c – 使用libav *库的FFMPEG音频转码

我正在使用ffmpeg库编写音频转码应用程序.
这是我的代码

/*
     * File:   main.cpp
     * Author: vinod
     * Compile with "g++ -std=c++11 -o audiotranscode main.cpp -lavformat -lavcodec -lavutil -lavfilter"
     * 
     */
 
 
    #if !defined PRId64 || PRI_MACROS_BROKEN
    #undef PRId64
    #define PRId64 "lld"
    #endif
 
    #define __STDC_FORMAT_MACROS
 
    #ifdef   __cplusplus
    extern "C" {
    #endif
 
    #include <stdio.h>
    #include <stdlib.h>
    #include <sys/types.h>
    #include <stdint.h>
    #include <libavutil/imgutils.h>
    #include <libavutil/samplefmt.h>
    #include <libavutil/frame.h>
    #include <libavutil/timestamp.h>
    #include <libavformat/avformat.h>
    #include <libavfilter/avfilter.h>
    #include <libavfilter/buffersrc.h>
    #include <libavfilter/buffersink.h>
    #include <libswscale/swscale.h>
    #include <libavutil/opt.h>
 
    #ifdef   __cplusplus
    }
    #endif
 
    #include <iostream>
    using namespace std;
 
    int select_stream,got_frame,got_packet;
 
    AVFormatContext *in_fmt_ctx = NULL,*out_fmt_ctx = NULL;
    AVCodec *dec_codec = NULL,* enc_codec = NULL;
    AVStream *audio_st = NULL;
    AVCodecContext *enc_ctx = NULL,*dec_ctx = NULL;
 
    AVFrame *pFrame = NULL,* pFrameFiltered = NULL;
 
    AVFilterGraph *filter_graph = NULL;
    AVFilterContext *buffersrc_ctx = NULL;
    AVFilterContext *buffersink_ctx = NULL;
 
    AVPacket packet;
 
    string inFileName = "/home/vinod/vinod/Media/univac.webm";
    string outFileName = "audio_extracted.m4a";
 
    int target_bit_rate = 128000,sample_rate = 22050,channels = 1;
    AVSampleFormat sample_fmt = AV_SAMPLE_FMT_S16;
    string filter_description = "aresample=22050,aformat=sample_fmts=s16:channel_layouts=mono";
 
    int log_averror(int errcode)
    {
            char *errbuf = (char *) calloc(AV_ERROR_MAX_STRING_SIZE,sizeof(char));
            av_strerror(errcode,errbuf,AV_ERROR_MAX_STRING_SIZE);
            std::cout << "Error - " << errbuf << std::endl;
            delete [] errbuf;
            return -1;
    }
 
    /**
     * Initialize conversion filter */
    int initialize_audio_filter()
    {
            char args[512];
            int ret;
            AVFilter *buffersrc = avfilter_get_by_name("abuffer");
            AVFilter *buffersink = avfilter_get_by_name("abuffersink");
            AVFilterInOut *outputs = avfilter_inout_alloc();
            AVFilterInOut *inputs = avfilter_inout_alloc();
            filter_graph = avfilter_graph_alloc();
            const enum AVSampleFormat out_sample_fmts[] = {sample_fmt,AV_SAMPLE_FMT_NONE};
            const int64_t out_channel_layouts[] = {av_get_default_channel_layout(out_fmt_ctx -> streams[0] -> codec -> channels),-1};
            const int out_sample_rates[] = {out_fmt_ctx -> streams[0] -> codec -> sample_rate,-1};
 
            if (!dec_ctx->channel_layout)
                    dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
 
            snprintf(args,sizeof(args),"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%" PRIx64,in_fmt_ctx -> streams[select_stream] -> time_base.num,in_fmt_ctx -> streams[select_stream] -> time_base.den,dec_ctx->sample_rate,av_get_sample_fmt_name(dec_ctx->sample_fmt),dec_ctx->channel_layout);
            ret = avfilter_graph_create_filter(&buffersrc_ctx,buffersrc,"in",args,NULL,filter_graph);
 
            if (ret < 0) {
                    av_log(NULL,AV_LOG_ERROR,"Cannot create buffer source\n");
                    return -1;
            }
 
            ret = avfilter_graph_create_filter(&buffersink_ctx,buffersink,"out","Cannot create buffer sink\n");
                    return ret;
            }
 
            ret = av_opt_set_int_list(buffersink_ctx,"sample_fmts",out_sample_fmts,-1,AV_OPT_SEARCH_CHILDREN);
 
            if (ret < 0) {
                    av_log(NULL,"Cannot set output sample format\n");
                    return ret;
            }
 
            ret = av_opt_set_int_list(buffersink_ctx,"channel_layouts",out_channel_layouts,"Cannot set output channel layout\n");
                    return ret;
            }
 
            ret = av_opt_set_int_list(buffersink_ctx,"sample_rates",out_sample_rates,"Cannot set output sample rate\n");
                    return ret;
            }
 
            /* Endpoints for the filter graph. */
            outputs -> name = av_strdup("in");
            outputs -> filter_ctx = buffersrc_ctx;
            outputs -> pad_idx = 0;
            outputs -> next = NULL;
            /* Endpoints for the filter graph. */
            inputs -> name = av_strdup("out");
            inputs -> filter_ctx = buffersink_ctx;
            inputs -> pad_idx = 0;
            inputs -> next = NULL;
            string filter_desc = filter_description;
 
            if ((ret = avfilter_graph_parse_ptr(filter_graph,filter_desc.c_str(),&inputs,&outputs,NULL)) < 0) {
                    log_averror(ret);
                    exit(1);
            }
 
            if ((ret = avfilter_graph_config(filter_graph,NULL)) < 0) {
                    log_averror(ret);
                    exit(1);
            }
 
            /* Print summary of the sink buffer
             * Note: args buffer is reused to store channel layout string */
            AVFilterLink *outlink = buffersink_ctx->inputs[0];
            av_get_channel_layout_string(args,outlink->channel_layout);
            av_log(NULL,AV_LOG_INFO,"Output: srate:%dHz fmt:%s chlayout:%s\n",(int) outlink->sample_rate,(char *) av_x_if_null(av_get_sample_fmt_name((AVSampleFormat) outlink->format),"?"),args);
            return 0;
    }
 
    /*
     *
     */
    int main(int argc,char **argv)
    {
            int ret;
            cout << "Hello World" << endl;
            printf("abcd");
            avcodec_register_all();
            av_register_all();
            avfilter_register_all();
 
            /* open input file,and allocate format context */
            if (avformat_open_input(&in_fmt_ctx,inFileName.c_str(),NULL) < 0) {
                    std::cout << "error opening input file - " << inFileName << std::endl;
                    return -1;
            }
 
            /* retrieve stream information */
            if (avformat_find_stream_info(in_fmt_ctx,NULL) < 0) {
                    std::cerr << "Could not find stream information in the input file " << inFileName << std::endl;
            }
 
            /* Dump format details */
            printf("\n ---------------------------------------------------------------------- \n");
            av_dump_format(in_fmt_ctx,0);
            printf("\n ---------------------------------------------------------------------- \n");
            /* Choose a audio stream */
            select_stream = av_find_best_stream(in_fmt_ctx,AVMEDIA_TYPE_AUdio,&dec_codec,0);
 
            if (select_stream == AVERROR_STREAM_NOT_FOUND) {
                    std::cerr << "No audio stream found" << std::endl;
                    return -1;
            }
 
            if (select_stream == AVERROR_DECODER_NOT_FOUND) {
                    std::cerr << "No suitable decoder found" << std::endl;
                    return -1;
            }
 
            dec_ctx = in_fmt_ctx -> streams[ select_stream] -> codec;
            av_opt_set_int(dec_ctx,"refcounted_frames",1,0);
 
            /* init the audio decoder */
            if ((ret = avcodec_open2(dec_ctx,dec_codec,NULL)) < 0) {
                    av_log(NULL,"Cannot open audio decoder\n");
                    return ret;
            }
 
            /* allocate output context */
            ret = avformat_alloc_output_context2(&out_fmt_ctx,outFileName.c_str());
 
            if (ret < 0) {
                    std::cerr << "Could not create output context for the file " << outFileName << std::endl;
                    return -1;
            }
 
            /* find the encoder */
            enum AVCodecID codec_id = out_fmt_ctx -> oformat -> audio_codec;
            enc_codec = avcodec_find_encoder(codec_id);
 
            if (!(enc_codec)) {
                    std::cerr << "Could not find encoder for - " << avcodec_get_name(codec_id) << std::endl;
                    return -1;
            }
 
            /* add a new stream */
            audio_st = avformat_new_stream(out_fmt_ctx,enc_codec);
 
            if (!audio_st) {
                    std::cerr << "Could not add audio stream - " << std::endl;
            }
 
            /* Initialise audio codec */
            audio_st -> id = out_fmt_ctx -> nb_streams - 1;
            enc_ctx = audio_st -> codec;
            enc_ctx -> codec_id = codec_id;
            enc_ctx -> codec_type = AVMEDIA_TYPE_AUdio;
            enc_ctx -> bit_rate = target_bit_rate;
            enc_ctx -> sample_rate = sample_rate;
            enc_ctx -> sample_fmt = sample_fmt;
            enc_ctx -> channels = channels;
            enc_ctx -> channel_layout = av_get_default_channel_layout(enc_ctx -> channels);
 
            /* Some formats want stream headers to be separate. */
            if (out_fmt_ctx -> oformat -> flags & AVFMT_GLOBALHEADER) {
                    enc_ctx -> flags |= CODEC_FLAG_GLOBAL_HEADER;
            }
 
            ret = avcodec_open2(out_fmt_ctx -> streams[0] -> codec,enc_codec,NULL);
 
            if (ret < 0) {
                    std::cerr << "Could not create codec context for the file " << outFileName << std::endl;
                    return -1;
            }
 
            /* Initialize filter */
            initialize_audio_filter();
 
            if (!(out_fmt_ctx -> oformat -> flags & AVFMT_NOFILE)) {
                    int ret = avio_open(& out_fmt_ctx -> pb,outFileName.c_str(),AVIO_FLAG_WRITE);
 
                    if (ret < 0) {
                            log_averror(ret);
                            return -1;
                    }
            }
 
            /* Write header */
            if (avformat_write_header(out_fmt_ctx,NULL) < 0) {
                    if (ret < 0) {
                            log_averror(ret);
                            return -1;
                    }
            }
 
            /* Allocate frame */
            pFrame = av_frame_alloc();
 
            if (!pFrame) {
                    std::cerr << "Could not allocate frame\n";
                    return -1;
            }
 
            pFrameFiltered = av_frame_alloc();
 
            if (!pFrameFiltered) {
                    std::cerr << "Could not allocate frame\n";
                    return -1;
            }
 
            av_init_packet(&packet);
            packet.data = NULL;
            packet.size = 0;
 
            /* Read packet from the stream */
            while (av_read_frame(in_fmt_ctx,&packet) >= 0) {
                    if (packet.stream_index == select_stream) {
                            avcodec_get_frame_defaults(pFrame);
                            ret = avcodec_decode_audio4(dec_ctx,pFrame,&got_frame,&packet);
 
                            if (ret < 0) {
                                    log_averror(ret);
                                    return ret;
                            }
 
                            printf("Decoded packet pts : %ld ",packet.pts);
                            printf("Frame Best Effor pts : %ld \n",pFrame->best_effort_timestamp);
                            /* Set frame pts */
                            pFrame -> pts = av_frame_get_best_effort_timestamp(pFrame);
 
                            if (got_frame) {
                                    /* push the decoded frame into the filtergraph */
                                    ret = av_buffersrc_add_frame_flags(buffersrc_ctx,AV_BUFFERSRC_FLAG_KEEP_REF);
 
                                    if (ret < 0) {
                                            log_averror(ret);
                                            return ret;
                                    }
 
                                    /* pull filtered frames from the filtergraph */
                                    while (1) {
                                            ret = av_buffersink_get_frame(buffersink_ctx,pFrameFiltered);
 
                                            if ((ret == AVERROR(EAGAIN)) || (ret == AVERROR_EOF)) {
                                                    break;
                                            }
 
                                            if (ret < 0) {
                                                    printf("Error while getting filtered frames from filtergraph\n");
                                                    log_averror(ret);
                                                    return -1;
                                            }
 
                                            /* Initialize the packets */
                                            AVPacket encodedPacket = {0};
                                            av_init_packet(&encodedPacket);
                                            ret = avcodec_encode_audio2(out_fmt_ctx -> streams[0] -> codec,&encodedPacket,pFrameFiltered,&got_packet);
 
                                            if (!ret && got_packet && encodedPacket.size) {
                                                    /* Set correct pts and dts */
                                                    if (encodedPacket.pts != AV_NOPTS_VALUE) {
                                                            encodedPacket.pts = av_rescale_q(encodedPacket.pts,buffersink_ctx -> inputs[0] -> time_base,out_fmt_ctx -> streams[0] -> time_base);
                                                    }
 
                                                    if (encodedPacket.dts != AV_NOPTS_VALUE) {
                                                            encodedPacket.dts = av_rescale_q(encodedPacket.dts,out_fmt_ctx -> streams[0] -> time_base);
                                                    }
 
                                                    printf("Encoded packet pts %ld\n",encodedPacket.pts);
                                                    /* Write the compressed frame to the media file. */
                                                    ret = av_interleaved_write_frame(out_fmt_ctx,&encodedPacket);
 
                                                    if (ret < 0) {
                                                            log_averror(ret);
                                                            return -1;
                                                    }
                                            } else if (ret < 0) {
                                                    log_averror(ret);
                                                    return -1;
                                            }
 
                                            av_frame_unref(pFrameFiltered);
                                    }
 
                                    av_frame_unref(pFrame);
                            }
                    }
            }
 
            /* Flush delayed frames from encoder*/
            got_packet=1;
            while (got_packet) {
                    AVPacket encodedPacket = {0};
                    av_init_packet(&encodedPacket);
                    ret = avcodec_encode_audio2(out_fmt_ctx -> streams[0] -> codec,&got_packet);
 
                    if (!ret && got_packet && encodedPacket.size) {
                            /* Set correct pts and dts */
                            if (encodedPacket.pts != AV_NOPTS_VALUE) {
                                    encodedPacket.pts = av_rescale_q(encodedPacket.pts,out_fmt_ctx -> streams[0] -> time_base);
                            }
 
                            if (encodedPacket.dts != AV_NOPTS_VALUE) {
                                    encodedPacket.dts = av_rescale_q(encodedPacket.dts,out_fmt_ctx -> streams[0] -> time_base);
                            }
 
                            printf("Encoded packet pts %ld\n",encodedPacket.pts);
                            /* Write the compressed frame to the media file. */
                            ret = av_interleaved_write_frame(out_fmt_ctx,&encodedPacket);
 
                            if (ret < 0) {
                                    log_averror(ret);
                                    return -1;
                            }
                    } else if (ret < 0) {
                            log_averror(ret);
                            return -1;
                    }
            }
 
            /* Write Trailer */
            av_write_trailer(out_fmt_ctx);
            avfilter_graph_free(&filter_graph);
 
            if (dec_ctx)
                    avcodec_close(dec_ctx);
 
            avformat_close_input(&in_fmt_ctx);
            av_frame_free(&pFrame);
            av_frame_free(&pFrameFiltered);
 
            if (!(out_fmt_ctx -> oformat -> flags & AVFMT_NOFILE))
                    avio_close(out_fmt_ctx -> pb);
            avcodec_close(out_fmt_ctx->streams[0]->codec);
            avformat_free_context(out_fmt_ctx);
            return 0;
    }

转码后的音频文件与输入的持续时间相同.但它完全嘈杂.谁能告诉我这里做错了什么！

解决方法

我已经找到了问题的所在,并且已经解决了.

当以大胆打开输出文件时,可以看到在音频信号中插入了不需要的静音.问题在于提供给编码器的“每帧采样数”.

不同的编解码器期望编码的帧大小不同.并且aac编码器期望大小为1024.这可以通过在执行avcodec_open2()之后观察enc_ctx-> frame_size来看出.

滤波器需要向编码器提供每通道1024个采样数的帧.
所以在我的代码中,pFrameFiltered每个通道需要有1024个样本数.如果它小于1024,编码器会附加零,使其成为1024个样本,然后对其进行编码.

这可以通过拥有我们自己的fifo队列或使用ffmpeg音频过滤器提供的过滤器来解决.我们需要使用过滤器asetnsamples = n = 1024：p = 0,如here所述.所以需要的更改是

`string filter_description = 
        "aresample=22050,aformat=sample_fmts=s16:channel_layouts=mono,asetnsamples=n=1024:p=0";`

只需在过滤器中使用n的值来更好地理解.检查avcodec_open2()设置的enc_ctx-> frame_size字段,并适当设置n的值.

c – 使用libav *库的FFMPEG音频转码

解决方法

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