使用x264和rtp进行麻烦同步libavformat / ffmpeg
-
12-12-2019 - |
题
我一直在研究一些带有实时饲料的流软件 从各种摄像机和网络上使用 H.264。要完成此操作,我直接使用x264编码器(有 “Zerolatency”预设)和喂养NALS可供选择 libavformat将包装成RTP(最终RTSP)。理想情况下,这是 应用应该尽可能实时。在大多数情况下, 这一直很好。
但是,不幸的是,有一些同步问题: 关于客户的任何视频播放似乎都展示了一些平滑的框架, 然后是短暂的暂停,然后更多的框架;重复。此外, 似乎大约4秒的延迟。这发生在 我尝试过的每个视频播放器:图腾,VLC和基本的GStreamer管道。我已经煮沸到了一定的小测试用例:
#include <stdio.h>
#include <stdint.h>
#include <unistd.h>
#include <x264.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#define WIDTH 640
#define HEIGHT 480
#define FPS 30
#define BITRATE 400000
#define RTP_ADDRESS "127.0.0.1"
#define RTP_PORT 49990
struct AVFormatContext* avctx;
struct x264_t* encoder;
struct SwsContext* imgctx;
uint8_t test = 0x80;
void create_sample_picture(x264_picture_t* picture)
{
// create a frame to store in
x264_picture_alloc(picture, X264_CSP_I420, WIDTH, HEIGHT);
// fake image generation
// disregard how wrong this is; just writing a quick test
int strides = WIDTH / 8;
uint8_t* data = malloc(WIDTH * HEIGHT * 3);
memset(data, test, WIDTH * HEIGHT * 3);
test = (test << 1) | (test >> (8 - 1));
// scale the image
sws_scale(imgctx, (const uint8_t* const*) &data, &strides, 0, HEIGHT,
picture->img.plane, picture->img.i_stride);
}
int encode_frame(x264_picture_t* picture, x264_nal_t** nals)
{
// encode a frame
x264_picture_t pic_out;
int num_nals;
int frame_size = x264_encoder_encode(encoder, nals, &num_nals, picture, &pic_out);
// ignore bad frames
if (frame_size < 0)
{
return frame_size;
}
return num_nals;
}
void stream_frame(uint8_t* payload, int size)
{
// initalize a packet
AVPacket p;
av_init_packet(&p);
p.data = payload;
p.size = size;
p.stream_index = 0;
p.flags = AV_PKT_FLAG_KEY;
p.pts = AV_NOPTS_VALUE;
p.dts = AV_NOPTS_VALUE;
// send it out
av_interleaved_write_frame(avctx, &p);
}
int main(int argc, char* argv[])
{
// initalize ffmpeg
av_register_all();
// set up image scaler
// (in-width, in-height, in-format, out-width, out-height, out-format, scaling-method, 0, 0, 0)
imgctx = sws_getContext(WIDTH, HEIGHT, PIX_FMT_MONOWHITE,
WIDTH, HEIGHT, PIX_FMT_YUV420P,
SWS_FAST_BILINEAR, NULL, NULL, NULL);
// set up encoder presets
x264_param_t param;
x264_param_default_preset(¶m, "ultrafast", "zerolatency");
param.i_threads = 3;
param.i_width = WIDTH;
param.i_height = HEIGHT;
param.i_fps_num = FPS;
param.i_fps_den = 1;
param.i_keyint_max = FPS;
param.b_intra_refresh = 0;
param.rc.i_bitrate = BITRATE;
param.b_repeat_headers = 1; // whether to repeat headers or write just once
param.b_annexb = 1; // place start codes (1) or sizes (0)
// initalize
x264_param_apply_profile(¶m, "high");
encoder = x264_encoder_open(¶m);
// at this point, x264_encoder_headers can be used, but it has had no effect
// set up streaming context. a lot of error handling has been ommitted
// for brevity, but this should be pretty standard.
avctx = avformat_alloc_context();
struct AVOutputFormat* fmt = av_guess_format("rtp", NULL, NULL);
avctx->oformat = fmt;
snprintf(avctx->filename, sizeof(avctx->filename), "rtp://%s:%d", RTP_ADDRESS, RTP_PORT);
if (url_fopen(&avctx->pb, avctx->filename, URL_WRONLY) < 0)
{
perror("url_fopen failed");
return 1;
}
struct AVStream* stream = av_new_stream(avctx, 1);
// initalize codec
AVCodecContext* c = stream->codec;
c->codec_id = CODEC_ID_H264;
c->codec_type = AVMEDIA_TYPE_VIDEO;
c->flags = CODEC_FLAG_GLOBAL_HEADER;
c->width = WIDTH;
c->height = HEIGHT;
c->time_base.den = FPS;
c->time_base.num = 1;
c->gop_size = FPS;
c->bit_rate = BITRATE;
avctx->flags = AVFMT_FLAG_RTP_HINT;
// write the header
av_write_header(avctx);
// make some frames
for (int frame = 0; frame < 10000; frame++)
{
// create a sample moving frame
x264_picture_t* pic = (x264_picture_t*) malloc(sizeof(x264_picture_t));
create_sample_picture(pic);
// encode the frame
x264_nal_t* nals;
int num_nals = encode_frame(pic, &nals);
if (num_nals < 0)
printf("invalid frame size: %d\n", num_nals);
// send out NALs
for (int i = 0; i < num_nals; i++)
{
stream_frame(nals[i].p_payload, nals[i].i_payload);
}
// free up resources
x264_picture_clean(pic);
free(pic);
// stream at approx 30 fps
printf("frame %d\n", frame);
usleep(33333);
}
return 0;
}
.
这个测试显示了白色背景上的黑线 应该平稳地向左移动。它已写成ffmpeg 0.6.5 但问题可以在 0.8 和 0.10 (到目前为止的测试)上再现。我在错误处理中拍了一些快捷方式,以使这个例子简称 可能仍然显示问题,所以请原谅一些 讨厌的代码。我还应该注意到这里没有使用SDP,我 尝试过使用类似结果的尝试。测试可以 编译:
gcc -g -std=gnu99 streamtest.c -lswscale -lavformat -lx264 -lm -lpthread -o streamtest
.
可以直接与Gtreamer一起玩:
gst-launch udpsrc port=49990 ! application/x-rtp,payload=96,clock-rate=90000 ! rtph264depay ! decodebin ! xvimagesink
.
你应该立即注意到口吃。一个常见的“修复”我 在互联网上看到的是将Sync= False添加到管道:
gst-launch udpsrc port=49990 ! application/x-rtp,payload=96,clock-rate=90000 ! rtph264depay ! decodebin ! xvimagesink sync=false
.
这会导致播放顺畅(和近实时),但是是一个 非解决方案,仅适用于GStreamer。我想解决这个问题 源问题。我已经能够与近乎相同的流 使用RAW FFMPEG的参数且没有任何问题:
ffmpeg -re -i sample.mp4 -vcodec libx264 -vpre ultrafast -vpre baseline -b 400000 -an -f rtp rtp://127.0.0.1:49990 -an
.
所以显然,我做错了什么。但它是什么?
解决方案
1)您没有为您发送给libx264的帧设置pts(您可能会看到“非严格单调的pts”警告) 2)您没有为您发送给LibaVFormat的RTP Muxer的数据包设置PTS / DTS(我不需要设置,但我猜它会更好。从源代码看起来像RTP使用PTS)。 3)imho useep(33333)很糟糕。它导致编码器在此时间(延迟增加)时,您可以在此期间编码下一个帧时,即使您仍然不需要通过RTP发送它,也可以在此时编码下一帧。
p.s。btw您未将param.rc.i_rc_method设置为x264_rc_abr,因此libx264将使用CRF 23,然后忽略“Param.rc.i_bitrate= bitRate”。在编码网络发送时,使用VBV也可能是好主意。