From 367000fa8d4da2425c80c9bcf470d4f53ed39632 Mon Sep 17 00:00:00 2001 From: "henrik.lundin@webrtc.org" Date: Fri, 14 Mar 2014 12:28:39 +0000 Subject: [PATCH] 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 --- webrtc/modules/audio_coding/neteq4/merge.cc | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/webrtc/modules/audio_coding/neteq4/merge.cc b/webrtc/modules/audio_coding/neteq4/merge.cc index 463b2ca78..4b5601657 100644 --- a/webrtc/modules/audio_coding/neteq4/merge.cc +++ b/webrtc/modules/audio_coding/neteq4/merge.cc @@ -248,7 +248,7 @@ void Merge::Downsample(const int16_t* input, int input_length, 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 @@ void Merge::Downsample(const int16_t* input, int input_length, } 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],