GPUImageMovieWriter frame presentationTime - iphone

I have a GPUImageColorDodgeBlend filter with two inputs connected:
A GPUImageVideoCamera which is getting frames from the iPhone video camera.
A GPUImageMovie which is an (MP4) video file that I want to have laid over the live camera feed.
The GPUImageColorDodgeBlend is then connected to two outputs:
A GPUImageImageView to provide a live preview of the blend in action.
A GPUImageMovieWriter to write the movie to storage once a record button is pressed.
Now, before the video starts recording, everything works OK 100% of the time. The GPUImageVideo is blended over the live camera video fine, and no issues or warnings are reported.
However, when the GPUImageMovieWriter starts recording, things start to go wrong randomly. About 80-90% of the time, the GPUImageMovieWriter works perfectly, there are no errors or warnings and the output video is written correctly.
However, about 10-20% of the time (and from what I can see, this is fairly random), things seem to go wrong during the recording process (although the on-screen preview continues to work fine).
Specifically, I start getting hundreds & hundreds of Program appending pixel buffer at time: errors.
This error originates from the - (void)newFrameReadyAtTime:(CMTime)frameTime atIndex:(NSInteger)textureIndex method in GPUImageWriter.
This issue is triggered by problems with the frameTime values that are reported to this method.
From what I can see, the problem is caused by the writer sometimes receiving frames numbered by the video camera (which tend to have extremely high time values like 64616612394291 with a timescale of 1000000000). But, then sometimes the writer gets frames numbered by the GPUImageMovie which are numbered much lower (like 200200 with a timescale of 30000).
It seems that GPUImageWriter is happy as long as the frame values are increasing, but once the frame value decreases, it stops writing and just emits Program appending pixel buffer at time: errors.
I seem to be doing something fairly common, and this hasn't been reported anywhere as a bug, so my questions are (answers to any or all of these are appreciated -- they don't all need to necessarily be answered sequentially as separate questions):
Where do the frameTime values come from -- why does it seem so arbitrary whether the frameTime is numbered according to the GPUImageVideoCamera source or the GPUImageMovie source? Why does it alternative between each -- shouldn't the frame numbering scheme be uniform across all frames?
Am I correct in thinking that this issue is caused by non-increasing frameTimes?
...if so, why does GPUImageView accept and display the frameTimes just fine on the screen 100% of the time, yet GPUImageMovieWriter requires them to be ordered?
...and if so, how can I ensure that the frameTimes that come in are valid? I tried adding if (frameTime.value < previousFrameTime.value) return; to skip any lesser-numbered frames which works -- most of the time. Unfortunately, when I set playsAtActualSpeed on the GPUImageMovie this tends to become far less effective as all the frames end up getting skipped after a certain point.
...or perhaps this is a bug, in which case I'll need to report it on GitHub -- but I'd be interested to know if there's something I've overlooked here in how the frameTimes work.

I've found a potential solution to this issue, which I've implemented as a hack for now, but could conceivably be extended to a proper solution.
I've traced the source of the timing back to GPUImageTwoInputFilter which essentially multiplexes the two input sources into a single output of frames.
In the method - (void)newFrameReadyAtTime:(CMTime)frameTime atIndex:(NSInteger)textureIndex, the filter waits until it has collected a frame from the first source (textureInput == 0) and the second, and then forwards on these frames to its targets.
The problem (the way I see it) is that the method simply uses the frameTime of whichever frame comes in second (excluding the cases of still images for which CMTIME_IS_INDEFINTE(frameTime) == YES which I'm not considering for now because I don't work with still images) which may not always be the same frame (for whatever reason).
The relevant code which checks for both frames and sends them on for processing is as follows:
if ((hasReceivedFirstFrame && hasReceivedSecondFrame) || updatedMovieFrameOppositeStillImage)
{
[super newFrameReadyAtTime:frameTime atIndex:0]; // this line has the problem
hasReceivedFirstFrame = NO;
hasReceivedSecondFrame = NO;
}
What I've done is adjusted the above code to [super newFrameReadyAtTime:firstFrameTime atIndex:0] so that it always uses the frameTime from the first input and totally ignores the frameTime from the second input. So far, it's all working fine like this. (Would still be interested for someone to let me know why this is written this way, given that GPUImageMovieWriter seems to insist on increasing frameTimes, which the method as-is doesn't guarantee.)
Caveat: This will almost certainly break entirely if you work only with still images, in which case you will have CMTIME_IS_INDEFINITE(frameTime) == YES for your first input'sframeTime.

Related

Syncing of buffer-transmission with ESP32, I2S MEMS-mic and SD-card (FreeRTOS, PlatformIO, ESP-PROG)

i know this forum dislikes "open" questions like this, nevertheless i'd like somebody to help untie the knot in my head, much appreciated.
The goal is simple:
read a stereo 32bit 44100 S/s I2S signal from 2 adafruit sph0645 mics
create a wav-header and store the data onto an SD-card
I've been at this for a few days now and i know that this will be much more complicated than i originally thought. Main reason: signal quality. Like most tutorials on this subject the simplest "hello world" for these mics is a looped polling for I2S-samples. Poll, fill buffer, output via serial or write to SD-card. This returns a choppy, noisy, sped up version of RL-audio. The filling of the internal DMA-buffers can be seen as constant, but the rest is mostly chaos, so
how to i sync these DMA-buffers with the rest of my code?
From experience with the STM32 HAL i'd imagine some register which can be set to throw an interrupt whenever a buffer is full, or an event which can be sent between tasks via queues. Examples on this subject either poll in a main loop with mono an abysmal sample-rate and bit depth or use pages of overkill code and never adress what it does, "just copy and it works", not good. Does the ESP32-Arduino framework provide some way to to this properly? The espressif-documentation isn't something to look forward to, since some of their I2S interface functions don't even work (if you are researching this topic as well, you too might have noticed that i2s_read only returns zeros). Just a hint into the right direction would help, i'm writing my own code anyway. Interrupts? Events? Timers? Polling for full buffers? Only you might know.
have a good one, thx
Thanks to https://github.com/atomic14/ i now have an answer for a syncing-method which works very well. This method has been tried by https://esp32.com/viewtopic.php?t=12546 who also didn't fully understand what was going on: the espressif i2s-interface offers a flag stored in an event which is triggererd every time one of the specified dma-buffers has received a full set of data, ergo, is full. It looks like this:
while(<your condition>){
i2s_event_t evt;
if (xQueueReceive(<your queue>, &evt, portMAX_DELAY) == pdPASS){
if (evt.type == I2S_EVENT_RX_DONE){
size_t bytesRead = 0;
do{
//read data via i2s_read or i2s_read_bytes
} while (bytesRead > 0);
No data is stored in this queue, but rather a flag which can then be used to synchronize dma-filling and further buffering/calculating/sending the read data.
HOWEVER this only works if you install the i2s driver in a specific setup. Instead of using
i2s_driver_install(I2S_NUM_0, &i2s_config, 0, NULL);
in your setup, you can activate the "affinity" for events by passing a queue-handle and a lenght:
i2s_driver_install(I2S_NUM_0, &i2s_config, 4, &<your queue>);
hope this helps getting started, it sure did help me.

Gapless playback in pyglet

I understood this page to mean that queuing in pyglet provides a gapless transition between audio tracks. But when I test it out, there is a noticeable gap. Has anyone here worked with gapless audio in pyglet?
Example:
player = pyglet.media.Player()
source1 = pyglet.media.load([file1]) # adding streaming=False doesn't fix the issue
source2 = pyglet.media.load([file2])
player.queue(source1)
player.queue(source2)
player.play()
player.seek([time]) # to avoid having to wait until the end of the track. removing this doesn't fix the gap issue
pyglet.app.run()
I would suggest you either edit your url1 and url2 into caching them locally if they're external sources. And then use Player().time to identify when you're about to reach the end. And then call player.next_source.
Or if it's local files and you don't want to programatically solve the problem you could chop up the audio files in something like Audacity to make them seamless on start/stop.
You could also experiment with having multiple players and layer them on top of each other. But if you're only interested in audio playback, there's other alternatives.
It turns out that there were 2 problems.
The first one: I should have used
source_group = pyglet.media.SourceGroup()
source_group.add(source1)
source_group.add(source2)
player.queue(source_group)
The second one: mp3 files are apparently slightly padded at the beginning and at the end, so that is where the gap is coming from. However, this does not seem to be an issue with any other file type.

Filtering an audio signal and then reading the meter without sending it to master

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

libspotify C sending zeros at the end of track

I'm using libspotify SDK, C library for win32.
I think to have a right setup, every session callback is registered. I don't understand why i can't receive the call for end_of_track, while music_delivery continues to be called with zero padding 22050 long frames.
I attempt to start playing first loading the track with sp_session_load; till it returns SP_ERROR_IS_LOADING I post a message on my message queue (synchronization method I've used, PostMessage win32 API) in order to reload again with same API sp_session_load. As soon as it returns SP_ERROR_OK I use the sp_session_play and the music_delivery starts immediately, with correct frames.
I don't know why at the end of track the libspotify runtime then start sending zero padded frames, instead of calling end_of_track callback.
In other conditions it works perfectly: I've used the sp_track obtained from a album browse, so the track is fully loaded at the moment I load to the current session for playing: with this track, it works fine with end_of_track called correctly. In the case with padding error, I search the track using its Spotify URI and got the results; in this case the track metadata are not still ready (at the play attempt) so I used that kind of "polling" on sp_session_load with PostMessage.
Can anybody help me?
I ran into the same problem and I think the issue was that I was consuming the data too fast without giving other threads time to do any work since I was spending all of my time in the music_delivery callback. I found that if I add some throttling and notify the main thread that it can wake up to do some processing, the extra zeros at the end of track is reduced to one delivery of 22,050 frames (or 500ms at 44.1kHz).
Here is an example of what I added to my callback, heavily borrowed from the jukebox.c example provided with the SDK:
/* Buffer 1 second of data, then notify the main thread to do some processing */
if (g_throttle > format->sample_rate) {
pthread_mutex_lock(&g_notify_mutex);
g_notify_do = 1;
pthread_cond_signal(&g_notify_cond);
pthread_mutex_unlock(&g_notify_mutex);
// Reset the throttle counter
g_throttle = 0;
return 0;
}
As I said, there was still 22,050 frames of zeros delivered before the track stopped, but I believe libspotify may purposely do this to ensure that the duration calculated by the number of frames received (song_duration_ms = total_frames_delivered / sample_rate * 1000) is greater than or equal to the duration reported by sp_track_duration. In my case, the track I was trying to stream was 172,000ms in duration, without the extra padding the duration calculated is 171,796ms, but with the padding it was 172,296ms.
Hope this helps.

Editing Timeline from CCB file in cocos

I did some research into this and couldn't really find anything, so if this is a repetitive question I apologize. but anyway I have made a CCB file in CocosBuilder and I would like to start the timeline, for example, at one second instead of playing from the beginning. Is there a way to do this? Thanks for the help guys.
Edit: i would like this to be done in the code.
I am using 2.2.1 Cocos2DX version. I think there is no option to play it from given interval. But you can tweak yourself to get it done. (Not simple one)
You have to go to CCBAnimationManager and there you get "mNodeSequences".
It is dictionary and you get difference properties there like "rotation position etc..."
values there.
Internally AnimationManager reads this value (These values are specified in your CCB)
and puts in runAction queue.
So you have to break it as you want.(Ex. 5 min timeline you have. But you want to start
from 1 min then you have run first 1 min Actions without delay and for remaining you
have properly calculate tween intervals.
It's long procedure and needs calculation. If you don't know any other simpler way try this. If you know pls let us know (Post it).