FFmpeg 混音学习【三】两路混流完整代码

最新推荐文章于 2024-07-18 16:08:26 发布
最新推荐文章于 2024-07-18 16:08:26 发布
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分类专栏： FFmpeg学习记录 文章标签： ffmpeg c++ filter aac
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本文链接：https://blog.csdn.net/qq_37984341/article/details/107404909
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前言

本文是简单的混流实现，输入为两个媒体文件（音频或带音频的视频文件），输出为混流后的aac文件。

完整代码

/*
 * Copyright (c) 2010 Nicolas George
 * Copyright (c) 2011 Stefano Sabatini
 * Copyright (c) 2012 Clément Bœsch
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */

/**
 * @file
 * API example for audio decoding and filtering
 * @example filtering_audio.c
 */

extern "C" {
#include <libavcodec/avcodec.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
}
//#include <unistd.h>

#include <string>
#include <iostream>

#define ENABLE_FILTERS 1

static const char* filter_descr ="[in0][in1]amix=inputs=2[out]";  

static AVFormatContext* fmt_ctx1;
static AVFormatContext* fmt_ctx2;

static AVCodecContext* dec_ctx1;
static AVCodecContext* dec_ctx2;

AVFilterContext* buffersink_ctx;
AVFilterContext* buffersrc_ctx1;
AVFilterContext* buffersrc_ctx2;

AVFormatContext* ofmt_ctx;
AVCodecContext* enc_ctx;  //输出文件的codec
int pkt_count = 0;
AVFrame* frame;

AVFilterGraph* filter_graph;
static int audio_stream_index_1 = -1;
static int audio_stream_index_2 = -1;


static int open_input_file_1(const char* filename) {
  int ret;
  AVCodec* dec;

  if ((ret = avformat_open_input(&fmt_ctx1, filename, NULL, NULL)) < 0) {
    av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
    return ret;
  }

  if ((ret = avformat_find_stream_info(fmt_ctx1, NULL)) < 0) {
    av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
    return ret;
  }

  /* select the audio stream */
  ret = av_find_best_stream(fmt_ctx1, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
  if (ret < 0) {
    av_log(NULL, AV_LOG_ERROR,
           "Cannot find an audio stream in the input file\n");
    return ret;
  }
  audio_stream_index_1 = ret;
  dec_ctx1 = fmt_ctx1->streams[audio_stream_index_1]->codec;
  av_opt_set_int(dec_ctx1, "refcounted_frames", 1, 0);

  /* init the audio decoder */
  if ((ret = avcodec_open2(dec_ctx1, dec, NULL)) < 0) {
    av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
    return ret;
  }

  return 0;
}

static int open_input_file_2(const char* filename) {
  int ret;
  AVCodec* dec;

  if ((ret = avformat_open_input(&fmt_ctx2, filename, NULL, NULL)) < 0) {
    av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
    return ret;
  }

  if ((ret = avformat_find_stream_info(fmt_ctx2, NULL)) < 0) {
    av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
    return ret;
  }

  /* select the audio stream */
  ret = av_find_best_stream(fmt_ctx2, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
  if (ret < 0) {
    av_log(NULL, AV_LOG_ERROR,
           "Cannot find an audio stream in the input file\n");
    return ret;
  }
  audio_stream_index_2 = ret;
  dec_ctx2 = fmt_ctx2->streams[audio_stream_index_2]->codec;
  av_opt_set_int(dec_ctx2, "refcounted_frames", 1, 0);

  /* init the audio decoder */
  if ((ret = avcodec_open2(dec_ctx2, dec, NULL)) < 0) {
    av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
    return ret;
  }

  return 0;
}

int encode_write_frame(AVFrame* filt_frame, unsigned int stream_index) {
  int ret;
  I_LOG("Write frame {}", filt_frame->nb_samples);
  ret = avcodec_send_frame(enc_ctx, filt_frame);
  if (ret < 0) {
    E_LOG("Error while sending a frame to the encoder: {}", ret);
    return -1;
  }

  while (ret >= 0) {
    AVPacket pkt;
    av_init_packet(&pkt);
    ret = avcodec_receive_packet(enc_ctx, &pkt);
    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
      break;
    else if (ret < 0) {
      E_LOG("Error while encoding a frame: {}", ret);
      return -1;
    }

    /* rescale output packet timestamp values from codec to stream timebase */
    av_packet_rescale_ts(&pkt, enc_ctx->time_base,
                         ofmt_ctx->streams[stream_index]->time_base);
    pkt.stream_index = stream_index;

    /* Write the compressed frame to the media file. */
    // log_packet(fmt_ctx, &pkt);

    pkt.pts = pkt_count * ofmt_ctx->streams[stream_index]->codec->frame_size;
    pkt.dts = pkt.pts;
    pkt.duration = ofmt_ctx->streams[stream_index]->codec->frame_size;

    pkt.pts = av_rescale_q_rnd(
        pkt.pts, ofmt_ctx->streams[stream_index]->codec->time_base,
        ofmt_ctx->streams[stream_index]->time_base,
        (AVRounding)(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));
    pkt.dts = pkt.pts;
    pkt.duration = av_rescale_q_rnd(
        pkt.duration, ofmt_ctx->streams[stream_index]->codec->time_base,
        ofmt_ctx->streams[stream_index]->time_base,
        (AVRounding)(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));

    ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
    ++pkt_count;
    av_packet_unref(&pkt);
    if (ret < 0) {
      E_LOG("Error while writing output packet: {}", ret);
      return -1;
    }
  }
  return ret;
}

int open_output_file(const char* filename) {
  AVStream* out_stream;
  AVCodec* encoder;
  int ret;
  ofmt_ctx = NULL;
  avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
  if (!ofmt_ctx) {
    av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
    return AVERROR_UNKNOWN;
  }
  {
    out_stream = avformat_new_stream(ofmt_ctx, NULL);
    out_stream->index = 0;
    if (!out_stream) {
      av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
      return AVERROR_UNKNOWN;
    }

    encoder = avcodec_find_encoder(AV_CODEC_ID_AAC);
    enc_ctx = avcodec_alloc_context3(encoder);
    if (!enc_ctx) {
      E_LOG("Could not alloc an encoding context\n");
      return -1;
    }
    enc_ctx = out_stream->codec;
    enc_ctx->codec_type = AVMEDIA_TYPE_AUDIO;
    enc_ctx->sample_rate = 48000;
    enc_ctx->channels = 2;
    enc_ctx->channel_layout = 3;
    enc_ctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
    // enc_ctx->bit_rate = 320000;
    enc_ctx->codec_tag = 0;
    AVRational ar = {1, enc_ctx->sample_rate};
    enc_ctx->time_base = ar;

    ret = avcodec_open2(enc_ctx, encoder, NULL);
    if (ret < 0) {
      av_log(NULL, AV_LOG_ERROR, "Cannot open audio encoder for stream \n");
      return ret;
    }

    ret = avcodec_parameters_from_context(out_stream->codecpar, enc_ctx);
    if (ret < 0) {
      E_LOG("Could not copy the stream parameters\n");
      return ret;
    }

    if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
      enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
  }

  av_dump_format(ofmt_ctx, 0, filename, 1);

  // if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
  ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
  if (ret < 0) {
    av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
    return ret;
  }
  // }
  /* init Mixer, write output file header */
  ret = avformat_write_header(ofmt_ctx, NULL);
  if (ret < 0) {
    av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
    return ret;
  }
  return 0;
}

static int init_filters(const char* filters_descr) {
  char args1[512];
  char args2[512];
  int ret = 0;
  const AVFilter* abuffersrc1 = avfilter_get_by_name("abuffer");
  const AVFilter* abuffersrc2 = avfilter_get_by_name("abuffer");
  const AVFilter* abuffersink = avfilter_get_by_name("abuffersink");

  AVFilterInOut* outputs1 = avfilter_inout_alloc();
  AVFilterInOut* outputs2 = avfilter_inout_alloc();
  AVFilterInOut* inputs = avfilter_inout_alloc();

  static const enum AVSampleFormat out_sample_fmts[] = {
      AV_SAMPLE_FMT_FLTP  ,// dec_ctx1->sample_fmt,
      (enum AVSampleFormat) - 1};  //{AV_SAMPLE_FMT_S16}
  static const int64_t out_channel_layouts[] =
      //{dec_ctx1->channel_layout,  -1};
      {3, -1};
  static const int out_sample_rates[] = {48000, -1};//{dec_ctx1->sample_rate, -1};
  const AVFilterLink* outlink;

  AVRational time_base_1 = fmt_ctx1->streams[audio_stream_index_1]->time_base;
  AVRational time_base_2 = fmt_ctx2->streams[audio_stream_index_2]->time_base;

  filter_graph = avfilter_graph_alloc();
  if (!outputs1 || !inputs || !filter_graph) {
    ret = AVERROR(ENOMEM);
    goto end;
  }

  /* buffer audio source: the decoded frames from the decoder will be inserted
   * here. */
  if (!dec_ctx1->channel_layout)
    dec_ctx1->channel_layout =
        av_get_default_channel_layout(dec_ctx1->channels);
  snprintf(
      args1, sizeof(args1),
      "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%" PRIx64,
      time_base_1.num, time_base_1.den, dec_ctx1->sample_rate,
      av_get_sample_fmt_name(dec_ctx1->sample_fmt), dec_ctx1->channel_layout);
  ret = avfilter_graph_create_filter(&buffersrc_ctx1, abuffersrc1, "in1", args1,
                                     NULL, filter_graph);
  if (ret < 0) {
    av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
    goto end;
  }

#if (ENABLE_FILTERS)
  /* buffer audio source: the decoded frames from the decoder will be inserted
   * here. */
  if (!dec_ctx2->channel_layout)
    dec_ctx2->channel_layout =
        av_get_default_channel_layout(dec_ctx2->channels);
  snprintf(
      args2, sizeof(args2),
      "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%" PRIx64,
      time_base_2.num, time_base_2.den, dec_ctx2->sample_rate,
      av_get_sample_fmt_name(dec_ctx2->sample_fmt), dec_ctx2->channel_layout);
  ret = avfilter_graph_create_filter(&buffersrc_ctx2, abuffersrc1, "in2", args2,
                                     NULL, filter_graph);
  if (ret < 0) {
    av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
    goto end;
  }
#endif
  /* buffer audio sink: to terminate the filter chain. */
  ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out", NULL,
                                     NULL, filter_graph);
  if (ret < 0) {
    av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
    goto end;
  }

  ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
                            AV_OPT_SEARCH_CHILDREN);
  if (ret < 0) {
    av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
    goto end;
  }

  ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts",
                            out_channel_layouts, -1, AV_OPT_SEARCH_CHILDREN);
  if (ret < 0) {
    av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
    goto end;
  }

  ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates,
                            -1, AV_OPT_SEARCH_CHILDREN);
  if (ret < 0) {
    av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
    goto end;
  }

  /*
   * Set the endpoints for the filter graph. The filter_graph will
   * be linked to the graph described by filters_descr.
   */

  /*
   * The buffer source output must be connected to the input pad of
   * the first filter described by filters_descr; since the first
   * filter input label is not specified, it is set to "in" by
   * default.
   */
  outputs1->name = av_strdup("in0");
  outputs1->filter_ctx = buffersrc_ctx1;
  outputs1->pad_idx = 0;
#if (ENABLE_FILTERS)
  outputs1->next = outputs2;

  outputs2->name = av_strdup("in1");
  outputs2->filter_ctx = buffersrc_ctx2;
  outputs2->pad_idx = 0;
  outputs2->next = NULL;
#else
  outputs1->next = NULL;
#endif
  /*
   * The buffer sink input must be connected to the output pad of
   * the last filter described by filters_descr; since the last
   * filter output label is not specified, it is set to "out" by
   * default.
   */
  inputs->name = av_strdup("out");
  inputs->filter_ctx = buffersink_ctx;
  inputs->pad_idx = 0;
  inputs->next = NULL;

  AVFilterInOut* filter_outputs[2];
  filter_outputs[0] = outputs1;
#if (ENABLE_FILTERS)
  filter_outputs[1] = outputs2;
#endif

  if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr, &inputs,
                                      &outputs1, NULL)) < 0)  // filter_outputs
  {
    av_log(NULL, AV_LOG_ERROR, "parse ptr fail, ret: %d\n", ret);
    goto end;
  }

  if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0) {
    av_log(NULL, AV_LOG_ERROR, "config graph fail, ret: %d\n", ret);
    goto end;
  }

  /* Print summary of the sink buffer
   * Note: args buffer is reused to store channel layout string */
  outlink = buffersink_ctx->inputs[0];
  av_get_channel_layout_string(args1, sizeof(args1), -1,
                               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((enum AVSampleFormat)outlink->format), "?"),
         args1);

end:
  avfilter_inout_free(&inputs);
  avfilter_inout_free(&outputs1);

  return ret;
}

int main(int argc, char** argv) {
  int ret;
  AVFrame* frame = av_frame_alloc();
  AVFrame* filt_frame = av_frame_alloc();
  int got_frame;

  if (!frame || !filt_frame) {
    perror("Could not allocate frame");
    exit(1);
  }
  /*
  if (argc != 2) {
      fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
      exit(1);
  }
  */


  if ((ret = open_input_file_1("./recv.aac")) < 0) { //打开第一个输入文件
    av_log(NULL, AV_LOG_ERROR, "open input file fail, ret: %d\n", ret);
    goto end;
  }
  if ((ret = open_input_file_2("./trip.aac")) < 0) { //打开第二个输入文件
    av_log(NULL, AV_LOG_ERROR, "open input file fail, ret: %d\n", ret);
    goto end;
  }
  if ((ret = init_filters(filter_descr)) < 0) {      //初始化filter
    av_log(NULL, AV_LOG_ERROR, "init filters fail, ret: %d\n", ret);
    goto end;
  }

  if ((ret = open_output_file("./AAC/mix.aac")) < 0) return ret; //打开输出文件

  AVPacket packet0, packet;
  AVPacket _packet0, _packet;

  /* read all packets */
  packet0.data = NULL;
  packet.data = NULL;

  _packet0.data = NULL;
  _packet.data = NULL;
  while (1) {
    if (!packet0.data) {
      if ((ret = av_read_frame(fmt_ctx1, &packet)) < 0) break;
      packet0 = packet;
    }

    if (packet.stream_index == audio_stream_index_1) {
      got_frame = 0;
      ret = avcodec_decode_audio4(dec_ctx1, frame, &got_frame, &packet);
      if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
        continue;
      }
      packet.size -= ret;
      packet.data += ret;

      if (got_frame) {
        /* push the audio data from decoded frame into the filtergraph */
        if (av_buffersrc_add_frame_flags(buffersrc_ctx1, frame, 0) < 0) {
          av_log(NULL, AV_LOG_ERROR,
                 "Error while feeding the audio filtergraph\n");
          break;
        }
      }

      if (packet.size <= 0) av_packet_unref(&packet0);
    } else {
      /* discard non-wanted packets */
      av_packet_unref(&packet0);
    }

    if (!_packet0.data) {
      if ((ret = av_read_frame(fmt_ctx2, &_packet)) < 0) break;
      _packet0 = _packet;
    }

    if (_packet.stream_index == audio_stream_index_2) {
      got_frame = 0;
      ret = avcodec_decode_audio4(dec_ctx2, frame, &got_frame, &_packet);
      if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
        continue;
      }
      _packet.size -= ret;
      _packet.data += ret;

      if (got_frame) {
        /* push the audio data from decoded frame into the filtergraph */
        if (av_buffersrc_add_frame_flags(buffersrc_ctx2, frame, 0) < 0) {
          av_log(NULL, AV_LOG_ERROR,
                 "Error while feeding the audio filtergraph\n");
          break;
        }
      }

      if (_packet.size <= 0) av_packet_unref(&_packet0);
    } else {
      /* discard non-wanted packets */
      av_packet_unref(&_packet0);
    }
    /* pull filtered audio from the filtergraph */
    if (got_frame) {
      while (1) {
        ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) break;
        if (ret < 0) {
          av_log(NULL, AV_LOG_ERROR, "buffersink get frame fail, ret: %d\n",
                 ret);
          goto end;
        }
        encode_write_frame(filt_frame, 0);
        av_frame_unref(filt_frame);
      }
    }
  }
end:
  avfilter_graph_free(&filter_graph);
  avcodec_close(dec_ctx1);
  avformat_close_input(&fmt_ctx1);
  avcodec_close(dec_ctx2);
  avformat_close_input(&fmt_ctx2);
  av_frame_free(&frame);
  av_frame_free(&filt_frame);

  if (ret < 0 && ret != AVERROR_EOF) {
    fprintf(stderr, "Error occurred: %s\n",ret);
    exit(1);
  }

  av_write_trailer(ofmt_ctx);
  if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
    avio_closep(&ofmt_ctx->pb);
  avformat_free_context(ofmt_ctx);
  return 0;
}

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一、实现思路 ffmpeg的滤波filter有amix这个混音器，可以借助它来做音频的多路混音。首先我们需要编译ffmpeg并安装它，可以具备编码功能，考虑可以加mp3或者aac编码库进来，最简单的方式是编码成pcm格式直接输出到文件，用VLC也可以播放。 二、编译ffmpeg $: git clone https://git.ffmpeg.org/ffmpeg.git $: cd ffmpeg ...
ffmpeg 音频混流--命令与代码说明
洛的博客
05-28 3827
  ffmpeg可使用filter实现各种复杂的操作，混音只是其中一小部分功能。代码实现因为参考已经说的较为详细，我这里补充一些我认为重要的。 1、命令实现 ffmpeg -i dragen.opus -i Monsters.opus -filter_complex "[0:a]aresample=8000[a0]; [1:a]adelay=delays=6400|6400,aresample=8000[a1];[a0][a1]amix=inputs=2" -b:a 20k -y mixer.aac -v
electron+ffmpeg+vue3 录屏推流 源代码 真实有效
05-23
标题中的“electron+ffmpeg+vue3 录屏推流 源代码 ...项目的源代码结构和实现细节可能包括如何集成Electron、Vue3与FFmpeg，如何处理录制和推流的逻辑，以及如何设计用户界面等方面，这些都是进一步学习和研究的重点。
基于FFMPEG设计的流媒体播放器(完整源码)
02-02
这是 基于FFMPEG设计的流媒体播放器(完整源码)，FFMPEG版本是FFMEPG4.2.2。 将工程下载下来之后，要解压到英文目录下打开项目。 （这是最新优化的Demo代码--2023） 这份播放器项目使用场景： 摄像头监控。 设计这份...
C# 利用ffmpeg进行推流
04-12
C# 利用ffmpeg进行推流，推流方式涉及：读取本地文件，以入读取指定内存数据；示例项目中使用开源项目 FFmpeg.AutoGen，其对ffmpeg的API做了封装，方便在C#代码中进行ffmpeg API的调用。
FFmpegDemo_C#推流_c++推流服务器_ffmpeg推流_ffmpeg_推流
09-11
这个FFmpegDemo项目很可能是包含示例代码和配置文件的压缩包，可能包括C++和C#两种语言的推流示例。通过查看和运行这些代码，开发者可以更好地理解如何在实际项目中使用FFmpeg进行推流操作。 总结来说，FFmpegDemo...
基于FFmpeg源码实现一个简单的混音demo，把文件a和文件b的音频混为一个音频流输出并存为文件
Fuguir™的博客
08-24 512
/* *一个简单的混音demo，把文件a和文件b的音频混为一个音频流输出并存为文件，只处理每个文件的第一个音频流 */ 命令行实现： ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT inputs The number of inputs. If unspecified, it defaults to 2.//输入...
【音视频】音频混流-avfilter（六）
hcmonk的博客
04-05 471
要实现两股音频混流功能，需要用到ffmpeg的filter库，它可以自定义过滤器并实现两股甚至多股音频的融合。 技术简介 使用ffmpeg的filter功能实现 使用模块（库） 使用ffmpeg的avfilter-7.dll 主要流程和代码 1、初始化音频混合器 int AudioMixer::init(const AUDIO_FILTER_CTX& outCtx, const AUDIO_FILTER_CTX* inCtx0, const AUDIO_FILTER_CTX* inCtx1) {
使用ffmpeg AVfilter 中的amix实现混音
好记性不如写博客！
08-03 9321
官方文档： 6.16 amix Mixes multiple audio inputs into a single output. Note that this filter only supports float samples (the amerge and pan audio filters support many formats). If the amix input has in
# ffmpeg 工具使用
tony365的博客
07-17 742
windows安装教程：https://blog.csdn.net/csdn_yudong/article/details/129182648。图片的文件名为"in000000.jpg"，从0开始依次递增。img%04d.jpg: 生成的图片命名格式。-f : 指定格式化的格式为image2。第二种方法生成的图片大小为260kb左右。-b 200k 指定码率为200k。fps=1/5: 每1s取5帧。%5d：以为5位数按正序编号。fps=5: 每5s取一帧。-i : 指定输入文件。-r : 帧数 10。
免费视频批量横转竖
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weixin_53505226的博客
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视频处理器 v1.3 是一款由是貔貅呀开发的视频编辑和处理工具，提供高效便捷的视频批量横转竖，
ffmpeg混音代码
09-06
要使用FFmpeg进行音频混音，可以使用以下代码： ```bash ffmpeg -i input1.mp3 -i input2.mp3 -filter_complex amix=inputs=2:duration=longest output.mp3 ``` 这个命令会将两个输入音频文件（`input1.mp3`和`input2.mp3`）进行混音，并将结果保存为`output.mp3`文件。`amix`过滤器用于实现音频混音，`inputs=2`表示有两个输入流，`duration=longest`表示输出音频的长度与最长的输入流的长度相同。 请注意，你需要将输入文件名替换为你自己的文件名，并确保FFmpeg已正确安装并在系统路径中可用。

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