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Sintel MP4 Surround 5.1 Full
13 mai 2011, par
Mis à jour : Février 2012
Langue : English
Type : Video
Autres articles (67)
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Websites made with MediaSPIP
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Creating farms of unique websites
13 avril 2011, parMediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...) -
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
Sur d’autres sites (6674)
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flv reencode to mp4 for iphone/ipod via ffmpeg and x264 (quality issue)
3 octobre 2011, par zeroasteriskThere are a lot of questions on this topic, and I've read most of them and most of the google search results I could come up with.
When I use FFMPEG to convert a FLV to a iphone3 compatble MP4 file, it just doesn't preserver enough of the quality. Yes, I've worked the hell out of
-sameqand-band-btsettings, text just isn't readable.Next I tried to split the video out and process it directly, using these instructions :
https://sites.google.com/site/linuxencoding/x264-encoding-guideThe problem is myplayer (via ffmpeg) was not able to determine the duration of the FLV (even though the metadata was set).
(I assume) Because of that unknown duration, when I create the MP4 file, the resulting x264 file plays through super-fast while the audio plays at the normal rate.
user@server:/tmp# mplayer -nosound -benchmark -sws 9 -vf dsize=640:480:0,scale=0:0,expand=640:480 -vo yuv4mpeg:file=>(x264 --demuxer y4m --crf 0 --preset slow --threads auto --output output.264 - 2>x264.log) 'input.flv'
MPlayer 1.0rc4-4.4.5 (C) 2000-2010 MPlayer Team
mplayer: could not connect to socket
mplayer: No such file or directory
Failed to open LIRC support. You will not be able to use your remote control.
Playing input.flv.
libavformat file format detected.
[flv @ 0x1202460]Estimating duration from bitrate, this may be inaccurate
[lavf] stream 0: video (vp6f), -vid 0
[lavf] stream 1: audio (nellymoser), -aid 0
VIDEO: [VP6F] 1680x992 0bpp 1000.000 fps 33.4 kbps ( 4.1 kbyte/s)
Clip info:
audiocodecid: 6
audiodatarate: 86
audiosamplerate: 44100
audiosamplesize: 16
audiosize: 6097005
canSeekToEnd: true
datasize: 8609138
duration: 567
framerate: 2
hasAudio: true
hasCuePoints: false
hasKeyframes: true
hasMetadata: true
hasVideo: true
height: 992
lasttimestamp: 567
metadatacreator: flvtool++ (Facebook, Motion project, dweatherford)
stereo: false
totalframes: 1043
videocodecid: 4
videodatarate: 33
videosize: 2316256
width: 1680
Using (default) progressive frame mode.Opening video filter: [expand w=640 h=480]
Expand: 640 x 480, -1 ; -1, osd: 0, aspect: 0.000000, round: 1
Opening video filter: [scale w=0 h=0]
Opening video filter: [dsize=640:480:0]
==========================================================================
Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
Selected video codec: [ffvp6f] vfm: ffmpeg (FFmpeg VP6 Flash)
==========================================================================
Audio: no sound
Starting playback...
Movie-Aspect is undefined - no prescaling applied.
[swscaler @ 0x7f0c738b9620]Lanczos scaler, from yuv420p to yuv420p using MMX2
VO: [yuv4mpeg] 640x480 => 641x480 Planar YV12I have also tried specifying FPS, but no change in results
user@server:/tmp# mplayer -nosound -fps 25-benchmark -sws 9 -vf dsize=640:480:0,scale=0:0,expand=640:480 -vo yuv4mpeg:file=>(x264 --demuxer y4m --fps 25 --crf 0 --preset slow --threads auto --output output.264 - 2>x264.log) 'input.flv'Can someone tell me how to either :
- fix my split A/V processing/timing/duration issues ?
- improve the
quality of the FFMPEG conversion of FLV to iphone3 compatible
format ?
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Decoding audio via Android using FFMpeg
31 octobre 2013, par Rob HauptI can play Wav files using the below code without issues. When trying to play the exact same media in Mp3 format I only get garbled junk. I believe I am fundamentally misunderstanding how the avcodec_decode_audio3 function works.
Since the Wav file contains PCM data when it is decoded it can go straight to the AudioTrack.write function. There must be some additional step to get Mp3s to work like this. I don't know what I'm missing, but I've been pulling my hair out for a week now.
Java Code
package com.rohaupt.RRD2;
import java.io.FileNotFoundException;
import java.io.FileOutputStream;
import java.io.IOException;
import android.app.Activity;
import android.media.AudioFormat;
import android.media.AudioManager;
import android.media.AudioTrack;
import android.media.MediaPlayer;
import android.os.Bundle;
import android.os.SystemClock;
public class player extends Activity
{
private AudioTrack track;
private FileOutputStream os;
/** Called when the activity is first created. */
@Override
public void onCreate(Bundle savedInstanceState)
{
super.onCreate(savedInstanceState);
setContentView(R.layout.main);
createEngine();
MediaPlayer mp = new MediaPlayer();
mp.start();
int bufSize = AudioTrack.getMinBufferSize(32000,
AudioFormat.CHANNEL_CONFIGURATION_STEREO,
AudioFormat.ENCODING_PCM_16BIT);
track = new AudioTrack(AudioManager.STREAM_MUSIC,
32000,
AudioFormat.CHANNEL_CONFIGURATION_STEREO,
AudioFormat.ENCODING_PCM_16BIT,
bufSize,
AudioTrack.MODE_STREAM);
byte[] bytes = new byte[bufSize];
try {
os = new FileOutputStream("/sdcard/a.out",false);
} catch (FileNotFoundException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
String result = loadFile("/sdcard/a.mp3",bytes);
try {
os.close();
} catch (IOException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
}
void playSound(byte[] buf, int size) {
//android.util.Log.v("ROHAUPT", "RAH Playing");
if(track.getPlayState()!=AudioTrack.PLAYSTATE_PLAYING)
track.play();
track.write(buf, 0, size);
try {
os.write(buf,0,size);
} catch (IOException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
}
private native void createEngine();
private native String loadFile(String file, byte[] array);
/** Load jni .so on initialization*/
static {
System.loadLibrary("avutil");
System.loadLibrary("avcore");
System.loadLibrary("avcodec");
System.loadLibrary("avformat");
System.loadLibrary("avdevice");
System.loadLibrary("swscale");
System.loadLibrary("avfilter");
System.loadLibrary("ffmpeg");
}
}C Code
#include
#include
#include
#include <android></android>log.h>
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#define DEBUG_TAG "ROHAUPT"
void Java_com_rohaupt_RRD2_player_createEngine(JNIEnv* env, jclass clazz)
{
avcodec_init();
av_register_all();
}
jstring Java_com_rohaupt_RRD2_player_loadFile(JNIEnv* env, jobject obj,jstring file,jbyteArray array)
{
jboolean isCopy;
int i;
int audioStream=-1;
int res;
int decoded = 0;
int out_size;
AVFormatContext *pFormatCtx;
AVCodecContext *aCodecCtx;
AVCodecContext *c= NULL;
AVCodec *aCodec;
AVPacket packet;
jclass cls = (*env)->GetObjectClass(env, obj);
jmethodID play = (*env)->GetMethodID(env, cls, "playSound", "([BI)V");//At the begining of your main function
const char * szfile = (*env)->GetStringUTFChars(env, file, &isCopy);
int16_t * pAudioBuffer = (int16_t *) av_malloc (AVCODEC_MAX_AUDIO_FRAME_SIZE*2+FF_INPUT_BUFFER_PADDING_SIZE);
int16_t * outBuffer = (int16_t *) av_malloc (AVCODEC_MAX_AUDIO_FRAME_SIZE*2+FF_INPUT_BUFFER_PADDING_SIZE);
__android_log_print(ANDROID_LOG_INFO, DEBUG_TAG, "RAH28 Starting");
res = av_open_input_file(&pFormatCtx, szfile, NULL, 0, NULL);
if(res!=0)
{
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH opening input failed with result: [%d]", res);
return file;
}
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH getting stream info");
res = av_find_stream_info(pFormatCtx);
if(res<0)
{
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH getting stream info failed with result: [%d]", res);
return file;
}
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH getting audio stream");
for(i=0; i < pFormatCtx->nb_streams; i++) {
if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO &&
audioStream < 0) {
audioStream=i;
}
}
if(audioStream==-1)
{
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH couldn't find audio stream");
return file;
}
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH audio stream found with result: [%d]", res);
aCodecCtx=pFormatCtx->streams[audioStream]->codec;
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH audio codec info loaded");
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH audio codec info [%d]", aCodecCtx->codec_id);
aCodec = avcodec_find_decoder(aCodecCtx->codec_id);
if(!aCodec) {
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH audio codec unsupported");
return file;
}
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH audio codec info found");
res = avcodec_open(aCodecCtx, aCodec);
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH audio codec loaded [%d] [%d]",aCodecCtx->sample_fmt,res);
//c=avcodec_alloc_context();
av_init_packet(&packet);
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH channels [%d] sample rate [%d] sample format [%d]",aCodecCtx->channels,aCodecCtx->sample_rate,aCodecCtx->sample_fmt);
int x,y;
x=0;y=0;
while (av_read_frame(pFormatCtx, &packet)>= 0) {
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH frame read: [%d] [%d]",x++,y);
if (aCodecCtx->codec_type == AVMEDIA_TYPE_AUDIO) {
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH audio ready");
int data_size = AVCODEC_MAX_AUDIO_FRAME_SIZE*2+FF_INPUT_BUFFER_PADDING_SIZE;
int size=packet.size;
y=0;
decoded = 0;
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH packet size: [%d]", size);
while(size > 0) {
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH decoding: [%d] [%d]",x,y++);
int len = avcodec_decode_audio3(aCodecCtx, pAudioBuffer, &data_size, &packet);
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH 1 size [%d] len [%d] data_size [%d] out_size [%d]",size,len,data_size,out_size);
jbyte *bytes = (*env)->GetByteArrayElements(env, array, NULL);
memcpy(bytes + decoded, pAudioBuffer, len); //
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH 2");
(*env)->ReleaseByteArrayElements(env, array, bytes, 0);
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH 3");
(*env)->CallVoidMethod(env, obj, play, array, len);
__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH 4");
size -= len;
decoded += len;
}
av_free_packet(&packet);
}
}
// Close the video file
av_close_input_file(pFormatCtx);
//__android_log_print(ANDROID_LOG_DEBUG, DEBUG_TAG, "RAH Finished Running result: [%d]", res);
(*env)->ReleaseStringUTFChars(env, file, szfile);
return file;
}To add some detail. When Calling this function with a Wav File I get the following Log Data
I/ROHAUPT ( 227): RAH28 Starting
D/ROHAUPT ( 227): RAH getting stream info
D/ROHAUPT ( 227): RAH getting audio stream
D/ROHAUPT ( 227): RAH audio stream found with result: [0]
D/ROHAUPT ( 227): RAH audio codec info loaded
D/ROHAUPT ( 227): RAH audio codec info [65536]
D/ROHAUPT ( 227): RAH audio codec info found
D/ROHAUPT ( 227): RAH audio codec loaded [1] [0]
D/ROHAUPT ( 227): RAH channels [2] sample rate [32000] sample format [1]
D/ROHAUPT ( 227): RAH frame read: [0] [0]
D/ROHAUPT ( 227): RAH audio ready
D/ROHAUPT ( 227): RAH packet size: [4096]
D/ROHAUPT ( 227): RAH decoding: [1] [0]
D/ROHAUPT ( 227): RAH 1 size [4096] len [4096] data_size [4096] out_size [0]
D/ROHAUPT ( 227): RAH 2
D/ROHAUPT ( 227): RAH 3
D/ROHAUPT ( 227): RAH 4
D/ROHAUPT ( 227): RAH frame read: [1] [1]
D/ROHAUPT ( 227): RAH audio ready
...
D/ROHAUPT ( 227): RAH frame read: [924] [1]
D/ROHAUPT ( 227): RAH audio ready
D/ROHAUPT ( 227): RAH packet size: [4096]
D/ROHAUPT ( 227): RAH decoding: [925] [0]
D/ROHAUPT ( 227): RAH 1 size [4096] len [4096] data_size [4096] out_size [0]
D/ROHAUPT ( 227): RAH 2
D/ROHAUPT ( 227): RAH 3
D/ROHAUPT ( 227): RAH 4
D/ROHAUPT ( 227): RAH frame read: [925] [1]
D/ROHAUPT ( 227): RAH audio ready
D/ROHAUPT ( 227): RAH packet size: [3584]
D/ROHAUPT ( 227): RAH decoding: [926] [0]
D/ROHAUPT ( 227): RAH 1 size [3584] len [3584] data_size [3584] out_size [0]
D/ROHAUPT ( 227): RAH 2
D/ROHAUPT ( 227): RAH 3
D/ROHAUPT ( 227): RAH 4When calling with an Mp3 file I get the following
I/ROHAUPT ( 280): RAH28 Starting
D/ROHAUPT ( 280): RAH getting stream info
D/ROHAUPT ( 280): RAH getting audio stream
D/ROHAUPT ( 280): RAH audio stream found with result: [0]
D/ROHAUPT ( 280): RAH audio codec info loaded
D/ROHAUPT ( 280): RAH audio codec info [86017]
D/ROHAUPT ( 280): RAH audio codec info found
D/ROHAUPT ( 280): RAH audio codec loaded [1] [0]
D/ROHAUPT ( 280): RAH channels [2] sample rate [32000] sample format [1]
D/ROHAUPT ( 280): RAH frame read: [0] [0]
D/ROHAUPT ( 280): RAH audio ready
D/ROHAUPT ( 280): RAH packet size: [432]
D/ROHAUPT ( 280): RAH decoding: [1] [0]
D/ROHAUPT ( 280): RAH 1 size [432] len [432] data_size [4608] out_size [0]
D/ROHAUPT ( 280): RAH 2
...
D/ROHAUPT ( 280): RAH frame read: [822] [1]
D/ROHAUPT ( 280): RAH audio ready
D/ROHAUPT ( 280): RAH packet size: [432]
D/ROHAUPT ( 280): RAH decoding: [823] [0]
D/ROHAUPT ( 280): RAH 1 size [432] len [432] data_size [4608] out_size [0]
D/ROHAUPT ( 280): RAH 2
D/ROHAUPT ( 280): RAH 3
D/ROHAUPT ( 280): RAH 4
D/ROHAUPT ( 280): RAH frame read: [823] [1]
D/ROHAUPT ( 280): RAH audio ready
D/ROHAUPT ( 280): RAH packet size: [432]
D/ROHAUPT ( 280): RAH decoding: [824] [0]
D/ROHAUPT ( 280): RAH 1 size [432] len [432] data_size [4608] out_size [0]
D/ROHAUPT ( 280): RAH 2
D/ROHAUPT ( 280): RAH 3
D/ROHAUPT ( 280): RAH 4 -
Anomalie #2333 (Nouveau) : Jointure / balise #TRI
15 septembre 2011, par Severo LipariLa balise #TRI est super pour présenter des listes de résultats, mais il y a un problème dans un cas précis : si on fait le tri sur un champ d’une table jointe (ici : spip_rubriques.profondeur), et qu’on n’utilise pas la balise correspondante dans la boucle (#PROFONDEUR) [(#TRItitre,"TITRE",ajax) |] (...)
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