I downloaded some init MP4 file(init.mp4) and a sequence of m4s files(like 744397965.m4s, 744397966.m4s, 744397967.m4s...) from a live stream http://vm2.dashif.org/livesim/testpic_2s/Manifest.mpd using Dash.js.
Then I tried to feed these files directly into SourceBuffer bind with a video element, no pictures been played and no error thrown, why?
Did you seek the video element to the earliest timestamp in the resultant buffer (if it is not zero) and then call play() ?
Related
I'm using the Google Drive API where I can gain access to 2 pieces of data that I need to display a jpg file oin my program. WebViewLink is the "large" size image while thumbnailLink is the "thumb" smaller size of the same image.
I'm having an issue with downloading the WebViewLink that I do not have with the thumbnailLink. Part of my code calls either exif_imagetype($filename) or getimagesize($filename) so I can retrieve the type, height & width etc for the $filename. This is successful for the thumbnailView but not the WebViewLink...
code snippet...
$WebViewLink = "https://drive.google.com/a/treering.com/file/d/blablabla";
$type = exif_imagetype($WebViewLink);
--- results in the error
"PHP Warning: exif_imagetype(): stream does not support seeking..."
where as...
$thumbnailLink = "https://lh6.googleusercontent.com/blablabla";
$type = exif_imagetype($thumbnailLink);
--- successful
where $type = 2 // a .jpg file
Not sure what I need to do to gain a usable WebViewLink... maybe use the "export" function to copy to a file on my server that is accessible, then use that exported file for the functions that fail above?
Thanks for any help.
John
I think you are using the wrong property to get the image of the file.
WebViewLink
A link for opening the file in a relevant Google editor or viewer in a browser.
thumbnailLink
A short-lived link to the file's thumbnail, if available. Typically lasts on the order of hours.
You can try using the iconLink():
A static, unauthenticated link to the file's icon.
Sample image of thumbnailLink:
Sample image of a iconLink:
It will still show relevant image about the file.
Hope it helps!
When playing a memory stream containing wav encoded audio, the playback starts with a sharp pop/crackle:
ms = new MemoryStream(File.ReadAllBytes(audio_filename));
[...]
dispose_audio();
sound_output = new DirectSoundOut();
IWaveProvider provider = new RawSourceWaveStream(ms, new WaveFormat());
sound_output.Init(provider);
sound_output.Play();
That pop/crackle does not occur when playing the wav file directly:
dispose_audio();
NAudio.Wave.WaveStream pcm = new WaveChannel32(new NAudio.Wave.WaveFileReader(audio_filename));
audio_stream = new BlockAlignReductionStream(pcm);
sound_output = new DirectSoundOut();
sound_output.Init(audio_stream);
sound_output.Play();
Same file is playing, but when the wav data are stored in a memory stream first, there is a somewhat loud pop at the beginning of the playback.
I am very much a newbie with NAudio and audio in general, so it's probably something silly, but I can't seem to figure it out.
You are playing the WAV file header as though it were audio. Instead of RawSourceWaveStream, you still need to use WaveFileReader, just pass in your memory stream.
I want to resample an audio file using NAudio and MFT on-the-fly.
For example, I have the following audio file:
File name: MyAudioFile.mp3
Duration: 10 sec
When this file is being played, I only want to resample that particular position to WAV in the desired format.
So, if the length of "MyAudioFile.mp3" is 10 sec, and the "current play position" is 2.5 sec, I want to resample only that portion of data into WAV format at the sampling rate of 48 KHz.
When the audio progresses further, again, only the "current play position" must be resampled.
I tried the following code:
WaveStream reader = new MediaFoundationReaderRT([path of
"MyAudioFile.mp3"]);
MemoryStream outMemStream = new MemoryStream(); //Decode to memory
stream
using (reader)
using (var resampler = new MediaFoundationResampler(reader,
resampler.WaveFormat))
{
WaveFileWriter.CreateWaveFile(outMemStream, resampler);
rsws = new RawSourceWaveStream(outMemStream, resampler.WaveFormat);
}
WaveChannel32 waveformInputStream = new WaveChannel32(rsws);
The resampling happens properly; however it resamples the whole audio file, which takes time.
What I am looking at is just resampling the "current play position" of the audio, and discard any other position information.
Thanks! Appreciate if you can provide some sample.
To resample on the fly, just pass the reader directly into MediaFoundationResampler. You will now have an ISampleProvider so you won't be able to use WaveChannel32, but really that is an obsolete class now, and you should be able to do anything you need with other ISampleProvider classes from NAudio.
My goal is to record audio in AAC format and send it over a network connection as a stream.
I'm using Audio Queue Services and have based my code on the SpeakHere example. I know that for writing to a file it uses the AudioFileWritePackets() function.
Here's the callback function:
void MyInputBufferHandler(void* inUserData,
AudioQueueRef inAQ,
AudioQueueBufferRef inBuffer,
const AudioTimeStamp * inStartTime,
UInt32 inNumPackets,
const AudioStreamPacketDescription* inPacketDesc) {
Recorder *aqr = (Recorder *)inUserData;
if (inNumPackets > 0) {
// write packets to file
AudioFileWritePackets(aqr->mRecordFile, FALSE,
inBuffer->mAudioDataByteSize,
inPacketDesc, aqr->mRecordPacket,
&inNumPackets, inBuffer->mAudioData);
aqr->mRecordPacket += inNumPackets;
}
// if we're not stopping, re-enqueue the buffe so that it gets filled again
if ([aqr IsRunning])
AudioQueueEnqueueBuffer(inAQ, inBuffer, 0, NULL);
}
At first I thought AudioFileWritePackets works by directly writing the content of inBuffer->mAudioData. However when I manually write just the contents of mAudioData to a file the decoding doesn't seem to work.
On examining and comparing the raw data of what AudioFileWritePackets writes to a file and of me just writing mAudioData to a file, they seem to have attached a header to it.
As I can't find out how AudioFileWritePackets() works inside, my question is, how can I write the recorded audio to a buffer (or stream, if you prefer to call it this way) instead of a file, so that I can later send it over a network connection?
Again, in short here's what I need: record audio in aac format and stream the audio over network connection.
Thanks! I've been searching my head blue...
P.S: please if you point me to existing projects, make sure they are what I need. I really feel like I've been looking for all possible projects out there >.<
This should help.
Check the "HandleInputBuffer" method in "SpeechToTextModule.m".
I am building a "video wall" app in flash AS3. I am importing a movie clip with a flvPlayback component nested within, then Adding it to the display list 12 times in a for loop (which is based on the length of an xml file.) The xml file aslo points to the .source of the flv instance.
This method is working, for displaying video content on all screens BUT, it only loops the last flvPlayback component. The rest just go back to the first frame of the video.
var vidURL = vidXML.video_item[i].#url
SS.video.source = vidURL;
SS.video.autoRewind = true;
SS.video.autoPlay = true;
SS.video.addEventListener(VideoEvent.COMPLETE, Loop);
function Loop(event:VideoEvent):void
{
SS.video.play();
}
I have tried refering to the SS + [i] to call the event to rewind as soon as it happens (as the videos are different lengths) but have had no luck.
Any help would be appreciated.
Thanks
Jono
Don't worry chaps...
using the "event.target.play()" is triggered when each video finishes, and rewinds them all nicely.
Sorry.
Jono