Moving/renaming webrtc/common.h.

This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.

- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.

BUG=webrtc:5879

Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}

webrtc/modules/audio_coding/test/APITest.cc

@@ -22,7 +22,6 @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/timeutils.h"
-#include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
@@ -48,7 +47,7 @@ void APITest::Wait(uint32_t waitLengthMs) {
   }
 }
 
-APITest::APITest(const Config& config)
+APITest::APITest()
     : _acmA(AudioCodingModule::Create(1)),
       _acmB(AudioCodingModule::Create(2)),
       _channel_A2B(NULL),

webrtc/modules/audio_coding/test/APITest.h

@@ -23,8 +23,6 @@
 
 namespace webrtc {
 
-class Config;
-
 enum APITESTAction {
   TEST_CHANGE_CODEC_ONLY = 0,
   DTX_TEST = 1
@@ -32,7 +30,7 @@ enum APITESTAction {
 
 class APITest : public ACMTest {
  public:
-  explicit APITest(const Config& config);
+  APITest();
   ~APITest();
 
   void Perform();

webrtc/modules/audio_coding/test/PacketLossTest.cc

@@ -13,7 +13,6 @@
 #include <memory>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {

webrtc/modules/audio_coding/test/TestAllCodecs.h

@@ -20,8 +20,6 @@
 
 namespace webrtc {
 
-class Config;
-
 class TestPack : public AudioPacketizationCallback {
  public:
   TestPack();

webrtc/modules/audio_coding/test/TestRedFec.cc

@@ -12,7 +12,6 @@
 
 #include <assert.h>
 
-#include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"

webrtc/modules/audio_coding/test/TestRedFec.h

@@ -20,8 +20,6 @@
 
 namespace webrtc {
 
-class Config;
-
 class TestRedFec : public ACMTest {
  public:
   explicit TestRedFec();

webrtc/modules/audio_coding/test/delay_test.cc

@@ -16,7 +16,6 @@
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"

webrtc/modules/audio_coding/test/utility.cc

@@ -16,7 +16,6 @@
 #include <string.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"

webrtc/modules/audio_coding/test/utility.h

@@ -128,10 +128,6 @@ class VADCallback : public ACMVADCallback {
   uint32_t _numFrameTypes[5];
 };
 
-void UseLegacyAcm(webrtc::Config* config);
-
-void UseNewAcm(webrtc::Config* config);
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_