audio_coding: remove "main" directory
This is the last piece of the old directory layout of the modules.
Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1481493004
Cr-Original-Commit-Position: refs/heads/master@{#10803}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 3e6db2321ccdc8738c9cecbe9bdab13d4f0f658d
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 69f090a..44ecae3 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -18,7 +18,7 @@
#include "webrtc/base/thread_annotations.h"
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 533e848..382ae51 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -11,10 +11,10 @@
source_set("rent_a_codec") {
sources = [
- "main/acm2/acm_codec_database.cc",
- "main/acm2/acm_codec_database.h",
- "main/acm2/rent_a_codec.cc",
- "main/acm2/rent_a_codec.h",
+ "acm2/acm_codec_database.cc",
+ "acm2/acm_codec_database.h",
+ "acm2/rent_a_codec.cc",
+ "acm2/rent_a_codec.h",
]
configs += [ "../..:common_config" ]
public_configs = [ "../..:common_inherited_config" ]
@@ -44,29 +44,29 @@
config("audio_coding_config") {
include_dirs = [
- "main/include",
+ "include",
"../include",
]
}
source_set("audio_coding") {
sources = [
- "main/acm2/acm_common_defs.h",
- "main/acm2/acm_receiver.cc",
- "main/acm2/acm_receiver.h",
- "main/acm2/acm_resampler.cc",
- "main/acm2/acm_resampler.h",
- "main/acm2/audio_coding_module.cc",
- "main/acm2/audio_coding_module_impl.cc",
- "main/acm2/audio_coding_module_impl.h",
- "main/acm2/call_statistics.cc",
- "main/acm2/call_statistics.h",
- "main/acm2/codec_manager.cc",
- "main/acm2/codec_manager.h",
- "main/acm2/initial_delay_manager.cc",
- "main/acm2/initial_delay_manager.h",
- "main/include/audio_coding_module.h",
- "main/include/audio_coding_module_typedefs.h",
+ "acm2/acm_common_defs.h",
+ "acm2/acm_receiver.cc",
+ "acm2/acm_receiver.h",
+ "acm2/acm_resampler.cc",
+ "acm2/acm_resampler.h",
+ "acm2/audio_coding_module.cc",
+ "acm2/audio_coding_module_impl.cc",
+ "acm2/audio_coding_module_impl.h",
+ "acm2/call_statistics.cc",
+ "acm2/call_statistics.h",
+ "acm2/codec_manager.cc",
+ "acm2/codec_manager.h",
+ "acm2/initial_delay_manager.cc",
+ "acm2/initial_delay_manager.h",
+ "include/audio_coding_module.h",
+ "include/audio_coding_module_typedefs.h",
]
defines = []
diff --git a/modules/audio_coding/main/acm2/acm_codec_database.cc b/modules/audio_coding/acm2/acm_codec_database.cc
similarity index 98%
rename from modules/audio_coding/main/acm2/acm_codec_database.cc
rename to modules/audio_coding/acm2/acm_codec_database.cc
index b54fc0b..8d4072f 100644
--- a/modules/audio_coding/main/acm2/acm_codec_database.cc
+++ b/modules/audio_coding/acm2/acm_codec_database.cc
@@ -15,12 +15,12 @@
// TODO(tlegrand): Change constant input pointers in all functions to constant
// references, where appropriate.
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/acm2/acm_codec_database.h"
#include <assert.h>
#include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
#include "webrtc/system_wrappers/include/trace.h"
namespace webrtc {
diff --git a/modules/audio_coding/main/acm2/acm_codec_database.h b/modules/audio_coding/acm2/acm_codec_database.h
similarity index 91%
rename from modules/audio_coding/main/acm2/acm_codec_database.h
rename to modules/audio_coding/acm2/acm_codec_database.h
index f9adda0..9e87238 100644
--- a/modules/audio_coding/main/acm2/acm_codec_database.h
+++ b/modules/audio_coding/acm2/acm_codec_database.h
@@ -13,12 +13,12 @@
* codecs.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
namespace webrtc {
@@ -80,4 +80,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
diff --git a/modules/audio_coding/main/acm2/acm_common_defs.h b/modules/audio_coding/acm2/acm_common_defs.h
similarity index 81%
rename from modules/audio_coding/main/acm2/acm_common_defs.h
rename to modules/audio_coding/acm2/acm_common_defs.h
index 23e3519..483bdd9 100644
--- a/modules/audio_coding/main/acm2/acm_common_defs.h
+++ b/modules/audio_coding/acm2/acm_common_defs.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_
#include "webrtc/engine_configurations.h"
@@ -29,4 +29,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_
diff --git a/modules/audio_coding/main/acm2/acm_neteq_unittest.cc b/modules/audio_coding/acm2/acm_neteq_unittest.cc
similarity index 100%
rename from modules/audio_coding/main/acm2/acm_neteq_unittest.cc
rename to modules/audio_coding/acm2/acm_neteq_unittest.cc
diff --git a/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc b/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
similarity index 97%
rename from modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc
rename to modules/audio_coding/acm2/acm_receive_test_oldapi.cc
index fdcfdfc..bb83e77 100644
--- a/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc
+++ b/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h"
#include <assert.h>
#include <stdio.h>
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
diff --git a/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h b/modules/audio_coding/acm2/acm_receive_test_oldapi.h
similarity index 92%
rename from modules/audio_coding/main/acm2/acm_receive_test_oldapi.h
rename to modules/audio_coding/acm2/acm_receive_test_oldapi.h
index 0b5671f..091513d 100644
--- a/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h
+++ b/modules/audio_coding/acm2/acm_receive_test_oldapi.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
@@ -91,4 +91,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_
diff --git a/modules/audio_coding/main/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
similarity index 98%
rename from modules/audio_coding/main/acm2/acm_receiver.cc
rename to modules/audio_coding/acm2/acm_receiver.cc
index 6c28933..036877c 100644
--- a/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
#include <stdlib.h> // malloc
@@ -21,8 +21,8 @@
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
diff --git a/modules/audio_coding/main/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
similarity index 95%
rename from modules/audio_coding/main/acm2/acm_receiver.h
rename to modules/audio_coding/acm2/acm_receiver.h
index bcedacd..d5a644d 100644
--- a/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
#include <map>
#include <vector>
@@ -20,10 +20,10 @@
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_audio/vad/include/webrtc_vad.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
-#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
@@ -302,4 +302,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
diff --git a/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc b/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
similarity index 98%
rename from modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
rename to modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
index 8f43ac4..8076687 100644
--- a/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
#include <algorithm> // std::min
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/test_suite.h"
diff --git a/modules/audio_coding/main/acm2/acm_resampler.cc b/modules/audio_coding/acm2/acm_resampler.cc
similarity index 96%
rename from modules/audio_coding/main/acm2/acm_resampler.cc
rename to modules/audio_coding/acm2/acm_resampler.cc
index cbcad85..e38cd94 100644
--- a/modules/audio_coding/main/acm2/acm_resampler.cc
+++ b/modules/audio_coding/acm2/acm_resampler.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
#include <assert.h>
#include <string.h>
diff --git a/modules/audio_coding/main/acm2/acm_resampler.h b/modules/audio_coding/acm2/acm_resampler.h
similarity index 83%
rename from modules/audio_coding/main/acm2/acm_resampler.h
rename to modules/audio_coding/acm2/acm_resampler.h
index a19b0c4..700fefa 100644
--- a/modules/audio_coding/main/acm2/acm_resampler.h
+++ b/modules/audio_coding/acm2/acm_resampler.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/typedefs.h"
@@ -36,4 +36,4 @@
} // namespace acm2
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
diff --git a/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc b/modules/audio_coding/acm2/acm_send_test_oldapi.cc
similarity index 97%
rename from modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
rename to modules/audio_coding/acm2/acm_send_test_oldapi.cc
index ac38dc0..3a89a77 100644
--- a/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
+++ b/modules/audio_coding/acm2/acm_send_test_oldapi.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h"
+#include "webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h"
#include <assert.h>
#include <stdio.h>
@@ -17,7 +17,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
diff --git a/modules/audio_coding/main/acm2/acm_send_test_oldapi.h b/modules/audio_coding/acm2/acm_send_test_oldapi.h
similarity index 91%
rename from modules/audio_coding/main/acm2/acm_send_test_oldapi.h
rename to modules/audio_coding/acm2/acm_send_test_oldapi.h
index 3e65ec6..ce68196 100644
--- a/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
+++ b/modules/audio_coding/acm2/acm_send_test_oldapi.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_
#include <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
#include "webrtc/system_wrappers/include/clock.h"
@@ -88,4 +88,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_
diff --git a/modules/audio_coding/main/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
similarity index 92%
rename from modules/audio_coding/main/acm2/audio_coding_module.cc
rename to modules/audio_coding/acm2/audio_coding_module.cc
index 889d620..034de32 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/base/checks.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
-#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/trace.h"
diff --git a/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/modules/audio_coding/acm2/audio_coding_module_impl.cc
similarity index 98%
rename from modules/audio_coding/main/acm2/audio_coding_module_impl.cc
rename to modules/audio_coding/acm2/audio_coding_module_impl.cc
index 5d18bda..5f61ef6 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_impl.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
+#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
#include <assert.h>
#include <stdlib.h>
@@ -17,10 +17,10 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/system_wrappers/include/metrics.h"
diff --git a/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/modules/audio_coding/acm2/audio_coding_module_impl.h
similarity index 95%
rename from modules/audio_coding/main/acm2/audio_coding_module_impl.h
rename to modules/audio_coding/acm2/audio_coding_module_impl.h
index c04ccf9..6006c68 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/modules/audio_coding/acm2/audio_coding_module_impl.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
#include <vector>
@@ -18,9 +18,9 @@
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/acm2/codec_manager.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
namespace webrtc {
@@ -277,4 +277,4 @@
} // namespace acm2
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
diff --git a/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
similarity index 99%
rename from modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
rename to modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index f14dcf3..39c14a8 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -21,10 +21,10 @@
#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h"
+#include "webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_checksum.h"
diff --git a/modules/audio_coding/main/acm2/call_statistics.cc b/modules/audio_coding/acm2/call_statistics.cc
similarity index 95%
rename from modules/audio_coding/main/acm2/call_statistics.cc
rename to modules/audio_coding/acm2/call_statistics.cc
index 4c3e9fc..4441932 100644
--- a/modules/audio_coding/main/acm2/call_statistics.cc
+++ b/modules/audio_coding/acm2/call_statistics.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
#include <assert.h>
diff --git a/modules/audio_coding/main/acm2/call_statistics.h b/modules/audio_coding/acm2/call_statistics.h
similarity index 90%
rename from modules/audio_coding/main/acm2/call_statistics.h
rename to modules/audio_coding/acm2/call_statistics.h
index e2df921..888afea 100644
--- a/modules/audio_coding/main/acm2/call_statistics.h
+++ b/modules/audio_coding/acm2/call_statistics.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
#include "webrtc/common_types.h"
#include "webrtc/modules/include/module_common_types.h"
@@ -60,4 +60,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
diff --git a/modules/audio_coding/main/acm2/call_statistics_unittest.cc b/modules/audio_coding/acm2/call_statistics_unittest.cc
similarity index 95%
rename from modules/audio_coding/main/acm2/call_statistics_unittest.cc
rename to modules/audio_coding/acm2/call_statistics_unittest.cc
index 2bee964..9ba0774 100644
--- a/modules/audio_coding/main/acm2/call_statistics_unittest.cc
+++ b/modules/audio_coding/acm2/call_statistics_unittest.cc
@@ -9,7 +9,7 @@
*/
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
+#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
namespace webrtc {
diff --git a/modules/audio_coding/main/acm2/codec_manager.cc b/modules/audio_coding/acm2/codec_manager.cc
similarity index 98%
rename from modules/audio_coding/main/acm2/codec_manager.cc
rename to modules/audio_coding/acm2/codec_manager.cc
index 7796786..a5a9e09 100644
--- a/modules/audio_coding/main/acm2/codec_manager.cc
+++ b/modules/audio_coding/acm2/codec_manager.cc
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/acm2/codec_manager.h"
+#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
#include "webrtc/base/checks.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/system_wrappers/include/trace.h"
namespace webrtc {
diff --git a/modules/audio_coding/main/acm2/codec_manager.h b/modules/audio_coding/acm2/codec_manager.h
similarity index 86%
rename from modules/audio_coding/main/acm2/codec_manager.h
rename to modules/audio_coding/acm2/codec_manager.h
index 7670bbd..61832e4 100644
--- a/modules/audio_coding/main/acm2/codec_manager.h
+++ b/modules/audio_coding/acm2/codec_manager.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_MANAGER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
#include <map>
@@ -17,8 +17,8 @@
#include "webrtc/base/optional.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "webrtc/common_types.h"
namespace webrtc {
@@ -78,4 +78,4 @@
} // namespace acm2
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_MANAGER_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
diff --git a/modules/audio_coding/main/acm2/codec_manager_unittest.cc b/modules/audio_coding/acm2/codec_manager_unittest.cc
similarity index 96%
rename from modules/audio_coding/main/acm2/codec_manager_unittest.cc
rename to modules/audio_coding/acm2/codec_manager_unittest.cc
index e930ca1..c09f256 100644
--- a/modules/audio_coding/main/acm2/codec_manager_unittest.cc
+++ b/modules/audio_coding/acm2/codec_manager_unittest.cc
@@ -10,7 +10,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
-#include "webrtc/modules/audio_coding/main/acm2/codec_manager.h"
+#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
namespace webrtc {
namespace acm2 {
diff --git a/modules/audio_coding/main/acm2/initial_delay_manager.cc b/modules/audio_coding/acm2/initial_delay_manager.cc
similarity index 98%
rename from modules/audio_coding/main/acm2/initial_delay_manager.cc
rename to modules/audio_coding/acm2/initial_delay_manager.cc
index 786fb2e..0c31b83 100644
--- a/modules/audio_coding/main/acm2/initial_delay_manager.cc
+++ b/modules/audio_coding/acm2/initial_delay_manager.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
+#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
namespace webrtc {
diff --git a/modules/audio_coding/main/acm2/initial_delay_manager.h b/modules/audio_coding/acm2/initial_delay_manager.h
similarity index 94%
rename from modules/audio_coding/main/acm2/initial_delay_manager.h
rename to modules/audio_coding/acm2/initial_delay_manager.h
index 6b50dd0..32dd126 100644
--- a/modules/audio_coding/main/acm2/initial_delay_manager.h
+++ b/modules/audio_coding/acm2/initial_delay_manager.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/include/module_common_types.h"
@@ -117,4 +117,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
diff --git a/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc b/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
similarity index 99%
rename from modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc
rename to modules/audio_coding/acm2/initial_delay_manager_unittest.cc
index e973593..d86d221 100644
--- a/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc
+++ b/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
@@ -11,7 +11,7 @@
#include <string.h>
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
+#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
namespace webrtc {
diff --git a/modules/audio_coding/main/acm2/rent_a_codec.cc b/modules/audio_coding/acm2/rent_a_codec.cc
similarity index 97%
rename from modules/audio_coding/main/acm2/rent_a_codec.cc
rename to modules/audio_coding/acm2/rent_a_codec.cc
index 229d367..4800249 100644
--- a/modules/audio_coding/main/acm2/rent_a_codec.cc
+++ b/modules/audio_coding/acm2/rent_a_codec.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
@@ -34,8 +34,8 @@
#ifdef WEBRTC_CODEC_RED
#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
#endif
-#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
namespace webrtc {
namespace acm2 {
diff --git a/modules/audio_coding/main/acm2/rent_a_codec.h b/modules/audio_coding/acm2/rent_a_codec.h
similarity index 96%
rename from modules/audio_coding/main/acm2/rent_a_codec.h
rename to modules/audio_coding/acm2/rent_a_codec.h
index 45d46bb..7035104 100644
--- a/modules/audio_coding/main/acm2/rent_a_codec.h
+++ b/modules/audio_coding/acm2/rent_a_codec.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_RENT_A_CODEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_RENT_A_CODEC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
#include <stddef.h>
#include <map>
@@ -20,7 +20,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "webrtc/typedefs.h"
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
@@ -246,4 +246,4 @@
} // namespace acm2
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_RENT_A_CODEC_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
diff --git a/modules/audio_coding/main/acm2/rent_a_codec_unittest.cc b/modules/audio_coding/acm2/rent_a_codec_unittest.cc
similarity index 98%
rename from modules/audio_coding/main/acm2/rent_a_codec_unittest.cc
rename to modules/audio_coding/acm2/rent_a_codec_unittest.cc
index ae6c98b..11c4bcb 100644
--- a/modules/audio_coding/main/acm2/rent_a_codec_unittest.cc
+++ b/modules/audio_coding/acm2/rent_a_codec_unittest.cc
@@ -11,7 +11,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/arraysize.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
-#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
namespace webrtc {
namespace acm2 {
diff --git a/modules/audio_coding/audio_coding.gypi b/modules/audio_coding/audio_coding.gypi
index bc3c48d..abdb1915 100644
--- a/modules/audio_coding/audio_coding.gypi
+++ b/modules/audio_coding/audio_coding.gypi
@@ -19,12 +19,195 @@
'codecs/isac/isacfix.gypi',
'codecs/pcm16b/pcm16b.gypi',
'codecs/red/red.gypi',
- 'main/audio_coding_module.gypi',
'neteq/neteq.gypi',
],
+ 'variables': {
+ 'audio_coding_dependencies': [
+ 'cng',
+ 'g711',
+ 'pcm16b',
+ '<(webrtc_root)/common.gyp:webrtc_common',
+ '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+ ],
+ 'audio_coding_defines': [],
+ 'conditions': [
+ ['include_opus==1', {
+ 'audio_coding_dependencies': ['webrtc_opus',],
+ 'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
+ }],
+ ['build_with_mozilla==0', {
+ 'conditions': [
+ ['target_arch=="arm"', {
+ 'audio_coding_dependencies': ['isac_fix',],
+ 'audio_coding_defines': ['WEBRTC_CODEC_ISACFX',],
+ }, {
+ 'audio_coding_dependencies': ['isac',],
+ 'audio_coding_defines': ['WEBRTC_CODEC_ISAC',],
+ }],
+ ],
+ 'audio_coding_dependencies': ['g722',],
+ 'audio_coding_defines': ['WEBRTC_CODEC_G722',],
+ }],
+ ['build_with_mozilla==0 and build_with_chromium==0', {
+ 'audio_coding_dependencies': ['ilbc', 'red',],
+ 'audio_coding_defines': ['WEBRTC_CODEC_ILBC', 'WEBRTC_CODEC_RED',],
+ }],
+ ],
+ },
+ 'targets': [
+ {
+ 'target_name': 'rent_a_codec',
+ 'type': 'static_library',
+ 'defines': [
+ '<@(audio_coding_defines)',
+ ],
+ 'dependencies': [
+ '<(webrtc_root)/common.gyp:webrtc_common',
+ ],
+ 'include_dirs': [
+ '<(webrtc_root)',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ '<(webrtc_root)',
+ ],
+ },
+ 'sources': [
+ 'acm2/acm_codec_database.cc',
+ 'acm2/acm_codec_database.h',
+ 'acm2/rent_a_codec.cc',
+ 'acm2/rent_a_codec.h',
+ ],
+ },
+ {
+ 'target_name': 'audio_coding_module',
+ 'type': 'static_library',
+ 'defines': [
+ '<@(audio_coding_defines)',
+ ],
+ 'dependencies': [
+ '<@(audio_coding_dependencies)',
+ '<(webrtc_root)/common.gyp:webrtc_common',
+ '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+ 'neteq',
+ 'rent_a_codec',
+ ],
+ 'include_dirs': [
+ 'include',
+ '../include',
+ '<(webrtc_root)',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'include',
+ '../include',
+ '<(webrtc_root)',
+ ],
+ },
+ 'conditions': [
+ ['include_opus==1', {
+ 'export_dependent_settings': ['webrtc_opus'],
+ }],
+ ],
+ 'sources': [
+ 'acm2/acm_common_defs.h',
+ 'acm2/acm_receiver.cc',
+ 'acm2/acm_receiver.h',
+ 'acm2/acm_resampler.cc',
+ 'acm2/acm_resampler.h',
+ 'acm2/audio_coding_module.cc',
+ 'acm2/audio_coding_module_impl.cc',
+ 'acm2/audio_coding_module_impl.h',
+ 'acm2/call_statistics.cc',
+ 'acm2/call_statistics.h',
+ 'acm2/codec_manager.cc',
+ 'acm2/codec_manager.h',
+ 'acm2/initial_delay_manager.cc',
+ 'acm2/initial_delay_manager.h',
+ 'include/audio_coding_module.h',
+ 'include/audio_coding_module_typedefs.h',
+ ],
+ },
+ ],
'conditions': [
['include_opus==1', {
'includes': ['codecs/opus/opus.gypi',],
}],
+ ['include_tests==1', {
+ 'targets': [
+ {
+ 'target_name': 'acm_receive_test',
+ 'type': 'static_library',
+ 'defines': [
+ '<@(audio_coding_defines)',
+ ],
+ 'dependencies': [
+ '<@(audio_coding_dependencies)',
+ 'audio_coding_module',
+ 'neteq_unittest_tools',
+ '<(DEPTH)/testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'acm2/acm_receive_test_oldapi.cc',
+ 'acm2/acm_receive_test_oldapi.h',
+ ],
+ }, # acm_receive_test
+ {
+ 'target_name': 'acm_send_test',
+ 'type': 'static_library',
+ 'defines': [
+ '<@(audio_coding_defines)',
+ ],
+ 'dependencies': [
+ '<@(audio_coding_dependencies)',
+ 'audio_coding_module',
+ 'neteq_unittest_tools',
+ '<(DEPTH)/testing/gtest.gyp:gtest',
+ ],
+ 'sources': [
+ 'acm2/acm_send_test_oldapi.cc',
+ 'acm2/acm_send_test_oldapi.h',
+ ],
+ }, # acm_send_test
+ {
+ 'target_name': 'delay_test',
+ 'type': 'executable',
+ 'dependencies': [
+ 'audio_coding_module',
+ '<(DEPTH)/testing/gtest.gyp:gtest',
+ '<(webrtc_root)/common.gyp:webrtc_common',
+ '<(webrtc_root)/test/test.gyp:test_support',
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+ '<(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)/common.gyp:webrtc_common',
+ '<(webrtc_root)/test/test.gyp:test_support',
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
+ '<(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/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
new file mode 100644
index 0000000..844bd57
--- /dev/null
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -0,0 +1,741 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
+
+#include <vector>
+
+#include "webrtc/base/optional.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// forward declarations
+struct CodecInst;
+struct WebRtcRTPHeader;
+class AudioDecoder;
+class AudioEncoder;
+class AudioFrame;
+class RTPFragmentationHeader;
+
+#define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz
+
+// Callback class used for sending data ready to be packetized
+class AudioPacketizationCallback {
+ public:
+ virtual ~AudioPacketizationCallback() {}
+
+ virtual int32_t SendData(FrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_len_bytes,
+ const RTPFragmentationHeader* fragmentation) = 0;
+};
+
+// Callback class used for reporting VAD decision
+class ACMVADCallback {
+ public:
+ virtual ~ACMVADCallback() {}
+
+ virtual int32_t InFrameType(FrameType frame_type) = 0;
+};
+
+class AudioCodingModule {
+ protected:
+ AudioCodingModule() {}
+
+ public:
+ struct Config {
+ Config() : id(0), neteq_config(), clock(Clock::GetRealTimeClock()) {
+ // Post-decode VAD is disabled by default in NetEq, however, Audio
+ // Conference Mixer relies on VAD decisions and fails without them.
+ neteq_config.enable_post_decode_vad = true;
+ }
+
+ int id;
+ NetEq::Config neteq_config;
+ Clock* clock;
+ };
+
+ ///////////////////////////////////////////////////////////////////////////
+ // Creation and destruction of a ACM.
+ //
+ // The second method is used for testing where a simulated clock can be
+ // injected into ACM. ACM will take the ownership of the object clock and
+ // delete it when destroyed.
+ //
+ static AudioCodingModule* Create(int id);
+ static AudioCodingModule* Create(int id, Clock* clock);
+ static AudioCodingModule* Create(const Config& config);
+ virtual ~AudioCodingModule() = default;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // Utility functions
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // uint8_t NumberOfCodecs()
+ // Returns number of supported codecs.
+ //
+ // Return value:
+ // number of supported codecs.
+ ///
+ static int NumberOfCodecs();
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t Codec()
+ // Get supported codec with list number.
+ //
+ // Input:
+ // -list_id : list number.
+ //
+ // Output:
+ // -codec : a structure where the parameters of the codec,
+ // given by list number is written to.
+ //
+ // Return value:
+ // -1 if the list number (list_id) is invalid.
+ // 0 if succeeded.
+ //
+ static int Codec(int list_id, CodecInst* codec);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t Codec()
+ // Get supported codec with the given codec name, sampling frequency, and
+ // a given number of channels.
+ //
+ // Input:
+ // -payload_name : name of the codec.
+ // -sampling_freq_hz : sampling frequency of the codec. Note! for RED
+ // a sampling frequency of -1 is a valid input.
+ // -channels : number of channels ( 1 - mono, 2 - stereo).
+ //
+ // Output:
+ // -codec : a structure where the function returns the
+ // default parameters of the codec.
+ //
+ // Return value:
+ // -1 if no codec matches the given parameters.
+ // 0 if succeeded.
+ //
+ static int Codec(const char* payload_name, CodecInst* codec,
+ int sampling_freq_hz, int channels);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t Codec()
+ //
+ // Returns the list number of the given codec name, sampling frequency, and
+ // a given number of channels.
+ //
+ // Input:
+ // -payload_name : name of the codec.
+ // -sampling_freq_hz : sampling frequency of the codec. Note! for RED
+ // a sampling frequency of -1 is a valid input.
+ // -channels : number of channels ( 1 - mono, 2 - stereo).
+ //
+ // Return value:
+ // if the codec is found, the index of the codec in the list,
+ // -1 if the codec is not found.
+ //
+ static int Codec(const char* payload_name, int sampling_freq_hz,
+ int channels);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // bool IsCodecValid()
+ // Checks the validity of the parameters of the given codec.
+ //
+ // Input:
+ // -codec : the structure which keeps the parameters of the
+ // codec.
+ //
+ // Return value:
+ // true if the parameters are valid,
+ // false if any parameter is not valid.
+ //
+ static bool IsCodecValid(const CodecInst& codec);
+
+ ///////////////////////////////////////////////////////////////////////////
+ // Sender
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t RegisterSendCodec()
+ // Registers a codec, specified by |send_codec|, as sending codec.
+ // This API can be called multiple of times to register Codec. The last codec
+ // registered overwrites the previous ones.
+ // The API can also be used to change payload type for CNG and RED, which are
+ // registered by default to default payload types.
+ // Note that registering CNG and RED won't overwrite speech codecs.
+ // This API can be called to set/change the send payload-type, frame-size
+ // or encoding rate (if applicable for the codec).
+ //
+ // Note: If a stereo codec is registered as send codec, VAD/DTX will
+ // automatically be turned off, since it is not supported for stereo sending.
+ //
+ // Note: If a secondary encoder is already registered, and the new send-codec
+ // has a sampling rate that does not match the secondary encoder, the
+ // secondary encoder will be unregistered.
+ //
+ // Input:
+ // -send_codec : Parameters of the codec to be registered, c.f.
+ // common_types.h for the definition of
+ // CodecInst.
+ //
+ // Return value:
+ // -1 if failed to initialize,
+ // 0 if succeeded.
+ //
+ virtual int32_t RegisterSendCodec(const CodecInst& send_codec) = 0;
+
+ // Registers |external_speech_encoder| as encoder. The new encoder will
+ // replace any previously registered speech encoder (internal or external).
+ virtual void RegisterExternalSendCodec(
+ AudioEncoder* external_speech_encoder) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t SendCodec()
+ // Get parameters for the codec currently registered as send codec.
+ //
+ // Return value:
+ // The send codec, or nothing if we don't have one
+ //
+ virtual rtc::Optional<CodecInst> SendCodec() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t SendFrequency()
+ // Get the sampling frequency of the current encoder in Hertz.
+ //
+ // Return value:
+ // positive; sampling frequency [Hz] of the current encoder.
+ // -1 if an error has happened.
+ //
+ virtual int32_t SendFrequency() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // Sets the bitrate to the specified value in bits/sec. If the value is not
+ // supported by the codec, it will choose another appropriate value.
+ virtual void SetBitRate(int bitrate_bps) = 0;
+
+ // int32_t RegisterTransportCallback()
+ // Register a transport callback which will be called to deliver
+ // the encoded buffers whenever Process() is called and a
+ // bit-stream is ready.
+ //
+ // Input:
+ // -transport : pointer to the callback class
+ // transport->SendData() is called whenever
+ // Process() is called and bit-stream is ready
+ // to deliver.
+ //
+ // Return value:
+ // -1 if the transport callback could not be registered
+ // 0 if registration is successful.
+ //
+ virtual int32_t RegisterTransportCallback(
+ AudioPacketizationCallback* transport) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t Add10MsData()
+ // Add 10MS of raw (PCM) audio data and encode it. If the sampling
+ // frequency of the audio does not match the sampling frequency of the
+ // current encoder ACM will resample the audio. If an encoded packet was
+ // produced, it will be delivered via the callback object registered using
+ // RegisterTransportCallback, and the return value from this function will
+ // be the number of bytes encoded.
+ //
+ // Input:
+ // -audio_frame : the input audio frame, containing raw audio
+ // sampling frequency etc.,
+ // c.f. module_common_types.h for definition of
+ // AudioFrame.
+ //
+ // Return value:
+ // >= 0 number of bytes encoded.
+ // -1 some error occurred.
+ //
+ virtual int32_t Add10MsData(const AudioFrame& audio_frame) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // (RED) Redundant Coding
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t SetREDStatus()
+ // configure RED status i.e. on/off.
+ //
+ // RFC 2198 describes a solution which has a single payload type which
+ // signifies a packet with redundancy. That packet then becomes a container,
+ // encapsulating multiple payloads into a single RTP packet.
+ // Such a scheme is flexible, since any amount of redundancy may be
+ // encapsulated within a single packet. There is, however, a small overhead
+ // since each encapsulated payload must be preceded by a header indicating
+ // the type of data enclosed.
+ //
+ // Input:
+ // -enable_red : if true RED is enabled, otherwise RED is
+ // disabled.
+ //
+ // Return value:
+ // -1 if failed to set RED status,
+ // 0 if succeeded.
+ //
+ virtual int32_t SetREDStatus(bool enable_red) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // bool REDStatus()
+ // Get RED status
+ //
+ // Return value:
+ // true if RED is enabled,
+ // false if RED is disabled.
+ //
+ virtual bool REDStatus() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // (FEC) Forward Error Correction (codec internal)
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t SetCodecFEC()
+ // Configures codec internal FEC status i.e. on/off. No effects on codecs that
+ // do not provide internal FEC.
+ //
+ // Input:
+ // -enable_fec : if true FEC will be enabled otherwise the FEC is
+ // disabled.
+ //
+ // Return value:
+ // -1 if failed, or the codec does not support FEC
+ // 0 if succeeded.
+ //
+ virtual int SetCodecFEC(bool enable_codec_fec) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // bool CodecFEC()
+ // Gets status of codec internal FEC.
+ //
+ // Return value:
+ // true if FEC is enabled,
+ // false if FEC is disabled.
+ //
+ virtual bool CodecFEC() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int SetPacketLossRate()
+ // Sets expected packet loss rate for encoding. Some encoders provide packet
+ // loss gnostic encoding to make stream less sensitive to packet losses,
+ // through e.g., FEC. No effects on codecs that do not provide such encoding.
+ //
+ // Input:
+ // -packet_loss_rate : expected packet loss rate (0 -- 100 inclusive).
+ //
+ // Return value
+ // -1 if failed to set packet loss rate,
+ // 0 if succeeded.
+ //
+ virtual int SetPacketLossRate(int packet_loss_rate) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // (VAD) Voice Activity Detection
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t SetVAD()
+ // If DTX is enabled & the codec does not have internal DTX/VAD
+ // WebRtc VAD will be automatically enabled and |enable_vad| is ignored.
+ //
+ // If DTX is disabled but VAD is enabled no DTX packets are send,
+ // regardless of whether the codec has internal DTX/VAD or not. In this
+ // case, WebRtc VAD is running to label frames as active/in-active.
+ //
+ // NOTE! VAD/DTX is not supported when sending stereo.
+ //
+ // Inputs:
+ // -enable_dtx : if true DTX is enabled,
+ // otherwise DTX is disabled.
+ // -enable_vad : if true VAD is enabled,
+ // otherwise VAD is disabled.
+ // -vad_mode : determines the aggressiveness of VAD. A more
+ // aggressive mode results in more frames labeled
+ // as in-active, c.f. definition of
+ // ACMVADMode in audio_coding_module_typedefs.h
+ // for valid values.
+ //
+ // Return value:
+ // -1 if failed to set up VAD/DTX,
+ // 0 if succeeded.
+ //
+ virtual int32_t SetVAD(const bool enable_dtx = true,
+ const bool enable_vad = false,
+ const ACMVADMode vad_mode = VADNormal) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t VAD()
+ // Get VAD status.
+ //
+ // Outputs:
+ // -dtx_enabled : is set to true if DTX is enabled, otherwise
+ // is set to false.
+ // -vad_enabled : is set to true if VAD is enabled, otherwise
+ // is set to false.
+ // -vad_mode : is set to the current aggressiveness of VAD.
+ //
+ // Return value:
+ // -1 if fails to retrieve the setting of DTX/VAD,
+ // 0 if succeeded.
+ //
+ virtual int32_t VAD(bool* dtx_enabled, bool* vad_enabled,
+ ACMVADMode* vad_mode) const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t RegisterVADCallback()
+ // Call this method to register a callback function which is called
+ // any time that ACM encounters an empty frame. That is a frame which is
+ // recognized inactive. Depending on the codec WebRtc VAD or internal codec
+ // VAD is employed to identify a frame as active/inactive.
+ //
+ // Input:
+ // -vad_callback : pointer to a callback function.
+ //
+ // Return value:
+ // -1 if failed to register the callback function.
+ // 0 if the callback function is registered successfully.
+ //
+ virtual int32_t RegisterVADCallback(ACMVADCallback* vad_callback) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // Receiver
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t InitializeReceiver()
+ // Any decoder-related state of ACM will be initialized to the
+ // same state when ACM is created. This will not interrupt or
+ // effect encoding functionality of ACM. ACM would lose all the
+ // decoding-related settings by calling this function.
+ // For instance, all registered codecs are deleted and have to be
+ // registered again.
+ //
+ // Return value:
+ // -1 if failed to initialize,
+ // 0 if succeeded.
+ //
+ virtual int32_t InitializeReceiver() = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t ReceiveFrequency()
+ // Get sampling frequency of the last received payload.
+ //
+ // Return value:
+ // non-negative the sampling frequency in Hertz.
+ // -1 if an error has occurred.
+ //
+ virtual int32_t ReceiveFrequency() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t PlayoutFrequency()
+ // Get sampling frequency of audio played out.
+ //
+ // Return value:
+ // the sampling frequency in Hertz.
+ //
+ virtual int32_t PlayoutFrequency() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t RegisterReceiveCodec()
+ // Register possible decoders, can be called multiple times for
+ // codecs, CNG-NB, CNG-WB, CNG-SWB, AVT and RED.
+ //
+ // Input:
+ // -receive_codec : parameters of the codec to be registered, c.f.
+ // common_types.h for the definition of
+ // CodecInst.
+ //
+ // Return value:
+ // -1 if failed to register the codec
+ // 0 if the codec registered successfully.
+ //
+ virtual int RegisterReceiveCodec(const CodecInst& receive_codec) = 0;
+
+ virtual int RegisterExternalReceiveCodec(int rtp_payload_type,
+ AudioDecoder* external_decoder,
+ int sample_rate_hz,
+ int num_channels) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t UnregisterReceiveCodec()
+ // Unregister the codec currently registered with a specific payload type
+ // from the list of possible receive codecs.
+ //
+ // Input:
+ // -payload_type : The number representing the payload type to
+ // unregister.
+ //
+ // Output:
+ // -1 if fails to unregister.
+ // 0 if the given codec is successfully unregistered.
+ //
+ virtual int UnregisterReceiveCodec(
+ uint8_t payload_type) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t ReceiveCodec()
+ // Get the codec associated with last received payload.
+ //
+ // Output:
+ // -curr_receive_codec : parameters of the codec associated with the last
+ // received payload, c.f. common_types.h for
+ // the definition of CodecInst.
+ //
+ // Return value:
+ // -1 if failed to retrieve the codec,
+ // 0 if the codec is successfully retrieved.
+ //
+ virtual int32_t ReceiveCodec(CodecInst* curr_receive_codec) const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t IncomingPacket()
+ // Call this function to insert a parsed RTP packet into ACM.
+ //
+ // Inputs:
+ // -incoming_payload : received payload.
+ // -payload_len_bytes : the length of payload in bytes.
+ // -rtp_info : the relevant information retrieved from RTP
+ // header.
+ //
+ // Return value:
+ // -1 if failed to push in the payload
+ // 0 if payload is successfully pushed in.
+ //
+ virtual int32_t IncomingPacket(const uint8_t* incoming_payload,
+ const size_t payload_len_bytes,
+ const WebRtcRTPHeader& rtp_info) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t IncomingPayload()
+ // Call this API to push incoming payloads when there is no rtp-info.
+ // The rtp-info will be created in ACM. One usage for this API is when
+ // pre-encoded files are pushed in ACM
+ //
+ // Inputs:
+ // -incoming_payload : received payload.
+ // -payload_len_byte : the length, in bytes, of the received payload.
+ // -payload_type : the payload-type. This specifies which codec has
+ // to be used to decode the payload.
+ // -timestamp : send timestamp of the payload. ACM starts with
+ // a random value and increment it by the
+ // packet-size, which is given when the codec in
+ // question is registered by RegisterReceiveCodec().
+ // Therefore, it is essential to have the timestamp
+ // if the frame-size differ from the registered
+ // value or if the incoming payload contains DTX
+ // packets.
+ //
+ // Return value:
+ // -1 if failed to push in the payload
+ // 0 if payload is successfully pushed in.
+ //
+ virtual int32_t IncomingPayload(const uint8_t* incoming_payload,
+ const size_t payload_len_byte,
+ const uint8_t payload_type,
+ const uint32_t timestamp = 0) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int SetMinimumPlayoutDelay()
+ // Set a minimum for the playout delay, used for lip-sync. NetEq maintains
+ // such a delay unless channel condition yields to a higher delay.
+ //
+ // Input:
+ // -time_ms : minimum delay in milliseconds.
+ //
+ // Return value:
+ // -1 if failed to set the delay,
+ // 0 if the minimum delay is set.
+ //
+ virtual int SetMinimumPlayoutDelay(int time_ms) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int SetMaximumPlayoutDelay()
+ // Set a maximum for the playout delay
+ //
+ // Input:
+ // -time_ms : maximum delay in milliseconds.
+ //
+ // Return value:
+ // -1 if failed to set the delay,
+ // 0 if the maximum delay is set.
+ //
+ virtual int SetMaximumPlayoutDelay(int time_ms) = 0;
+
+ //
+ // The shortest latency, in milliseconds, required by jitter buffer. This
+ // is computed based on inter-arrival times and playout mode of NetEq. The
+ // actual delay is the maximum of least-required-delay and the minimum-delay
+ // specified by SetMinumumPlayoutDelay() API.
+ //
+ virtual int LeastRequiredDelayMs() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t PlayoutTimestamp()
+ // The send timestamp of an RTP packet is associated with the decoded
+ // audio of the packet in question. This function returns the timestamp of
+ // the latest audio obtained by calling PlayoutData10ms().
+ //
+ // Input:
+ // -timestamp : a reference to a uint32_t to receive the
+ // timestamp.
+ // Return value:
+ // 0 if the output is a correct timestamp.
+ // -1 if failed to output the correct timestamp.
+ //
+ // TODO(tlegrand): Change function to return the timestamp.
+ virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t PlayoutData10Ms(
+ // Get 10 milliseconds of raw audio data for playout, at the given sampling
+ // frequency. ACM will perform a resampling if required.
+ //
+ // Input:
+ // -desired_freq_hz : the desired sampling frequency, in Hertz, of the
+ // output audio. If set to -1, the function returns
+ // the audio at the current sampling frequency.
+ //
+ // Output:
+ // -audio_frame : output audio frame which contains raw audio data
+ // and other relevant parameters, c.f.
+ // module_common_types.h for the definition of
+ // AudioFrame.
+ //
+ // Return value:
+ // -1 if the function fails,
+ // 0 if the function succeeds.
+ //
+ virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz,
+ AudioFrame* audio_frame) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // Codec specific
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int SetOpusApplication()
+ // Sets the intended application if current send codec is Opus. Opus uses this
+ // to optimize the encoding for applications like VOIP and music. Currently,
+ // two modes are supported: kVoip and kAudio.
+ //
+ // Input:
+ // - application : intended application.
+ //
+ // Return value:
+ // -1 if current send codec is not Opus or error occurred in setting the
+ // Opus application mode.
+ // 0 if the Opus application mode is successfully set.
+ //
+ virtual int SetOpusApplication(OpusApplicationMode application) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int SetOpusMaxPlaybackRate()
+ // If current send codec is Opus, informs it about maximum playback rate the
+ // receiver will render. Opus can use this information to optimize the bit
+ // rate and increase the computation efficiency.
+ //
+ // Input:
+ // -frequency_hz : maximum playback rate in Hz.
+ //
+ // Return value:
+ // -1 if current send codec is not Opus or
+ // error occurred in setting the maximum playback rate,
+ // 0 if maximum bandwidth is set successfully.
+ //
+ virtual int SetOpusMaxPlaybackRate(int frequency_hz) = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // EnableOpusDtx()
+ // Enable the DTX, if current send codec is Opus.
+ //
+ // Return value:
+ // -1 if current send codec is not Opus or error occurred in enabling the
+ // Opus DTX.
+ // 0 if Opus DTX is enabled successfully.
+ //
+ virtual int EnableOpusDtx() = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int DisableOpusDtx()
+ // If current send codec is Opus, disables its internal DTX.
+ //
+ // Return value:
+ // -1 if current send codec is not Opus or error occurred in disabling DTX.
+ // 0 if Opus DTX is disabled successfully.
+ //
+ virtual int DisableOpusDtx() = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
+ // statistics
+ //
+
+ ///////////////////////////////////////////////////////////////////////////
+ // int32_t GetNetworkStatistics()
+ // Get network statistics. Note that the internal statistics of NetEq are
+ // reset by this call.
+ //
+ // Input:
+ // -network_statistics : a structure that contains network statistics.
+ //
+ // Return value:
+ // -1 if failed to set the network statistics,
+ // 0 if statistics are set successfully.
+ //
+ virtual int32_t GetNetworkStatistics(
+ NetworkStatistics* network_statistics) = 0;
+
+ //
+ // Enable NACK and set the maximum size of the NACK list. If NACK is already
+ // enable then the maximum NACK list size is modified accordingly.
+ //
+ // If the sequence number of last received packet is N, the sequence numbers
+ // of NACK list are in the range of [N - |max_nack_list_size|, N).
+ //
+ // |max_nack_list_size| should be positive (none zero) and less than or
+ // equal to |Nack::kNackListSizeLimit|. Otherwise, No change is applied and -1
+ // is returned. 0 is returned at success.
+ //
+ virtual int EnableNack(size_t max_nack_list_size) = 0;
+
+ // Disable NACK.
+ virtual void DisableNack() = 0;
+
+ //
+ // Get a list of packets to be retransmitted. |round_trip_time_ms| is an
+ // estimate of the round-trip-time (in milliseconds). Missing packets which
+ // will be playout in a shorter time than the round-trip-time (with respect
+ // to the time this API is called) will not be included in the list.
+ //
+ // Negative |round_trip_time_ms| results is an error message and empty list
+ // is returned.
+ //
+ virtual std::vector<uint16_t> GetNackList(
+ int64_t round_trip_time_ms) const = 0;
+
+ virtual void GetDecodingCallStatistics(
+ AudioDecodingCallStats* call_stats) const = 0;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
diff --git a/modules/audio_coding/include/audio_coding_module_typedefs.h b/modules/audio_coding/include/audio_coding_module_typedefs.h
new file mode 100644
index 0000000..280d6bf
--- /dev/null
+++ b/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+
+#include <map>
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+///////////////////////////////////////////////////////////////////////////
+// enum ACMVADMode
+// An enumerator for aggressiveness of VAD
+// -VADNormal : least aggressive mode.
+// -VADLowBitrate : more aggressive than "VADNormal" to save on
+// bit-rate.
+// -VADAggr : an aggressive mode.
+// -VADVeryAggr : the most agressive mode.
+//
+enum ACMVADMode {
+ VADNormal = 0,
+ VADLowBitrate = 1,
+ VADAggr = 2,
+ VADVeryAggr = 3
+};
+
+///////////////////////////////////////////////////////////////////////////
+//
+// Enumeration of Opus mode for intended application.
+//
+// kVoip : optimized for voice signals.
+// kAudio : optimized for non-voice signals like music.
+//
+enum OpusApplicationMode {
+ kVoip = 0,
+ kAudio = 1,
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
diff --git a/modules/audio_coding/main/acm2/OWNERS b/modules/audio_coding/main/acm2/OWNERS
deleted file mode 100644
index 3ee6b4b..0000000
--- a/modules/audio_coding/main/acm2/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gyp=*
-per-file *.gypi=*
diff --git a/modules/audio_coding/main/audio_coding_module.gypi b/modules/audio_coding/main/audio_coding_module.gypi
deleted file mode 100644
index 061ffaa..0000000
--- a/modules/audio_coding/main/audio_coding_module.gypi
+++ /dev/null
@@ -1,196 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'variables': {
- 'audio_coding_dependencies': [
- 'cng',
- 'g711',
- 'pcm16b',
- '<(webrtc_root)/common.gyp:webrtc_common',
- '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
- ],
- 'audio_coding_defines': [],
- 'conditions': [
- ['include_opus==1', {
- 'audio_coding_dependencies': ['webrtc_opus',],
- 'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
- }],
- ['build_with_mozilla==0', {
- 'conditions': [
- ['target_arch=="arm"', {
- 'audio_coding_dependencies': ['isac_fix',],
- 'audio_coding_defines': ['WEBRTC_CODEC_ISACFX',],
- }, {
- 'audio_coding_dependencies': ['isac',],
- 'audio_coding_defines': ['WEBRTC_CODEC_ISAC',],
- }],
- ],
- 'audio_coding_dependencies': ['g722',],
- 'audio_coding_defines': ['WEBRTC_CODEC_G722',],
- }],
- ['build_with_mozilla==0 and build_with_chromium==0', {
- 'audio_coding_dependencies': ['ilbc', 'red',],
- 'audio_coding_defines': ['WEBRTC_CODEC_ILBC', 'WEBRTC_CODEC_RED',],
- }],
- ],
- },
- 'targets': [
- {
- 'target_name': 'rent_a_codec',
- 'type': 'static_library',
- 'defines': [
- '<@(audio_coding_defines)',
- ],
- 'dependencies': [
- '<(webrtc_root)/common.gyp:webrtc_common',
- ],
- 'include_dirs': [
- '<(webrtc_root)',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '<(webrtc_root)',
- ],
- },
- 'sources': [
- 'acm2/acm_codec_database.cc',
- 'acm2/acm_codec_database.h',
- 'acm2/rent_a_codec.cc',
- 'acm2/rent_a_codec.h',
- ],
- },
- {
- 'target_name': 'audio_coding_module',
- 'type': 'static_library',
- 'defines': [
- '<@(audio_coding_defines)',
- ],
- 'dependencies': [
- '<@(audio_coding_dependencies)',
- '<(webrtc_root)/common.gyp:webrtc_common',
- '<(webrtc_root)/webrtc.gyp:rtc_event_log',
- 'neteq',
- 'rent_a_codec',
- ],
- 'include_dirs': [
- 'include',
- '../../include',
- '<(webrtc_root)',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- '../../include',
- '<(webrtc_root)',
- ],
- },
- 'conditions': [
- ['include_opus==1', {
- 'export_dependent_settings': ['webrtc_opus'],
- }],
- ],
- 'sources': [
- 'acm2/acm_common_defs.h',
- 'acm2/acm_receiver.cc',
- 'acm2/acm_receiver.h',
- 'acm2/acm_resampler.cc',
- 'acm2/acm_resampler.h',
- 'acm2/audio_coding_module.cc',
- 'acm2/audio_coding_module_impl.cc',
- 'acm2/audio_coding_module_impl.h',
- 'acm2/call_statistics.cc',
- 'acm2/call_statistics.h',
- 'acm2/codec_manager.cc',
- 'acm2/codec_manager.h',
- 'acm2/initial_delay_manager.cc',
- 'acm2/initial_delay_manager.h',
- 'include/audio_coding_module.h',
- 'include/audio_coding_module_typedefs.h',
- ],
- },
- ],
- 'conditions': [
- ['include_tests==1', {
- 'targets': [
- {
- 'target_name': 'acm_receive_test',
- 'type': 'static_library',
- 'defines': [
- '<@(audio_coding_defines)',
- ],
- 'dependencies': [
- '<@(audio_coding_dependencies)',
- 'audio_coding_module',
- 'neteq_unittest_tools',
- '<(DEPTH)/testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'acm2/acm_receive_test_oldapi.cc',
- 'acm2/acm_receive_test_oldapi.h',
- ],
- }, # acm_receive_test
- {
- 'target_name': 'acm_send_test',
- 'type': 'static_library',
- 'defines': [
- '<@(audio_coding_defines)',
- ],
- 'dependencies': [
- '<@(audio_coding_dependencies)',
- 'audio_coding_module',
- 'neteq_unittest_tools',
- '<(DEPTH)/testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'acm2/acm_send_test_oldapi.cc',
- 'acm2/acm_send_test_oldapi.h',
- ],
- }, # acm_send_test
- {
- 'target_name': 'delay_test',
- 'type': 'executable',
- 'dependencies': [
- 'audio_coding_module',
- '<(DEPTH)/testing/gtest.gyp:gtest',
- '<(webrtc_root)/common.gyp:webrtc_common',
- '<(webrtc_root)/test/test.gyp:test_support',
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
- '<(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)/common.gyp:webrtc_common',
- '<(webrtc_root)/test/test.gyp:test_support',
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
- '<(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/modules/audio_coding/main/include/audio_coding_module.h b/modules/audio_coding/main/include/audio_coding_module.h
index fc3ddd5..03f4087 100644
--- a/modules/audio_coding/main/include/audio_coding_module.h
+++ b/modules/audio_coding/main/include/audio_coding_module.h
@@ -8,14 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
+
+#pragma message("WARNING: audio_coding/main/include is DEPRECATED; use audio_coding/include")
#include <vector>
#include "webrtc/base/optional.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/include/module.h"
#include "webrtc/system_wrappers/include/clock.h"
@@ -738,4 +740,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
diff --git a/modules/audio_coding/main/include/audio_coding_module_typedefs.h b/modules/audio_coding/main/include/audio_coding_module_typedefs.h
index 1ca6f9d..e1ec30a 100644
--- a/modules/audio_coding/main/include/audio_coding_module_typedefs.h
+++ b/modules/audio_coding/main/include/audio_coding_module_typedefs.h
@@ -8,8 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+
+#pragma message("WARNING: audio_coding/main/include is DEPRECATED; use audio_coding/include")
#include <map>
@@ -48,4 +50,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
diff --git a/modules/audio_coding/neteq/audio_decoder_impl.h b/modules/audio_coding/neteq/audio_decoder_impl.h
index d1aae4a..bc8bdd9 100644
--- a/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/modules/audio_coding/neteq/audio_decoder_impl.h
@@ -20,7 +20,7 @@
#ifdef WEBRTC_CODEC_G722
#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
#endif
-#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/nack.h b/modules/audio_coding/neteq/nack.h
index 116b7e2..17fef46 100644
--- a/modules/audio_coding/neteq/nack.h
+++ b/modules/audio_coding/neteq/nack.h
@@ -15,7 +15,7 @@
#include <map>
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "webrtc/test/testsupport/gtest_prod_util.h"
//
diff --git a/modules/audio_coding/neteq/nack_unittest.cc b/modules/audio_coding/neteq/nack_unittest.cc
index 853af94..53b19dc 100644
--- a/modules/audio_coding/neteq/nack_unittest.cc
+++ b/modules/audio_coding/neteq/nack_unittest.cc
@@ -17,7 +17,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
namespace webrtc {
namespace {
diff --git a/modules/audio_coding/main/test/ACMTest.h b/modules/audio_coding/test/ACMTest.h
similarity index 74%
rename from modules/audio_coding/main/test/ACMTest.h
rename to modules/audio_coding/test/ACMTest.h
index f73961f..d7e87d3 100644
--- a/modules/audio_coding/main/test/ACMTest.h
+++ b/modules/audio_coding/test/ACMTest.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_
class ACMTest {
public:
@@ -18,4 +18,4 @@
virtual void Perform() = 0;
};
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_
diff --git a/modules/audio_coding/main/test/APITest.cc b/modules/audio_coding/test/APITest.cc
similarity index 99%
rename from modules/audio_coding/main/test/APITest.cc
rename to modules/audio_coding/test/APITest.cc
index 88ad7e2..59a5a3a 100644
--- a/modules/audio_coding/main/test/APITest.cc
+++ b/modules/audio_coding/test/APITest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/test/APITest.h"
+#include "webrtc/modules/audio_coding/test/APITest.h"
#include <ctype.h>
#include <stdio.h>
@@ -24,8 +24,8 @@
#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/system_wrappers/include/trace.h"
diff --git a/modules/audio_coding/main/test/APITest.h b/modules/audio_coding/test/APITest.h
similarity index 87%
rename from modules/audio_coding/main/test/APITest.h
rename to modules/audio_coding/test/APITest.h
index d4c5b1e..a1937c2 100644
--- a/modules/audio_coding/main/test/APITest.h
+++ b/modules/audio_coding/test/APITest.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
@@ -160,4 +160,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
diff --git a/modules/audio_coding/main/test/Channel.cc b/modules/audio_coding/test/Channel.cc
similarity index 99%
rename from modules/audio_coding/main/test/Channel.cc
rename to modules/audio_coding/test/Channel.cc
index 02bd783..31521fe 100644
--- a/modules/audio_coding/main/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
#include <assert.h>
#include <iostream>
diff --git a/modules/audio_coding/main/test/Channel.h b/modules/audio_coding/test/Channel.h
similarity index 92%
rename from modules/audio_coding/main/test/Channel.h
rename to modules/audio_coding/test/Channel.h
index ff6937e..b047aa9 100644
--- a/modules/audio_coding/main/test/Channel.h
+++ b/modules/audio_coding/test/Channel.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
#include <stdio.h>
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
@@ -127,4 +127,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
diff --git a/modules/audio_coding/main/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
similarity index 97%
rename from modules/audio_coding/main/test/EncodeDecodeTest.cc
rename to modules/audio_coding/test/EncodeDecodeTest.cc
index d68e575..ef45705 100644
--- a/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
#include <sstream>
#include <stdio.h>
@@ -17,9 +17,9 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
diff --git a/modules/audio_coding/main/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h
similarity index 86%
rename from modules/audio_coding/main/test/EncodeDecodeTest.h
rename to modules/audio_coding/test/EncodeDecodeTest.h
index 4ad92ce..3881062 100644
--- a/modules/audio_coding/main/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
#include <stdio.h>
#include <string.h>
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/RTPFile.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/RTPFile.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -120,4 +120,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
diff --git a/modules/audio_coding/main/test/PCMFile.cc b/modules/audio_coding/test/PCMFile.cc
similarity index 100%
rename from modules/audio_coding/main/test/PCMFile.cc
rename to modules/audio_coding/test/PCMFile.cc
diff --git a/modules/audio_coding/main/test/PCMFile.h b/modules/audio_coding/test/PCMFile.h
similarity index 90%
rename from modules/audio_coding/main/test/PCMFile.h
rename to modules/audio_coding/test/PCMFile.h
index 785ed66..9365180 100644
--- a/modules/audio_coding/main/test/PCMFile.h
+++ b/modules/audio_coding/test/PCMFile.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
#include <stdio.h>
#include <stdlib.h>
@@ -65,4 +65,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
diff --git a/modules/audio_coding/main/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
similarity index 98%
rename from modules/audio_coding/main/test/PacketLossTest.cc
rename to modules/audio_coding/test/PacketLossTest.cc
index f7c96fa..ad3e834 100644
--- a/modules/audio_coding/main/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/test/PacketLossTest.h"
+#include "webrtc/modules/audio_coding/test/PacketLossTest.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common.h"
diff --git a/modules/audio_coding/main/test/PacketLossTest.h b/modules/audio_coding/test/PacketLossTest.h
similarity index 86%
rename from modules/audio_coding/main/test/PacketLossTest.h
rename to modules/audio_coding/test/PacketLossTest.h
index d25dea2..f3570ae 100644
--- a/modules/audio_coding/main/test/PacketLossTest.h
+++ b/modules/audio_coding/test/PacketLossTest.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
#include <string>
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
namespace webrtc {
@@ -64,4 +64,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
diff --git a/modules/audio_coding/main/test/RTPFile.cc b/modules/audio_coding/test/RTPFile.cc
similarity index 100%
rename from modules/audio_coding/main/test/RTPFile.cc
rename to modules/audio_coding/test/RTPFile.cc
diff --git a/modules/audio_coding/main/test/RTPFile.h b/modules/audio_coding/test/RTPFile.h
similarity index 92%
rename from modules/audio_coding/main/test/RTPFile.h
rename to modules/audio_coding/test/RTPFile.h
index 6bad755..696d41e 100644
--- a/modules/audio_coding/main/test/RTPFile.h
+++ b/modules/audio_coding/test/RTPFile.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
#include <stdio.h>
#include <queue>
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
#include "webrtc/typedefs.h"
@@ -123,4 +123,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
diff --git a/modules/audio_coding/main/test/SpatialAudio.cc b/modules/audio_coding/test/SpatialAudio.cc
similarity index 98%
rename from modules/audio_coding/main/test/SpatialAudio.cc
rename to modules/audio_coding/test/SpatialAudio.cc
index 17d4fc8..c9f8080 100644
--- a/modules/audio_coding/main/test/SpatialAudio.cc
+++ b/modules/audio_coding/test/SpatialAudio.cc
@@ -14,7 +14,7 @@
#include <math.h>
#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/test/SpatialAudio.h"
+#include "webrtc/modules/audio_coding/test/SpatialAudio.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
diff --git a/modules/audio_coding/main/test/SpatialAudio.h b/modules/audio_coding/test/SpatialAudio.h
similarity index 66%
rename from modules/audio_coding/main/test/SpatialAudio.h
rename to modules/audio_coding/test/SpatialAudio.h
index fc25897..3548cc9 100644
--- a/modules/audio_coding/main/test/SpatialAudio.h
+++ b/modules/audio_coding/test/SpatialAudio.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
#define MAX_FILE_NAME_LENGTH_BYTE 500
@@ -44,4 +44,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
diff --git a/modules/audio_coding/main/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
similarity index 97%
rename from modules/audio_coding/main/test/TestAllCodecs.cc
rename to modules/audio_coding/test/TestAllCodecs.cc
index e9e4f2b..21ce7c1 100644
--- a/modules/audio_coding/main/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/test/TestAllCodecs.h"
+#include "webrtc/modules/audio_coding/test/TestAllCodecs.h"
#include <cstdio>
#include <limits>
@@ -18,9 +18,9 @@
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"
diff --git a/modules/audio_coding/main/test/TestAllCodecs.h b/modules/audio_coding/test/TestAllCodecs.h
similarity index 85%
rename from modules/audio_coding/main/test/TestAllCodecs.h
rename to modules/audio_coding/test/TestAllCodecs.h
index 1cdc0cb..e79bd69 100644
--- a/modules/audio_coding/main/test/TestAllCodecs.h
+++ b/modules/audio_coding/test/TestAllCodecs.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -81,4 +81,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
diff --git a/modules/audio_coding/main/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
similarity index 98%
rename from modules/audio_coding/main/test/TestRedFec.cc
rename to modules/audio_coding/test/TestRedFec.cc
index 0627ae2..d544026 100644
--- a/modules/audio_coding/main/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/test/TestRedFec.h"
+#include "webrtc/modules/audio_coding/test/TestRedFec.h"
#include <assert.h>
#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
diff --git a/modules/audio_coding/main/test/TestRedFec.h b/modules/audio_coding/test/TestRedFec.h
similarity index 78%
rename from modules/audio_coding/main/test/TestRedFec.h
rename to modules/audio_coding/test/TestRedFec.h
index ac0b6cd..6343d8e 100644
--- a/modules/audio_coding/main/test/TestRedFec.h
+++ b/modules/audio_coding/test/TestRedFec.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TESTREDFEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TESTREDFEC_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
#include <string>
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
namespace webrtc {
@@ -48,4 +48,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TESTREDFEC_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
diff --git a/modules/audio_coding/main/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
similarity index 98%
rename from modules/audio_coding/main/test/TestStereo.cc
rename to modules/audio_coding/test/TestStereo.cc
index bb38fac..19f027b 100644
--- a/modules/audio_coding/main/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
#include <assert.h>
@@ -17,8 +17,8 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
diff --git a/modules/audio_coding/main/test/TestStereo.h b/modules/audio_coding/test/TestStereo.h
similarity index 88%
rename from modules/audio_coding/main/test/TestStereo.h
rename to modules/audio_coding/test/TestStereo.h
index b56e995..4526be6 100644
--- a/modules/audio_coding/main/test/TestStereo.h
+++ b/modules/audio_coding/test/TestStereo.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
#include <math.h>
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
#define PCMA_AND_PCMU
@@ -114,4 +114,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
diff --git a/modules/audio_coding/main/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc
similarity index 97%
rename from modules/audio_coding/main/test/TestVADDTX.cc
rename to modules/audio_coding/test/TestVADDTX.cc
index bba7b91..98b1224 100644
--- a/modules/audio_coding/main/test/TestVADDTX.cc
+++ b/modules/audio_coding/test/TestVADDTX.cc
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/test/TestVADDTX.h"
+#include "webrtc/modules/audio_coding/test/TestVADDTX.h"
#include <string>
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/test/testsupport/fileutils.h"
namespace webrtc {
diff --git a/modules/audio_coding/main/test/TestVADDTX.h b/modules/audio_coding/test/TestVADDTX.h
similarity index 85%
rename from modules/audio_coding/main/test/TestVADDTX.h
rename to modules/audio_coding/test/TestVADDTX.h
index 07596e2..1e7f0ef 100644
--- a/modules/audio_coding/main/test/TestVADDTX.h
+++ b/modules/audio_coding/test/TestVADDTX.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
namespace webrtc {
@@ -99,4 +99,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
diff --git a/modules/audio_coding/main/test/Tester.cc b/modules/audio_coding/test/Tester.cc
similarity index 87%
rename from modules/audio_coding/main/test/Tester.cc
rename to modules/audio_coding/test/Tester.cc
index 7302e5d..3ff3dd8 100644
--- a/modules/audio_coding/main/test/Tester.cc
+++ b/modules/audio_coding/test/Tester.cc
@@ -13,17 +13,17 @@
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/APITest.h"
-#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
-#include "webrtc/modules/audio_coding/main/test/iSACTest.h"
-#include "webrtc/modules/audio_coding/main/test/opus_test.h"
-#include "webrtc/modules/audio_coding/main/test/PacketLossTest.h"
-#include "webrtc/modules/audio_coding/main/test/TestAllCodecs.h"
-#include "webrtc/modules/audio_coding/main/test/TestRedFec.h"
-#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
-#include "webrtc/modules/audio_coding/main/test/TestVADDTX.h"
-#include "webrtc/modules/audio_coding/main/test/TwoWayCommunication.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/APITest.h"
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/test/iSACTest.h"
+#include "webrtc/modules/audio_coding/test/opus_test.h"
+#include "webrtc/modules/audio_coding/test/PacketLossTest.h"
+#include "webrtc/modules/audio_coding/test/TestAllCodecs.h"
+#include "webrtc/modules/audio_coding/test/TestRedFec.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/test/TestVADDTX.h"
+#include "webrtc/modules/audio_coding/test/TwoWayCommunication.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
diff --git a/modules/audio_coding/main/test/TimedTrace.cc b/modules/audio_coding/test/TimedTrace.cc
similarity index 100%
rename from modules/audio_coding/main/test/TimedTrace.cc
rename to modules/audio_coding/test/TimedTrace.cc
diff --git a/modules/audio_coding/main/test/TimedTrace.h b/modules/audio_coding/test/TimedTrace.h
similarity index 82%
rename from modules/audio_coding/main/test/TimedTrace.h
rename to modules/audio_coding/test/TimedTrace.h
index ef9609a..0793eb0 100644
--- a/modules/audio_coding/main/test/TimedTrace.h
+++ b/modules/audio_coding/test/TimedTrace.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef TIMED_TRACE_H
-#define TIMED_TRACE_H
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_
#include "webrtc/typedefs.h"
@@ -33,4 +33,4 @@
};
-#endif
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_
diff --git a/modules/audio_coding/main/test/TwoWayCommunication.cc b/modules/audio_coding/test/TwoWayCommunication.cc
similarity index 98%
rename from modules/audio_coding/main/test/TwoWayCommunication.cc
rename to modules/audio_coding/test/TwoWayCommunication.cc
index 725cbf7..56e136b 100644
--- a/modules/audio_coding/main/test/TwoWayCommunication.cc
+++ b/modules/audio_coding/test/TwoWayCommunication.cc
@@ -21,8 +21,8 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
diff --git a/modules/audio_coding/main/test/TwoWayCommunication.h b/modules/audio_coding/test/TwoWayCommunication.h
similarity index 69%
rename from modules/audio_coding/main/test/TwoWayCommunication.h
rename to modules/audio_coding/test/TwoWayCommunication.h
index bf969fe..7763993 100644
--- a/modules/audio_coding/main/test/TwoWayCommunication.h
+++ b/modules/audio_coding/test/TwoWayCommunication.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
namespace webrtc {
@@ -57,4 +57,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
diff --git a/modules/audio_coding/main/test/delay_test.cc b/modules/audio_coding/test/delay_test.cc
similarity index 95%
rename from modules/audio_coding/main/test/delay_test.cc
rename to modules/audio_coding/test/delay_test.cc
index ce08c0f..a8c137f 100644
--- a/modules/audio_coding/main/test/delay_test.cc
+++ b/modules/audio_coding/test/delay_test.cc
@@ -19,12 +19,12 @@
#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/test/testsupport/fileutils.h"
diff --git a/modules/audio_coding/main/test/iSACTest.cc b/modules/audio_coding/test/iSACTest.cc
similarity index 98%
rename from modules/audio_coding/main/test/iSACTest.cc
rename to modules/audio_coding/test/iSACTest.cc
index 203e12b..09744b1 100644
--- a/modules/audio_coding/main/test/iSACTest.cc
+++ b/modules/audio_coding/test/iSACTest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/test/iSACTest.h"
+#include "webrtc/modules/audio_coding/test/iSACTest.h"
#include <ctype.h>
#include <stdio.h>
@@ -23,8 +23,8 @@
#include <time.h>
#endif
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/system_wrappers/include/trace.h"
diff --git a/modules/audio_coding/main/test/iSACTest.h b/modules/audio_coding/test/iSACTest.h
similarity index 76%
rename from modules/audio_coding/main/test/iSACTest.h
rename to modules/audio_coding/test/iSACTest.h
index 0693d93..c5bb515 100644
--- a/modules/audio_coding/main/test/iSACTest.h
+++ b/modules/audio_coding/test/iSACTest.h
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_
#include <string.h>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
#define MAX_FILE_NAME_LENGTH_BYTE 500
#define NO_OF_CLIENTS 15
@@ -76,4 +76,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_
diff --git a/modules/audio_coding/main/test/insert_packet_with_timing.cc b/modules/audio_coding/test/insert_packet_with_timing.cc
similarity index 97%
rename from modules/audio_coding/main/test/insert_packet_with_timing.cc
rename to modules/audio_coding/test/insert_packet_with_timing.cc
index 857381d..481df55 100644
--- a/modules/audio_coding/main/test/insert_packet_with_timing.cc
+++ b/modules/audio_coding/test/insert_packet_with_timing.cc
@@ -14,9 +14,9 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/testsupport/fileutils.h"
diff --git a/modules/audio_coding/main/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
similarity index 97%
rename from modules/audio_coding/main/test/opus_test.cc
rename to modules/audio_coding/test/opus_test.cc
index 27cc40a..3372a2a 100644
--- a/modules/audio_coding/main/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/main/test/opus_test.h"
+#include "webrtc/modules/audio_coding/test/opus_test.h"
#include <assert.h>
@@ -18,9 +18,9 @@
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
diff --git a/modules/audio_coding/main/test/opus_test.h b/modules/audio_coding/test/opus_test.h
similarity index 72%
rename from modules/audio_coding/main/test/opus_test.h
rename to modules/audio_coding/test/opus_test.h
index 0b96009..090c8fa 100644
--- a/modules/audio_coding/main/test/opus_test.h
+++ b/modules/audio_coding/test/opus_test.h
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
#include <math.h>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/main/test/Channel.h"
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/main/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
namespace webrtc {
@@ -54,4 +54,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
diff --git a/modules/audio_coding/main/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
similarity index 97%
rename from modules/audio_coding/main/test/target_delay_unittest.cc
rename to modules/audio_coding/test/target_delay_unittest.cc
index afc0e10..d7c0411 100644
--- a/modules/audio_coding/main/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -12,8 +12,8 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/test/utility.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/testsupport/fileutils.h"
diff --git a/modules/audio_coding/main/test/utility.cc b/modules/audio_coding/test/utility.cc
similarity index 97%
rename from modules/audio_coding/main/test/utility.cc
rename to modules/audio_coding/test/utility.cc
index 34af5e7..89368bc 100644
--- a/modules/audio_coding/main/test/utility.cc
+++ b/modules/audio_coding/test/utility.cc
@@ -18,8 +18,8 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
#define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13
diff --git a/modules/audio_coding/main/test/utility.h b/modules/audio_coding/test/utility.h
similarity index 94%
rename from modules/audio_coding/main/test/utility.h
rename to modules/audio_coding/test/utility.h
index e936ec1..23869be 100644
--- a/modules/audio_coding/main/test/utility.h
+++ b/modules/audio_coding/test/utility.h
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
namespace webrtc {
@@ -136,4 +136,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
+#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
diff --git a/modules/modules.gyp b/modules/modules.gyp
index 0c981e6..599a931 100644
--- a/modules/modules.gyp
+++ b/modules/modules.gyp
@@ -71,24 +71,24 @@
'<@(audio_coding_defines)',
],
'sources': [
- 'audio_coding/main/test/APITest.cc',
- 'audio_coding/main/test/Channel.cc',
- 'audio_coding/main/test/EncodeDecodeTest.cc',
- 'audio_coding/main/test/PCMFile.cc',
- 'audio_coding/main/test/PacketLossTest.cc',
- 'audio_coding/main/test/RTPFile.cc',
- 'audio_coding/main/test/SpatialAudio.cc',
- 'audio_coding/main/test/TestAllCodecs.cc',
- 'audio_coding/main/test/TestRedFec.cc',
- 'audio_coding/main/test/TestStereo.cc',
- 'audio_coding/main/test/TestVADDTX.cc',
- 'audio_coding/main/test/Tester.cc',
- 'audio_coding/main/test/TimedTrace.cc',
- 'audio_coding/main/test/TwoWayCommunication.cc',
- 'audio_coding/main/test/iSACTest.cc',
- 'audio_coding/main/test/opus_test.cc',
- 'audio_coding/main/test/target_delay_unittest.cc',
- 'audio_coding/main/test/utility.cc',
+ 'audio_coding/test/APITest.cc',
+ 'audio_coding/test/Channel.cc',
+ 'audio_coding/test/EncodeDecodeTest.cc',
+ 'audio_coding/test/PCMFile.cc',
+ 'audio_coding/test/PacketLossTest.cc',
+ 'audio_coding/test/RTPFile.cc',
+ 'audio_coding/test/SpatialAudio.cc',
+ 'audio_coding/test/TestAllCodecs.cc',
+ 'audio_coding/test/TestRedFec.cc',
+ 'audio_coding/test/TestStereo.cc',
+ 'audio_coding/test/TestVADDTX.cc',
+ 'audio_coding/test/Tester.cc',
+ 'audio_coding/test/TimedTrace.cc',
+ 'audio_coding/test/TwoWayCommunication.cc',
+ 'audio_coding/test/iSACTest.cc',
+ 'audio_coding/test/opus_test.cc',
+ 'audio_coding/test/target_delay_unittest.cc',
+ 'audio_coding/test/utility.cc',
'rtp_rtcp/test/testFec/test_fec.cc',
'video_coding/codecs/test/videoprocessor_integrationtest.cc',
'video_coding/codecs/vp8/test/vp8_impl_unittest.cc',
@@ -156,12 +156,12 @@
],
'sources': [
'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc',
- 'audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc',
- 'audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc',
- 'audio_coding/main/acm2/call_statistics_unittest.cc',
- 'audio_coding/main/acm2/codec_manager_unittest.cc',
- 'audio_coding/main/acm2/initial_delay_manager_unittest.cc',
- 'audio_coding/main/acm2/rent_a_codec_unittest.cc',
+ 'audio_coding/acm2/acm_receiver_unittest_oldapi.cc',
+ 'audio_coding/acm2/audio_coding_module_unittest_oldapi.cc',
+ 'audio_coding/acm2/call_statistics_unittest.cc',
+ 'audio_coding/acm2/codec_manager_unittest.cc',
+ 'audio_coding/acm2/initial_delay_manager_unittest.cc',
+ 'audio_coding/acm2/rent_a_codec_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',
diff --git a/modules/utility/source/coder.h b/modules/utility/source/coder.h
index 4270e9b..abfa87e 100644
--- a/modules/utility/source/coder.h
+++ b/modules/utility/source/coder.h
@@ -13,7 +13,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index ba18aaa..0e509d2 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -14,7 +14,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "webrtc/modules/audio_processing/rms_level.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
diff --git a/voice_engine/voe_base_impl.cc b/voice_engine/voe_base_impl.cc
index 677e9b1..2b5587d 100644
--- a/voice_engine/voe_base_impl.cc
+++ b/voice_engine/voe_base_impl.cc
@@ -13,7 +13,7 @@
#include "webrtc/base/format_macros.h"
#include "webrtc/common.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_device/audio_device_impl.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
diff --git a/voice_engine/voe_codec_impl.cc b/voice_engine/voe_codec_impl.cc
index 3ab02a6..162f1c2 100644
--- a/voice_engine/voe_codec_impl.cc
+++ b/voice_engine/voe_codec_impl.cc
@@ -10,7 +10,7 @@
#include "webrtc/voice_engine/voe_codec_impl.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/voice_engine/channel.h"
diff --git a/voice_engine/voe_neteq_stats_impl.cc b/voice_engine/voe_neteq_stats_impl.cc
index 00e04d8..807325b 100644
--- a/voice_engine/voe_neteq_stats_impl.cc
+++ b/voice_engine/voe_neteq_stats_impl.cc
@@ -10,7 +10,7 @@
#include "webrtc/voice_engine/voe_neteq_stats_impl.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/voice_engine/channel.h"
diff --git a/voice_engine/voice_engine_impl.cc b/voice_engine/voice_engine_impl.cc
index d9c5744..8df05cc 100644
--- a/voice_engine/voice_engine_impl.cc
+++ b/voice_engine/voice_engine_impl.cc
@@ -16,7 +16,7 @@
#endif
#include "webrtc/base/checks.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/voice_engine/channel_proxy.h"
