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Change audio pipelines to process in exactly 10ms blocks (240 samples instead of 256).

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This produces an integral number of samples when resampled to 44.1k or 48k, allowing a 44.1k back-end to work correctly without extra buffering or dynamic buffers sizes.
This commit is contained in:
Ken Cooke 2016-08-18 08:27:20 -07:00
parent 7202d132c5
commit f80304d68e
1 changed files with 2 additions and 2 deletions
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4
libraries/audio/src/AudioConstants.h
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@ -28,9 +28,9 @@ namespace AudioConstants {
const int MAX_CODEC_NAME_LENGTH = 30;
const int MAX_CODEC_NAME_LENGTH_ON_WIRE = MAX_CODEC_NAME_LENGTH + sizeof(uint32_t);
const int NETWORK_FRAME_BYTES_STEREO = 1024;
const int NETWORK_FRAME_BYTES_STEREO = 960;
const int NETWORK_FRAME_SAMPLES_STEREO = NETWORK_FRAME_BYTES_STEREO / sizeof(AudioSample);
const int NETWORK_FRAME_BYTES_PER_CHANNEL = 512;
const int NETWORK_FRAME_BYTES_PER_CHANNEL = NETWORK_FRAME_BYTES_STEREO / 2;
const int NETWORK_FRAME_SAMPLES_PER_CHANNEL = NETWORK_FRAME_BYTES_PER_CHANNEL / sizeof(AudioSample);
const float NETWORK_FRAME_SECS = (AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL / float(AudioConstants::SAMPLE_RATE));
const float NETWORK_FRAME_MSECS = NETWORK_FRAME_SECS * 1000.0f;