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audio_coding: remove "main" directory

Browse Source 
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-Commit-Position: refs/heads/master@{#10803}
This commit is contained in:
kjellander
2015-11-26 04:44:54 -08:00
committed by Commit bot
parent a3c20bb9a0
commit 3e6db2321c
85 changed files with 1276 additions and 498 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/call/call_perf_tests.cc
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@ -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"

42
webrtc/modules/audio_coding/BUILD.gn
View File

@ -11,10 +11,10 @@ import("../../build/webrtc.gni")
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 @@ source_set("rent_a_codec") {
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 = []

4
webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc → webrtc/modules/audio_coding/acm2/acm_codec_database.cc
View File

@ -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 {

8
webrtc/modules/audio_coding/main/acm2/acm_codec_database.h → webrtc/modules/audio_coding/acm2/acm_codec_database.h
View File

@ -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 @@ class ACMCodecDB {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_

6
webrtc/modules/audio_coding/main/acm2/acm_common_defs.h → webrtc/modules/audio_coding/acm2/acm_common_defs.h
View File

@ -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 @@ const int kIsacPacSize960 = 960;
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_

0
webrtc/modules/audio_coding/main/acm2/acm_neteq_unittest.cc → webrtc/modules/audio_coding/acm2/acm_neteq_unittest.cc
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4
webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc → webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
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@ -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"

6
webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h → webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h
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@ -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 @@ class AcmReceiveTestToggleOutputFreqOldApi : public AcmReceiveTestOldApi {
} // 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_

6
webrtc/modules/audio_coding/main/acm2/acm_receiver.cc → webrtc/modules/audio_coding/acm2/acm_receiver.cc
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@ -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"

14
webrtc/modules/audio_coding/main/acm2/acm_receiver.h → webrtc/modules/audio_coding/acm2/acm_receiver.h
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@ -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 @@ class AcmReceiver {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_

6
webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc → webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
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@ -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"

2
webrtc/modules/audio_coding/main/acm2/acm_resampler.cc → webrtc/modules/audio_coding/acm2/acm_resampler.cc
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@ -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>

6
webrtc/modules/audio_coding/main/acm2/acm_resampler.h → webrtc/modules/audio_coding/acm2/acm_resampler.h
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@ -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 @@ class ACMResampler {
} // namespace acm2
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_

4
webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc → webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc
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@ -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"

8
webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h → webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
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@ -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 @@ class AcmSendTestOldApi : public AudioPacketizationCallback,
} // 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_

6
webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc → webrtc/modules/audio_coding/acm2/audio_coding_module.cc
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@ -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"

10
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc → webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
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@ -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"

12
webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h → webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
View File

@ -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 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
} // 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_

8
webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc → webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
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@ -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"

2
webrtc/modules/audio_coding/main/acm2/call_statistics.cc → webrtc/modules/audio_coding/acm2/call_statistics.cc
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@ -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>

6
webrtc/modules/audio_coding/main/acm2/call_statistics.h → webrtc/modules/audio_coding/acm2/call_statistics.h
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@ -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 @@ class CallStatistics {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_

2
webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc → webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc
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@ -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 {

4
webrtc/modules/audio_coding/main/acm2/codec_manager.cc → webrtc/modules/audio_coding/acm2/codec_manager.cc
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@ -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 {

10
webrtc/modules/audio_coding/main/acm2/codec_manager.h → webrtc/modules/audio_coding/acm2/codec_manager.h
View File

@ -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 @@ class CodecManager final {
} // namespace acm2
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_MANAGER_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_

2
webrtc/modules/audio_coding/main/acm2/codec_manager_unittest.cc → webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc
View File

@ -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 {

2
webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc → webrtc/modules/audio_coding/acm2/initial_delay_manager.cc
View File

@ -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 {

6
webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h → webrtc/modules/audio_coding/acm2/initial_delay_manager.h
View File

@ -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 @@ class InitialDelayManager {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_

2
webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc → webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
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@ -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 {

6
webrtc/modules/audio_coding/main/acm2/rent_a_codec.cc → webrtc/modules/audio_coding/acm2/rent_a_codec.cc
View File

@ -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 {

8
webrtc/modules/audio_coding/main/acm2/rent_a_codec.h → webrtc/modules/audio_coding/acm2/rent_a_codec.h
View File

@ -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 @@ class RentACodec {
} // 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_

2
webrtc/modules/audio_coding/main/acm2/rent_a_codec_unittest.cc → webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc
View File

@ -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 {

185
webrtc/modules/audio_coding/audio_coding.gypi
View File

@ -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
],
}],
],
}

741
webrtc/modules/audio_coding/include/audio_coding_module.h Normal file
View File

@ -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_

51
webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h Normal file
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@ -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_

5
webrtc/modules/audio_coding/main/acm2/OWNERS
View File

@ -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=*

196
webrtc/modules/audio_coding/main/audio_coding_module.gypi
View File

@ -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
],
}],
],
}

10
webrtc/modules/audio_coding/main/include/audio_coding_module.h
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@ -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 @@ class AudioCodingModule {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_

8
webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h
View File

@ -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 @@ enum OpusApplicationMode {
} // 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_

2
webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
View File

@ -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 {

2
webrtc/modules/audio_coding/neteq/nack.h
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@ -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"
//

2
webrtc/modules/audio_coding/neteq/nack_unittest.cc
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@ -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 {

6
webrtc/modules/audio_coding/main/test/ACMTest.h → webrtc/modules/audio_coding/test/ACMTest.h
View File

@ -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 @@ class ACMTest {
virtual void Perform() = 0;
};
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_

6
webrtc/modules/audio_coding/main/test/APITest.cc → webrtc/modules/audio_coding/test/APITest.cc
View File

@ -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"

16
webrtc/modules/audio_coding/main/test/APITest.h → webrtc/modules/audio_coding/test/APITest.h
View File

@ -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 @@ class APITest : public ACMTest {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_

2
webrtc/modules/audio_coding/main/test/Channel.cc → webrtc/modules/audio_coding/test/Channel.cc
View File

@ -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>

8
webrtc/modules/audio_coding/main/test/Channel.h → webrtc/modules/audio_coding/test/Channel.h
View File

@ -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 @@ class Channel : public AudioPacketizationCallback {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_

8
webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc → webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
View File

@ -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"

14
webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h → webrtc/modules/audio_coding/test/EncodeDecodeTest.h
View File

@ -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 @@ class EncodeDecodeTest : public ACMTest {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_

0
webrtc/modules/audio_coding/main/test/PCMFile.cc → webrtc/modules/audio_coding/test/PCMFile.cc
View File

6
webrtc/modules/audio_coding/main/test/PCMFile.h → webrtc/modules/audio_coding/test/PCMFile.h
View File

@ -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 @@ class PCMFile {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_

2
webrtc/modules/audio_coding/main/test/PacketLossTest.cc → webrtc/modules/audio_coding/test/PacketLossTest.cc
View File

@ -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"

8
webrtc/modules/audio_coding/main/test/PacketLossTest.h → webrtc/modules/audio_coding/test/PacketLossTest.h
View File

@ -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 @@ class PacketLossTest : public ACMTest {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_

0
webrtc/modules/audio_coding/main/test/RTPFile.cc → webrtc/modules/audio_coding/test/RTPFile.cc
View File

8
webrtc/modules/audio_coding/main/test/RTPFile.h → webrtc/modules/audio_coding/test/RTPFile.h
View File

@ -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 @@ class RTPFile : public RTPStream {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_

2
webrtc/modules/audio_coding/main/test/SpatialAudio.cc → webrtc/modules/audio_coding/test/SpatialAudio.cc
View File

@ -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"

16
webrtc/modules/audio_coding/main/test/SpatialAudio.h → webrtc/modules/audio_coding/test/SpatialAudio.h
View File

@ -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 @@ class SpatialAudio : public ACMTest {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_

8
webrtc/modules/audio_coding/main/test/TestAllCodecs.cc → webrtc/modules/audio_coding/test/TestAllCodecs.cc
View File

@ -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"

12
webrtc/modules/audio_coding/main/test/TestAllCodecs.h → webrtc/modules/audio_coding/test/TestAllCodecs.h
View File

@ -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 @@ class TestAllCodecs : public ACMTest {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_

6
webrtc/modules/audio_coding/main/test/TestRedFec.cc → webrtc/modules/audio_coding/test/TestRedFec.cc
View File

@ -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"

12
webrtc/modules/audio_coding/main/test/TestRedFec.h → webrtc/modules/audio_coding/test/TestRedFec.h
View File

@ -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 @@ class TestRedFec : public ACMTest {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TESTREDFEC_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_

6
webrtc/modules/audio_coding/main/test/TestStereo.cc → webrtc/modules/audio_coding/test/TestStereo.cc
View File

@ -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"

12
webrtc/modules/audio_coding/main/test/TestStereo.h → webrtc/modules/audio_coding/test/TestStereo.h
View File

@ -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 @@ class TestStereo : public ACMTest {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_

6
webrtc/modules/audio_coding/main/test/TestVADDTX.cc → webrtc/modules/audio_coding/test/TestVADDTX.cc
View File

@ -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 {

14
webrtc/modules/audio_coding/main/test/TestVADDTX.h → webrtc/modules/audio_coding/test/TestVADDTX.h
View File

@ -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 @@ class TestOpusDtx final : public TestVadDtx {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_

22
webrtc/modules/audio_coding/main/test/Tester.cc → webrtc/modules/audio_coding/test/Tester.cc
View File

@ -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"

0
webrtc/modules/audio_coding/main/test/TimedTrace.cc → webrtc/modules/audio_coding/test/TimedTrace.cc
View File

6
webrtc/modules/audio_coding/main/test/TimedTrace.h → webrtc/modules/audio_coding/test/TimedTrace.h
View File

@ -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 @@ class TimedTrace {
};
#endif
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_

4
webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc → webrtc/modules/audio_coding/test/TwoWayCommunication.cc
View File

@ -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"

16
webrtc/modules/audio_coding/main/test/TwoWayCommunication.h → webrtc/modules/audio_coding/test/TwoWayCommunication.h
View File

@ -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 @@ class TwoWayCommunication : public ACMTest {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_

12
webrtc/modules/audio_coding/main/test/delay_test.cc → webrtc/modules/audio_coding/test/delay_test.cc
View File

@ -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"

6
webrtc/modules/audio_coding/main/test/iSACTest.cc → webrtc/modules/audio_coding/test/iSACTest.cc
View File

@ -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"

16
webrtc/modules/audio_coding/main/test/iSACTest.h → webrtc/modules/audio_coding/test/iSACTest.h
View File

@ -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 @@ class ISACTest : public ACMTest {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_

6
webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc → webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
View File

@ -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"

8
webrtc/modules/audio_coding/main/test/opus_test.cc → webrtc/modules/audio_coding/test/opus_test.cc
View File

@ -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"

16
webrtc/modules/audio_coding/main/test/opus_test.h → webrtc/modules/audio_coding/test/opus_test.h
View File

@ -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 @@ class OpusTest : public ACMTest {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_

4
webrtc/modules/audio_coding/main/test/target_delay_unittest.cc → webrtc/modules/audio_coding/test/target_delay_unittest.cc
View File

@ -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"

4
webrtc/modules/audio_coding/main/test/utility.cc → webrtc/modules/audio_coding/test/utility.cc
View File

@ -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

8
webrtc/modules/audio_coding/main/test/utility.h → webrtc/modules/audio_coding/test/utility.h
View File

@ -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 @@ void UseNewAcm(webrtc::Config* config);
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_
#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_

48
webrtc/modules/modules.gyp
View File

@ -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',

2
webrtc/modules/utility/source/coder.h
View File

@ -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 {

2
webrtc/voice_engine/channel.h
View File

@ -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"

2
webrtc/voice_engine/voe_base_impl.cc
View File

@ -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"

2
webrtc/voice_engine/voe_codec_impl.cc
View File

@ -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"

2
webrtc/voice_engine/voe_neteq_stats_impl.cc
View File

@ -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"

2
webrtc/voice_engine/voice_engine_impl.cc
View File

@ -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"