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Publier une image simplement
13 avril 2011, par ,
Mis à jour : Février 2012
Langue : français
Type : Video
Autres articles (66)
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Participer à sa traduction
10 avril 2011Vous pouvez nous aider à améliorer les locutions utilisées dans le logiciel ou à traduire celui-ci dans n’importe qu’elle nouvelle langue permettant sa diffusion à de nouvelles communautés linguistiques.
Pour ce faire, on utilise l’interface de traduction de SPIP où l’ensemble des modules de langue de MediaSPIP sont à disposition. ll vous suffit de vous inscrire sur la liste de discussion des traducteurs pour demander plus d’informations.
Actuellement MediaSPIP n’est disponible qu’en français et (...) -
D’autres logiciels intéressants
12 avril 2011, parOn ne revendique pas d’être les seuls à faire ce que l’on fait ... et on ne revendique surtout pas d’être les meilleurs non plus ... Ce que l’on fait, on essaie juste de le faire bien, et de mieux en mieux...
La liste suivante correspond à des logiciels qui tendent peu ou prou à faire comme MediaSPIP ou que MediaSPIP tente peu ou prou à faire pareil, peu importe ...
On ne les connais pas, on ne les a pas essayé, mais vous pouvez peut être y jeter un coup d’oeil.
Videopress
Site Internet : (...) -
Personnaliser les catégories
21 juin 2013, parFormulaire de création d’une catégorie
Pour ceux qui connaissent bien SPIP, une catégorie peut être assimilée à une rubrique.
Dans le cas d’un document de type catégorie, les champs proposés par défaut sont : Texte
On peut modifier ce formulaire dans la partie :
Administration > Configuration des masques de formulaire.
Dans le cas d’un document de type média, les champs non affichés par défaut sont : Descriptif rapide
Par ailleurs, c’est dans cette partie configuration qu’on peut indiquer le (...)
Sur d’autres sites (11008)
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uncertain behaviour of xml parser and ffmpeg streaming
8 octobre 2013, par user2775836I am making an ios application to display live streaming from ip camera using ffmpeg libraries.I am also using http api requests and responses.the response is in xml format ,hence i am using xml parser to parse the response.I have two view controllers(first and second).I am initializing the xml parser at view did disappear function of the first view controller also i am calling the function to stop the streaming in view did disappear after xml parsing function.While navigating from first view controller to second view controller the xml parser gets initiated but the parsing response does not get completed .when i come back from second to first view controller and then navigate again to second view controller ,at that time the full response is parsed.Why the response gets parsed second time not first time and why is it parsing half first time.Please help.
The code is as follows :- (void)viewDidDisappear:(BOOL)animated {
[self parsing]; //call to parsing function
[self.h264dec stopDecode]; //call to stop streaming
self.h264dec = nil;
}
parsing function is:
-(void)parsing
{
NSString *urlString = [NSString stringWithFormat:@"http://www.example.com"];
NSURL *url = [NSURL URLWithString:urlString];
NSMutableURLRequest *theRequest = [NSMutableURLRequest requestWithURL:url];
[theRequest addValue: @"application/x-www-form-urlencoded; charset=utf-8" forHTTPHeaderField:@"Content-Type"];
[theRequest setHTTPMethod:@"GET"];
NSURLConnection *connection = [[NSURLConnection alloc] initWithRequest:theRequest delegate:self];
if( connection )
{
mutableData = [[NSMutableData alloc] init];
NSLog(@"connection successful");
}
}
-(void) connection:(NSURLConnection *) connection didReceiveResponse:(NSURLResponse *)response
{
NSLog(@"receive response");
}
-(void) connection:(NSURLConnection *)connection didReceiveData:(NSData *)data
{
[mutableData appendData:data];
NSLog(@"DIDRECEIVE");
NSLog(@"the mutable data is: %@",mutableData);
}
-(void) connection:(NSURLConnection *)connection didFailWithError:(NSError *)error
{
return;
}
-(void)connectionDidFinishLoading:(NSURLConnection *)connection
{
NSLog(@"finish loading");
// You can do your functions here. If your repines is in XML you have to parse the response using NSXMLParser. If your response in JSON you have use SBJSON.
NSXMLParser *parser = [[NSXMLParser alloc] initWithData:mutableData];
[parser setDelegate:self];
[parser parse];
}
- (void)parser:(NSXMLParser *)parser didStartElement:(NSString *)elementName namespaceURI:(NSString *)namespaceURI qualifiedName:(NSString *)qualifiedName attributes:(NSDictionary *)attributeDict
{
if([elementName isEqualToString:@"Response"]){
NSLog(@"item found");
xmlStringFileObject =[[XMLStringFile alloc]init];
}
}
- (void)parser:(NSXMLParser *)parser foundCharacters:(NSString *)string
{
[nodecontent appendString:[string stringByTrimmingCharactersInSet:[NSCharacterSet whitespaceAndNewlineCharacterSet]]];
NSLog(@"node content = %@",nodecontent);
}
//bellow delegate method specify when it encounter end tag of specific that tag
- (void)parser:(NSXMLParser *)parser didEndElement:(NSString *)elementName namespaceURI:(NSString *)namespaceURI qualifiedName:(NSString *)qName
{
//I am saving my nodecontent data inside the property of XMLString File class
if([elementName isEqualToString:@"test"]){
xmlStringFileObject.test=nodecontent;
NSLog(@"test:%@",xmlStringFileObject.test);
}
else if([elementName isEqualToString:@"resolution"]){
xmlStringFileObject.resolution=nodecontent;
NSLog(@"resolution:%@",xmlStringFileObject.resolution);
AppDelegate *app = (AppDelegate*) [[UIApplication sharedApplication]delegate];
app.res = xmlStringFileObject.resolution;
NSLog(@"THE APPDELEGATE RES VALUE IS:%@",app.res);
}
//finally when we reaches the end of tag i am adding data inside the NSMutableArray
if([elementName isEqualToString:@"Response"]){
[rssOutputData addObject:xmlStringFileObject];
xmlStringFileObject = nil;
}
//release the data from mutable string variable
//reallocate the memory to get new content data from file
nodecontent=[[NSMutableString alloc]init];
} -
FFMPEG Video to Audio Conversion Results in Different Durations
10 juin 2020, par Eric JI am trying to covert an MP4 file into a mono WAV file sampled at 16,000 Hz.



When I run below code, the duration goes from 00:09:59.99 (MP4) to 00:09:57.64 (WAV). Its original, longer version goes from 00:48:37.46 (MP4) to 00:48:23.38 (WAV).



ffmpeg -i .mp4 -ac 1 -ar 16000 .wav




I've also tried below code. The result is much worse, going from 00:09:59.99 (MP4) to 00:12:56.29 (AAC).



ffmpeg -I .mp4 -vn -acodec copy .aac




Attaching the log :



Report written to "ffmpeg-20200610-093115.log"
Command line:
ffmpeg -i short.mp4 -ac 1 -ar 16000 short.wav -report
ffmpeg version 4.1.1 Copyright (c) 2000-2019 the FFmpeg developers
 built with Apple LLVM version 10.0.0 (clang-1000.11.45.5)
 configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1.1 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags='-I/Library/Java/JavaVirtualMachines/openjdk-11.0.2.jdk/Contents/Home/include -I/Library/Java/JavaVirtualMachines/openjdk-11.0.2.jdk/Contents/Home/include/darwin' --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-videotoolbox --disable-libjack --disable-indev=jack --enable-libaom --enable-libsoxr
 libavutil 56. 22.100 / 56. 22.100
 libavcodec 58. 35.100 / 58. 35.100
 libavformat 58. 20.100 / 58. 20.100
 libavdevice 58. 5.100 / 58. 5.100
 libavfilter 7. 40.101 / 7. 40.101
 libavresample 4. 0. 0 / 4. 0. 0
 libswscale 5. 3.100 / 5. 3.100
 libswresample 3. 3.100 / 3. 3.100
 libpostproc 55. 3.100 / 55. 3.100
Splitting the commandline.
Reading option '-i' ... matched as input url with argument 'short.mp4'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '1'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '16000'.
Reading option 'short.wav' ... matched as output url.
Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option report (generate a report) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url short.mp4.
Successfully parsed a group of options.
Opening an input file: short.mp4.
[NULL @ 0x7f98a3008200] Opening 'short.mp4' for reading
[file @ 0x7f98a2904440] Setting default whitelist 'file,crypto'
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] ISO: File Type Major Brand: mp42
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] Unknown dref type 0x206c7275 size 12
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] Processing st: 0, edit list 0 - media time: 0, duration: 7679872
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] Unknown dref type 0x206c7275 size 12
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] Processing st: 1, edit list 0 - media time: 1024, duration: 26459559
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] drop a frame at curr_cts: 0 @ 0
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] Before avformat_find_stream_info() pos: 11213917 bytes read:318782 seeks:1 nb_streams:2
[h264 @ 0x7f98a3808800] nal_unit_type: 7(SPS), nal_ref_idc: 3
[h264 @ 0x7f98a3808800] nal_unit_type: 8(PPS), nal_ref_idc: 3
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] demuxer injecting skip 1024 / discard 0
[aac @ 0x7f98a1008c00] skip 1024 / discard 0 samples due to side data
[h264 @ 0x7f98a3808800] nal_unit_type: 6(SEI), nal_ref_idc: 0
[h264 @ 0x7f98a3808800] nal_unit_type: 5(IDR), nal_ref_idc: 3
[h264 @ 0x7f98a3808800] Format yuv420p chosen by get_format().
[h264 @ 0x7f98a3808800] Reinit context to 640x368, pix_fmt: yuv420p
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] All info found
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f98a3008200] After avformat_find_stream_info() pos: 21961 bytes read:351550 seeks:2 frames:46
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'short.mp4':
 Metadata:
 major_brand : mp42
 minor_version : 1
 compatible_brands: isommp41mp42
 creation_time : 2020-06-10T16:12:17.000000Z
 Duration: 00:09:59.99, start: 0.000000, bitrate: 149 kb/s
 Stream #0:0(eng), 1, 1/12800: Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 DAR 16:9], 47 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default)
 Metadata:
 creation_time : 2020-06-10T16:12:17.000000Z
 handler_name : Core Media Video
 Stream #0:1(eng), 45, 1/44100: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 98 kb/s (default)
 Metadata:
 creation_time : 2020-06-10T16:12:17.000000Z
 handler_name : Core Media Audio
Successfully opened the file.
Parsing a group of options: output url short.wav.
Applying option ac (set number of audio channels) with argument 1.
Applying option ar (set audio sampling rate (in Hz)) with argument 16000.
Successfully parsed a group of options.
Opening an output file: short.wav.
[file @ 0x7f98a0c1db40] Setting default whitelist 'file,crypto'
Successfully opened the file.
Stream mapping:
 Stream #0:1 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[aac @ 0x7f98a100de00] skip 1024 / discard 0 samples due to side data
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
detected 12 logical cores
[graph_0_in_0_1 @ 0x7f98a0e2c4c0] Setting 'time_base' to value '1/44100'
[graph_0_in_0_1 @ 0x7f98a0e2c4c0] Setting 'sample_rate' to value '44100'
[graph_0_in_0_1 @ 0x7f98a0e2c4c0] Setting 'sample_fmt' to value 'fltp'
[graph_0_in_0_1 @ 0x7f98a0e2c4c0] Setting 'channel_layout' to value '0x4'
[graph_0_in_0_1 @ 0x7f98a0e2c4c0] tb:1/44100 samplefmt:fltp samplerate:44100 chlayout:0x4
[format_out_0_0 @ 0x7f98a0e2cb80] Setting 'sample_fmts' to value 's16'
[format_out_0_0 @ 0x7f98a0e2cb80] Setting 'sample_rates' to value '16000'
[format_out_0_0 @ 0x7f98a0e2cb80] Setting 'channel_layouts' to value '0x4'
[format_out_0_0 @ 0x7f98a0e2cb80] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_0'
[AVFilterGraph @ 0x7f98a0c16ac0] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto_resampler_0 @ 0x7f98a0e2d540] [SWR @ 0x7f98a28e1000] Using fltp internally between filters
[auto_resampler_0 @ 0x7f98a0e2d540] ch:1 chl:mono fmt:fltp r:44100Hz -> ch:1 chl:mono fmt:s16 r:16000Hz
Output #0, wav, to 'short.wav':
 Metadata:
 major_brand : mp42
 minor_version : 1
 compatible_brands: isommp41mp42
 ISFT : Lavf58.20.100
 Stream #0:0(eng), 0, 1/16000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, s16, 256 kb/s (default)
 Metadata:
 creation_time : 2020-06-10T16:12:17.000000Z
 handler_name : Core Media Audio
 encoder : Lavc58.35.100 pcm_s16le
size= 17152kB time=00:09:16.63 bitrate= 252.4kbits/s speed=1.11e+03x 
[out_0_0 @ 0x7f98a0e2c700] EOF on sink link out_0_0:default.
No more output streams to write to, finishing.
size= 18676kB time=00:09:59.99 bitrate= 255.0kbits/s speed=1.11e+03x 
video:0kB audio:18676kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000408%
Input file #0 (short.mp4):
 Input stream #0:0 (video): 1 packets read (3689 bytes); 
 Input stream #0:1 (audio): 25739 packets read (7375414 bytes); 25738 frames decoded (26355712 samples); 
 Total: 25740 packets (7379103 bytes) demuxed
Output file #0 (short.wav):
 Output stream #0:0 (audio): 25739 frames encoded (9562163 samples); 25739 packets muxed (19124326 bytes); 
 Total: 25739 packets (19124326 bytes) muxed
25738 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x7f98a0c1dc40] Statistics: 4 seeks, 76 writeouts
[AVIOContext @ 0x7f98a29045c0] Statistics: 10902846 bytes read, 29 seeks



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How to improve the performance of Audio Queue Services when playing audio ?
11 décembre 2014, par 谢小进I want to play
RTMP
’sH.264
/AAC
usingFFmpeg
for decoding and playing video,Audio Queue Services
for playing audio. I have successfully done them all, but still some tough issues, for example high memory allocation when playing audio. I have debugged and found thatAudio Queue Services
lead to high memory allocation, and then crash ! Does anybody know how to improve the memory performance ? Here is my code forAudio Queue Services
playing audio.//
// RTMPAudioPlayer.h
//
#import <foundation></foundation>Foundation.h>
#import <audiotoolbox></audiotoolbox>AudioToolbox.h>
@interface AQBuffer : NSObject
@property (nonatomic) AudioQueueBufferRef buffer;
@end
@interface RTMPAudioPlayer : NSObject {
AudioQueueRef queue;
AudioStreamBasicDescription dataFormat;
NSMutableArray *buffers;
NSMutableArray *reusableBuffers;
}
- (id)initWithSampleRate:(int)sampleRate channels:(int)channels bitsPerChannel:(int)bitsPerChannel;
- (void)start;
- (void)stop;
- (void)putData:(NSData *)data;
@end
//
// RTMPAudioPlayer.m
//
#import "RTMPAudioPlayer.h"
static const int kNumberBuffers = 3;
static const int kBufferSize = 0xA000;
@implementation AQBuffer
@end
@implementation RTMPAudioPlayer
void AQOutputCallback(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inCompleteAQBuffer) {
RTMPAudioPlayer *THIS = (__bridge RTMPAudioPlayer *)inUserData;
[THIS handleAQOutputCallback:inAQ buffer:inCompleteAQBuffer];
}
- (void)handleAQOutputCallback:(AudioQueueRef)audioQueue buffer:(AudioQueueBufferRef)buffer {
for (int i = 0; i < [buffers count]; ++i) {
if (buffer == [buffers[i] buffer]) {
[reusableBuffers addObject:buffers[i]];
break;
}
}
}
- (id)initWithSampleRate:(int)sampleRate channels:(int)channels bitsPerChannel:(int)bitsPerChannel {
self = [super init];
if (self) {
memset(&dataFormat, 0, sizeof(dataFormat));
dataFormat.mSampleRate = sampleRate;
dataFormat.mFormatID = kAudioFormatLinearPCM;
dataFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
dataFormat.mBitsPerChannel = bitsPerChannel;
dataFormat.mChannelsPerFrame = channels;
dataFormat.mFramesPerPacket = 1;
dataFormat.mBytesPerFrame = (dataFormat.mBitsPerChannel / 8) * dataFormat.mChannelsPerFrame;
dataFormat.mBytesPerPacket = dataFormat.mBytesPerFrame * dataFormat.mFramesPerPacket;
}
return self;
}
- (void)start {
OSStatus status = AudioQueueNewOutput(&dataFormat, AQOutputCallback, (__bridge void *)self, NULL, NULL, 0, &queue);
if (status == noErr) {
buffers = [NSMutableArray array];
reusableBuffers = [NSMutableArray array];
for (int i = 0; i < kNumberBuffers; i++) {
AudioQueueBufferRef buffer;
status = AudioQueueAllocateBuffer(queue, kBufferSize, &buffer);
if (status == noErr) {
AQBuffer *bufferObj = [[AQBuffer alloc] init];
bufferObj.buffer = buffer;
[buffers addObject:bufferObj];
[reusableBuffers addObject:bufferObj];
} else {
AudioQueueDispose(queue, true);
queue = NULL;
break;
}
}
AudioQueueStart(queue, NULL);
} else {
queue = NULL;
}
}
- (void)stop {
if (queue) {
AudioQueueStop(queue, true);
}
}
- (void)putData:(NSData *)data {
AQBuffer *bufferObj = [reusableBuffers firstObject];
[reusableBuffers removeObject:bufferObj];
AudioQueueBufferRef buffer;
OSStatus status = AudioQueueAllocateBuffer(queue, kBufferSize, &buffer);
if (status == noErr) {
bufferObj = [[AQBuffer alloc] init];
bufferObj.buffer = buffer;
memcpy(bufferObj.buffer->mAudioData, [data bytes], [data length]);
bufferObj.buffer->mAudioDataByteSize = [data length];
AudioQueueEnqueueBuffer(queue, bufferObj.buffer, 0, NULL);
}
}
@end