commit fd11bbfb56b42f82e18a744a414325db7a56013f
author henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon Sep 30 20:38:44 2013 +0000
committer henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon Sep 30 20:38:44 2013 +0000
tree 9812ba5413ce303c3526b460c50eb68db605aa83
parent 6ad6a07fd3e0279ee4018c4ecb66d3041d27392d

NetEq4: Removing templatization for AudioMultiVector

This saves approx 6% runtime for neteq4_speed_test.
$ time out/Release/neteq4_speed_test --runtime_ms=50000000

BUG=1363
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2320006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4885 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/neteq4/neteq_impl.cc

diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
index 289b67f..cf0a1db 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
@@ -1685,7 +1685,7 @@
     overdub_length = output_size_samples_ - static_cast<int>(out_index);
   }
 
-  AudioMultiVector<int16_t> dtmf_output(num_channels);
+  AudioMultiVector dtmf_output(num_channels);
   int dtmf_return_value = 0;
   if (!dtmf_tone_generator_->initialized()) {
     dtmf_return_value = dtmf_tone_generator_->Init(fs_hz_, dtmf_event.event_no,
@@ -1811,7 +1811,7 @@
   vad_->Init();
 
   // Delete algorithm buffer and create a new one.
-  algorithm_buffer_.reset(new AudioMultiVector<int16_t>(channels));
+  algorithm_buffer_.reset(new AudioMultiVector(channels));
 
   // Delete sync buffer and create a new one.
   sync_buffer_.reset(new SyncBuffer(channels, kSyncBufferSize * fs_mult_));