How can I set a sound's volume?
Is it possible to play two sounds at once?
Can I know if the music is over using 'if~'?
Pass a value between 0 and 1 to the set_volume method of the sound, e.g.: sound.set_volume(0.5).
Just call the play methods of the sounds successively.
sound1.play()
sound2.play()
The default number of channels for sound playback is 8, but it can be changed with pygame.mixer.set_num_channels.
pygame.mixer.music.get_busy
Related
I want to built a SoundWave sampling an audio stream.
I read that a good method is to get amplitude of the audio stream and represent it with a Polygon. But, suppose we have and AudioGraph with just a DeviceInputNode and a FileOutpuNode (a simple recorder).
How can I get the amplitude from a node of the AudioGraph?
What is the best way to periodize this sampling? Is a DispatcherTimer good enough?
Any help will be appreciated.
First, everything you care about is kind of here:
uwp AudioGraph audio processing
But since you have a different starting point, I'll explain some more core things.
An AudioGraph node is already periodized for you -- it's generally how audio works. I think Win10 defaults to periods of 10ms and/or 20ms, but this can be set (theoretically) via the AudioGraphSettings.DesiredSamplesPerQuantum setting, with the AudioGraphSettings.QuantumSizeSelectionMode = QuantumSizeSelectionMode.ClosestToDesired; I believe the success of this functionality actually depends on your audio hardware and not the OS specifically. My PC can only do 480 and 960. This number is how many samples of the audio signal to accumulate per channel (mono is one channel, stereo is two channels, etc...), and this number will also set the callback timing as a by-product.
Win10 and most devices default to 48000Hz sample rate, which means they are measuring/output data that many times per second. So with my QuantumSize of 480 for every frame of audio, i am getting 48000/480 or 100 frames every second, which means i'm getting them every 10 milliseconds by default. If you set your quantum to 960 samples per frame, you would get 50 frames every second, or a frame every 20ms.
To get a callback into that frame of audio every quantum, you need to register an event into the AudioGraph.QuantumProcessed handler. You can directly reference the link above for how to do that.
So by default, a frame of data is stored in an array of 480 floats from [-1,+1]. And to get the amplitude, you just average the absolute value of this data.
This part, including handling multiple channels of audio, is explained more thoroughly in my other post.
Have fun!
i created an app which plays a playlist of small tracks every thing was working fine , till windows phone 8.1 update
the problem is -> there is weird tick sound" at track end
so i tried to play the track in xbox music player it also has the same tick ... i tried to play the audio at my pc and android device the audio is okay, so i think it's a wp8.1 issue or a comparability issue with my mp3 tracks
so, is there any specifications for the mp3 to be compatible with wp8.1?
or any work around in code, i was thinking a bout muting the sound before the track end , by the way i'm using AudioPlayerAgent
All audio rendering processes encounter this same challenge/problem. Root cause : sound is a curve and as it varies above/below centerline, (typically varies from -1 to 0 to +1 where centerline is 0), if it ends not close enough to the centerline this pop/tick sound happens, (speaker is left in the lurge not at 0 and will physically instantaneously return to 0 producing the tick). Solution : either the player ~helps~ the sound by artificially forcing the hand by ending the clip at the centerline or do similar as a preprocess step in the source media. This ending transition can happen quickly, yet not instantaneously, or you'd be back where you started with the instantaneous transition to 0. Silence is just when the media supplies a series of zeros (IE. at centerline).
During capturing from some audio and video sources and encoding at AVI container for synchronizing audio & video I set audio as a master stream and this gave best result for synchronizing.
http://msdn.microsoft.com/en-us/library/windows/desktop/dd312034(v=vs.85).aspx
But this method gives a higher FPS value as a result. About 40 or 50 instead of 30 FPS.
If this media file just playback - all OK, but if try to recode at different software to another video format appears out of sync.
How can I programmatically set dwScale and dwRate values in the AVISTREAMHEADER structure at AVI muxing?
How can I programmatically set dwScale and dwRate values in the AVISTREAMHEADER structure at AVI muxing?
MSDN:
This method works by adjusting the dwScale and dwRate values in the AVISTREAMHEADER structure.
You requested that multiplexer manages the scale/rate values, so you cannot adjust them. You should be seeing more odd things in your file, not just higher FPS. The file itself is perhaps out of sync and as soon as you process it with other applciations that don't do playback fine tuning, you start seeing issues. You might be having video media type showing one frame rate and effectively the rate is different.
Now I am working on a musical project in which i need accurate timings.I already used NSTimer,NSdate,but iam geting delay while playing the beats(beats i.e tik tok)So i have decided to use Audio Queue API to play my sound file present in the main bundle that is of .wav format, Its been 2 weeks i am struggling with this, Can anybody please help me out of this problem.
Use uncompressed sounds and a single Audio Queue, continuously running. Don't stop it between beats. Instead mix in raw PCM samples of each Tock sound (etc.) starting some exact number of samples apart. Play silence (zeros) in between. That will produce sub-millisecond accurate timing.
I am trying to write an application(I'm a gui first timer) for my son, he has autism. There is a video player in the top half and a text entry area in the bottom. When letters are typed sounds are produced to mimic the words in the video.
There have been other posts on this site in regard to playing sounds on key presses, using gstreamer as a system call. I have also tried libcanberra but both seem to have significant delays between sounds. I can write the app in python or C but will likely do at least some of it in C.
I also want to mention that the video portion is being played by gstreamer. I tried to create two instances of gstreamer, to avoid expensive system calls but the audio instance seemed to kill the app when called.
If anyone has any tips on creating faster responding sounds I would really appreciate it.
You can upload a raw audio sample directly to PulseAudio so there will be no decoding and (perhaps save) extra switches by using the following function from Canberra:
http://developer.gnome.org/libcanberra/unstable/libcanberra-canberra.html#ca-context-cache
The next ca_context_play() will use it.
However, the biggest problem you'll encounter with this scenario (with simultaneous video playback) is that the audio device might be configured with large latency with PulseAudio (up to 1/2s or more for normal playback). It may be reasonable to file a bug to libcanberra to support a LOW_LATENCY flag, as it currently doesn't attempt to minimize delay for sound events afaik. That would be great to have.
GStreamer pulsesink could probably get low latency too (it has some properties for that), but I am afraid it won't be as lightweight as libcanberra, and you won't be able to cache a sample for instance. Ideally, GStreamer could also learn to cache samples, or pre-fill PulseAudio...