if (key == 'k')
{
player = minim.loadFile("abc.wav");
player.amp(volume[levelOfVolume]);
}
I try to use Minim to play an audio, and alsp i want to set a key which can change the volume, however, the volume function is not working
I think you should try setGain() or shiftGain(), as described in the links^.
Related
I'm quite new to Processing.
I'm trying to make Processing randomly play a video after I clear the screen by mouseclick, so I create an array that contain 3 videos and play one at a time.
Holding 'Spacebar' will play a video and release it will stop the video. Mouseclick will clear the screen to an image. The question is how can it randomize to another video if I press spacebar again after clear the screen.
I've been searching all over the internet but couldn't find any solution for my coding or if my logic is wrong, please help me.
Here's my code.
int value = 0;
PImage photo;
import processing.video.*;
int n = 3; //number of videos
float vidN = random(0, n+1);
int x = int (vidN);
Movie[] video = new Movie[3];
//int rand = 0;
int index = 0;
void setup() {
size(800, 500);
frameRate(30);
video = new Movie[3];
video[0] = new Movie (this, "01.mp4");
video[1] = new Movie (this, "02.mp4");
video[2] = new Movie (this, "03.mp4");
photo = loadImage("1.jpg");
}
void draw() {
}
void movieEvent(Movie video) {
video.read();
}
void keyPressed() {
if (key == ' ') {
image(video[x], 0, 0);
video[x].play();
}
}
void mouseClicked() {
if (value == 0) {
video[x].jump(0);
video[x].stop();
background(0);
image(photo, 0, 0);
}
}
You have this bit of logic in your code which picks a random integer:
float vidN = random(0, n+1);
int x = int (vidN);
In theory, if you want to randomise to another video when the spacebar is pressed again you can re-use this bit of logic:
void keyPressed() {
if (key == ' ') {
x = int(random(n+1));
image(video[x], 0, 0);
video[x].play();
}
}
(Above I've used shorthand of the two lines declaring vidN and x, but the logic is the same. If the logic is harder to follow since two operations on the same line (picking a random float between 0,n+1 and rounding down to an integer value), feel free to expand back to two lines: readability is more important).
As side notes, these bit of logic look a bit off:
the if (value == 0) condition will always be true since value never changes, making both value and the condition redundant. (Perhaps you plan to use for something else later ? If so, you could save separate sketches, but start with the simplest version and exclude anything you don't need, otherwise, in general, remove any bit of code you don't need. It will be easier to read, follow and change.)
Currently your logic says that whenever you click current video resets to the start and stops playing and when you hit the spacebar. Once you add the logic to randomise the video that the most recent frame of the current video (just randomised) will display (image(video[x], 0, 0);), then that video will play. Unless you click to stop the current video, previously started videos (via play()) will play in the background (e.g. if they have audio you'll hear them in the background even if you only see one static frame from the last time space was pressed).
Maybe this is the behaviour you want ? You've explained a localised section of what you want to achieve, but not overall what the whole of the program you posted should do. That would help others provide suggestions regarding logic.
In general, try to break the problem down to simple steps that you can test in isolation. Once you've found a solid solution for each part, you can add each part into a main sketch one at a time, testing each time you add something. (This way if something goes wrong it's easy to isolate/fix).
Kevin Workman's How To Program is a great article on this.
As a mental excercise it will help to read through the code line by line and imagine what it might do. Then run it and see if the code behaves as you predicted/intended. Slowly and surely this will get better and better. Have fun learning!
I am trying to make a custom media sink for video playback in an OpenGL application (without the various WGL_NV_DX_INTEROP, as I am not sure if all my target devices support this).
What I have done so far is to write a custom stream sink that accepts RGB32 samples and set up playback with a media session, however i encountered a problem with initial testing of playing an mp4 file:
one (or more) of the MFTs in the generated topology keep failing with an error code MF_E_TRANSFORM_NEED_MORE_INPUT, therefore my stream sink never receives samples
After a few samples have been requested, the media session receives the event MF_E_ATTRIBUTENOTFOUND, but I still don't know where it is coming from
If, however, I configure the stream sink to receive NV12 samples, everything seems to work fine.
My best guess is the color converter MFT generated by the TopologyLoader needs some more configuration, but I don't know how to do that, considering that I need to keep this entire process indipendent from the original file types.
I've made a minimal test case, that demonstrate the use of a custom video renderer with a classical Media Session.
I use big_buck_bunny_720p_50mb.mp4, and i don't see any problems using RGB32 format.
Sample code here : https://github.com/mofo7777/Stackoverflow under MinimalSinkRenderer.
EDIT
Your program works well with big_buck_bunny_720p_50mb.mp4. I think that your mp4 file is the problem. Share it, if you can.
I just made a few changes :
You Stop on MESessionEnded, and you Close on MESessionStopped.
case MediaEventType.MESessionEnded:
Debug.WriteLine("MediaSession:SesssionEndedEvent");
hr = mediaSession.Stop();
break;
case MediaEventType.MESessionClosed:
Debug.WriteLine("MediaSession:SessionClosedEvent");
receiveSessionEvent = false;
break;
case MediaEventType.MESessionStopped:
Debug.WriteLine("MediaSession:SesssionStoppedEvent");
hr = mediaSession.Close();
break;
default:
Debug.WriteLine("MediaSession:Event: " + eventType);
break;
Adding this to wait for the sound, and to check sample is ok :
internal HResult ProcessSample(IMFSample s)
{
//Debug.WriteLine("Received sample!");
CurrentFrame++;
if (s != null)
{
long llSampleTime = 0;
HResult hr = s.GetSampleTime(out llSampleTime);
if (hr == HResult.S_OK && ((CurrentFrame % 50) == 0))
{
TimeSpan ts = TimeSpan.FromMilliseconds(llSampleTime / (10000000 / 1000));
Debug.WriteLine("Frame {0} : {1}", CurrentFrame.ToString(), ts.ToString());
}
// Do not call SafeRelease here, it is done by the caller, it is a parameter
//SafeRelease(s);
}
System.Threading.Thread.Sleep(26);
return HResult.S_OK;
}
In
public HResult SetPresentationClock(IMFPresentationClock pPresentationClock)
adding
SafeRelease(PresentationClock);
before
if (pPresentationClock != null)
PresentationClock = pPresentationClock;
We've been working with AudioUnits in Core Audio. It is simultaniously a very powerful audio framework, and one of the worst documented which makes it both a joy and a frustration to work with.
We want to accomplish something we know iPads had been able to do since iOS 6.0 - Multiple audio inputs.
So far - from the 2012 Developer Talk - It appears you have to set the audio session to MultiRoute. We've done this. If I plug in an a soundcard from a keyboard. I can see that there are two inputs. Great. We're then told that we need to set a ChannelMap on a Remote I/O unit.
To what? Well... here's where it gets vague. We need to set all the channels we don't want to -1 and the channels we want to 0 and 1 (for stereo input or for mono?).
We attempt this and... nothing. Sound still plays through on the 'last in wins' principle. Microphone if everything plugged out, soundcard if that's the one plugged in. But we can't switch between them.
This setup code is always run before the other function listed
func setupAudioSession() {
self.audioSession = AVAudioSession.sharedInstance()
do {
try audioSession.setCategory(AVAudioSessionCategoryMultiRoute, with: [.mixWithOthers])
try audioSession.setActive(true)
audioSessionWasSetup = true
} catch let error {
//TODO: Implement something here
print(error)
audioSessionWasSetup = false
}
}
We then have a remote I/O with an associated audiograph set up. This has been tested and works beautifully. But we need to be able to set where it's pulling sound from.
I've attempted to do it with this, but not only doesn't it have any effect... nothing happens.
Am I missing something?
private func setChannelMap(onAudioUnit audioUnit: AudioUnit?, toChannel channelIndex: Int = 0) {
var channelMap: [Int32] = []
if audioUnit == nil {
return
}
var numberOfInputChannels: UInt32 = 4 // Two stereo inputs? - I'm just guessing here
let mapSize: UInt32 = numberOfInputChannels * UInt32(MemoryLayout<Int32>.size);
for _ in 0...(numberOfInputChannels) {
channelMap.append(-1)
}
channelMap[2 * channelIndex] = 0;
channelMap[2 * channelIndex + 1] = 1;
let status = AudioUnitSetProperty(audioUnit!,
kAudioOutputUnitProperty_ChannelMap,
kAudioUnitScope_Input,
0,
&channelMap,
mapSize);
self.checkError(status, "Failed to set Channel Map on input unit")
}
There isn't any documentation on this at all as far as I've been able to find. Nor any code examples.
I hope you can help us.
I'm attempting to port an old open-source FMOD 3 game (Candy Crisis) to the latest version of FMOD Ex 4 on OS X. Its sound needs are very simple—it plays WAVs, sometimes changing their frequency or speaker mix, and also plays MOD tracker music, sometimes changing the speed. I'm finding that the game works fine at first, but over the course of a few minutes, it starts truncating sounds early, then the music loses channels and eventually stops, then over time all sound ceases. I can cause the problem to reproduce more quickly if I lower the number of channels available to FMOD.
I can get the truncated/missing sounds issue to occur even if I never play a music file, but music definitely seems to make things worse. I have also tried commenting out the code which adjusts the sound frequency and speaker mix, and that was not the issue.
I am calling update() every frame.
Here's the entirety of my interactions with FMOD to play WAVs:
void InitSound( void )
{
FMOD_RESULT result = FMOD::System_Create(&g_fmod);
FMOD_ERRCHECK(result);
unsigned int version;
result = g_fmod->getVersion(&version);
FMOD_ERRCHECK(result);
if (version < FMOD_VERSION)
{
printf("Error! You are using an old version of FMOD %08x. This program requires %08x\n", version, FMOD_VERSION);
abort();
}
result = g_fmod->init(8 /* was originally 64, but 8 repros the issue faster */, FMOD_INIT_NORMAL, 0);
FMOD_ERRCHECK(result);
for (int index=0; index<kNumSounds; index++)
{
result = g_fmod->createSound(QuickResourceName("snd", index+128, ".wav"), FMOD_DEFAULT, 0, &s_sound[index]);
FMOD_ERRCHECK(result);
}
}
void PlayMono( short which )
{
if (soundOn)
{
FMOD_RESULT result = g_fmod->playSound(FMOD_CHANNEL_FREE, s_sound[which], false, NULL);
FMOD_ERRCHECK(result);
}
}
void PlayStereoFrequency( short player, short which, short freq )
{
if (soundOn)
{
FMOD::Channel* channel = NULL;
FMOD_RESULT result = g_fmod->playSound(FMOD_CHANNEL_FREE, s_sound[which], true, &channel);
FMOD_ERRCHECK(result);
result = channel->setSpeakerMix(player, 1.0f - player, 0, 0, 0, 0, 0, 0);
FMOD_ERRCHECK(result);
float channelFrequency;
result = s_sound[which]->getDefaults(&channelFrequency, NULL, NULL, NULL);
FMOD_ERRCHECK(result);
result = channel->setFrequency((channelFrequency * (16 + freq)) / 16);
FMOD_ERRCHECK(result);
result = channel->setPaused(false);
FMOD_ERRCHECK(result);
}
}
void UpdateSound()
{
g_fmod->update();
}
And here's how I play MODs.
void ChooseMusic( short which )
{
if( musicSelection >= 0 && musicSelection <= k_songs )
{
s_musicChannel->stop();
s_musicChannel = NULL;
s_musicModule->release();
s_musicModule = NULL;
musicSelection = -1;
}
if (which >= 0 && which <= k_songs)
{
FMOD_RESULT result = g_fmod->createSound(QuickResourceName("mod", which+128, ""), FMOD_DEFAULT, 0, &s_musicModule);
FMOD_ERRCHECK(result);
result = g_fmod->playSound(FMOD_CHANNEL_FREE, s_musicModule, true, &s_musicChannel);
FMOD_ERRCHECK(result);
EnableMusic(musicOn);
s_musicModule->setLoopCount(-1);
s_musicChannel->setPaused(false);
musicSelection = which;
s_musicPaused = 0;
}
}
If someone wants to experiment with this, let me know and I'll upload the project somewhere. My gut feeling is that FMOD is busted but I'd love to be proven wrong.
Sounds like your music needs to be set as higher priority than your other sounds. Remember, lower numbers are more important. I think you can just set the priority on the channel.
Every time I play the following WAV, FMOD loses one channel permanently. I am able to reproduce this channel-losing behavior in the "playsound" example if I replace the existing jaguar.wav with my file.
https://drive.google.com/file/d/0B1eDRY8sV_a9SXMyNktXbWZOYWs/view?usp=sharing
I contacted Firelight and got this response. Apparently WAVs can include a looping command! I had no idea.
Hello John,
I've taken a look at the two files you have provided. Both files end
with a 2 sample infinite loop region.
FMOD 4 (and FMOD 5 for that matter) will see the loop region in the
file and automatically enable FMOD_LOOP_NORMAL if you haven't
specified any loop mode. Assuming you want one-shot behavior just pass
in FMOD_LOOP_OFF when you create the sound.
Kind regards, Mathew Block | Senior Platform Engineer
Technically this behavior contradicts the documented behavior of FMOD_DEFAULT (which is specified to imply FMOD_LOOP_OFF) so they are planning to improve the documentation here.
Based on the wave sample you supplied, FMOD is behaving correctly as it appears you've figured out. The sample has a loop that is honored by FMOD and the last samples are simply repeated forever. While useless, this is correct and the variance in the samples is so slight as to not be audible. While not part of the original spec for wave format, extended information was added later to support meta data such as author, title, comments and multiple loop points.
Your best bet is to examine all your source assets for those that contain loop information. Simply playing all sounds without loop information is probably not the best workaround. Some loops may be intentional. Those that are will have code that stops them. Typically, in a game, the entire waveform is looped when looping is desired. You can then write or use a tool that will strip the loop information. If you do write your own tool, I'd recommend resampling the audio to the native output sampling rate of the hardware. You'd need to insure your resampler was sample accurate (no time shift) and did not introduce noise.
Historically, some game systems had a section at the end of the sound with silence and a loop point set on this region. The short reason for this was to reduce popping that might occur at the end of a sound in a hardware audio channel.
Curiosly, the last 16 samples of your .wav look like garbage and I'm wondering if the .wav assets you're using were converted from a source meant for a game console and that's where the bogus loop information came from as well.
This would have been a comment but my lowly rep does not allow it.
This is supposed to be possible on Mac OS X by overwriting the sample rate in the AudioStreamBasicDescription then create a new output queue.
I've been able to retrieve the default sample rate and write a new one (ie. replace 44100 with 48000) but this is not resulting in any pitch change in the output signal.
err = AudioFileGetProperty(mAudioFile, kAudioFilePropertyDataFormat, &size, &mDataFormat);
if (err != noErr)
NSLog(#"Couldn't determine the audio file format");
Float64 mySampleRate = mDataFormat.mSampleRate; //the initial rate
if (inRate != 1) {
//write a new value
mDataFormat.mSampleRate = inRate;
//then
err = AudioQueueNewOutput etc.
Any suggestions would be greatly appreciated.
Changing the sample rate doesn't change the pitch of the audio. You may perceive that something playing back faster has a higher pitch. However that's perception rather than reality.
To change pitch, you'll need to process the audio data through a Digital Signal Processing (DSP) library. Alternatively, take a look at running it through an AudioUnit:
Audio Unit Programming Guide