I'm trying to use GDCL MP4 Muxer with my RTSP Source Filter. They work fine together except after stopping the graph, muxer doesn't finilize the file and write the reqiured tables to the end of file via file writer (some parts are written starting from moov but not the time table values). When I try another RTSP source filter (which I don't have its source codes), table values are created with GDCL MP4 Muxer.
But when I try Elecard's MP4 Muxer, it works fine with my RTSP Source Filter. So, there is an incompatibility. I examined GDCL's source codes but couldn't find what it was expecting from me. I already calculate and set timestamp values to samples using SetTime method. But GDCL still doesn't finilaze file. Is it caused by missing information or missing signal when graph stops? What can be the problem, any ideas?
One thing you should be aware of regarding Geraint's MP4 Mux is that it is checking incoming media samples to have both start and stop time. You might be having only .tStart/AM_SAMPLE_TIMEVALID which still makes sense for video, but this would be a problem.
So the samples have to have stop time, or you need to fix this in multiplexer code.
A typical symptom for the problem is that generated files are empty or of zero duration.
Related
I have fragmented mp4 which I want to send users through HLS. It's ok if I just send it as is. But I need opportunity to reorder fragments in this video.
For example initial video, which looks like this:
original video format
I want reorganize fragments and get this:
expected video format
I try make it locally, and it's work in VLS player (HLS). For this I modified sequence number for fragments in moof (mfhd). But when I try play it remotely (HLS) it does not work. I think, that some players (js) expect some additional information from each fragment, probably for example time offset. But I can not find which atom (box) contain this information. I spent a lot of time searching and I'm still at the very beginning of the problem.
I tried to modify the fragment sequence number, but it doesn't work.
The "Track Fragment Media Decode Time Box" (tfdt) stores the baseMediaDecodeTime which is the accumulative decode time.
Consider the following...
baseMediaDecodeTime must increase monotonically for each chunk.
This means you must update (replace) the tfdt entry of chunk with expected next tftd entry.
When you naively reorder the chunks, the baseMediaDecodeTime will be invalid.
The "Track Fragment Media Decode Time Box" (tfdt) is located inside each moof header at:
moof --> traf --> tfdt
I would like to fill an array from a stream for around ten seconds.{I wish to do some processing on the data)So far I can:
(a) obtain the microphone stream using mediaRecorder
(b) use analyser and analyser.getFloatTimeDomainData(dataArray) to obtain an array but it is size limited to only a little over half a second of data.I can also successfully output the data after processing back onto a stream and to outDestination.
(c) I have also experimented with obtaining a 'chunks' array from mediaRecorder directly but the problem then is that I can't find any mime type that would give me a simple array of values - ie an uncompressed sample by sample single channel set of value - ie a longer version of 'dataArray' in (b).
I am wondering if I am missing a simple way round this problem?
Solutions I have seen tend to use step (b) and do regular polls then reassemble a longer array - however it seems the timing is a bit tricky ..
I'v also seen suggestions to use audio workouts - I might have to do this but would prefer a simpler solution!
Or again, if someone knows how to drive mediaRecorder to output the chunks array in a simple array format FLOAT32.of one channel.That would do the trick.
Or maybe I'm missing something simpler?
I have code showing those steps that have been successful and will upload if anyone requests.
I'm trying to filter a signal and then analyse the values of the filtered signal using Tone.js / Web-Audio API.
I'm expecting to get values of the filtered signal, but I only get -Infinity, meaning that my connections between the nodes are wrong. I've made a small fiddle demonstrating this, however in my use-case I do not want to send this node to the destination of the context - I only want to analyse it, not hear it.
osc.connect(filter)
filter.connect(gainNode)
gainNode.connect(meter)
console.log(meter.getLevel())
I guess you tested the code in Chrome because there is a problem with Chrome which causes it to not process anything until it is connected to the destination. When using Tone.js that means you need to call .toMaster() at the end of your chain. I updated you fiddle to make it work: https://jsfiddle.net/8f7abzoL/.
In Firefox calling .toMaster() is not necessary therefore the following works in Firefox as well: https://jsfiddle.net/yrjgfdtz/.
After some digging I've found out that I need to have a scriptProcessorNode - which is apparently no longer recommended - so looking into Audio Worklet Nodes
I'm attempting to stream a H.264 video feed to a web browser. Media Foundation is used for encoding a fragmented MPEG4 stream (MFCreateFMPEG4MediaSink with MFTranscodeContainerType_FMPEG4, MF_LOW_LATENCY and MF_READWRITE_ENABLE_HARDWARE_TRANSFORMS enabled). The stream is then connected to a web server through IMFByteStream.
Streaming of the H.264 video works fine when it's being consumed by a <video src=".."/> tag. However, the resulting latency is ~2sec, which is too much for the application in question. My suspicion is that client-side buffering causes most of the latency. Therefore, I'm experimenting with Media Source Extensions (MSE) for programmatic control over the in-browser streaming. Chrome does, however, fail with the following error when consuming the same MPEG4 stream through MSE:
Failure parsing MP4: TFHD base-data-offset not allowed by MSE. See
https://www.w3.org/TR/mse-byte-stream-format-isobmff/#movie-fragment-relative-addressing
mp4dump of a moof/mdat fragment in the MPEG4 stream. This clearly shows that the TFHD contains an "illegal" base data offset parameter:
[moof] size=8+200
[mfhd] size=12+4
sequence number = 3
[traf] size=8+176
[tfhd] size=12+16, flags=1
track ID = 1
base data offset = 36690
[trun] size=12+136, version=1, flags=f01
sample count = 8
data offset = 0
[mdat] size=8+1624
I'm using Chrome 65.0.3325.181 (Official Build) (32-bit), running on Win10 version 1709 (16299.309).
Is there any way of generating a MSE-compatible H.264/MPEG4 video stream using Media Foundation?
Status Update:
Based on roman-r advise, I managed to fix the problem myself by intercepting the generated MPEG4 stream and perform the following modifications:
Modify Track Fragment Header Box (tfhd):
remove base_data_offset parameter (reduces stream size by 8bytes)
set default-base-is-moof flag
Add missing Track Fragment Decode Time (tfdt) (increases stream size by 20bytes)
set baseMediaDecodeTime parameter
Modify Track fragment Run box (trun):
adjust data_offset parameter
The field descriptions are documented in https://www.iso.org/standard/68960.html (free download).
Switching to MSE-based video streaming reduced the latency from ~2.0 to 0.7 sec. The latency was furthermore reduced to 0-1 frames by calling IMFSinkWriter::NotifyEndOfSegment after each IMFSinkWriter::WriteSample call.
There's a sample implementation available on https://github.com/forderud/AppWebStream
I was getting the same error (Failure parsing MP4: TFHD base-data-offset not allowed by MSE) when trying to play a fmp4 via MSE. The fmp4 had been created from a mp4 using the following ffmpeg comand:
ffmpeg -i myvideo.mp4 -g 52 -vcodec copy -f mp4 -movflags frag_keyframe+empty_moov myfmp4video.mp4
Based on this question I was able to find out that to have the fmp4 working in Chrome I had to add the "default_base_moof" flag. So, after creating the fmp4 with the following command:
ffmpeg -i myvideo.mp4 -g 52 -vcodec copy -f mp4 -movflags frag_keyframe+empty_moov+default_base_moof myfmp4video.mp4
I was able to play successfully the video using Media Source Extensions.
This Mozilla article helped to find out that missing flag:
https://developer.mozilla.org/en-US/docs/Web/API/Media_Source_Extensions_API/Transcoding_assets_for_MSE
The mentioned 0.7 sec latency (in your Status Update) is caused by the Media Foundation's MFTranscodeContainerType_FMPEG4 containterizer which gathers and outputs each roughly 1/3 seconds (from unknown reason) of frames in one MP4 moof/mdat box pair. This means that you need to wait 19 frames before getting any output from MFTranscodeContainerType_FMPEG4 at 60 FPS.
To output single MP4 moof/mdat per each frame, simply lie that MF_MT_FRAME_RATE is 1 FPS (or anything higher than 1/3 sec). To play the video at the correct speed, use Media Source Extensions' <video>.playbackRate or rather update timescale (i.e. multiply by real FPS) of mvhd and mdhd boxes in your MP4 stream interceptor to get the correctly timed MP4 stream.
Doing that, the latency can be squeezed to under 20 ms. This is barely recognizable when you see the output side by side on localhost in chains such as Unity (research) -> NvEnc -> MFTranscodeContainerType_FMPEG4 -> WebSocket -> Chrome Media Source Extensions display.
Note that MFTranscodeContainerType_FMPEG4 still introduces 1 frame delay (1st frame in, no output, 2nd frame in, 1st frame out, ...), hence the 20 ms latency at 60 FPS. The only solution to that seems to be writing own FMPEG4 containerizer. But that is order of magnitude more complex than intercepting of Media Foundation's MP4 streams.
The problem was solved by following roman-r's advise, and modifying the generated MPEG4 stream. See answer above.
Another way to do this is again using the same code #Fredrik mentioned but I write my own IMFByteStream and and I check the chunks written to the IMFByteStream.
FFMpeg writes the atoms almost once at a time. So you can check the atom name and do the mods. It is the same thing. I wish there was an MSE compliant windows sinker.
Is there one that can generate .ts files for HLS?
We are seeing this when "tying" two video files together.
Example we have Ad video that is segmented and content file which is also segmented.
We create a new file which has both Ad and content segment information together. However we are seeing an issue where either the Ad content is truncated or the content starts having A/V sync issues.
Both ad and content are segmented the same way , 5 sec segmentation. however since Ads are variable length the result file may have left over segment something like:
#EXTM3U
#EXT-X-TARGETDURATION:5
#EXT-X-MEDIA-SEQUENCE:0
#EXTINF:5,
fileSequence6.ts
#EXTINF:5,
fileSequence7.ts
#EXTINF:4,
fileSequence8.ts
#EXTINF:5,
fileSequence0.ts
#EXTINF:5,
fileSequence1.ts
#EXTINF:5,
fileSequence2.ts
#EXTINF:3,
fileSequence3.ts
Is this the proper way to play 2 files one after the other without rebuffering?
should generate-variant-plist be used to a play list of 2 files?
When you have a break in the stream to switch to a commercial, ad, or alternate video source then you want to introduce the discontinuity tag before the start of the next segment, for example:
#EXTM3U
#EXT-X-TARGETDURATION:5
#EXT-X-MEDIA-SEQUENCE:0
#EXTINF:5,
movie0.ts
#EXTINF:2,
movie1.ts
#EXT-X-DISCONTINUITY
#EXTINF:5,
commercial0.ts
#EXTINF:5,
commercial1.ts
#EXTINF:3,
commercial2.ts
This gets a little more complicated if you encrypt the streams because they use progressive encryption based on the prior segments encryption state and the sequence number which come together to form an "Initialization Vector". If you break the stream you have to reset the initialization vector so that the encryption/decryption can continue uninterrupted. This is an involved process so best to just search on Initialization Vector in Apple's docs.