我知道ffmpeg命令行很简单，但是如何编程实现？我不擅长这个，这里有一些来自互联网的代码，用于将.mp4转换为.ts，我做了一些改变，但是音频流问题依旧：</p>
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include "libavformat/avformat.h"
#include "libavcodec/avcodec.h"
#include "libavutil/avutil.h"
#include "libavutil/rational.h"
#include "libavdevice/avdevice.h"
#include "libavutil/mathematics.h"
#include "libswscale/swscale.h"
static AVStream* add_output_stream(AVFormatContext* output_format_context, AVStream* input_stream)
{
AVCodecContext* input_codec_context = NULL;
AVCodecContext* output_codec_context = NULL;
AVStream* output_stream = NULL;
output_stream = av_new_stream(output_format_context, 0);
if (!output_stream)
{
printf("Call av_new_stream function failed\n");
return NULL;
}
input_codec_context = input_stream->codec;
output_codec_context = output_stream->codec;
output_codec_context->codec_id = input_codec_context->codec_id;
output_codec_context->codec_type = input_codec_context->codec_type;
output_codec_context->codec_tag = input_codec_context->codec_tag;
output_codec_context->bit_rate = input_codec_context->bit_rate;
output_codec_context->extradata = input_codec_context->extradata;
output_codec_context->extradata_size = input_codec_context->extradata_size;
if (av_q2d(input_codec_context->time_base) * input_codec_context->ticks_per_frame > av_q2d(input_stream->time_base) && av_q2d(input_stream->time_base) < 1.0 / 1000)
{
output_codec_context->time_base = input_codec_context->time_base;
output_codec_context->time_base.num *= input_codec_context->ticks_per_frame;
}
else
{
output_codec_context->time_base = input_stream->time_base;
}
switch (input_codec_context->codec_type)
{
case AVMEDIA_TYPE_AUDIO:
output_codec_context->channel_layout = input_codec_context->channel_layout;
output_codec_context->sample_rate = input_codec_context->sample_rate;
output_codec_context->channels = input_codec_context->channels;
output_codec_context->frame_size = input_codec_context->frame_size;
if ((input_codec_context->block_align == 1 && input_codec_context->codec_id == CODEC_ID_MP3) || input_codec_context->codec_id == CODEC_ID_AC3)
{
output_codec_context->block_align = 0;
}
else
{
output_codec_context->block_align = input_codec_context->block_align;
}
break;
case AVMEDIA_TYPE_VIDEO:
output_codec_context->pix_fmt = input_codec_context->pix_fmt;
output_codec_context->width = input_codec_context->width;
output_codec_context->height = input_codec_context->height;
output_codec_context->has_b_frames = input_codec_context->has_b_frames;
if (output_format_context->oformat->flags & AVFMT_GLOBALHEADER)
{
output_codec_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
}
break;
default:
break;
}
return output_stream;
}
//[[** from ffmpeg.c
static void write_frame(AVFormatContext *s, AVPacket *pkt, AVCodecContext *avctx, AVBitStreamFilterContext *bsfc){
int ret;
while(bsfc){
AVPacket new_pkt= *pkt;
int a= av_bitstream_filter_filter(bsfc, avctx, NULL,
&new_pkt.data, &new_pkt.size,
pkt->data, pkt->size,
pkt->flags & AV_PKT_FLAG_KEY);
if(a>0){
av_free_packet(pkt);
new_pkt.destruct= av_destruct_packet;
} else if(a<0){
fprintf(stderr, "%s failed for stream %d, codec %s\n",
bsfc->filter->name, pkt->stream_index,
avctx->codec ? avctx->codec->name : "copy");
//print_error("", a);
//if (exit_on_error)
// ffmpeg_exit(1);
}
*pkt= new_pkt;
bsfc= bsfc->next;
}
ret= av_interleaved_write_frame(s, pkt);
if(ret < 0){
//print_error("av_interleaved_write_frame()", ret);
fprintf(stderr, "av_interleaved_write_frame(%d)\n", ret);
exit(1);
}
}
//]]**
int main(int argc, char* argv[])
{
const char* input;
const char* output;
const char* output_prefix = NULL;
char* segment_duration_check = 0;
const char* index = NULL;
char* tmp_index = NULL;
const char* http_prefix = NULL;
long max_tsfiles = NULL;
double prev_segment_time = 0;
double segment_duration = 0;
AVInputFormat* ifmt = NULL;
AVOutputFormat* ofmt = NULL;
AVFormatContext* ic = NULL;
AVFormatContext* oc = NULL;
AVStream* video_st = NULL;
AVStream* audio_st = NULL;
AVCodec* codec = NULL;
AVDictionary* pAVDictionary = NULL;
long frame_count = 0;
if (argc != 3) {
fprintf(stderr, "Usage: %s inputfile outputfile\n", argv[0]);
exit(1);
}
input = argv[1];
output = argv[2];
av_register_all();
char szError[256] = {0};
int nRet = avformat_open_input(&ic, input, ifmt, &pAVDictionary);
if (nRet != 0)
{
av_strerror(nRet, szError, 256);
printf(szError);
printf("\n");
printf("Call avformat_open_input function failed!\n");
return 0;
}
if (av_find_stream_info(ic) < 0)
{
printf("Call av_find_stream_info function failed!\n");
return 0;
}
ofmt = av_guess_format("mpegts", NULL, NULL);
if (!ofmt)
{
printf("Call av_guess_format function failed!\n");
return 0;
}
oc = avformat_alloc_context();
if (!oc)
{
printf("Call av_guess_format function failed!\n");
return 0;
}
oc->oformat = ofmt;
int video_index = -1, audio_index = -1;
for (unsigned int i = 0; i < ic->nb_streams && (video_index < 0 || audio_index < 0); i++)
{
switch (ic->streams[i]->codec->codec_type)
{
case AVMEDIA_TYPE_VIDEO:
video_index = i;
ic->streams[i]->discard = AVDISCARD_NONE;
video_st = add_output_stream(oc, ic->streams[i]);
break;
case AVMEDIA_TYPE_AUDIO:
audio_index = i;
ic->streams[i]->discard = AVDISCARD_NONE;
audio_st = add_output_stream(oc, ic->streams[i]);
break;
default:
ic->streams[i]->discard = AVDISCARD_ALL;
break;
}
}
codec = avcodec_find_decoder(video_st->codec->codec_id);
if (codec == NULL)
{
printf("Call avcodec_find_decoder function failed!\n");
return 0;
}
if (avcodec_open(video_st->codec, codec) < 0)
{
printf("Call avcodec_open function failed !\n");
return 0;
}
if (avio_open(&oc->pb, output, AVIO_FLAG_WRITE) < 0)
{
return 0;
}
if (avformat_write_header(oc, &pAVDictionary))
{
printf("Call avformat_write_header function failed.\n");
return 0;
}
//[[++
AVBitStreamFilterContext *bsfc = av_bitstream_filter_init("h264_mp4toannexb");
//AVBitStreamFilterContext *absfc = av_bitstream_filter_init("aac_adtstoasc");
if (!bsfc) {
fprintf(stderr, "bsf init error!\n");
return -1;
}
//]]++
int decode_done = 0;
do
{
double segment_time = 0;
AVPacket packet;
decode_done = av_read_frame(ic, &packet);
if (decode_done < 0)
break;
if (av_dup_packet(&packet) < 0)
{
printf("Call av_dup_packet function failed\n");
av_free_packet(&packet);
break;
}
//[[**
if (packet.stream_index == audio_index) {
segment_time = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
nRet = av_interleaved_write_frame(oc, &packet);
} else if (packet.stream_index == video_index) {
if (packet.flags & AV_PKT_FLAG_KEY) {
segment_time = (double)video_st->pts.val * video_st->time_base.num / video_st->time_base.den;
} else {
segment_time = prev_segment_time;
}
//nRet = av_interleaved_write_frame(oc, &packet);
write_frame(oc, &packet, video_st->codec, bsfc);
}
//]]**
if (nRet < 0)
{
printf("Call av_interleaved_write_frame function failed: %d\n", nRet);
}
else if (nRet > 0)
{
printf("End of stream requested\n");
av_free_packet(&packet);
break;
}
av_free_packet(&packet);
frame_count++;
}while(!decode_done);
av_write_trailer(oc);
printf("frame_count = %d\n", frame_count);
av_bitstream_filter_close(bsfc);
avcodec_close(video_st->codec);
for(unsigned int k = 0; k < oc->nb_streams; k++)
{
av_freep(&oc->streams[k]->codec);
av_freep(&oc->streams[k]);
}
av_free(oc);
//getchar();
return 0;
}
编译这段代码，得到一个名为 的可执行文件muxts，然后：
$ ./muxts vid1.mp4 vid1.ts
没有打印错误信息，但是音频流不同步并且有噪音。使用ffmpeg检查.ts文件：
$ ffmpeg -i vid1.ts
ffmpeg version 0.8.14-tessus, Copyright (c) 2000-2013 the FFmpeg developers
built on Jul 29 2013 17:05:18 with llvm_gcc 4.2.1 (Based on Apple Inc. build 5658) (LLVM build 2336.1.00)
configuration: --prefix=/usr/local --arch=x86_64 --as=yasm --extra-version=tessus --enable-gpl --enable-nonfree --enable-version3 --disable-ffplay --enable-libvorbis --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-bzlib --enable-zlib --enable-postproc --enable-filters --enable-runtime-cpudetect --enable-debug=3 --disable-optimizations
libavutil 51. 9. 1 / 51. 9. 1
libavcodec 53. 8. 0 / 53. 8. 0
libavformat 53. 5. 0 / 53. 5. 0
libavdevice 53. 1. 1 / 53. 1. 1
libavfilter 2. 23. 0 / 2. 23. 0
libswscale 2. 0. 0 / 2. 0. 0
libpostproc 51. 2. 0 / 51. 2. 0
Seems stream 0 codec frame rate differs from container frame rate: 180000.00 (180000/1) -> 90000.00 (180000/2)
Input #0, mpegts, from 'vid1.ts':
Duration: 00:00:03.75, start: 0.000000, bitrate: 3656 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0.0[0x100]: Video: h264 (Baseline), yuv420p, 640x480, 90k tbr, 90k tbn, 180k tbc
Stream #0.1[0x101]: Audio: aac, 48000 Hz, mono, s16, 190 kb/s
At least one output file must be specified
我应该怎么办？
如果此问题已解决，我如何将多个 .ts 文件合并为单个 .mp4 文件？