From 1f9baab753be55a7c6d31c84a5470fe646936edd Mon Sep 17 00:00:00 2001
From: kwiberg <kwiberg@webrtc.org>
Date: Wed, 16 Sep 2015 19:29:43 -0700
Subject: [PATCH] Remove the preprocessor symbol WEBRTC_CODEC_AVT (it was
 always defined)

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1338283002

Cr-Commit-Position: refs/heads/master@{#9960}
---
 webrtc/engine_configurations.h                              | 4 ----
 webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc | 6 ------
 webrtc/modules/audio_coding/main/acm2/acm_codec_database.h  | 5 -----
 .../audio_coding/main/acm2/audio_coding_module_impl.cc      | 2 --
 webrtc/voice_engine/channel.cc                              | 6 +-----
 5 files changed, 1 insertion(+), 22 deletions(-)

diff --git a/webrtc/engine_configurations.h b/webrtc/engine_configurations.h
index 3b06c35a30..5963892e5a 100644
--- a/webrtc/engine_configurations.h
+++ b/webrtc/engine_configurations.h
@@ -31,10 +31,6 @@
 #define WEBRTC_CODEC_G722
 #endif  // !WEBRTC_MOZILLA_BUILD
 
-// AVT is included in all builds, along with G.711, NetEQ and CNG
-// (which are mandatory and don't have any defines).
-#define WEBRTC_CODEC_AVT
-
 // iLBC and Redundancy coding are excluded from Chromium and Mozilla
 // builds to reduce binary size.
 #if !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_MOZILLA_BUILD)
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
index 4abd369656..78316669f6 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc
@@ -75,9 +75,7 @@ const CodecInst ACMCodecDB::database_[] = {
 #ifdef ENABLE_48000_HZ
   {100, "CN", 48000, 1440, 1, 0},
 #endif
-#ifdef WEBRTC_CODEC_AVT
   {106, "telephone-event", 8000, 240, 1, 0},
-#endif
 #ifdef WEBRTC_CODEC_RED
   {127, "red", 8000, 0, 1, 0},
 #endif
@@ -134,9 +132,7 @@ const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
 #ifdef ENABLE_48000_HZ
     {1, {1440}, 1440, 1, false},
 #endif
-#ifdef WEBRTC_CODEC_AVT
     {1, {240}, 240, 1, false},
-#endif
 #ifdef WEBRTC_CODEC_RED
     {1, {0}, 0, 1, false},
 #endif
@@ -188,9 +184,7 @@ const NetEqDecoder ACMCodecDB::neteq_decoders_[] = {
 #ifdef ENABLE_48000_HZ
     , kDecoderCNGswb48kHz
 #endif
-#ifdef WEBRTC_CODEC_AVT
     , kDecoderAVT
-#endif
 #ifdef WEBRTC_CODEC_RED
     , kDecoderRED
 #endif
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
index 925c25fc42..f34d755ccd 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h
@@ -71,9 +71,7 @@ class ACMCodecDB {
 #ifdef ENABLE_48000_HZ
     , kCNFB
 #endif
-#ifdef WEBRTC_CODEC_AVT
     , kAVT
-#endif
 #ifdef WEBRTC_CODEC_RED
     , kRED
 #endif
@@ -103,9 +101,6 @@ class ACMCodecDB {
   // Mono and stereo
   enum {kOpus = -1};
 #endif
-#ifndef WEBRTC_CODEC_AVT
-  enum {kAVT = -1};
-#endif
 #ifndef WEBRTC_CODEC_RED
   enum {kRED = -1};
 #endif
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index be0fbf1d3e..5aa320b8b6 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -1130,13 +1130,11 @@ bool AudioCodingImpl::MapCodecTypeToParameters(int codec_type,
       *sample_rate_hz = 8000;
       *channels = 1;
       break;
-#ifdef WEBRTC_CODEC_AVT
     case acm2::ACMCodecDB::kAVT:
       *codec_name = "telephone-event";
       *sample_rate_hz = 8000;
       *channels = 1;
       break;
-#endif
     default:
       FATAL() << "Codec type " << codec_type << " not supported.";
   }
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index d602bb4fdf..fb29dfabab 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -932,12 +932,8 @@ Channel::Init()
     // --- ACM initialization
 
     if ((audio_coding_->InitializeReceiver() == -1)
-#ifdef WEBRTC_CODEC_AVT
         // out-of-band Dtmf tones are played out by default
-        || (audio_coding_->SetDtmfPlayoutStatus(true) == -1)
-#endif
-       )
-    {
+        || (audio_coding_->SetDtmfPlayoutStatus(true) == -1)) {
         _engineStatisticsPtr->SetLastError(
             VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
             "Channel::Init() unable to initialize the ACM - 1");