Based on the setting of the configuration parameters of MPEG-2 codec I realized that it can encode HD video format but I am still in doubt. Is anybody sure if it is possible?
Why are you doubting it?
http://en.wikipedia.org/wiki/MPEG-2
https://www.google.com/?q=MPEG-2%20codec%20+%20HD
Related
I want to make a app that record video using webcam,
I made the logic like get each frame as bitmap and store it to file using
AForge VideoFileWriter WriteVideoFrame function,
When I open then file using VideoFileWriter Open function,
writer.Open(path, VideoWidth, VideoHeight, frameRate, VideoCodec.H264, bitRate);
It is hard to determine the bitRate, when the bitrate is wrong, the whole program die without any error.
I think the bitrate is related to video frame width, height, framerate, bitcount as well as codec,
But I not sure the specific formular to calculate.
I want to compress the video using h264 codec.
Can anyone help me to find out the solution?
Thank you very much.
In my previous question I'd asked for a convenient codec which supports intra frames and prevents onion skin animation. Now I'm using the UtVideo codec.
Codec which supports intra-frame
Which UtVideo version shall I use?
And how get I the right setting for capturing a video frame by frame?
All variants are good but correspond to different YUV matrix/sampling: normally you should know what your capture source is and select the same.
If you capture screen there should be UtVideo RGB option, not sure why you dont have it. Try latest installer http://umezawa.dyndns.info/wordpress/?p=6107
I'm trying to decode array of bytes from DSP TrueSpeech to PCM.
When we convert this array as part of streaming (divide it to packets) we can hear some strange "Beep" tones after the decoding.
We tried to decode the entire WAV file in one piece and we didn't get those Beeps.
Currently we are using Alvas.net for it, but we tried also with NAudio and got the same reaults?
My questions:
1)Is anyone familiar with this kind of behavior?
2)Do you have an idea what can we do?
Thanks
Ziv
How are you performing the decode? Often codecs maintain internal state, so it's important that you don't keep closing and re-opening the codec for each block of audio that you receive. In NAudio, that means just one AcmStream/WaveFormatConversionStream that everything you receive is passed through.
Also, make sure it is only compressed audio that is being passed into the codec. Sometimes when you receive audio over the network it is contained within some kind of larger packet that contains timing or encoding metadata (e.g. RTP).
At the bottom line, we have the packet data(array of bytes) which we are sending to decode (return as PCM) and then we're writing the new decoded array of bytes in to the new WAV file.
We're defiantly going to try your suggestion regarding the stream with NAudio.
Regarding the bytes we're working on, they don't contain any garbage. We've wrote a tester that stream the file directly (without network) and got the same beep results.
Our solution is working so well with many other codecs (GSM and etc..) and only in true speech we're having this problem.
Therefore it seems to be like some behavior of True Speech codec, but we didn't find any documentation about it.
Thanks Again
Ziv
I have an original video coded at 20Mbps, 1920x1080, 30fps and want to convert it down to be 640x480 30fps at a range of (3 different) bitrates for use by Adobe Live Streaming.
Should I use ffmpeg to resize and encode at the 3 bitrates then use f4fpackager to create the f4m f4f and f4x files or just use ffmpeg to reduce the resolution and then f4fpackager to encode the relevant bitrates?
I've had several tries so far, but when encoded the videos seem to play at a much larger bitrate than they've been encoded at. For example, if I set up the OSMF to play from my webserver, I'd be expecting my best encoded video to play at 1,500kbps but it's way above that.
Has anyone had any experience of encoding for use like this?
I'm using the following options to f4fpackager
--bitrate=1428 --segment-duration 30 --fragment-duration 2
f4fpackager doesn't do any encoding, it does 2 things:
- fragment the mp4 files (mp4 -> f4f)
- generate a Manifest (f4m) file referencing all you fragmented files (f4f)
So the process is:
- transcode your source file in all the size/bitrate that you want to provide (eg: 1920x01080#4Mbps, 1280x720#2Mbps, etc)
- use f4fpackager to convert the mp4 to f4f (this is the fragmentation step)
- use f4fpackager to generate the Manifest.f4m referencing the files that you generated in the previous step
the --bitrate option of f4fpackager should match the value that you use with ffmpeg, this parameter is used to generate the manifest file with the correct bitrate value of each quality
FMLE = Flash live media encoder 3.0
i have posted this question on Adobe Forum, but not sure if they have people on that forum with programming experience.
I am a developer writing a video capture and audio capture device. The devices already work in other encoders. The devices are written in directshow. I am integrating with FMLE and encountered this problem.
The audio device doesnt have a usable volume bar in FMLE. The FMLE error is "The selected audio device "censored (company secret)" doesn't allow setting volume intensity. Disabling the volume slider control."
my audio device implements these interfaces along with the standard directshow filter interfaces
IBasicAudio
IAMAudioInputMixer
I put tracepoints in queryinterface and found FMLE query's for (my comments in comment string)
{IID_IUnknown}
{IID_IPersistPropertyBag}
{IID_IBaseFilter}
{IID_IAMOpenProgress}
{IID_IAMDeviceRemoval}
{IID_IMediaFilter}
{IID_IAMBufferNegotiation}
{IID_IAMStreamConfig}
{IID_IPin}
{IID_IReferenceClock}
{IID_IMediaSeeking}
{IID_IMediaPosition}
{IID_IVideoWindow} // WTF ?? query video window ?
{IID_IBasicAudio}
{2DD74950-A890-11D1-ABE8-00A0C905F375} // i think this is async stream,
What am i missing ? FMLE doesnt use IAMAudioInputMixer ?
Anyone know the exact interface which FMLE uses for Volume intensity ? . .I assumed it was IBasicAudio, but it doesnt seem to call any methods in there.
Answer provided by Ram Gupta of adobe forum.
"FMLE does not query for CLSID_AudioInputMixerProperties interface.
FMLE enumerates all the pin of audio source filter(using EnumPins) and then it extracts each pin info using QueryPinInfo Function.
FMLE searches for the audio filter Pin whose direction is PINDIR_INPUT(using QueryPinInfo) and then it queries for IAMAudioInputMixer interface to set the volume level.
Could you pls chk if the following functions are properly implemented
-->get_enable: it should set its parameter value to true.
-->put_MixLevel
-->QueryPinInfo:"
This solution did work. My problem was that because i never declared an input pin (since i dont have any directshow related input).