commit 056f9738bf7a3d16da45398239656e165c4e0851
author Fredrik Solenberg <solenberg@webrtc.org> Mon Dec 03 15:22:16 2018 +0100
committer Commit Bot <commit-bot@chromium.org> Mon Dec 03 15:16:20 2018 +0000
tree 4dc9545e3453d52604787e4ee9fd0c749fc3a941
parent 450b548cade5498bbdef6eddce0703f52296bf65

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}

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) {