ffmpeg SDK与示例程序

/* * Libavformat API example: Output a media file in any supported * libavformat format. The default codecs are used. * * Copyright (c) 2003 Fabrice Bellard * * Permission is hereby granted, free of charge, to any person obtaining a copy * of this software and associated documentation files (the "Software"), to deal * in the Software without restriction, including without limitation the rights * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell * copies of the Software, and to permit persons to whom the Software is * furnished to do so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in * all copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN * THE SOFTWARE. */ #include <stdlib.h> #include <stdio.h> #include <string.h> #include <math.h> #ifndef M_PI #define M_PI 3.14159265358979323846 #endif #include "libavformat/avformat.h" #include "libswscale/swscale.h" #undef exit /* 5 seconds stream duration */ #define STREAM_DURATION 5.0 #define STREAM_FRAME_RATE 25 /* 25 images/s */ #define STREAM_NB_FRAMES ((int)(STREAM_DURATION * STREAM_FRAME_RATE)) #define STREAM_PIX_FMT PIX_FMT_YUV420P /* default pix_fmt */ static int sws_flags = SWS_BICUBIC; /**************************************************************/ /* audio output */ float t, tincr, tincr2; int16_t *samples; uint8_t *audio_outbuf; int audio_outbuf_size; int audio_input_frame_size; /* * add an audio output stream */ static AVStream *add_audio_stream(AVFormatContext *oc, int codec_id) { AVCodecContext *c; AVStream *st; st = av_new_stream(oc, 1); if (!st) { fprintf(stderr, "Could not alloc stream\n"); exit(1); } c = st->codec; c->codec_id = codec_id; c->codec_type = CODEC_TYPE_AUDIO; /* put sample parameters */ c->bit_rate = 64000; c->sample_rate = 44100; c->channels = 2; return st; } static void open_audio(AVFormatContext *oc, AVStream *st) { AVCodecContext *c; AVCodec *codec; c = st->codec; /* find the audio encoder */ codec = avcodec_find_encoder(c->codec_id); if (!codec) { fprintf(stderr, "codec not found\n"); exit(1); } /* open it */ if (avcodec_open(c, codec) < 0) { fprintf(stderr, "could not open codec\n"); 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; audio_outbuf_size = 10000; audio_outbuf = av_malloc(audio_outbuf_size); /* ugly hack for PCM codecs (will be removed ASAP with new PCM support to compute the input frame size in samples */ if (c->frame_size <= 1) { audio_input_frame_size = audio_outbuf_size / c->channels; switch(st->codec->codec_id) { case CODEC_ID_PCM_S16LE: case CODEC_ID_PCM_S16BE: case CODEC_ID_PCM_U16LE: case CODEC_ID_PCM_U16BE: audio_input_frame_size >>= 1; break; default: break; } } else { audio_input_frame_size = c->frame_size; } samples = av_malloc(audio_input_frame_size * 2 * c->channels); } /* 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; av_init_packet(&pkt); c = st->codec; get_audio_frame(samples, audio_input_frame_size, c->channels); pkt.size= avcodec_encode_audio(c, audio_outbuf, audio_outbuf_size, samples); if (c->coded_frame->pts != AV_NOPTS_VALUE) pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base); pkt.flags |= PKT_FLAG_KEY; pkt.stream_index= st->index; pkt.data= audio_outbuf; /* write the compressed frame in the media file */ if (av_interleaved_write_frame(oc, &pkt) != 0) { fprintf(stderr, "Error while writing audio frame\n"); exit(1); } } static void close_audio(AVFormatContext *oc, AVStream *st) { avcodec_close(st->codec); av_free(samples); av_free(audio_outbuf); } /**************************************************************/ /* video output */ AVFrame *picture, *tmp_picture; uint8_t *video_outbuf; int frame_count, video_outbuf_size; /* add a video output stream */ static AVStream *add_video_stream(AVFormatContext *oc, int codec_id) { AVCodecContext *c; AVStream *st; st = av_new_stream(oc, 0); if (!st) { fprintf(stderr, "Could not alloc stream\n"); exit(1); } c = st->codec; c->codec_id = codec_id; c->codec_type = CODEC_TYPE_VIDEO; /* put sample parameters */ c->bit_rate = 400000; /* resolution must be a multiple of two */ c->width = 352; c->height = 288; /* time base: this is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented. for fixed-fps content, timebase should be 1/framerate and timestamp increments should be identically 1. */ c->time_base.den = STREAM_FRAME_RATE; c->time_base.num = 1; c->gop_size = 12; /* emit one intra frame every twelve frames at most */ c->pix_fmt = STREAM_PIX_FMT; if (c->codec_id == CODEC_ID_MPEG2VIDEO) { /* just for testing, we also add B frames */ c->max_b_frames = 2; } if (c->codec_id == CODEC_ID_MPEG1VIDEO){ /* Needed to avoid using macroblocks in which some coeffs overflow. This does not happen with normal video, it just happens here as the motion of the chroma plane does not match the luma plane. */ c->mb_decision=2; } // some formats want stream headers to be separate if(oc->oformat->flags & AVFMT_GLOBALHEADER) c->flags |= CODEC_FLAG_GLOBAL_HEADER; return st; } static AVFrame *alloc_picture(int pix_fmt, int width, int height) { AVFrame *picture; uint8_t *picture_buf; int size; picture = avcodec_alloc_frame(); if (!picture) return NULL; size = avpicture_get_size(pix_fmt, width, height); picture_buf = av_malloc(size); if (!picture_buf) { av_free(picture); return NULL; } avpicture_fill((AVPicture *)picture, picture_buf, pix_fmt, width, height); return picture; } static void open_video(AVFormatContext *oc, AVStream *st) { AVCodec *codec; AVCodecContext *c; c = st->codec; /* find the video encoder */ codec = avcodec_find_encoder(c->codec_id); if (!codec) { fprintf(stderr, "codec not found\n"); exit(1); } /* open the codec */ if (avcodec_open(c, codec) < 0) { fprintf(stderr, "could not open codec\n"); exit(1); } video_outbuf = NULL; if (!(oc->oformat->flags & AVFMT_RAWPICTURE)) { /* allocate output buffer */ /* XXX: API change will be done */ /* buffers passed into lav* can be allocated any way you prefer, as long as they're aligned enough for the architecture, and they're fr