Remove ACM1 and NetEq3 related targets from modules.gyp

Make necessary changes to compile.

BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6008 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
index d4c6f5c..eca909c 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
@@ -13,15 +13,11 @@
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
 #include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
-#include "webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h"
 #include "webrtc/system_wrappers/interface/clock.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
 namespace webrtc {
 
-const char kLegacyAcmVersion[] = "acm1";
-const char kExperimentalAcmVersion[] = "acm2";
-
 // Create module
 AudioCodingModule* AudioCodingModule::Create(int id) {
   return Create(id, Clock::GetRealTimeClock());
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
index f51c3bf..07fe727 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi
@@ -7,9 +7,29 @@
 # be found in the AUTHORS file in the root of the source tree.
 
 {
+  'variables': {
+    'audio_coding_dependencies': [
+      'CNG',
+      'G711',
+      'G722',
+      'iLBC',
+      'iSAC',
+      'iSACFix',
+      'PCM16B',
+      '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
+      '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+    ],
+    'audio_coding_defines': [],
+    'conditions': [
+      ['include_opus==1', {
+        'audio_coding_dependencies': ['webrtc_opus',],
+        'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
+      }],
+    ],
+  },
   'targets': [
     {
-      'target_name': 'acm2',
+      'target_name': 'audio_coding_module',
       'type': 'static_library',
       'defines': [
         '<@(audio_coding_defines)',
@@ -93,4 +113,43 @@
       ],
     },
   ],
+  'conditions': [
+    ['include_tests==1', {
+      'targets': [
+        {
+          'target_name': 'delay_test',
+          'type': 'executable',
+          'dependencies': [
+            'audio_coding_module',
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+            '<(webrtc_root)/test/test.gyp:test_support',
+            '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+          ],
+          'sources': [
+             '../test/delay_test.cc',
+             '../test/Channel.cc',
+             '../test/PCMFile.cc',
+             '../test/utility.cc',
+           ],
+        }, # delay_test
+        {
+          'target_name': 'insert_packet_with_timing',
+          'type': 'executable',
+          'dependencies': [
+            'audio_coding_module',
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+            '<(webrtc_root)/test/test.gyp:test_support',
+            '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+          ],
+          'sources': [
+             '../test/insert_packet_with_timing.cc',
+             '../test/Channel.cc',
+             '../test/PCMFile.cc',
+           ],
+        }, # delay_test
+      ],
+    }],
+  ],
 }
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 936f1f1..6133d23 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
@@ -1978,10 +1978,6 @@
   return receiver_.LeastRequiredDelayMs();
 }
 
-const char* AudioCodingModuleImpl::Version() const {
-  return kExperimentalAcmVersion;
-}
-
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const {
   receiver_.GetDecodingCallStatistics(call_stats);
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index d53c98f..157fc01 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -35,8 +35,6 @@
   explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
   ~AudioCodingModuleImpl();
 
-  virtual const char* Version() const;
-
   // Change the unique identifier of this object.
   virtual int32_t ChangeUniqueId(const int32_t id);
 
diff --git a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
index 89b3376..9c7e562 100644
--- a/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
@@ -74,10 +74,6 @@
       const uint16_t delayMS) = 0;  // average delay in ms
 };
 
-// Version string for testing, to distinguish instances of ACM1 from ACM2.
-extern const char kLegacyAcmVersion[];
-extern const char kExperimentalAcmVersion[];
-
 class AudioCodingModule: public Module {
  protected:
   AudioCodingModule() {}
@@ -190,11 +186,6 @@
   //
   static bool IsCodecValid(const CodecInst& codec);
 
-  // Returns the version of ACM. This facilitates distinguishing instances of
-  // ACM1 from ACM2 while testing. This API will be removed when ACM1 is
-  // completely removed.
-  virtual const char* Version() const = 0;
-
   ///////////////////////////////////////////////////////////////////////////
   //   Sender
   //
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index 8fd0714..444c665 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -16,8 +16,7 @@
     'audio_coding/codecs/isac/main/source/isac.gypi',
     'audio_coding/codecs/isac/fix/source/isacfix.gypi',
     'audio_coding/codecs/pcm16b/pcm16b.gypi',
-    'audio_coding/main/source/audio_coding_module.gypi',
-    'audio_coding/neteq/neteq.gypi',
+    'audio_coding/main/acm2/audio_coding_module.gypi',
     'audio_coding/neteq4/neteq.gypi',
     'audio_conference_mixer/source/audio_conference_mixer.gypi',
     'audio_device/audio_device.gypi',
@@ -76,7 +75,6 @@
             'desktop_capture',
             'iSACFix',
             'media_file',
-            'NetEq',
             'NetEq4',
             'NetEq4TestTools',
             'neteq_unittest_tools',
@@ -107,7 +105,6 @@
             'audio_coding/main/acm2/call_statistics_unittest.cc',
             'audio_coding/main/acm2/initial_delay_manager_unittest.cc',
             'audio_coding/main/acm2/nack_unittest.cc',
-            'audio_coding/main/source/acm_neteq_unittest.cc',
             'audio_coding/codecs/cng/cng_unittest.cc',
             'audio_coding/codecs/isac/fix/source/filters_unittest.cc',
             'audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc',