av_frame_free(&frame);
             printf("can not decoder a frame");
             break;
         }
         av_free_packet(&pkt_in);

         if (got_frame)
         {
             frame->pts = av_frame_get_best_effort_timestamp(frame);

             frame_out = av_frame_alloc();

             if (0 != TransSample(frame, frame_out))
             {
                 printf("can not swr the audio data!\n");
                 break;
             }

             av_init_packet(&pkt_out);
             int ret = avcodec_encode_audio2(out_fmt_ctx->streams[0]->codec, &pkt_out, frame_out, &got_picture);
             av_free(frame_out->data[0]);
             av_frame_free(&frame_out);
             if (got_picture )
             {
                 pkt_out.stream_index = out_stream->index;
                 pkt_out.dts = av_rescale_q_rnd(pkt_out.dts,
                     out_fmt_ctx->streams[out_stream->index]->codec->time_base,
                     out_fmt_ctx->streams[out_stream->index]->time_base,
                     (AVRounding)(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));

                 pkt_out.pts = av_rescale_q_rnd(pkt_out.pts,
                     out_fmt_ctx->streams[out_stream->index]->codec->time_base,
                     out_fmt_ctx->streams[out_stream->index]->time_base,
                     (AVRounding)(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));

                 pkt_out.duration = av_rescale_q(pkt_out.duration,
                     out_fmt_ctx->streams[out_stream->index]->codec->time_base,
                     out_fmt_ctx->streams[out_stream->index]->time_base);

                 av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
                 /* mux encoded frame */
                 ret = av_interleaved_write_frame(out_fmt_ctx, &pkt_out);
                 av_free_packet(&pkt_out);
                 if (ret < 0)
                 {
                     printf("write a frame failed!\n");
                     break;
                 }
                 printf("success write a frame:index %d\n", frame_index++);
             }

         }
     }
     ret = flush_encoder(out_fmt_ctx, out_stream->index);

     if (ret < 0)
     {
         printf("Flushing encoder failed");
         return -1;
     }

     //write file trailer
     av_write_trailer(out_fmt_ctx);

     //clean
     avcodec_close(out_stream->codec);
     avcodec_close(in_fmt_ctx->streams[audio_index]->codec);

 end:
     avformat_close_input(&in_fmt_ctx);

     if (out_fmt_ctx && !(out_fmt_ctx->oformat->flags & AVFMT_NOFILE))
     {
         avio_close(out_fmt_ctx->pb);
     }
     avformat_free_context(out_fmt_ctx);
     getchar();
     return 0;
 }

3、工程下载地址：

http://download.csdn.net/detail/dancing_night/8685669