Is it possible to rip an audio steam with Variable Bit Rate encoding and re encode it on the fly as it is being recorded with Constant Bit Rate encoding?
I am downloading an audio stream in AAC format with VBR encoding using cURL.
The duration of a VBR encoded file will be calculated by its byte length, resulting in a discrepancy for the duration on different players. The result of this duration discrepancy does not allow me to seek and slice precisely. I would need to re encode it somehow with a constant bit rate to get the seeking to work properly.
The audio stream is hours long so re encoding it afterwards takes way too much time and processing power.
Is there anything I can do about this?
Perhaps I can specify some settings in cURL to achieve a constant recording bit rate?
Related
I would like to use ffprobe to look at the information of media files. However, the files are not on my local disk, and I have to read from a remote storage. I can read the first n bytes, write them to a temporary file and use ffprobe to read the information. I would like to know the least such n.
I tested with a few files, and 512KB worked with all the files that I tested. However, I am not sure if that will work for all media files.
ffprobe (and ffmpeg) aims to parse two things when opening an input:
the input container's header
payload data from each stream, enough to ascertain salient stream parameters like codec attributes and frame rate.
The header size is generally proportional to the number of packets in the file i.e. a 3 hour MP4 file will have a larger header than a 3 min MP4.
(if the header is at the end of the file, then access to the first 512 kB won't help)
From each stream, ffmpeg will decode packets till its stream attributes have been populated. The amount of bytes consumed will depend on stream bitrate, and how many streams are present.
So, the strict response to 'I am not sure if that will work for all media files' is it won't.
I am beginner of video codec. not an video codec expert
I just want to know base on the same criteria, Comparing H254 encoding/decoding which is more efficiency.
Thanks
Decoding is more efficient. To be useful, decoding must run in real time, where encoding does not (except in videophone / conferencing applications).
How much more efficient? An encoder can generate motion vectors. The more compute power used on generating those motion vectors, the more accurate they are. And, the more accurate they are, the more bandwidth is available for the difference frames, so the quality goes up.
So, the kind of encoding used to generate video for streaming or distribution on DVD or BD discs can run many times slower than real time on server farms. But decoding for that kind of program is useless unless it runs in real time.
Even in the case of real-time encoding it takes more power (actual milliwatts, compute cycles, etc) than decoding.
It's true of H.264, H.265, VP8, VP9, and other video codecs.
I'm using NVENC SDK to encode OpenGL frames and stream them over RTSP. NVENC gives me encoded data in the form of several NAL units. In order to stream them with Live555 I need to find the start code (0x00 0x00 0x01) and remove it. I want to avoid this operation.
NVENC has a sliceOffset attribute which I can consult, but it indicates slices, not NAL units. It only points the ending of the SPS and PPS headers, where the actual data starts. I understand that a slice is not equal to a NAL (correct me if I'm wrong). I'm already forcing single slices for encoded data.
Is any of the following possible?
Force NVENC to encode individual NAL units
Force NVENC to indicate where the NAL units in each encoded data block are
Make Live555 accept the sequence parameters for streaming
There seems to be a point where every person trying to do H.264 over RTSP/RTP comes down to this question. Well here are my two cents:
1) There is a concept of an access unit. An access unit is a set of NAL units (may be as well be only one) that represent an encoded frame. That is the level of logic you should work at. If you are saying that you want the encoder to give you individual NAL unit's, then what behavior do you expect when the encoding procedure results in multiple NAL units from one raw frame (e.g. SPS + PPS + coded picture). That being said, there are ways to configure the encoder to reduce the number of NAL units in an access unit (like not including the AUD NAL, not repeating SPS/PPS, exclude SEI NAL's) - with that knowledge you can actually know what to expect and kind of force the encoder to give you single NAL per frame (of course this will not work for all frames, but with the knowledge you have about the decoder you can handle that). I'm not an expert on the NVENC API, I've also just started using it, but at least as for Intel Quick Sync, turning off AUD,SEI and disabling repetition of PPS/SPS gave me roughly around 1 NAL per frame for frames 2...N.
2) Won't be able to answer this since as I mentioned I'm not familiar with the API but I highly doubt this.
3) SPS and PPS should be in the first access unit (the first bit-stream you get from the encoder) and you could just find the right NAL's in the bit-stream and extract them, or there may be a special API call to obtain them from the encoder.
All that being said, I don't think it is that hard to actually run through the bit-stream, parse the start codes and extract the NAL unit's and feed them to Live555 one by one. Of course, if the encoder offers to output the bit-stream in the AVCC format (compared to the start codes or Annex B it uses interleaved length value between the NAL units so you can just jump to the next one without looking for the prefix) then you should use it. When it is just RTP it's easy enough to implement the transport yourself, since I've had bad luck with GStreamer that did not have proper support for FU-A packetization, in case of RTSP the overhead of the transport infrastructure is bigger and it is reasonable to use a 3rd party library like Live555.
Anyone know how configure the HLS media server for reduce a little bit the delay of live streaming video?
what types of parameters i need to change?
I had heard that you could do some tuning using parameters like this: HLSMediaFileDuration
Thanks in advance
A Http Live Streaming system typically has an encoder which produces segments of a certain number of seconds and a media server (web server) which serves playlists containing a list of URLs to these segments to player applications.
Media Files = Segments = .ts files = MPEG2-TS files (in HLS speak)
There are some ways to reduce the delay in HLS:
Reduce the encoded segment length from Apple's recommended 10 seconds to 5 seconds or less. Reducing segment length increases network overhead and load on the web server.
Use lower bitrates, larger .ts files take longer to upload and download. If you use multi-bitrate streams, make sure the first bitrate listed in the playlist is a little lower than the bitrate most of your users use. This will reduce the time to start playing back the stream
Get the segments from the encoder to the web server faster. Upload while still encoding if possible. Update the playlist as soon as the segment has finished uploading
Also remember that the higher the delay the better the quality of your stream (low delay = lower quality). With larger segments, there is less overhead so more space for video data. Taking a longer time to encode results in better quality. More buffering results in less chance of video streams stuttering on playback.
HLS is all about trading quality of playback for longer delay, so you will never be able to use HLS for things like video conferencing. Typical delay in HLS is 30-60 sec, minimum in practice is around 15 sec. If you want low delay use RTP for streaming, but good luck getting good quality on low or variable speed networks.
Please specify which media server you use. Generally speaking, yes - changing chunk size will definitely affect delay time. The less is the first chunk, the quicker the video will be shown in the player.
Actually, Apple recommend to divide your file to small chunks this equal length of file, integers.
In practice, there is huge difference between players. Some of them parse manifest changing this values.
Known practice is to pre-cache in memory first chunks in low & medium resolution (Or try to download them in background of app/page - Amazon does this, though their video is MSS)
I was having the same problem and the keys for me were:
Lower the segment length. I set it to 2s because I'm streaming on a local network. In other type of networks, you need to be careful with the overhead that a low segment length adds that can impact your playback quality.
In your manifest, make sure the #EXT-X-TARGETDURATION is accurate. From here:
The EXT-X-TARGETDURATION tag specifies the maximum Media Segment
duration. The EXTINF duration of each Media Segment in the Playlist
file, when rounded to the nearest integer, MUST be less than or equal
to the target duration; longer segments can trigger playback stalls
or other errors. It applies to the entire Playlist file.
For some reason, the #EXT-X-TARGETDURATION in my manifest was set to 5 and I was seeing a 16-20s delay. After changing that value to 2, which is the correct one according to my segments' length, I am now seeing delays of 6-10s.
In summary, you should expect a delay of at least 3X your #EXT-X-TARGETDURATION. So, lowering the segment length and the #EXT-X-TARGETDURATION value should help reducing the delay.
I am developing an AIR application for desktop that simulate a drum set. Pressing the keyboard will result in a corresponding drum sound played in the application. I have placed music notes in the application so the user will try to play a particular song.
Now I want to record the whole performance and export it to a video file, say flv. I have already succeed in recording the video using this encoder:
http://www.zeropointnine.com/blog/updated-flv-encoder-alchem/
However, this encoder does not have the ability to record sound automatically. I need to find a way to get the sound in ByteArray at that particular frame, and pass it to the encoder. Each frame may have different Sound objects playing at the same time, so I need to get the bytes of the final sound.
I am aware that SoundMixer.computeSpectrum() can return the current sound in bytes. However, the ByteArray returned has a fixed length of 512, which does not fit in the requirement of the encoder. After a bit of testing, with a sample rate 44khz 8 bit stero, the encoder expects the audio byte data array to have a length of 5880. The data returned by SoundMixer.computeSpectrum() is much much shorter than the encoder required.
My application is running at 60FPS, and recording at 15FPS.
So my question is: Is there any way I can obtain the audio bytes in the current frame, which is mixed by more than one Sound objects, and has the data length enough for the encoder to work? If there is no API to do that, I will have to mix the audio and get the result bytes by myself, how can that be done?