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}
diff --git a/webrtc/engine_configurations.h b/webrtc/engine_configurations.h
index 3b06c35..5963892 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 4abd369..7831666 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 @@
#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 @@
#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 @@
#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 925c25f..f34d755 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 @@
#ifdef ENABLE_48000_HZ
, kCNFB
#endif
-#ifdef WEBRTC_CODEC_AVT
, kAVT
-#endif
#ifdef WEBRTC_CODEC_RED
, kRED
#endif
@@ -103,9 +101,6 @@
// 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 be0fbf1..5aa320b 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 @@
*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 d602bb4..fb29dfa 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -932,12 +932,8 @@
// --- 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");
