FFMPEG发送RTSP编码流C ++ [英] FFMPEG to send RTSP encoded stream C++
本文介绍了FFMPEG发送RTSP编码流C ++的处理方法，对大家解决问题具有一定的参考价值，需要的朋友们下面随着小编来一起学习吧！
问题描述
我试图找出一种方法来获取我在FFMEG中创建的编码的H264图像，并使用FFMPEG通过RTSP发送它，是否有一些示例代码或教程，显示如何做到这一点。我试过在网上搜索，但没有什么，我可以找到那里。
I am trying to figure out a way to take an encoded H264 image that I have created in FFMEG and send it out via RTSP using FFMPEG, is there some sample code or a tutorial out there that shows how to do this. I tried searching the web, but there is nothing that I could find out there.
任何帮助是非常感激。提前感谢。
Any help is greatly appreciated. Thanks in advance.
推荐答案
我在docs / examples / muxing.c中做了一些更改，这里是我的代码，传输音频和视频使用mpeg4 rtsp的流
I made few changes in docs/examples/muxing.c here is my code which transmits audio and video streams using mpeg4 rtsp
#include <stdio.h>
#include <vector>
#include <windows.h>
#include <iostream>
#include <fstream>
#include <time.h>
#define _XOPEN_SOURCE 600 /* for usleep */
extern "C"
{
#ifndef __STDC_CONSTANT_MACROS
#define __STDC_CONSTANT_MACROS
#endif
#include <libavcodec\avcodec.h>
#include <libavutil/imgutils.h>
#include <libavutil/samplefmt.h>
//#include <libavutil/timestamp.h>
#include <libswscale\swscale.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
#include <libavutil/imgutils.h>
#include <libavutil/mathematics.h>
#include <libavutil/samplefmt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <math.h>
#include <libavutil/opt.h>
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/imgutils.h>
#include <libavutil/mathematics.h>
#include <libavutil/samplefmt.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
}
using namespace std;
#pragma comment(lib, "dev\\avcodec.lib")
#pragma comment(lib, "dev\\avformat.lib")
#pragma comment(lib, "dev\\avfilter.lib")
//#pragma comment(lib, "dev\\avcodec.lib")
#pragma comment(lib, "dev\\avdevice.lib")
//#pragma comment(lib, "dev\\avfilter.lib")
//#pragma comment(lib, "dev\\avformat.lib")
#pragma comment(lib, "dev\\avutil.lib")
#pragma comment(lib, "dev\\postproc.lib")
#pragma comment(lib, "dev\\swresample.lib")
#pragma comment(lib, "dev\\swscale.lib")
#ifdef _MSC_VER
//#define snprintf c99_snprintf
//inline int c99_vsnprintf(char* str, size_t size, const char* format, va_list ap)
//{
// int count = -1;
//
// if (size != 0)
// count = _vsnprintf_s(str, size, _TRUNCATE, format, ap);
// if (count == -1)
// count = _vscprintf(format, ap);
//
// return count;
//}
//inline int c99_snprintf(char* str, size_t size, const char* format, ...)
//{
// int count;
// va_list ap;
//
// va_start(ap, format);
// count = c99_vsnprintf(str, size, format, ap);
// va_end(ap);
//
// return count;
//}
#endif
static int audio_is_eof, video_is_eof;
#define STREAM_DURATION 50.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */
static int sws_flags = SWS_BICUBIC;
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
/*printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
av_ts_make_string(pkt->pts), av_ts_make_string(pkt->pts, time_base),
av_ts_make_string(pkt->dts), av_ts_make_string(pkt->dts, time_base),
av_ts_make_string(pkt->duration), av_ts_make_string(pkt->duration, time_base),
pkt->stream_index);*/
}
static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
{
/* rescale output packet timestamp values from codec to stream timebase */
pkt->pts = av_rescale_q_rnd(pkt->pts, *time_base, st->time_base, AVRounding(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
pkt->dts = av_rescale_q_rnd(pkt->dts, *time_base, st->time_base, AVRounding(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
pkt->duration = av_rescale_q(pkt->duration, *time_base, st->time_base);
pkt->stream_index = st->index;
/* Write the compressed frame to the media file. */
log_packet(fmt_ctx, pkt);
return av_interleaved_write_frame(fmt_ctx, pkt);
}
/* 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 = (*codec)->sample_fmts ?
(*codec)->sample_fmts[0] : 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;
if(codec_id == CODEC_ID_H264)
cout<<"Codec ID "<<(AVCodecID)codec_id<<endl;
c->bit_rate = 400000;
/* Resolution must be a multiple of two. */
c->width = 352;
c->height = 288;
/* timebase: 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
* identical to 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 == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_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;
}
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;
AVFrame *audio_frame;
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;
int samples_count;
struct SwrContext *swr_ctx = NULL;
static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
AVCodecContext *c;
int ret;
c = st->codec;
/* allocate and init a re-usable frame */
audio_frame = av_frame_alloc();
if (!audio_frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
/* open it */
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open audio 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;
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, AV_SAMPLE_FMT_S16, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\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;
/* 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);
}
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);
}
} else {
dst_samples_data = src_samples_data;
}
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, int flush)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
int got_packet, ret, dst_nb_samples;
av_init_packet(&pkt);
c = st->codec;
if (!flush) {
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_nb_samples = src_nb_samples;
}
AVRational t= {1, c->sample_rate};
audio_frame->nb_samples = dst_nb_samples;
audio_frame->pts = av_rescale_q(samples_count,t , c->time_base);
avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
dst_samples_data[0], dst_samples_size, 0);
samples_count += dst_nb_samples;
}
ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : audio_frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame:");
exit(1);
}
if (!got_packet) {
if (flush)
audio_is_eof = 1;
return;
}
ret = write_frame(oc, &c->time_base, st, &pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing audio frame: ");
exit(1);
}
}
static void close_audio(AVFormatContext *oc, AVStream *st)
{
avcodec_close(st->codec);
if (dst_samples_data != src_samples_data) {
av_free(dst_samples_data[0]);
av_free(dst_samples_data);
}
av_free(src_samples_data[0]);
av_free(src_samples_data);
av_frame_free(&audio_frame);
}
/**************************************************************/
/* 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: ");
exit(1);
}
/* allocate and init a re-usable frame */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame->format = c->pix_fmt;
frame->width = c->width;
frame->height = c->height;
/* 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: ");
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:");
exit(1);
}
}
/* copy data and linesize picture pointers to frame */
*((AVPicture *)frame) = dst_picture;
}
/* Prepare a dummy image. */
static void fill_yuv_image(AVPicture *pict, int frame_index,
int width, int height)
{
int x, y, i;
i = frame_index;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
}
}
}
static void write_video_frame(AVFormatContext *oc, AVStream *st, int flush)
{
int ret;
static struct SwsContext *sws_ctx;
AVCodecContext *c = st->codec;
if (!flush) {
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
/* as we only generate a YUV420P picture, we must convert it
* to the codec pixel format if needed */
if (!sws_ctx) {
sws_ctx = sws_getContext(c->width, c->height, AV_PIX_FMT_YUV420P,
c->width, c->height, c->pix_fmt,
sws_flags, NULL, NULL, NULL);
if (!sws_ctx) {
fprintf(stderr,
"Could not initialize the conversion context\n");
exit(1);
}
}
fill_yuv_image(&src_picture, frame_count, c->width, c->height);
sws_scale(sws_ctx,
(const uint8_t * const *)src_picture.data, src_picture.linesize,
0, c->height, dst_picture.data, dst_picture.linesize);
} else {
fill_yuv_image(&dst_picture, frame_count, c->width, c->height);
}
}
if (oc->oformat->flags & AVFMT_RAWPICTURE && !flush) {
/* 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 */
frame->pts = frame_count;
ret = avcodec_encode_video2(c, &pkt, flush ? NULL : frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding video frame:");
exit(1);
}
/* If size is zero, it means the image was buffered. */
if (got_packet) {
//cout<<"got Packet"<<endl;
ret = write_frame(oc, &c->time_base, st, &pkt);
} else {
//cout<<"EOF\n";
if (flush)
video_is_eof = 1;
ret = 0;
}
}
if (ret < 0) {
fprintf(stderr, "Error while writing video frame: ");
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_frame_free(&frame);
}
/**************************************************************/
/* media file output */
int main(int argc, char **argv)
{
const char *filename = "rtsp://127.0.0.1:8554/live.sdp";
AVOutputFormat *fmt;
AVFormatContext *oc;
AVStream *audio_st, *video_st;
AVCodec *audio_codec, *video_codec;
double audio_time, video_time;
int flush, ret;
/* Initialize libavcodec, and register all codecs and formats. */
av_register_all();
avformat_network_init();
/* allocate the output media context */
avformat_alloc_output_context2(&oc, NULL, "rtsp", 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;
if(!fmt)
cout<<"Error creating outformat\n";
/* Add the audio and video streams using the default format codecs
* and initialize the codecs. */
/*if(av_opt_set(fmt,"rtsp_transport","tcp",0) < 0)
cout<<"Opt not set\n";*/
video_st = NULL;
audio_st = NULL;
cout<<"Codec = "<<avcodec_get_name(fmt->video_codec)<<endl;
if (fmt->video_codec != AV_CODEC_ID_NONE)
{
video_st = add_stream(oc, &video_codec, fmt->video_codec);
}
if (fmt->audio_codec != AV_CODEC_ID_NONE)
audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
/* 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);
char errorBuff[80];
if (!(fmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open outfile '%s': %s",filename,av_make_error_string(errorBuff,80,ret));
return 1;
}
}
ret = avformat_write_header(oc, NULL);
if (ret < 0) {
fprintf(stderr, "Error occurred when writing header: %s",av_make_error_string(errorBuff,80,ret));
return 1;
}
flush = 0;
while ((video_st && !video_is_eof) || (audio_st && !audio_is_eof)) {
/* Compute current audio and video time. */
audio_time = (audio_st && !audio_is_eof) ? audio_st->pts.val * av_q2d(audio_st->time_base) : INFINITY;
video_time = (video_st && !video_is_eof) ? video_st->pts.val * av_q2d(video_st->time_base) : INFINITY;
if (!flush &&
(!audio_st || audio_time >= STREAM_DURATION) &&
(!video_st || video_time >= STREAM_DURATION)) {
flush = 1;
}
/* write interleaved audio and video frames */
if (audio_st && !audio_is_eof && audio_time <= video_time) {
write_audio_frame(oc, audio_st, flush);
} else if (video_st && !video_is_eof && video_time < audio_time) {
write_video_frame(oc, video_st, flush);
}
}
/* Write the trailer, if any. The trailer must be written before you
* close the CodecContexts open when you wrote the header; otherwise
* av_write_trailer() may try to use memory that was freed on
* av_codec_close(). */
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))
/* Close the output file. */
avio_close(oc->pb);
/* free the stream */
avformat_free_context(oc);
return 0;
}
上述代码充当客户端，因此在执行代码之前，将接收流的服务器。我为此使用了以下命令
The above code acts as a client so before executing your code you need to start a server which would receive the stream. I used the following command for this purpose
ffplay -rtsp_flags listen -i rtsp://127.0.0.1:8554/live.sdp
这篇关于FFMPEG发送RTSP编码流C ++的文章就介绍到这了，希望我们推荐的答案对大家有所帮助，也希望大家多多支持IT屋！
查看全文
