audio-panning

OpenAL 2d panning C++

由 生来就可爱ヽ(ⅴ＜●) 提交于 2020-01-14 05:11:12
问题 I'm trying to figure out how to get openAL to pan in 2D (by manipulating the 3D positioning). Ideally I want to achieve panning such that the Left or Right channel can be fully engaged with the other channel completely silent. It seems that Open AL handles 3d distances and falloffs nicely, but I'm struggling to emulate this kind of 2D panning. I'm using alDistanceModel(AL_LINEAR_DISTANCE_CLAMPED) float sourcePosition[3] = {0.99f,0.f,0.f}; alSourcefv(sourceID, AL_POSITION, sourcePosition);

Gradually Change Web Audio API Panner

由 拈花ヽ惹草 提交于 2019-12-23 14:58:17
问题 I'm trying to use a simple HTML range input to control the panning of my Web Audio API audio but I can only get 3 "positions" for my audio output: -Center -100% to the left -100% to the right. I would like to have something in between does positions, like 20% left and 80% right and so on... The code that I'm using is: //Creating the node var pannerNode = context.createPanner(); //Getting the value from the HTML input and using it on the position X value document.getElementById('panInput')

How to adjust audio panning with RtAudio

由 ﹥＞﹥吖頭↗ 提交于 2019-12-19 09:43:12
问题 I use Rtaudio library and I would like to implement an audio program where I can control the panning (e.g. shifting the sound from the left channel to the right channel). In my specific case, I use a duplex mode (you can find an example here: duplex mode). It means that I link the microphone input to the speaker output. Should I apply a filter on the output buffer? What kind of filter? Can anyone help me? 回答1: To reduce the signal on the left, simply multiply every sample on the left by a

How to adjust audio panning with RtAudio

由 非 Y 不嫁゛ 提交于 2019-12-01 09:08:08
I use Rtaudio library and I would like to implement an audio program where I can control the panning (e.g. shifting the sound from the left channel to the right channel). In my specific case, I use a duplex mode (you can find an example here: duplex mode ). It means that I link the microphone input to the speaker output. Should I apply a filter on the output buffer? What kind of filter? Can anyone help me? To reduce the signal on the left, simply multiply every sample on the left by a number less than or equal to 1, let's call it l. Similarly for the right, we'll call that number r. Generally
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