I am writing an application for transcoding sound using ffmpeg libraries. Here is my code
#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", NULL, NULL, filter_graph); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "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, AV_LOG_ERROR, "Cannot set output sample format\n"); return ret; } 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"); return ret; } 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"); 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, sizeof(args), -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((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, 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, inFileName.c_str(), 0); printf("\n ---------------------------------------------------------------------- \n"); /* Choose a audio stream */ select_stream = av_find_best_stream(in_fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &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, AV_LOG_ERROR, "Cannot open audio decoder\n"); return ret; } /* allocate output context */ ret = avformat_alloc_output_context2(&out_fmt_ctx, NULL, NULL, 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, pFrame, 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, buffersink_ctx -> inputs[0] -> time_base, 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, &encodedPacket, NULL, &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, buffersink_ctx -> inputs[0] -> time_base, 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; }
An audio file after transcoding the same length as the input. But its completely noisy. Can someone tell me what I'm doing wrong here?