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", 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;
}
转码后的音频文件与输入的持续时间相同。但它完全嘈杂。有人能告诉我这里我做错了什么！
答案 0 :(得分：14)
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。 aac编码器的大小为1024.这可以通过在执行enc_ctx->frame_size后观察avcodec_open2()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。所以需要的改动是
`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，并相应地设置n的值。