Remove legacy weirdness in Merge::Downsample
In practice, this will have only marginal effect. The length_limit
was increased from 6.7 ms to 10 ms. This is compared with the
input_length, which is equal to the decoded frame size. Thus,
this change will only affect encoded frame sizes in this range
(including 10 ms).
BUG=2696
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5700 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/merge.cc b/webrtc/modules/audio_coding/neteq4/merge.cc
index 463b2ca..4b56016 100644
--- a/webrtc/modules/audio_coding/neteq4/merge.cc
+++ b/webrtc/modules/audio_coding/neteq4/merge.cc
@@ -248,7 +248,7 @@
int num_coefficients;
int decimation_factor = fs_hz_ / 4000;
static const int kCompensateDelay = 0;
- int length_limit = fs_hz_ / 100;
+ int length_limit = fs_hz_ / 100; // 10 ms in samples.
if (fs_hz_ == 8000) {
filter_coefficients = DspHelper::kDownsample8kHzTbl;
num_coefficients = 3;
@@ -261,8 +261,6 @@
} else { // fs_hz_ == 48000
filter_coefficients = DspHelper::kDownsample48kHzTbl;
num_coefficients = 7;
- // TODO(hlundin) Why is |length_limit| not 480 (legacy)?
- length_limit = 320;
}
int signal_offset = num_coefficients - 1;
WebRtcSpl_DownsampleFast(&expanded_signal[signal_offset],