commit ec0f45be1144078338805b8bd525b29e8e736085
author Fredrik Solenberg <solenberg@webrtc.org> Mon Dec 03 15:50:44 2018 +0000
committer Commit Bot <commit-bot@chromium.org> Mon Dec 03 15:50:51 2018 +0000
tree 63daac9a0b8ed19a2397140f6f66c2e4059dd594
parent 9d54bd889898baa02a2b3920cbaf80fa3c70e598

Revert "Remove CodecInst pt.1"

This reverts commit 056f9738bf7a3d16da45398239656e165c4e0851.

Reason for revert: breaks downstream

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}

TBR=solenberg@webrtc.org,kwiberg@webrtc.org

Change-Id: I51d666969bcd63e2b7cb7d669ec2f59b5f8f9dde
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7626
Reviewed-on: https://webrtc-review.googlesource.com/c/112906
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25881}

modules/audio_coding/test/Tester.cc

diff --git a/modules/audio_coding/test/Tester.cc b/modules/audio_coding/test/Tester.cc
index 750656f..85926f1 100644
--- a/modules/audio_coding/test/Tester.cc
+++ b/modules/audio_coding/test/Tester.cc
@@ -25,8 +25,12 @@
 #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().Perform();
+  webrtc::TestAllCodecs(ACM_TEST_MODE).Perform();
 }
 
 #if defined(WEBRTC_ANDROID)
@@ -34,7 +38,7 @@
 #else
 TEST(AudioCodingModuleTest, TestEncodeDecode) {
 #endif
-  webrtc::EncodeDecodeTest().Perform();
+  webrtc::EncodeDecodeTest(ACM_TEST_MODE).Perform();
 }
 
 TEST(AudioCodingModuleTest, TestRedFec) {
@@ -46,7 +50,7 @@
 #else
 TEST(AudioCodingModuleTest, TestIsac) {
 #endif
-  webrtc::ISACTest().Perform();
+  webrtc::ISACTest(ACM_TEST_MODE).Perform();
 }
 
 #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
@@ -66,7 +70,7 @@
 #else
 TEST(AudioCodingModuleTest, TestStereo) {
 #endif
-  webrtc::TestStereo().Perform();
+  webrtc::TestStereo(ACM_TEST_MODE).Perform();
 }
 
 TEST(AudioCodingModuleTest, TestWebRtcVadDtx) {