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Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir -
Librairies et binaires spécifiques au traitement vidéo et sonore
31 janvier 2010, parLes logiciels et librairies suivantes sont utilisées par SPIPmotion d’une manière ou d’une autre.
Binaires obligatoires FFMpeg : encodeur principal, permet de transcoder presque tous les types de fichiers vidéo et sonores dans les formats lisibles sur Internet. CF ce tutoriel pour son installation ; Oggz-tools : outils d’inspection de fichiers ogg ; Mediainfo : récupération d’informations depuis la plupart des formats vidéos et sonores ;
Binaires complémentaires et facultatifs flvtool2 : (...) -
Support audio et vidéo HTML5
10 avril 2011MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...)
Sur d’autres sites (4000)
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HTTP Livestreaming with ffmpeg
25 août 2016, par Hugo14453Some context : I have an MKV file, I am attempting to stream it to http://localhost:8090/test.flv as an flv file.
The stream begins and then immediately ends.
The command I am using is :
sudo ffmpeg -re -i input.mkv -c:v libx264 -maxrate 1000k -bufsize 2000k -an -bsf:v h264_mp4toannexb -g 50 http://localhost:8090/test.flv
A breakdown of what I believe these options do incase this post becomes useful for someone else :
sudo
Run as root
ffmpeg
The stream command thingy
-re
Stream in real-time
-i input.mkv
Input option and path to input file
-c:v libx264
Use codec libx264 for conversion
-maxrate 1000k -bufsize 2000k
No idea, some options for conversion, seems to help
-an -bsf:v h264_mp4toannexb
Audio options I think, not sure really. Also seems to help
-g 50
Still no idea, maybe frame rateframerateframerateframerate ?
http://localhost:8090/test.flv
Output using http protocol to localhost on port 8090 as a file called test.flv
Anyway the actual issue I have is that it begins to stream for about a second and then immediately ends.
The mpeg command result :
ffmpeg version N-80901-gfebc862 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable-libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable-libvidstab
libavutil 55. 28.100 / 55. 28.100
libavcodec 57. 48.101 / 57. 48.101
libavformat 57. 41.100 / 57. 41.100
libavdevice 57. 0.102 / 57. 0.102
libavfilter 6. 47.100 / 6. 47.100
libavresample 3. 0. 0 / 3. 0. 0
libswscale 4. 1.100 / 4. 1.100
libswresample 2. 1.100 / 2. 1.100
libpostproc 54. 0.100 / 54. 0.100
Input #0, matroska,webm, from 'input.mkv':
Metadata:
encoder : libebml v1.3.0 + libmatroska v1.4.0
creation_time : 1970-01-01 00:00:02
Duration: 00:01:32.26, start: 0.000000, bitrate: 4432 kb/s
Stream #0:0(eng): Video: h264 (High 10), yuv420p10le, 1920x1080 [SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default)
Stream #0:1(nor): Audio: flac, 48000 Hz, stereo, s16 (default)
[libx264 @ 0x2e1c380] using SAR=1/1
[libx264 @ 0x2e1c380] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX
[libx264 @ 0x2e1c380] profile High, level 4.0
[libx264 @ 0x2e1c380] 264 - core 148 r2643 5c65704 - H.264/MPEG-4 AVC codec - Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=1 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=50 keyint_min=5 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=1000 vbv_bufsize=2000 crf_max=0.0 nal_hrd=none filler=0 ip_ratio=1.40 aq=1:1.00
[flv @ 0x2e3f0a0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, flv, to 'http://localhost:8090/test.flv':
Metadata:
encoder : Lavf57.41.100
Stream #0:0(eng): Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=-1--1, 23.98 fps, 1k tbn, 23.98 tbc (default)
Metadata:
encoder : Lavc57.48.101 libx264
Side data:
cpb: bitrate max/min/avg: 1000000/0/0 buffer size: 2000000 vbv_delay: -1
Stream mapping:
Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
Press [q] to stop, [?] for help
Killed 26 fps= 26 q=0.0 size= 0kB time=00:00:00.00 bitrate=N/A speed= 0xThe ffserver outputs :
Sat Aug 20 12:40:11 2016 File '/test.flv' not found
Sat Aug 20 12:40:11 2016 [SERVER IP] - - [POST] "/test.flv HTTP/1.1" 404 189The config file is :
#Sample ffserver configuration file
# Port on which the server is listening. You must select a different
# port from your standard HTTP web server if it is running on the same
# computer.
Port 8090
# Address on which the server is bound. Only useful if you have
# several network interfaces.
BindAddress 0.0.0.0
# Number of simultaneous HTTP connections that can be handled. It has
# to be defined *before* the MaxClients parameter, since it defines the
# MaxClients maximum limit.
MaxHTTPConnections 2000
# Number of simultaneous requests that can be handled. Since FFServer
# is very fast, it is more likely that you will want to leave this high
# and use MaxBandwidth, below.
MaxClients 1000
# This the maximum amount of kbit/sec that you are prepared to
# consume when streaming to clients.
MaxBandwidth 1000
# Access log file (uses standard Apache log file format)
# '-' is the standard output.
CustomLog -
# Suppress that if you want to launch ffserver as a daemon.
#NoDaemon
##################################################################
# Definition of the live feeds. Each live feed contains one video
# and/or audio sequence coming from an ffmpeg encoder or another
# ffserver. This sequence may be encoded simultaneously with several
# codecs at several resolutions.
<feed>
ACL allow 192.168.0.0 192.168.255.255
# You must use 'ffmpeg' to send a live feed to ffserver. In this
# example, you can type:
#
#ffmpeg http://localhost:8090/test.ffm
# ffserver can also do time shifting. It means that it can stream any
# previously recorded live stream. The request should contain:
# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
# a path where the feed is stored on disk. You also specify the
# maximum size of the feed, where zero means unlimited. Default:
# File=/tmp/feed_name.ffm FileMaxSize=5M
File /tmp/feed1.ffm
FileMaxSize 200m
# You could specify
# ReadOnlyFile /saved/specialvideo.ffm
# This marks the file as readonly and it will not be deleted or updated.
# Specify launch in order to start ffmpeg automatically.
# First ffmpeg must be defined with an appropriate path if needed,
# after that options can follow, but avoid adding the http:// field
#Launch ffmpeg
# Only allow connections from localhost to the feed.
ACL allow 127.0.0.1
</feed>
##################################################################
# Now you can define each stream which will be generated from the
# original audio and video stream. Each format has a filename (here
# 'test1.mpg'). FFServer will send this stream when answering a
# request containing this filename.
<stream>
# coming from live feed 'feed1'
Feed feed1.ffm
# Format of the stream : you can choose among:
# mpeg : MPEG-1 multiplexed video and audio
# mpegvideo : only MPEG-1 video
# mp2 : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)
# ogg : Ogg format (Vorbis audio codec)
# rm : RealNetworks-compatible stream. Multiplexed audio and video.
# ra : RealNetworks-compatible stream. Audio only.
# mpjpeg : Multipart JPEG (works with Netscape without any plugin)
# jpeg : Generate a single JPEG image.
# asf : ASF compatible streaming (Windows Media Player format).
# swf : Macromedia Flash compatible stream
# avi : AVI format (MPEG-4 video, MPEG audio sound)
Format mpeg
# Bitrate for the audio stream. Codecs usually support only a few
# different bitrates.
AudioBitRate 32
# Number of audio channels: 1 = mono, 2 = stereo
AudioChannels 2
# Sampling frequency for audio. When using low bitrates, you should
# lower this frequency to 22050 or 11025. The supported frequencies
# depend on the selected audio codec.
AudioSampleRate 44100
# Bitrate for the video stream
VideoBitRate 64
# Ratecontrol buffer size
VideoBufferSize 40
# Number of frames per second
VideoFrameRate 3
# Size of the video frame: WxH (default: 160x128)
# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,
# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,
# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,
# hd1080
VideoSize hd1080
# Transmit only intra frames (useful for low bitrates, but kills frame rate).
#VideoIntraOnly
# If non-intra only, an intra frame is transmitted every VideoGopSize
# frames. Video synchronization can only begin at an intra frame.
VideoGopSize 12
# More MPEG-4 parameters
# VideoHighQuality
# Video4MotionVector
# Choose your codecs:
#AudioCodec mp2
#VideoCodec mpeg1video
# Suppress audio
#NoAudio
# Suppress video
#NoVideo
#VideoQMin 3
#VideoQMax 31
# Set this to the number of seconds backwards in time to start. Note that
# most players will buffer 5-10 seconds of video, and also you need to allow
# for a keyframe to appear in the data stream.
#Preroll 15
# ACL:
# You can allow ranges of addresses (or single addresses)
ACL ALLOW localhost
# You can deny ranges of addresses (or single addresses)
#ACL DENY <first address="address">
# You can repeat the ACL allow/deny as often as you like. It is on a per
# stream basis. The first match defines the action. If there are no matches,
# then the default is the inverse of the last ACL statement.
#
# Thus 'ACL allow localhost' only allows access from localhost.
# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
# allow everybody else.
</first></stream>
##################################################################
# Example streams
# Multipart JPEG
#<stream>
#Feed feed1.ffm
#Format mpjpeg
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#Strict -1
#</stream>
# Single JPEG
#<stream>
#Feed feed1.ffm
#Format jpeg
#VideoFrameRate 2
#VideoIntraOnly
##VideoSize 352x240
#NoAudio
#Strict -1
#</stream>
# Flash
#<stream>
#Feed feed1.ffm
#Format swf
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#</stream>
# ASF compatible
<stream>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</stream>
# MP3 audio
#<stream>
#Feed feed1.ffm
#Format mp2
#AudioCodec mp3
#AudioBitRate 64
#AudioChannels 1
#AudioSampleRate 44100
#NoVideo
#</stream>
# Ogg Vorbis audio
#<stream>
#Feed feed1.ffm
#Title "Stream title"
#AudioBitRate 64
#AudioChannels 2
#AudioSampleRate 44100
#NoVideo
#</stream>
# Real with audio only at 32 kbits
#<stream>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#NoVideo
#NoAudio
#</stream>
# Real with audio and video at 64 kbits
#<stream>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#VideoBitRate 128
#VideoFrameRate 25
#VideoGopSize 25
#NoAudio
#</stream>
##################################################################
# A stream coming from a file: you only need to set the input
# filename and optionally a new format. Supported conversions:
# AVI -> ASF
#<stream>
#File "/usr/local/httpd/htdocs/tlive.rm"
#NoAudio
#</stream>
#<stream>
#File "/usr/local/httpd/htdocs/test.asf"
#NoAudio
#Author "Me"
#Copyright "Super MegaCorp"
#Title "Test stream from disk"
#Comment "Test comment"
#</stream>
##################################################################
# RTSP examples
#
# You can access this stream with the RTSP URL:
# rtsp://localhost:5454/test1-rtsp.mpg
#
# A non-standard RTSP redirector is also created. Its URL is:
# http://localhost:8090/test1-rtsp.rtsp
#<stream>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#</stream>
# Transcode an incoming live feed to another live feed,
# using libx264 and video presets
#<stream>
#Format rtp
#Feed feed1.ffm
#VideoCodec libx264
#VideoFrameRate 24
#VideoBitRate 100
#VideoSize 480x272
#AVPresetVideo default
#AVPresetVideo baseline
#AVOptionVideo flags +global_header
#
#AudioCodec libfaac
#AudioBitRate 32
#AudioChannels 2
#AudioSampleRate 22050
#AVOptionAudio flags +global_header
#</stream>
##################################################################
# SDP/multicast examples
#
# If you want to send your stream in multicast, you must set the
# multicast address with MulticastAddress. The port and the TTL can
# also be set.
#
# An SDP file is automatically generated by ffserver by adding the
# 'sdp' extension to the stream name (here
# http://localhost:8090/test1-sdp.sdp). You should usually give this
# file to your player to play the stream.
#
# The 'NoLoop' option can be used to avoid looping when the stream is
# terminated.
#<stream>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#MulticastAddress 224.124.0.1
#MulticastPort 5000
#MulticastTTL 16
#NoLoop
#</stream>
##################################################################
# Special streams
# Server status
<stream>
Format status
# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255
#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
</stream>
# Redirect index.html to the appropriate site
<redirect>
URL http://www.ffmpeg.org/
</redirect>
#http://www.ffmpeg.org/Any help is greatly appreciated, I will do my best draw a picture of the best answer based on their username.
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Translating Return To Ringworld
17 août 2016, par Multimedia Mike — Game HackingAs indicated in my previous post, the Translator has expressed interest in applying his hobby towards another DOS adventure game from the mid 1990s : Return to Ringworld (henceforth R2RW) by Tsunami Media. This represents significantly more work than the previous outing, Phantasmagoria.
Return to Ringworld Title Screen
I have been largely successful thus far in crafting translation tools. I have pushed the fruits of these labors to a Github repository named improved-spoon (named using Github’s random name generator because I wanted something more interesting than ‘game-hacking-tools’).
Further, I have recorded everything I have learned about the game’s resource format (named RLB) at the XentaxWiki.
New Challenges
The previous project mostly involved scribbling subtitle text on an endless series of video files by leveraging a separate software library which took care of rendering fonts. In contrast, R2RW has at least 30k words of English text contained in various blocks which require translation. Further, the game encodes its own fonts (9 of them) which stubbornly refuse to be useful for rendering text in nearly any other language.Thus, the immediate 2 challenges are :
- Translating volumes of text to Spanish
- Expanding the fonts to represent Spanish characters
Normally, “figuring out the file format data structures involved” is on the list as well. Thankfully, understanding the formats is not a huge challenge since the folks at the ScummVM project already did all the heavy lifting of reverse engineering the file formats.
The Pitch
Here was the plan :- Create a tool that can dump out the interesting data from the game’s master resource file.
- Create a tool that can perform the elaborate file copy described in the previous post. The new file should be bit for bit compatible with the original file.
- Modify the rewriting tool to repack some modified strings into the new resource file.
- Unpack the fonts and figure out a way to add new characters.
- Repack the new fonts into the resource file.
- Repack message strings with Spanish characters.
Showing The Work : Modifying Strings
First, I created the tool to unpack blocks of message string resources. I elected to dump the strings to disk as JSON data since it’s easy to write and read JSON using Python, and it’s quick to check if any mistakes have crept in.The next step is to find a string to focus on. So I started the game and looked for the first string I could trigger :
This shows up in the JSON string dump as :
"Spanish" : " !0205Your quarters on the Lance of Truth are spartan, in accord with your mercenary lifestyle.", "English" : " !0205Your quarters on the Lance of Truth are spartan, in accord with your mercenary lifestyle." ,
As you can see, many of the strings are encoded with an ID key as part of the string which should probably be left unmodified. I changed the Spanish string :
"Spanish" : " !0205Hey, is this thing on ?", "English" : " !0205Your quarters on the Lance of Truth are spartan, in accord with your mercenary lifestyle." ,
And then I wrote the repacking tool to substitute this message block for the original one. Look ! The engine liked it !
Little steps, little steps.
Showing The Work : Modifying Fonts
The next little step is to find a place to put the new characters. First, a problem definition : The immediate goal is to translate the game into Spanish. The current fonts encoded in the game resource only support 128 characters, corresponding to 7-bit ASCII. In order to properly express Spanish, 16 new characters are required : á, é, Ã, ó, ú, ü, ñ (each in upper and lower case for a total of 14 characters) as well as the inverted punctuation symbols : ¿, ¡.Again, ScummVM already documents (via code) the font coding format. So I quickly determined that each of the 9 fonts is comprised of 128 individual bitmaps with either 1 or 2 bits per pixel. I wrote a tool to unpack each character into an individual portable grey map (PGM) image. These can be edited with graphics editors or with text editors since they are just text files.
Where to put the 16 new Spanish characters ? ASCII characters 1-31 are non-printable, so my first theory was that these characters would be empty and could be repurposed. However, after dumping and inspecting, I learned that they represent the same set of characters as seen in DOS Code Page 437. So that’s a no-go (so I assumed ; I didn’t check if any existing strings leveraged those characters).
My next plan was hope that I could extend the font beyond index 127 and use positions 128-143. This worked superbly. This is the new example string :
"Spanish" : " !0205¿Ves esto ? ¡La puntuacion se hace girar !", "English" : " !0205Your quarters on the Lance of Truth are spartan, in accord with your mercenary lifestyle." ,
Fortunately, JSON understands UTF-8 and after mapping the 16 necessary characters down to the numeric range of 128-143, I repacked the new fonts and the new string :
Translation : “See this ? The punctuation is rotated !”
Another victory. Notice that there are no diacritics in this string. None are required for this translation (according to Google Translate). But adding the diacritics to the 14 characters isn’t my department. My tool does help by prepopulating [aeiounAEIOUN] into the right positions to make editing easier for the Translator. But the tool does make the effort to rotate the punctuation since that is easy to automate.
Next Steps and Residual Weirdness
There is another method for storing ASCII text inside the R2RW resource called strip resources. These store conversation scripts. There are plenty of fields in the data structures that I don’t fully understand. So, following the lessons I learned from my previous translation outing, I was determined to modify as little as possible. This means copying over most of the original data structures intact, but changing the field representing the relative offset that points to the corresponding string. This works well since the strings are invariably stored NULL-terminated in a concatenated manner.I wanted to document for the record that the format that R2RW uses has some weirdness in they way it handles residual bytes in a resource. The variant of the resource format that R2RW uses requires every block to be aligned on a 16-byte boundary. If there is space between the logical end of the resource and the start of the next resource, there are random bytes in that space. This leads me to believe that these bytes were originally recorded from stale/uninitialized memory. This frustrates me because when I write the initial file copy tool which unpacks and repacks each block, I want the new file to be identical to the original. However, these apparent nonsense bytes at the end thwart that effort.
But leaving those bytes as 0 produces an acceptable resource file.
Text On Static Images
There is one last resource type we are working on translating. There are various bits of text that are rendered as images. For example, from the intro :
It’s possible to locate and extract the exact image that is overlaid on this scene, though without the colors :
The palettes are stored in a separate resource type. So it seems the challenge is to figure out the palette in use for these frames and render a transparent image that uses the same palette, then repack the new text-image into the new resource file.
-
FFmpeg RTP example error
20 juillet 2016, par JohnnylinCan anyone show me an example code of how to use FFmpeg to encode a ".mp4" video and then output to network stream using RTP(rtp ://127.0.0.1:6666). I have searched for it on google but most of them are just command line. Thanks very much.
UPDATED :
extern "C"{
#include <libavcodec></libavcodec>avcodec.h>
#include <libavformat></libavformat>avformat.h>
#include <libswscale></libswscale>swscale.h>
#include <libavutil></libavutil>avutil.h>
#include <libavutil></libavutil>opt.h>
#include <libavutil></libavutil>time.h>
}
#include
#include <iostream>
#include <string>
using namespace std;
// compatibility with newer API
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(55,28,1)
#define av_frame_alloc avcodec_alloc_frame
#define av_frame_free avcodec_free_frame
#endif
#define RNDTO2(X) ( ( (X) & 0xFFFFFFFE ))
#define RNDTO32(X) ( ( (X) % 32 ) ? ( ( (X) + 32 ) & 0xFFFFFFE0 ) : (X) )
//avcodec_alloc_frame is an old name
int main(int argc, char *argv[]) {
// Initalizing these to NULL prevents segfaults!
AVFormatContext *pFormatCtx = NULL;
int i, videoStream;
AVCodecContext *pCodecCtxOrig = NULL;
AVCodecContext *pCodecCtx = NULL;
AVCodec *pCodec = NULL;
AVFrame *pFrame = NULL;
AVPacket packet;
int frameFinished;
int numBytes;
struct SwsContext *sws_ctx = NULL;
int frame_index=0;
if(argc < 2) {
printf("Please provide a movie file\n");
return -1;
}
// Register all formats and codecs
av_register_all();
//initialize network video
avformat_network_init();
int errorStatus = 0;
char errorLog[128] = { 0 };
av_log_set_level(AV_LOG_TRACE);
//------------------ This is for local video file --------------
// Open video file
if(avformat_open_input(&pFormatCtx, argv[1], NULL, NULL)!=0)
return -1; // Couldn't open file
// Retrieve stream information
if(avformat_find_stream_info(pFormatCtx, NULL)<0)
return -1; // Couldn't find stream information
// Find the first video stream
videoStream=-1;
for(i=0; inb_streams; i++)
if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_VIDEO) {
videoStream=i;
break;
}
// Dump information about file onto standard error
av_dump_format(pFormatCtx, 0, argv[1], 0);
if(videoStream==-1)
return -1; // Didn't find a video stream
// Get a pointer to the codec context for the video stream
pCodecCtxOrig=pFormatCtx->streams[videoStream]->codec;
// Find the decoder for the video stream
pCodec=avcodec_find_decoder(pCodecCtxOrig->codec_id);
if(pCodec==NULL) {
fprintf(stderr, "Unsupported codec!\n");
return -1; // Codec not found
}
// Copy context
pCodecCtx = avcodec_alloc_context3(pCodec);
if(avcodec_copy_context(pCodecCtx, pCodecCtxOrig) != 0) {
fprintf(stderr, "Couldn't copy codec context");
return -1; // Error copying codec context
}
// Open codec
if(avcodec_open2(pCodecCtx, pCodec, NULL)<0)
return -1; // Could not open codec
// Allocate video frame
pFrame=av_frame_alloc();
// use nvidia codec
AVCodec* en_codec = avcodec_find_encoder_by_name("nvenc");
//AVCodec* en_codec = avcodec_find_encoder(AV_CODEC_ID_H264);
AVCodecContext* nv_codec_context = avcodec_alloc_context3(en_codec);
nv_codec_context->bit_rate = 1244179;
nv_codec_context->width = 1920;
nv_codec_context->height = 1080;
nv_codec_context->time_base.num = 1;
nv_codec_context->time_base.den = 30;
nv_codec_context->gop_size = 10;
nv_codec_context->max_b_frames = 1;
nv_codec_context->keyint_min = 1;
nv_codec_context->i_quant_factor = (float)0.71;
nv_codec_context->b_frame_strategy = 20;
nv_codec_context->qcompress = (float)0.6;
nv_codec_context->qmin = 20;
nv_codec_context->qmax = 51;
nv_codec_context->max_qdiff = 4;
nv_codec_context->refs = 4;
nv_codec_context->trellis = 1;
nv_codec_context->pix_fmt = AV_PIX_FMT_YUV420P;
//nv_codec_context->codec_id = AV_CODEC_ID_H264;
//nv_codec_context->codec_type = AVMEDIA_TYPE_VIDEO;
if (avcodec_open2(nv_codec_context, en_codec,NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}else printf("\nH264 codec opened\n");
/******stream*******/
string m_output("rtp://147.8.179.229:6666");
AVFormatContext* m_formatContext = NULL;
AVStream* m_stream = NULL;
if (avformat_alloc_output_context2(&m_formatContext, NULL, "H264", m_output.c_str()) < 0) {
cerr << "Cannot allocate output context: "
<< av_make_error_string(errorLog, 128, errorStatus) << endl;
return -1;
}
//AVCodec* tmp_codec = avcodec_find_encoder(AV_CODEC_ID_H264);
m_stream = avformat_new_stream(m_formatContext, en_codec);
if (!m_stream) {
cerr << "Cannot create a new stream: "
<< av_make_error_string(errorLog, 128, errorStatus) << endl;
return -1;
}
av_dump_format(m_formatContext, 0, m_output.c_str(), 1);
m_stream->codec->pix_fmt = AV_PIX_FMT_YUV420P;
m_stream->codec->width = 1920;
m_stream->codec->height = 1080;
//m_stream->codec->codec_id = AV_CODEC_ID_H264;
m_stream->codec->bit_rate = 40000;
m_stream->codec->time_base.den = 30;
m_stream->codec->time_base.num = 1;
m_stream->time_base.den = 30;
m_stream->time_base.num = 1;
m_stream->codec->codec_tag = 0;
if (m_formatContext->oformat->flags & AVFMT_GLOBALHEADER)
m_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
if (!(m_formatContext->oformat->flags & AVFMT_NOFILE))
errorStatus = avio_open(&m_formatContext->pb, m_output.c_str(), AVIO_FLAG_WRITE);
if ((errorStatus) < 0) {
cerr << "Cannot open output: "
<< av_make_error_string(errorLog, 128, errorStatus) << endl;
return -1;
}
if (avformat_write_header(m_formatContext, NULL) < 0) {
cerr << "Cannot write header to stream: "
<< av_make_error_string(errorLog, 128, errorStatus) << endl;
return -1;
}
/******stream*******/
FILE *fp_yuv = NULL;
fp_yuv=fopen("output.yuv","wb+");
// Read frames and save first five frames to disk
i=0;
while(av_read_frame(pFormatCtx, &packet)>=0) {
// Is this a packet from the video stream?
if(packet.stream_index==videoStream) {
// Decode video frame
avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished, &packet);
// Did we get a video frame?
if(frameFinished) {
AVPacket Outpacket;
int got_packet_ptr;
av_init_packet(&Outpacket);
Outpacket.data = NULL;
Outpacket.size = 0;
int out_size = avcodec_encode_video2(nv_codec_context, &Outpacket, pFrame, &got_packet_ptr);
Outpacket.pts = av_rescale_q(pFrame->pts, m_stream->codec->time_base, m_stream->time_base);
fwrite(Outpacket.data,1,Outpacket.size,fp_yuv);
//av_write_frame(m_formatContext, &Outpacket);
av_interleaved_write_frame(m_formatContext, &Outpacket);
//Free the packet that was allocated by encoder
av_packet_unref(&Outpacket);
}
}
// Free the packet that was allocated by av_read_frame
av_packet_unref(&packet);
}
fclose(fp_yuv);
// Free the YUV frame
av_frame_free(&pFrame);
// Close the codecs
avcodec_close(pCodecCtx);
avcodec_close(pCodecCtxOrig);
avcodec_close(m_stream->codec);
avformat_free_context(m_formatContext);
// Close the video file
avformat_close_input(&pFormatCtx);
return 0;
}
</string></iostream>Above is my code for decoding a .mp4 file, outputting to a network stream through RTP. But it does not work. On the client side, I call VLC library to decode the stream ( VlcMedia(url, _instance) where url is rtp/h264 ://@147.8.179.229:6666 ). When I change the protocal to UDP, it works. However, it is not stable. The frames jumps and probably the frame lost or something wrong with the pts setting ?
UPDATED :
When I use UDP, I got this error message.
When I use RTP, I got this error message :
Not sure about how to solve this problem, Can anyone help ? Thanks very much.