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G711/AAC与H264/H265封装成MP4_ffmpeg4.4.1 将g711直接封装到mp4文件中

作者：神奇cpp | 2024-07-30 03:05:36

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ffmpeg4.4.1 将g711直接封装到mp4文件中

提示：本文提供一种将音频文件和视频文件封装成MP4文件的解决方案

文章目录

一、背景描述

FFmpeg支持将 AAC+H264/H265 封装成 MP4，但是不支持G711+H264/H265封装成MP4封装成MP4，所以如果音频类型是G711，则要先转换成AAC再进行封装。

二、解决方案

首先，当音频类型是AAC时，可以直接用FFmpeg来进行封装。FFmpeg的编译可以参考：Windows编译和使用ffmpeg。注意编译FFmpeg的时候，不要裁剪过多东西，不然可能导致调用一些接口的时候不成功（如果发现代码调用FFmpeg接口失败，可以先试一下未裁剪版本的FFmpeg，验证是不是由于过度裁剪导致的异常）。以下是雷霄骅实现的音视频封装代码，可以封装AAC+H264/H265：

/**
 * 最简单的基于FFmpeg的视音频复用器
 * Simplest FFmpeg Muxer
 *
 * 雷霄骅 Lei Xiaohua
 * leixiaohua1020@126.com
 * 中国传媒大学/数字电视技术
 * Communication University of China / Digital TV Technology
 * http://blog.csdn.net/leixiaohua1020
 *
 * 本程序可以将视频码流和音频码流打包到一种封装格式中。
 * 程序中将AAC编码的音频码流和H.264编码的视频码流打包成
 * MPEG2TS封装格式的文件。
 * 需要注意的是本程序并不改变视音频的编码格式。
 *
 * This software mux a video bitstream and a audio bitstream
 * together into a file.
 * In this example, it mux a H.264 bitstream (in MPEG2TS) and
 * a AAC bitstream file together into MP4 format file.
 *
 */

#include <stdio.h>

#define __STDC_CONSTANT_MACROS

#ifdef _WIN32
// Windows
extern "C" {
#include "libavformat/avformat.h"
};
#else
// Linux...
#ifdef __cplusplus
extern "C" {
#endif
#include <libavformat/avformat.h>
#ifdef __cplusplus
};
#endif
#endif

/*
FIX: H.264 in some container format (FLV, MP4, MKV etc.) need
"h264_mp4toannexb" bitstream filter (BSF)
  *Add SPS,PPS in front of IDR frame
  *Add start code ("0,0,0,1") in front of NALU
H.264 in some container (MPEG2TS) don't need this BSF.
*/
//'1': Use H.264 Bitstream Filter
#define USE_H264BSF 0

/*
FIX:AAC in some container format (FLV, MP4, MKV etc.) need
"aac_adtstoasc" bitstream filter (BSF)
*/
//'1': Use AAC Bitstream Filter
#define USE_AACBSF 0

int main(int argc, char *argv[]) {
    getchar();
    AVOutputFormat *ofmt = NULL;
    // Input AVFormatContext and Output AVFormatContext
    AVFormatContext *ifmt_ctx_v = NULL, *ifmt_ctx_a = NULL, *ofmt_ctx = NULL;
    AVPacket pkt;
    int ret, i;
    int videoindex_v = -1, videoindex_out = -1;
    int audioindex_a = -1, audioindex_out = -1;
    int frame_index = 0;
    int64_t cur_pts_v = 0, cur_pts_a = 0;

    // const char *in_filename_v = "cuc_ieschool.ts";//Input file URL
    const char *in_filename_v = "cuc_ieschool.h264";
    // const char *in_filename_v = "cuc_ieschool.h265";
    // const char *in_filename_a = "cuc_ieschool.mp3";
    // const char *in_filename_a = "gowest.m4a";
    const char *in_filename_a = "gowest.aac";
    //const char *in_filename_a = "huoyuanjia.mp3";

    const char *out_filename = "cuc_ieschool.mp4";  // Output file URL
    //const char *out_filename = "cuc_ieschool.ts";

	av_register_all();
    // Input
    if ((ret = avformat_open_input(&ifmt_ctx_v, in_filename_v, 0, 0)) < 0) {
        printf("Could not open input file.");
        goto end;
    }
    if ((ret = avformat_find_stream_info(ifmt_ctx_v, 0)) < 0) {
        printf("Failed to retrieve input stream information");
        goto end;
    }

    if ((ret = avformat_open_input(&ifmt_ctx_a, in_filename_a, 0, 0)) < 0) {
        printf("Could not open input file.");
        goto end;
    }
    if ((ret = avformat_find_stream_info(ifmt_ctx_a, 0)) < 0) {
        printf("Failed to retrieve input stream information");
        goto end;
    }
    printf("===========Input Information==========\n");
    av_dump_format(ifmt_ctx_v, 0, in_filename_v, 0);
    av_dump_format(ifmt_ctx_a, 0, in_filename_a, 0);
    printf("======================================\n");
    // Output
    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
    if (!ofmt_ctx) {
        printf("Could not create output context\n");
        ret = AVERROR_UNKNOWN;
        goto end;
    }
    ofmt = ofmt_ctx->oformat;

    for (i = 0; i < ifmt_ctx_v->nb_streams; i++) {
        // Create output AVStream according to input AVStream
        if (ifmt_ctx_v->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
            AVStream *in_stream = ifmt_ctx_v->streams[i];
            AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
            videoindex_v = i;
            if (!out_stream) {
                printf("Failed allocating output stream\n");
                ret = AVERROR_UNKNOWN;
                goto end;
            }
            videoindex_out = out_stream->index;
            // Copy the settings of AVCodecContext
            if (avcodec_copy_context(out_stream->codec, in_stream->codec) < 0) {
                printf("Failed to copy context from input to output stream codec context\n");
                goto end;
            }
            out_stream->codec->codec_tag = 0;
            if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER) out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
            break;
        }
    }

    for (i = 0; i < ifmt_ctx_a->nb_streams; i++) {
        // Create output AVStream according to input AVStream
        if (ifmt_ctx_a->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
            AVStream *in_stream = ifmt_ctx_a->streams[i];
            AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
            audioindex_a = i;
            if (!out_stream) {
                printf("Failed allocating output stream\n");
                ret = AVERROR_UNKNOWN;
                goto end;
            }
            audioindex_out = out_stream->index;
            // Copy the settings of AVCodecContext
            if (avcodec_copy_context(out_stream->codec, in_stream->codec) < 0) {
                printf("Failed to copy context from input to output stream codec context\n");
                goto end;
            }
            out_stream->codec->codec_tag = 0;
            if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER) out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;

            break;
        }
    }

    printf("==========Output Information==========\n");
    av_dump_format(ofmt_ctx, 0, out_filename, 1);
    printf("======================================\n");
    // Open output file
    if (!(ofmt->flags & AVFMT_NOFILE)) {
        if (avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE) < 0) {
            printf("Could not open output file '%s'", out_filename);
            goto end;
        }
    }
    // Write file header
    if (avformat_write_header(ofmt_ctx, NULL) < 0) {
        printf("Error occurred when opening output file\n");
        goto end;
    }

    // FIX
#if USE_H264BSF
    AVBitStreamFilterContext *h264bsfc = av_bitstream_filter_init("h264_mp4toannexb");
#endif
#if USE_AACBSF
    AVBitStreamFilterContext *aacbsfc = av_bitstream_filter_init("aac_adtstoasc");
#endif

    while (1) {
        AVFormatContext *ifmt_ctx;
        int stream_index = 0;
        AVStream *in_stream, *out_stream;

        // Get an AVPacket
        if (av_compare_ts(cur_pts_v, ifmt_ctx_v->streams[videoindex_v]->time_base, cur_pts_a, ifmt_ctx_a->streams[audioindex_a]->time_base) <= 0) {
            ifmt_ctx = ifmt_ctx_v;
            stream_index = videoindex_out;

            if (av_read_frame(ifmt_ctx, &pkt) >= 0) {
                do {
                    in_stream = ifmt_ctx->streams[pkt.stream_index];
                    out_stream = ofmt_ctx->streams[stream_index];

                    if (pkt.stream_index == videoindex_v) {
                        // FIX：No PTS (Example: Raw H.264)
                        // Simple Write PTS
                        if (pkt.pts == AV_NOPTS_VALUE) {
                            // Write PTS
                            AVRational time_base1 = in_stream->time_base;
                            // Duration between 2 frames (us)
                            int64_t calc_duration = (double)AV_TIME_BASE / av_q2d(in_stream->r_frame_rate);
                            // Parameters
                            pkt.pts = (double)(frame_index * calc_duration) / (double)(av_q2d(time_base1) * AV_TIME_BASE);
                            pkt.dts = pkt.pts;
                            pkt.duration = (double)calc_duration / (double)(av_q2d(time_base1) * AV_TIME_BASE);
                            frame_index++;
                        }

                        cur_pts_v = pkt.pts;
                        break;
                    }
                } while (av_read_frame(ifmt_ctx, &pkt) >= 0);
            } else {
                break;
            }
        } else {
            ifmt_ctx = ifmt_ctx_a;
            stream_index = audioindex_out;
            if (av_read_frame(ifmt_ctx, &pkt) >= 0) {
                do {
                    in_stream = ifmt_ctx->streams[pkt.stream_index];
                    out_stream = ofmt_ctx->streams[stream_index];

                    if (pkt.stream_index == audioindex_a) {
                        // FIX：No PTS
                        // Simple Write PTS
                        if (pkt.pts == AV_NOPTS_VALUE) {
                            // Write PTS
                            AVRational time_base1 = in_stream->time_base;
                            // Duration between 2 frames (us)
                            int64_t calc_duration = (double)AV_TIME_BASE / av_q2d(in_stream->r_frame_rate);
                            // Parameters
                            pkt.pts = (double)(frame_index * calc_duration) / (double)(av_q2d(time_base1) * AV_TIME_BASE);
                            pkt.dts = pkt.pts;
                            pkt.duration = (double)calc_duration / (double)(av_q2d(time_base1) * AV_TIME_BASE);
                            frame_index++;
                        }
                        cur_pts_a = pkt.pts;

                        break;
                    }
                } while (av_read_frame(ifmt_ctx, &pkt) >= 0);
            } else {
                break;
            }
        }

        // FIX:Bitstream Filter
#if USE_H264BSF
        av_bitstream_filter_filter(h264bsfc, in_stream->codec, NULL, &pkt.data, &pkt.size, pkt.data, pkt.size, 0);
#endif
#if USE_AACBSF
        av_bitstream_filter_filter(aacbsfc, out_stream->codec, NULL, &pkt.data, &pkt.size, pkt.data, pkt.size, 0);
#endif

        // Convert PTS/DTS
        pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, (AVRounding)(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));
        pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, (AVRounding)(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));
        pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
        pkt.pos = -1;
        pkt.stream_index = stream_index;

        printf("Write 1 Packet. size:%5d\tpts:%lld\n", pkt.size, pkt.pts);
        // Write
        if (av_interleaved_write_frame(ofmt_ctx, &pkt) < 0) {
            printf("Error muxing packet\n");
            break;
        }
        av_free_packet(&pkt);
    }
    // Write file trailer
    av_write_trailer(ofmt_ctx);

#if USE_H264BSF
    av_bitstream_filter_close(h264bsfc);
#endif
#if USE_AACBSF
    av_bitstream_filter_close(aacbsfc);
#endif

end:
    avformat_close_input(&ifmt_ctx_v);
    avformat_close_input(&ifmt_ctx_a);
    /* close output */
    if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE)) avio_close(ofmt_ctx->pb);
    avformat_free_context(ofmt_ctx);
    if (ret < 0 && ret != AVERROR_EOF) {
        printf("Error occurred.\n");
        return -1;
    }
    getchar();
    return 0;
}

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接下来看一下，当音频类型不是AAC时，要先转换成AAC。代码基于EasyAACEncoder开源库（点击下载），该库是EasyDarwin开源流媒体服务团队整理、开发的一款音频转码到AAC的工具库，目前支持G711a/G711u/G726/PCM等音频格式的转码，跨平台，支持Windows/Linux/arm。

#include <iostream>
#include <stdio.h>
#include <stdlib.h>
#include "EasyAACEncoderAPI.h"
#pragma comment(lib, "libEasyAACEncoder.lib")

using namespace std;

int main() {
    InitParam initParam;
    initParam.u32AudioSamplerate = 8000;
    initParam.ucAudioChannel = 1;
    initParam.u32PCMBitSize = 16;
    initParam.ucAudioCodec = Law_ALaw;
    //initParam.ucAudioCodec = Law_ULaw;

    Easy_Handle handle = Easy_AACEncoder_Init(initParam);
    char* infilename = "g711.g711a";  //标准
    char* outAacname = "g711.aac";

    FILE* fpIn = fopen(infilename, "rb");
    if (NULL == fpIn) {
        printf("%s:[%d] open %s file failed\n", __FUNCTION__, __LINE__, infilename);
        return -1;
    }

    FILE* fpOut = fopen(outAacname, "wb");
    if(NULL == fpOut) {
        printf("%s:[%d] open %s file failed\n", __FUNCTION__, __LINE__, outAacname);
        return -1;
    }

    int gBytesRead = 0;
    int bG711ABufferSize = 500;
    int bAACBufferSize = 4 * bG711ABufferSize;  // 提供足够大的缓冲区
    unsigned char *pbG711ABuffer = (unsigned char *)malloc(bG711ABufferSize * sizeof(unsigned char));
    unsigned char *pbAACBuffer = (unsigned char*)malloc(bAACBufferSize * sizeof(unsigned char));  
    unsigned int out_len = 0;

    while((gBytesRead = fread(pbG711ABuffer, 1, bG711ABufferSize, fpIn)) >0) {    
        if(Easy_AACEncoder_Encode(handle, pbG711ABuffer, gBytesRead, pbAACBuffer, &out_len) > 0) {
            fwrite(pbAACBuffer, 1, out_len, fpOut);
        }
    }

    Easy_AACEncoder_Release(handle);

    free(pbG711ABuffer);
    free(pbAACBuffer);
    fclose(fpIn);
    fclose(fpOut);
    return 0;
}

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转成AAC后，就可以继续封装成MP4了。

参考文章

最简单的基于FFmpeg的封装格式处理：视音频复用器（muxer）

开源项目：简单、高效、稳定的开源音频编码库

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