一、ffmpeg版本说明
ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-39) configuration: --disable-debug --enable-libx264 --enable-libx265 --enable-gpl --enable-shared --prefix=/usr/local/ffmpeg libavutil 56. 51.100 / 56. 51.100 libavcodec 58. 91.100 / 58. 91.100 libavformat 58. 45.100 / 58. 45.100 libavdevice 58. 10.100 / 58. 10.100 libavfilter 7. 85.100 / 7. 85.100 libswscale 5. 7.100 / 5. 7.100 libswresample 3. 7.100 / 3. 7.100 libpostproc 55. 7.100 / 55. 7.100 Hyper fast Audio and Video encoder usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}...
二、代码使用说明
输入:3.mp3,4.mp3 或者3.wav,4.wav
输出:temp.pcm
本代码是在官方文档中的例子中filtering_audio.c修改而成。
三、混音代码
/** * @file * API example for audio decoding and filtering * @example filtering_audio.c */ #include <unistd.h> #include <libavcodec/avcodec.h> #include <libavformat/avformat.h> #include <libavfilter/buffersink.h> #include <libavfilter/buffersrc.h> #include <libavutil/opt.h> #define ENABLE_FILTERS 1 static const char *filter_descr = "[in0][in1]amix=inputs=2[out]"; static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -"; 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; 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 "); return ret; } if ((ret = avformat_find_stream_info(fmt_ctx1, NULL)) < 0) { av_log(NULL, AV_LOG_ERROR, "Cannot find stream information "); 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 "); return ret; } audio_stream_index_1 = ret; /* create decoding context */ dec_ctx1 = avcodec_alloc_context3(dec); if (!dec_ctx1) return AVERROR(ENOMEM); avcodec_parameters_to_context(dec_ctx1, fmt_ctx1->streams[audio_stream_index_1]->codecpar); /* init the audio decoder */ if ((ret = avcodec_open2(dec_ctx1, dec, NULL)) < 0) { av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder "); 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 "); return ret; } if ((ret = avformat_find_stream_info(fmt_ctx2, NULL)) < 0) { av_log(NULL, AV_LOG_ERROR, "Cannot find stream information "); 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 "); return ret; } audio_stream_index_2 = ret; /* create decoding context */ dec_ctx2 = avcodec_alloc_context3(dec); if (!dec_ctx2) return AVERROR(ENOMEM); avcodec_parameters_to_context(dec_ctx2, fmt_ctx2->streams[audio_stream_index_2]->codecpar); /* init the audio decoder */ if ((ret = avcodec_open2(dec_ctx2, dec, NULL)) < 0) { av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder "); 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_S16, -1 }; static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 }; static const int out_sample_rates[] = { 8000, -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 "); 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 "); 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 "); 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 "); 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 "); 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 "); 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) goto end; if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0) 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 ", (int)outlink->sample_rate, (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"), args1); end: avfilter_inout_free(&inputs); avfilter_inout_free(&outputs1); return ret; } static void print_frame(const AVFrame *frame) #if 1 { FILE *file = NULL; const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout); const uint16_t *p = (uint16_t*)frame->data[0]; const uint16_t *p_end = p + n; file = fopen("tmp.pcm", "ab+"); if (NULL == file){ perror("fopen tmp.mp3 error "); return; } fwrite(frame->data[0], n * 2, 1, file); fclose(file); file = NULL; } #else { const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout); const uint16_t *p = (uint16_t*)frame->data[0]; const uint16_t *p_end = p + n; while (p < p_end) { fputc(*p & 0xff, stdout); fputc(*p>>8 & 0xff, stdout); p++; } fflush(stdout); } #endif int main(int argc, char **argv) { int ret; AVPacket packet1; AVPacket packet2; AVFrame *frame = av_frame_alloc(); AVFrame *filt_frame = av_frame_alloc(); if (!frame || !filt_frame) { perror("Could not allocate frame"); exit(1); } if ((ret = open_input_file_1("3_0.1.wav")) < 0) goto end; if ((ret = open_input_file_2("4.wav")) < 0) goto end; if ((ret = init_filters(filter_descr)) < 0) goto end; /* read all packets */ while (1) { if ((ret = av_read_frame(fmt_ctx1, &packet1)) < 0) break; if (packet1.stream_index == audio_stream_index_1) { ret = avcodec_send_packet(dec_ctx1, &packet1); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder "); break; } while (ret >= 0) { ret = avcodec_receive_frame(dec_ctx1, frame); if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) { break; } else if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder "); goto end; } if (ret >= 0) { /* push the audio data from decoded frame into the filtergraph */ if (av_buffersrc_add_frame_flags(buffersrc_ctx1, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) { av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph "); break; } /* pull filtered audio from the filtergraph */ while (1) { ret = av_buffersink_get_frame(buffersink_ctx, filt_frame); if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) break; if (ret < 0) goto end; print_frame(filt_frame); av_frame_unref(filt_frame); } av_frame_unref(frame); } } } if ((ret = av_read_frame(fmt_ctx2, &packet2)) < 0) break; if (packet2.stream_index == audio_stream_index_2) { ret = avcodec_send_packet(dec_ctx2, &packet2); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder "); break; } while (ret >= 0) { ret = avcodec_receive_frame(dec_ctx2, frame); if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) { break; }else if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder "); goto end; } if (ret >= 0) { if (av_buffersrc_add_frame_flags(buffersrc_ctx2, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) { av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph "); break; } while (1) { ret = av_buffersink_get_frame(buffersink_ctx, filt_frame); if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) break; if (ret < 0) goto end; print_frame(filt_frame); av_frame_unref(filt_frame); } av_frame_unref(frame); } } } av_packet_unref(&packet1); av_packet_unref(&packet2); } end: avfilter_graph_free(&filter_graph); avcodec_free_context(&dec_ctx1); avcodec_free_context(&dec_ctx2); avformat_close_input(&fmt_ctx2); avformat_close_input(&fmt_ctx1); av_frame_free(&frame); av_frame_free(&filt_frame); if (ret < 0 && ret != AVERROR_EOF) { fprintf(stderr, "Error occurred: %s ", av_err2str(ret)); exit(1); } exit(0); }