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?
Related
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?
How to plot the detected faces on to the video frames from the result json of face detection and tracking. I mean, how to calculate the frame number for particular event in the Json file.
This give some details in case you hadn't seen it.
In essence, the video is divided into one or more fragments, and each fragment is divided in to intervals. There will one event per interval. The times and durations of fragments and intervals are expressed in ticks, which you can convert to time by dividing by the timescale. You can map frames to/from times using the framerate.
I've been searching all over and can't find a solution. I have a 25 FPS video that I'm playing on OSMF, but OSMF insists on playing with 29-31 FPS. This causes the video to play ~15% faster than real time. The result is extremely noticeable if you open the same video in VLC and play it side by side.
The problem comes in when I try to do a live stream. It will eat through the buffer and catch up to real time then the stream crashes because there's no new video waiting.
I've tried tracing the code to find out where the frames are actually output to the screen, but I hit a dead end at the SWC file. I also have tried searching online but I can't find anything about limiting the FPS - everyone is just interested in increasing it.
I'd rather play at 15 FPS and drop 10 frames per second than catch up to real time and crash tragically.
Edit - after an entire weekend spent staring at this issue I've made some incredible headway. First and foremost, the only way to limit the FPS in OSMF is by sending a custom FLV header with the timestamp set appropriately (1000 / FPS difference between each frame)
Realizing this I could solve this issue I'm having temporarily by manually setting the timestamps based on an internal counter. Each time a frame is processed set timestamp = last_timestamp + 40;. The problem is that I don't know if video will always be 25 FPS. Some day I may have 30 FPS or even 60 FPS video streams. To make this more robust I decided to decode the MPEG-2 header (read the PTS value) and convert it to an FLV header.
Now here's the issue… This video file (theoretically 25 FPS) plays perfectly in QuickTime. As a result I know the headers are fine because an expensive piece of software with billions of dollars behind it properly calculated the frame rate. But when I read the PTS from the header (as per this SO post) and divide by 90 (convert 90Khz clock to millisecond timestamp) each timestamp is 33 or 34 milliseconds apart - the 29~31 FPS I was getting.
So why is the PTS giving me timestamps that are 33-34 milliseconds apart when I know the video is 25 FPS (40 milliseconds apart)? More importantly, how is QuickTime reading the MPEG-2 header so that everything plays correctly?
First, to answer my original question:
Reducing FPS in OSMF
You must create your own implementation of the file handler which parses the video header and modifies the timestamps. There is no way to tell OSMF to play in "slow motion" or "fast forward". For an example of creating your own file handler, look at the FLVParser class. Notice that there are separate classes for parsing video and audio tags. The headers must be updated in EACH of these to ensure that video and audio play back in sync.
When a file is passed to the video parser, each timestamp is considered relative to the first. So if the first timestamp is 1234 then this will be set as "time zero" and all future timestamps will be relative to this. This is important. If you skip a video tag and the first timestamp you send is from later in the video, it will use the wrong value for "time zero" and things will not be sync'd properly.
This bring us to…
My issue
First and foremost, the durations in the M3U8 did not match up with the sum of the differences of timestamps. Starting at the first timestamp and labeling it "time zero", then looking at the last time stamp and subtracting time zero from it, the resulting time span was not equal to the expected duration of the TS file.
It turns out when VLC or QuickTime encounter this situation (the sum of PTS values does not equal the duration of the video) they generate new headers on the fly. Take the duration, divide by the number of frames, there's your new offset between PTS values.
That was my first problem. Once that was solved I was no longer gaining a second every 10 seconds, but instead gaining a second every 2 minutes. Turns out I was also loading the next TS file a bit too early (off-by-one error) which was causing me to drop a packet from each TS. This led to losing one frame every 10 seconds.
Additional Info
I ran into another issue with FPS shortly after this which I have found a solution for. Anyone experiencing OSMF playing videos too quickly, I urge you to grep your code for bufferTimeMax.
When bufferTimeMax > 0 and bufferLength >= bufferTimeMax, audio plays faster until bufferLength reaches bufferTime. If a live stream is video-only, video plays faster until bufferLength reaches bufferTime.
Depending on how much playback is lagging (the difference between bufferLength and bufferTime), Flash Player controls the rate of catch-up between 1.5% and 6.25%. If the stream contains audio, faster playback is achieved by frequency domain downsampling which minimizes audible distortion.
6.25% = 0.975 seconds per 15 (gaining a second every 10~20)
You would need to set the FPS manually.
In Flash Professional, you can change the framerate in Document Properties.
In Flash Builder, you can set the framerate using metadata, as described here and here.
Thus...
[SWF(frameRate='15')]
I am recording sound using Microphone class. After completing the record, I am getting a byte array.
Now I want to use this byte array and play the sound. Is this possible?
Thanks.
See this manual
In short, make the microphone generate 44.1 kHz samples (mono), duplicate them in your own sample data procedure, and store them in your Sound object. 44kHz are mandatory to not have your sound pitched, as it's always played back as 44.1kHz sound. The resampling is to make mono sound into stereo sound, as you can't have mono sounds in Flash.
I need to "look" at a NetStream and determine if I'm receiving audio. From what I investigated, i may use the property audioBytesPerSecond from NetStreamInfo:
"(audioBytesPerSecond) Specifies the rate at which the NetStream audio
buffer is filled in bytes per second. The value is calculated as a
smooth average for the audio data received in the last second."
I also learned that NetStream may have contain some overhead bytes from the network so, which is the minimum reasonable audioBytesPerSecond value to determine if NetStream is playing audio (and not just noise, for example)?
Can this analysis be done this way?
Thanks in advance!
Yes you can do it this way. It's rather subjective, however.
Try to find a threshold that works for you. We used 5 kilobits/sec in the past. If the amount of data falls below this value, they are likely not sending any audio. Note, we were using the stream.info.byteCount property (you might want a slightly lower value if you're using auiodBytesPerSecond).
This is pretty easy to observe if you speak into the microphone and periodically check audioBytesPerSecond or the other counters/statistics that are available.