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| author | Mathias Agopian <mathias@google.com> | 2012-10-26 13:48:42 -0700 |
|---|---|---|
| committer | Mathias Agopian <mathias@google.com> | 2012-10-26 14:58:42 -0700 |
| commit | 443e69625d598ea578e2c838960778ce498fd773 (patch) | |
| tree | 4d021ec94a7766aaed15c1d93f3fa7b7d9e89ba1 /services/audioflinger/AudioResamplerSinc.h | |
| parent | b6cf07ba69018fd4633e1b9f99646f8482c6d76d (diff) | |
| download | frameworks_av-443e69625d598ea578e2c838960778ce498fd773.zip frameworks_av-443e69625d598ea578e2c838960778ce498fd773.tar.gz frameworks_av-443e69625d598ea578e2c838960778ce498fd773.tar.bz2 | |
improve SINC resampler coefficients
- we increase the interpolation precision from 4 to 7 bits
this doesn't increase CPU power required, it only increases the
size of the filter table but significantly reduces the noise
introduced by the quantization of the impulse response.
- the parameters of the filter are set such that aliasing is
rejected at 80 dB below 20 KHz. Because we don't use a lot of
coefficient (to save compute power), there are quite a bit of
attenuation in the pass-band: starting at 9KHz for the
down-sampler (48 to 44.1), and starting at 13 KHz for the
up-sampler (44.1 to 48) -- the transition band is about 15 KHz.
Change-Id: I855548d2aab8a0fb0d2a2da3a364b6842d7d3838
Diffstat (limited to 'services/audioflinger/AudioResamplerSinc.h')
| -rw-r--r-- | services/audioflinger/AudioResamplerSinc.h | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h index 25fc025..48bc747 100644 --- a/services/audioflinger/AudioResamplerSinc.h +++ b/services/audioflinger/AudioResamplerSinc.h @@ -71,7 +71,7 @@ private: // ---------------------------------------------------------------------------- static const int32_t RESAMPLE_FIR_NUM_COEF = 8; - static const int32_t RESAMPLE_FIR_LERP_INT_BITS = 4; + static const int32_t RESAMPLE_FIR_LERP_INT_BITS = 7; struct Constants { // we have 16 coefs samples per zero-crossing |