commit 657b296ff5448e62bd4d0c3daeee1900cbb168f0
author Fredrik Solenberg <solenberg@webrtc.org> Wed Dec 05 10:30:25 2018 +0100
committer Commit Bot <commit-bot@chromium.org> Wed Dec 05 10:38:23 2018 +0000
tree 0d9087b305b5e5a044711f0dc9f71ad2a1d5b9e4
parent 5b1477839d8569291b88dfe950089d0ebf34bc8f

Reland "Remove CodecInst pt.1"

This is a reland of 056f9738bf7a3d16da45398239656e165c4e0851

Original change's description:
> Remove CodecInst pt.1
> 
> Update audio_coding tests to not use CodecInst.
> 
> Bug: webrtc:7626
> Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/112594
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25879}

Bug: webrtc:7626
Change-Id: I5d6ca0baf6230bfe9bf95c2c25496d2a56812d90
Reviewed-on: https://webrtc-review.googlesource.com/c/112942
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25902}

modules/audio_coding/test/Tester.cc

diff --git a/modules/audio_coding/test/Tester.cc b/modules/audio_coding/test/Tester.cc
index 85926f1..750656f 100644
--- a/modules/audio_coding/test/Tester.cc
+++ b/modules/audio_coding/test/Tester.cc
@@ -25,12 +25,8 @@
 #include "test/gtest.h"
 #include "test/testsupport/fileutils.h"
 
-// This parameter is used to describe how to run the tests. It is normally
-// set to 0, and all tests are run in quite mode.
-#define ACM_TEST_MODE 0
-
 TEST(AudioCodingModuleTest, TestAllCodecs) {
-  webrtc::TestAllCodecs(ACM_TEST_MODE).Perform();
+  webrtc::TestAllCodecs().Perform();
 }
 
 #if defined(WEBRTC_ANDROID)
@@ -38,7 +34,7 @@
 #else
 TEST(AudioCodingModuleTest, TestEncodeDecode) {
 #endif
-  webrtc::EncodeDecodeTest(ACM_TEST_MODE).Perform();
+  webrtc::EncodeDecodeTest().Perform();
 }
 
 TEST(AudioCodingModuleTest, TestRedFec) {
@@ -50,7 +46,7 @@
 #else
 TEST(AudioCodingModuleTest, TestIsac) {
 #endif
-  webrtc::ISACTest(ACM_TEST_MODE).Perform();
+  webrtc::ISACTest().Perform();
 }
 
 #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
@@ -70,7 +66,7 @@
 #else
 TEST(AudioCodingModuleTest, TestStereo) {
 #endif
-  webrtc::TestStereo(ACM_TEST_MODE).Perform();
+  webrtc::TestStereo().Perform();
 }
 
 TEST(AudioCodingModuleTest, TestWebRtcVadDtx) {