My MP4 file issue is a bit complicated, so I have created a very simple scenario to help me diagnose it step by step.
I can create a working MP4 file that works flawlessly. The following is its structure shown by Mp4Explorer:
For debugging purpose, I removed the media data box mdat, and al stts, stsz,stss, stsc, stco boxes and kept everything else the same. This means the MP4 file has no media data. It just has some metadata.
This file is named error.mp4. If I run:
ffprobe "error.mp4"
I get the following error:
[mov,mp4,m4a,3gp,3g2,mj2 # 00000286e763ef00] Could not find codec parameters for stream 0 (Video: h264 (avc1 / 0x31637661), none, 1280x800): unspecified pixel format
Consider increasing the value for the 'analyzeduration' (0) and 'probesize' (5000000) options
Could anyone shed some light on this error? Why is it "Could not find codec parameters for stream 0"? If I remove the video track and leave the audio track as is, ffprobe will be happy with no errors.
Related
I followed the top answer in this StackOverflow post to use ffmpeg-python extract a .wav file from a YouTube URL (into the pcm_s16le codec), which was played successfully by my local audio player (Mac's Music).
However, as I tried to read it using scipy.io's wavefile,
samplerate, data = wavfile.read(wav_fname)
the following error message is thrown:
"WavFileWarning: Reached EOF prematurely; finished at 1192015 bytes, expected 4294967303 bytes from header."
May anyone suggest what's going on?
I have successfully extracted a .wav file which is successfully read by my local music player. However, it is failed to be recognized by scipy.io's wavefile. And I am not sure why.
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?
How to create an Mp4 file from H264 raw data that I am receiving from a live streamer (no predefined duration or moov atom), unfortunately can't use FFMPEG, I have to write my own code using live555. Can somebody help me with Mp4 container and how h264 data has to be pushed into it.? Thank you in advance : )
There are several operations to be made to store H.264 raw data into MP4, among them:
create box structures, in particular the moov box
store the NAL units in a mdatbox, possibly storing non-VCL NAL units in the moovbox
replace start codes with length fields
It also depends on your requirements. If you want to do the conversion on-the-fly, you have to use fragmented mp4. If you can store the H264 and then do the conversion, you may use non-fragmented mp4. In particular using MP4Box:
MP4Box -add file.264 file.mp4
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.
I am trying to encode video in h.264 that when split with Apples HTTP Live Streaming tools media file segmenter will pass the media file validator I am getting two errors on the split MPEG-TS file
WARNING: Media segment contains a video track but does not contain any IDR access unit with a SPS and a PPS.
WARNING: 7 samples (17.073 %) do not have timestamps in track 257 (avc1).
After hours of research I think the "IDR" warning relates to not having keyframes in the right place on the segmented MPEG-TS file so in my ffmpeg command I set -keyint_min 1 to ensure keyframes where at every frame, but this didn't work.
Although it would be great to get an answer, if anyone can shed any light on what a "IDR access unit with a SPS and a PPS" is or what the timestamps warning means I would be very grateful, thanks.
Fix can be found on this thread https://devforums.apple.com/thread/45830?tstart=15