I'm trying to change the max buffer length on my video streaming in clappr video player.
I know that in HLS format the way to do it is like this:
player = new Clappr.Player({
playback: {
hlsjsConfig: {
maxMaxBufferLength: 30
}}})
And it's realy working for HLS videos,
I'm looking for equivalent way to do it with MPEG-dash foramt
How are you playing DASH in Clappr?
If you are using Shaka, https://github.com/clappr/dash-shaka-playback, set it up as shown at https://github.com/clappr/dash-shaka-playback, using the buffer settings you require as described at https://github.com/google/shaka-player/blob/master/docs/tutorials/network-and-buffering-config.md#buffering-configuration
Eg:
player = new Clappr.Player({
source: '//storage.googleapis.com/shaka-demo-assets/angel-one/dash.mpd',
plugins: [DashShakaPlayback],
shakaConfiguration: {
preferredAudioLanguage: 'pt-BR',
streaming: {
bufferingGoal: 30,
rebufferingGoal: 15,
bufferBehind: 60
}
}
});
Related
Firefox is limited in its audio resampling ability for audio mediastreams. If the input media stream's sample rate is not the same as the AudioCotext's, then it complains :
DOMException: AudioContext.createMediaStreamSource: Connecting AudioNodes from AudioContexts with different sample-rate is currently not supported.
For example if we get an audio stream like so :
navigator.mediaDevices.getUserMedia(constraints).then(stream => {
let context = new (window.AudioContext || window.webkitAudioContext)({sampleRate : 48000});
let audioInput = this.context.createMediaStreamSource(stream);
});
Firefox will complain about mismatching sample rates - if they are different between the audio context and the hardware device's settings in the audio subsystem.
I can't find a way to get the sample rate from the audio track in the stream. I've tried :
let tracks = stream.getAudioTracks();
let settings = tracks[0].getSettings();
let constraints = tracks[0].getConstraints();
But none of these objects have the streams's sampleRate in them.
Is there another way to enquire an audio track's/stream's sample rate ?
I'm writing a filter in ffmpeg and I have to get the duration or number of frames in filter_frame function. But the parameters of "filter_frame" function only consist of AVFilterLink and AVFrame:
static int filter_frame(AVFilterLink* link, AVFrame* in) {
AVFilterContext* avctx = link->dst;
ExtractContext* privCtx = (ExtractContext*)avctx->priv;
AVFilterLink* outlink = avctx->outputs[0];
//here I want to get the duration of total number of frames:
}
I can't find attributes which is related to duration of frame numbers in either class. I have googles but all answers are about getting duration or frame numbers using ffmpeg.exe, which don't meet my requirements because I have to get them in self-written filter in FFMPEG. Thanks for your help!
It seems that MediaSource and Progressive playback use the different demuxer. ChunkDemuxer is used for MediaSource, ShellDemuxer is used for Progressive playback.
In ShellParser.cpp implementation:
PipelineStatus ShellParser::Construct(
scoped_refptr<ShellDataSourceReader> reader,
scoped_refptr<ShellParser>* parser,
const scoped_refptr<MediaLog>& media_log) {
DCHECK(parser);
DCHECK(media_log);
*parser = NULL;
// download first 16 bytes of stream to determine file type and extract basic
// container-specific stream configuration information
uint8 header[kInitialHeaderSize];
int bytes_read = reader->BlockingRead(0, kInitialHeaderSize, header);
if (bytes_read != kInitialHeaderSize) {
return DEMUXER_ERROR_COULD_NOT_PARSE;
}
// attempt to construct mp4 parser from this header
return ShellMP4Parser::Construct(reader, header, parser, media_log);
}
It seems that Cobalt can only demux MP4 container(Only ShellMP4Parser) for progressive playback.
Is it known status for Cobalt ?how can we support webm progressive playback on the device?
Cobalt will not support WebM/VP9 progressive playback. We changed the Progressive Conformance test to change VP9 to H264. This will be pushed soon.
https://github.com/youtube/js_mse_eme/commit/d7767e13be7ed8b8bdb2efda39337a4a2fb121ba
I get a input frame stream through a socket, it is a mono 32-bit IEEE floating point PCM stream sampled at 16 kHz.
I get this with the following code : audio File sample
With Audacity i can visualize this and i see a regular cuts between my audio flux:
var audioCtx = new(window.AudioContext || window.webkitAudioContext)();
var audioBuffer = audioCtx.createBuffer(1, 256, 16000);
var BufferfloatArray;
var source = audioCtx.createBufferSource();
source.buffer = audioBuffer;
var gainNode = audioCtx.createGain();
gainNode.gain.value = 0.1;
gainNode.connect(audioCtx.destination);
source.connect(gainNode);
source.start(0);
socket.on('audioFrame', function(raw) {
var context = audioCtx;
BufferfloatArray = new Float32Array(raw);
var src = context.createBufferSource();
audioBuffer.getChannelData(0).set(BufferfloatArray);
src.buffer = audioBuffer;
src.connect(gainNode);
src.start(0);
});
I think it is because of the sample rate of my raw buffer (16000) is different of the sample rate of my Audio context (44100), what do you think ?
This is not a sample rate problem, because the AudioBufferSourceNode resamples the audio to the AudioContext's rate when playing.
What you should do here, is to have a little queue of buffers you feed using the network, and then, play your buffers normally like you do, but from the buffer queue, taking extra care to schedule them (using the first parameter of the start method of the AudioBufferSourceNode) at the right time, so that the end of the previous buffer is exactly the start of the next one. You can use the AudioBuffer.duration parameter to achieve this (duration is in seconds).
I am using the following code snippets to record screen, and in most situations recorded wmv file is clear enough, but for some part of video it is not very clear (grey color for some parts). What I record is ppt with full screen mode. I am using Windows Media Encoder 9.
Here is my code snippet,
IWMEncSourceGroup SrcGrp;
IWMEncSourceGroupCollection SrcGrpColl;
SrcGrpColl = encoder.SourceGroupCollection;
SrcGrp = (IWMEncSourceGroup)SrcGrpColl.Add("SG_1");
IWMEncVideoSource2 SrcVid;
IWMEncSource SrcAud;
SrcVid = (IWMEncVideoSource2)SrcGrp.AddSource(WMENC_SOURCE_TYPE.WMENC_VIDEO);
SrcAud = SrcGrp.AddSource(WMENC_SOURCE_TYPE.WMENC_AUDIO);
SrcVid.SetInput("ScreenCap://ScreenCapture1", "", "");
SrcAud.SetInput("Device://Default_Audio_Device", "", "");
// Specify a file object in which to save encoded content.
IWMEncFile File = encoder.File;
string CurrentFileName = Guid.NewGuid().ToString();
File.LocalFileName = CurrentFileName;
CurrentFileName = File.LocalFileName;
// Choose a profile from the collection.
IWMEncProfileCollection ProColl = encoder.ProfileCollection;
IWMEncProfile Pro;
for (int i = 0; i < ProColl.Count; i++)
{
Pro = ProColl.Item(i);
if (Pro.Name == "Screen Video/Audio High (CBR)")
{
SrcGrp.set_Profile(Pro);
break;
}
}
encoder.Start();
thanks in advance,
George
I would guess that it's a problem with your encoder profile or settings, and not a problem with the code. If you're using the default "Screen Video/Audio High (CBR)" profile in WME9, it's using a video bitrate of 250Kbps, which is pretty low. I'd suggest creating a custom profile in the Windows Media Encoder Profile Editor Utility. Something like this:
awesomesc.prx
Name: Awesome Screen Profile
Audio: WMA 9.2 CBR (32kbps, 44kHz, mono CBR)
Video: WMV 9 Screen Quality VBR (Video size Same as video input, Frame rate 10fps, Key frame interval 3sec, Video quality 90)
Then just change the code to match the custom profile's name.
if (Pro.Name == "Awesome Screen Profile")
The encoder settings would take a much longer post to go through, but if you have not changed them from the defaults, you should be OK.
The Quality-based VBR algorithm can be pretty amazing, and will likely produce a surprisingly low average bitrate, but if VBR won't work for your needs, you can use the Windows Media Encoder Profile Editor utility to import the schia.prx profile that you're using and tweak the settings to find a higher CBR bitrate that produces acceptable quality.
"Screen Video/Audio Medium (CBR)"
it solved my problem