使用ffmpeg实现转码样例(代码实现)

使用ffmpeg实现转码样例(代码实现)

使用ffmpeg转码主要工作如下:

Demux -> Decoding -> Encoding -> Muxing

其中接口调用如下：

av_register_all();
avformat_open_input
avformat_find_stream_info
open_codec_context
av_image_alloc
avcodec_alloc_frame
avformat_alloc_output_context2
avcodec_open2
avcodec_alloc_frame
avpicture_alloc
avpicture_alloc
avformat_write_header
av_init_packet
av_read_frame
avcodec_decode_video2
av_image_copy
avcodec_encode_video2
av_interleaved_write_frame
av_write_trailer

下面的代码为主要将视频转码，封装为h264编码格式的mp4文件，音频为mp3，但是主要操作并不处理音频文件。代码如下

#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>

#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>

/* 5 seconds stream duration */
#define STREAM_DURATION   200.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_NB_FRAMES ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
#define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */

static AVFormatContext *input_fmt_ctx = NULL;
static int sws_flags = SWS_BICUBIC;
static AVPacket input_pkt;
static int video_stream_idx = -1;
static uint8_t *video_dst_data[4] = {NULL};
static int video_dst_linesize[4];
static AVStream *input_video_stream;
static AVFrame *input_frame;
static AVCodecContext *video_dec_ctx = NULL;
static int video_dst_bufsize;


static int open_codec_context(int *stream_idx,
 AVFormatContext *fmt_ctx, enum AVMediaType type)
{
 int ret;
 AVStream *st;
 AVCodecContext *dec_ctx = NULL;
 AVCodec *dec = NULL;

 ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
 if (ret < 0) {
 return ret;
 } else {
 *stream_idx = ret;
 st = fmt_ctx->streams[*stream_idx];

 /* find decoder for the stream */
 dec_ctx = st->codec;
 dec = avcodec_find_decoder(dec_ctx->codec_id);
 if (!dec) {
 fprintf(stderr, "Failed to find %s codec\n",
 av_get_media_type_string(type));
 return ret;
 }

 if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
 fprintf(stderr, "Failed to open %s codec\n",
 av_get_media_type_string(type));
 return ret;
 }
 }

 return 0;
}

/* Add an output stream. */
static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
 enum AVCodecID codec_id)
{
 AVCodecContext *c;
 AVStream *st;

 /* find the encoder */
 *codec = avcodec_find_encoder(codec_id);
 if (!(*codec)) {
 fprintf(stderr, "Could not find encoder for '%s'\n",
 avcodec_get_name(codec_id));
 exit(1);
 }

 st = avformat_new_stream(oc, *codec);
 if (!st) {
 fprintf(stderr, "Could not allocate stream\n");
 exit(1);
 }
 st->id = oc->nb_streams-1;
 c = st->codec;

 switch ((*codec)->type) {
 case AVMEDIA_TYPE_AUDIO:
 c->sample_fmt = AV_SAMPLE_FMT_FLTP;
 c->bit_rate = 64000;
 c->sample_rate = 44100;
 c->channels = 2;
 break;

 case AVMEDIA_TYPE_VIDEO:
 c->codec_id = codec_id;

 c->bit_rate = 400000;
 c->time_base.den = STREAM_FRAME_RATE;
 c->time_base.num = 1;
 c->gop_size = 12;
 c->pix_fmt = STREAM_PIX_FMT;
 break;

 default:
 break;
 }

 /* Some formats want stream headers to be separate. */
 if (oc->oformat->flags & AVFMT_GLOBALHEADER)
 c->flags |= CODEC_FLAG_GLOBAL_HEADER;

 return st;
}

/**************************************************************/
/* audio output */

static float t, tincr, tincr2;

static uint8_t **src_samples_data;
static int src_samples_linesize;
static int src_nb_samples;

static int max_dst_nb_samples;
uint8_t **dst_samples_data;
int dst_samples_linesize;
int dst_samples_size;

struct SwrContext *swr_ctx = NULL;

static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
 AVCodecContext *c;
 int ret;

 c = st->codec;

 /* open it */
 ret = avcodec_open2(c, codec, NULL);
 if (ret < 0) {
 fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
 exit(1);
 }

 /* init signal generator */
 t = 0;
 tincr = 2 * M_PI * 110.0 / c->sample_rate;
 /* increment frequency by 110 Hz per second */
 tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;

 src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
 10000 : c->frame_size;

 ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
 src_nb_samples, c->sample_fmt, 0);
 if (ret < 0) {
 fprintf(stderr, "Could not allocate source samples\n");
 exit(1);
 }

 /* create resampler context */
 if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
 swr_ctx = swr_alloc();
 if (!swr_ctx) {
 fprintf(stderr, "Could not allocate resampler context\n");
 exit(1);
 }

 /* set options */
 av_opt_set_int (swr_ctx, "in_channel_count", c->channels, 0);
 av_opt_set_int (swr_ctx, "in_sample_rate", c->sample_rate, 0);
 av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
 av_opt_set_int (swr_ctx, "out_channel_count", c->channels, 0);
 av_opt_set_int (swr_ctx, "out_sample_rate", c->sample_rate, 0);
 av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", c->sample_fmt, 0);

 /* initialize the resampling context */
 if ((ret = swr_init(swr_ctx)) < 0) {
 fprintf(stderr, "Failed to initialize the resampling context\n");
 exit(1);
 }
 }

 /* compute the number of converted samples: buffering is avoided
  * ensuring that the output buffer will contain at least all the
  * converted input samples */
 max_dst_nb_samples = src_nb_samples;
 ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
 max_dst_nb_samples, c->sample_fmt, 0);
 if (ret < 0) {
 fprintf(stderr, "Could not allocate destination samples\n");
 exit(1);
 }
 dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
 c->sample_fmt, 0);
}

/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
 * 'nb_channels' channels. */
static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
{
 int j, i, v;
 int16_t *q;

 q = samples;
 for (j = 0; j < frame_size; j++) {
 v = (int)(sin(t) * 10000);
 for (i = 0; i < nb_channels; i++)
 *q++ = v;
 t += tincr;
 tincr += tincr2;
 }
}

static void write_audio_frame(AVFormatContext *oc, AVStream *st)
{
 AVCodecContext *c;
 AVPacket pkt = { 0 }; // data and size must be 0;
 AVFrame *frame = avcodec_alloc_frame();
 int got_packet, ret, dst_nb_samples;

 av_init_packet(&pkt);
 c = st->codec;

 get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);

 /* convert samples from native format to destination codec format, using the resampler */
 if (swr_ctx) {
 /* compute destination number of samples */
 dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
 c->sample_rate, c->sample_rate, AV_ROUND_UP);
 if (dst_nb_samples > max_dst_nb_samples) {
 av_free(dst_samples_data[0]);
 ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
 dst_nb_samples, c->sample_fmt, 0);
 if (ret < 0)
 exit(1);
 max_dst_nb_samples = dst_nb_samples;
 dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
 c->sample_fmt, 0);
 }

 /* convert to destination format */
 ret = swr_convert(swr_ctx,
 dst_samples_data, dst_nb_samples,
 (const uint8_t **)src_samples_data, src_nb_samples);
 if (ret < 0) {
 fprintf(stderr, "Error while converting\n");
 exit(1);
 }
 } else {
 dst_samples_data[0] = src_samples_data[0];
 dst_nb_samples = src_nb_samples;
 }

 frame->nb_samples = dst_nb_samples;
 avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
 dst_samples_data[0], dst_samples_size, 0);

 ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
 if (ret < 0) {
 fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
 exit(1);
 }

 if (!got_packet)
 return;

 pkt.stream_index = st->index;

 /* Write the compressed frame to the media file. */
 ret = av_interleaved_write_frame(oc, &pkt);
 if (ret != 0) {
 fprintf(stderr, "Error while writing audio frame: %s\n",
 av_err2str(ret));
 exit(1);
 }
 avcodec_free_frame(&frame);
}

static void close_audio(AVFormatContext *oc, AVStream *st)
{
 avcodec_close(st->codec);
 av_free(src_samples_data[0]);
 av_free(dst_samples_data[0]);
}

/***********************bbs.ChinaFFmpeg.com****孙悟空***********************/
/* video output */

static AVFrame *frame;
static AVPicture src_picture, dst_picture;
static int frame_count;

static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
 int ret;
 AVCodecContext *c = st->codec;

 /* open the codec */
 ret = avcodec_open2(c, codec, NULL);
 if (ret < 0) {
 fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
 exit(1);
 }

 /* allocate and init a re-usable frame */
 frame = avcodec_alloc_frame();
 if (!frame) {
 fprintf(stderr, "Could not allocate video frame\n");
 exit(1);
 }

 /* Allocate the encoded raw picture. */
 ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
 if (ret < 0) {
 fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret));
 exit(1);
 }

 /* If the output format is not YUV420P, then a temporary YUV420P
  * picture is needed too. It is then converted to the required
  * output format. */
 if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
 ret = avpicture_alloc(&src_picture, AV_PIX_FMT_YUV420P, c->width, c->height);
 if (ret < 0) {
 fprintf(stderr, "Could not allocate temporary picture: %s\n",
 av_err2str(ret));
 exit(1);
 }
 }

 /* copy data and linesize picture pointers to frame */
 *((AVPicture *)frame) = dst_picture;
}

static void write_video_frame(AVFormatContext *oc, AVStream *st)
{
 int ret;
 static struct SwsContext *sws_ctx;
 AVCodecContext *c = st->codec;

 // fill_yuv_image(&dst_picture, frame_count, c->width, c->height);


 if (oc->oformat->flags & AVFMT_RAWPICTURE) {
 /* Raw video case - directly store the picture in the packet */
 AVPacket pkt;
 av_init_packet(&pkt);

 pkt.flags |= AV_PKT_FLAG_KEY;
 pkt.stream_index = st->index;
 pkt.data = dst_picture.data[0];
 pkt.size = sizeof(AVPicture);

 ret = av_interleaved_write_frame(oc, &pkt);
 } else {
 AVPacket pkt = { 0 };
 int got_packet;
 av_init_packet(&pkt);

 /* encode the image */
 ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
 if (ret < 0) {
 fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
 exit(1);
 }
 /* If size is zero, it means the image was buffered. */

 if (!ret && got_packet && pkt.size) {
 pkt.stream_index = st->index;

 /* Write the compressed frame to the media file. */
 ret = av_interleaved_write_frame(oc, &pkt);
 } else {
 ret = 0;
 }
 }
 if (ret != 0) {
 fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
 exit(1);
 }
 frame_count++;
}

static void close_video(AVFormatContext *oc, AVStream *st)
{
 avcodec_close(st->codec);
 av_free(src_picture.data[0]);
 av_free(dst_picture.data[0]);
 av_free(frame);
}

/**************************************************************/
/* media file output */

int main(int argc, char **argv)
{
 const char *filename;
 const char *input_file;
 AVOutputFormat *fmt;
 AVFormatContext *oc;
 AVStream *audio_st, *video_st;
 AVCodec *audio_codec, *video_codec;
 double audio_time, video_time;
 int ret;
 int decoded;
 int got_frame;

 /* Initialize libavcodec, and register all codecs and formats. */
 av_register_all();

 if (argc != 3) {
 printf("usage: %s output_file inputfile\n"
 "API example program to output a media file with libavformat.\n"
 "This program generates a synthetic audio and video stream, encodes and\n"
 "muxes them into a file named output_file.\n"
 "The output format is automatically guessed according to the file extension.\n"
 "Raw images can also be output by using '%%d' in the filename.\n"
 "\n", argv[0]);
 return 1;
 }

 input_file = argv[2];
 filename = argv[1];

 if (avformat_open_input( &input_fmt_ctx, input_file, NULL, NULL) < 0) {
 fprintf(stderr, "Could not open source file %s\n", input_file);
 exit(-1);
 }

 /* retrieve stream information */
 if (avformat_find_stream_info(input_fmt_ctx, NULL) < 0) {
 fprintf(stderr, "Could not find stream information\n");
 exit(1);
 }

 if (open_codec_context(&video_stream_idx, input_fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
 input_video_stream = input_fmt_ctx->streams[video_stream_idx];
 video_dec_ctx = input_video_stream->codec;
 ret = av_image_alloc(video_dst_data, video_dst_linesize,
 video_dec_ctx->width, video_dec_ctx->height,
 video_dec_ctx->pix_fmt, 1);
 if (ret < 0) {
 fprintf(stderr, "Could not allocate raw video buffer\n");
 }
 video_dst_bufsize = ret;
 }


 input_frame = avcodec_alloc_frame();
 if (!input_frame) {
 fprintf(stderr, "Could not allocate frame\n");
 ret = AVERROR(ENOMEM);
 exit(-1);
 }

 /* allocate the output media context */
 avformat_alloc_output_context2(&oc, NULL, NULL, filename);
 if (!oc) {
 printf("Could not deduce output format from file extension: using MPEG.\n");
 avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
 }
 if (!oc) {
 return 1;
 }
 fmt = oc->oformat;

 video_st = NULL;
 audio_st = NULL;

 fmt->video_codec = AV_CODEC_ID_H264;
 if (fmt->video_codec != AV_CODEC_ID_NONE) {
 video_st = add_stream(oc, &video_codec, AV_CODEC_ID_H264);
 }
 if (video_st->codec) {
 video_st->codec->width = video_dec_ctx->width;
 video_st->codec->height = video_dec_ctx->height;
 }
 fmt->audio_codec = AV_CODEC_ID_MP3;
 if (fmt->audio_codec != AV_CODEC_ID_NONE) {
 audio_st = add_stream(oc, &audio_codec, AV_CODEC_ID_MP3);
 }

 /* Now that all the parameters are set, we can open the audio and
  * video codecs and allocate the necessary encode buffers. */
 if (video_st)
 open_video(oc, video_codec, video_st);
 if (audio_st)
 open_audio(oc, audio_codec, audio_st);

 av_dump_format(oc, 0, filename, 1);


 /* open the output file, if needed */
 if (!(fmt->flags & AVFMT_NOFILE)) {
 ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
 if (ret < 0) {
 fprintf(stderr, "Could not open '%s': %s\n", filename,
 av_err2str(ret));
 return 1;
 }
 }

 /* Write the stream header, if any. */
 ret = avformat_write_header(oc, NULL);
 if (ret < 0) {
 fprintf(stderr, "Error occurred when opening output file: %s\n",
 av_err2str(ret));
 return 1;
 }

 av_init_packet(&input_pkt);
 input_pkt.data = NULL;
 input_pkt.size = 0;


 if (frame)
 frame->pts = 0;
 for (;;) {
 /* Compute current audio and video time. */
 audio_time = audio_st ? audio_st->pts.val * av_q2d(audio_st->time_base) : 0.0;
 video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 0.0;

 if ((!audio_st || audio_time >= STREAM_DURATION) &&
 (!video_st || video_time >= STREAM_DURATION))
 break;

 /* write interleaved audio and video frames */
 if (!video_st || (video_st && audio_st && audio_time < video_time)) {
 write_audio_frame(oc, audio_st);
 } else {
 av_read_frame(input_fmt_ctx, &input_pkt);

 decoded = input_pkt.size;
 if (input_pkt.stream_index == video_stream_idx) {
 /* decode video frame */
 ret = avcodec_decode_video2(video_dec_ctx, input_frame, &got_frame, &input_pkt);
 if (ret < 0) {
 fprintf(stderr, "Error decoding video frame\n");
 return ret;
 }

 if (got_frame) {
 av_image_copy(&dst_picture, video_dst_linesize,
 (const uint8_t **)(input_frame->data), input_frame->linesize,
 video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
 }
 }

 write_video_frame(oc, video_st);
 frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
 }
 }

 av_write_trailer(oc);

 /* Close each codec. */
 if (video_st)
 close_video(oc, video_st);
 if (audio_st)
 close_audio(oc, audio_st);

 if (!(fmt->flags & AVFMT_NOFILE))
 avio_close(oc->pb);

 avformat_free_context(oc);
 av_free(input_frame);
 av_free(video_dst_data[0]);
 avcodec_close(video_dec_ctx);
 return 0;
}

以上代码为从dox/example/muxing.c中修改得到

编译命令如下：

gcc -g doc/examples/muxing.c -o muxing -lavcodec -lavdevice -lavfilter -lavformat -lavutil -lswscale -lswresample -lpostproc -lx264 -lmp3lame -lz -liconv -lbz2

执行命令如下：

./muxing output.mp4 ~/Movies/input.avi

执行后效果如下：

查看转码完成后的多媒体文件的信息：

查看转码后的文件的视频: