Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #5 id:80001 of https://codereview.chromium.org/2037623002/ )
Reason for revert:
voice_engine_unittests: FilePlayerTest.PlayWavPcm16File and FilePlayerTest.PlayWavPcmuFile fail on 32-bit android (android_rel and android-dbg try bots, Android32 Tests (L Nexus5) and Android32 Tests (L Nexus7.2) build bots).
Not sure why this would happen, since I just moved the test without modifying it. Some test filtering that no longer manages to disable them? Anyway, reverting until I know how to fix.
This was actually caught by the try bots, but I missed it because I was manually ignoring them because of an error with the bots. :-(
Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> R=perkj@webrtc.org, solenberg@webrtc.org
>
> Committed: https://chromium.googlesource.com/external/webrtc/+/65874b163ee5d6eb460c53590702ed98ca399605
TBR=perkj@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2092633002
Cr-Original-Commit-Position: refs/heads/master@{#13267}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: e7edea97592733649ac067355da122b047b128cd
diff --git a/modules/BUILD.gn b/modules/BUILD.gn
index c86bae1..4eeaa59 100644
--- a/modules/BUILD.gn
+++ b/modules/BUILD.gn
@@ -223,6 +223,7 @@
"rtp_rtcp/test/testAPI/test_api_rtcp.cc",
"rtp_rtcp/test/testAPI/test_api_video.cc",
"utility/source/audio_frame_operations_unittest.cc",
+ "utility/source/file_player_unittests.cc",
"utility/source/process_thread_impl_unittest.cc",
"video_coding/codecs/test/packet_manipulator_unittest.cc",
"video_coding/codecs/test/stats_unittest.cc",
diff --git a/modules/modules.gyp b/modules/modules.gyp
index f0fc128..e9a250c 100644
--- a/modules/modules.gyp
+++ b/modules/modules.gyp
@@ -351,6 +351,7 @@
'rtp_rtcp/test/testAPI/test_api_rtcp.cc',
'rtp_rtcp/test/testAPI/test_api_video.cc',
'utility/source/audio_frame_operations_unittest.cc',
+ 'utility/source/file_player_unittests.cc',
'utility/source/process_thread_impl_unittest.cc',
'video_coding/codecs/test/packet_manipulator_unittest.cc',
'video_coding/codecs/test/stats_unittest.cc',
diff --git a/modules/utility/BUILD.gn b/modules/utility/BUILD.gn
index c3c9f0a..5437e4f 100644
--- a/modules/utility/BUILD.gn
+++ b/modules/utility/BUILD.gn
@@ -11,10 +11,18 @@
source_set("utility") {
sources = [
"include/audio_frame_operations.h",
+ "include/file_player.h",
+ "include/file_recorder.h",
"include/helpers_android.h",
"include/jvm_android.h",
"include/process_thread.h",
"source/audio_frame_operations.cc",
+ "source/coder.cc",
+ "source/coder.h",
+ "source/file_player_impl.cc",
+ "source/file_player_impl.h",
+ "source/file_recorder_impl.cc",
+ "source/file_recorder_impl.h",
"source/helpers_android.cc",
"source/helpers_ios.mm",
"source/jvm_android.cc",
diff --git a/voice_engine/file_player.h b/modules/utility/include/file_player.h
similarity index 93%
rename from voice_engine/file_player.h
rename to modules/utility/include/file_player.h
index 898d66c..b064e30 100644
--- a/voice_engine/file_player.h
+++ b/modules/utility/include/file_player.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
-#define WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
@@ -83,5 +83,4 @@
};
} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
diff --git a/voice_engine/file_recorder.h b/modules/utility/include/file_recorder.h
similarity index 91%
rename from voice_engine/file_recorder.h
rename to modules/utility/include/file_recorder.h
index 001a449..92c91bd 100644
--- a/voice_engine/file_recorder.h
+++ b/modules/utility/include/file_recorder.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_
-#define WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
@@ -61,5 +61,4 @@
};
} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
diff --git a/voice_engine/coder.cc b/modules/utility/source/coder.cc
similarity index 98%
rename from voice_engine/coder.cc
rename to modules/utility/source/coder.cc
index ab724e5..f2ae43e 100644
--- a/voice_engine/coder.cc
+++ b/modules/utility/source/coder.cc
@@ -8,11 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/voice_engine/coder.h"
-
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/source/coder.h"
namespace webrtc {
namespace {
diff --git a/voice_engine/coder.h b/modules/utility/source/coder.h
similarity index 92%
rename from voice_engine/coder.h
rename to modules/utility/source/coder.h
index 41a7c59..5f44190 100644
--- a/voice_engine/coder.h
+++ b/modules/utility/source/coder.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_VOICE_ENGINE_CODER_H_
-#define WEBRTC_VOICE_ENGINE_CODER_H_
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
+#define WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
#include <memory>
@@ -65,4 +65,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_VOICE_ENGINE_CODER_H_
+#endif // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
diff --git a/voice_engine/file_player_impl.cc b/modules/utility/source/file_player_impl.cc
similarity index 99%
rename from voice_engine/file_player_impl.cc
rename to modules/utility/source/file_player_impl.cc
index c1239d3..e783a7e 100644
--- a/voice_engine/file_player_impl.cc
+++ b/modules/utility/source/file_player_impl.cc
@@ -8,8 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/voice_engine/file_player_impl.h"
-
+#include "webrtc/modules/utility/source/file_player_impl.h"
#include "webrtc/system_wrappers/include/logging.h"
namespace webrtc {
diff --git a/voice_engine/file_player_impl.h b/modules/utility/source/file_player_impl.h
similarity index 88%
rename from voice_engine/file_player_impl.h
rename to modules/utility/source/file_player_impl.h
index 82d7daf..62887da 100644
--- a/voice_engine/file_player_impl.h
+++ b/modules/utility/source/file_player_impl.h
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_VOICE_ENGINE_FILE_PLAYER_IMPL_H_
-#define WEBRTC_VOICE_ENGINE_FILE_PLAYER_IMPL_H_
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
+#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/media_file/media_file.h"
#include "webrtc/modules/media_file/media_file_defines.h"
+#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/modules/utility/source/coder.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/typedefs.h"
-#include "webrtc/voice_engine/coder.h"
-#include "webrtc/voice_engine/file_player.h"
namespace webrtc {
class FilePlayerImpl : public FilePlayer
@@ -75,5 +75,4 @@
float _scaling;
};
} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_IMPL_H_
+#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
diff --git a/voice_engine/file_player_unittests.cc b/modules/utility/source/file_player_unittests.cc
similarity index 98%
rename from voice_engine/file_player_unittests.cc
rename to modules/utility/source/file_player_unittests.cc
index dd440fb..58471e5 100644
--- a/voice_engine/file_player_unittests.cc
+++ b/modules/utility/source/file_player_unittests.cc
@@ -10,6 +10,8 @@
// Unit tests for FilePlayer.
+#include "webrtc/modules/utility/include/file_player.h"
+
#include <stdio.h>
#include <string>
@@ -18,7 +20,6 @@
#include "webrtc/base/md5digest.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/voice_engine/file_player.h"
DEFINE_bool(file_player_output, false, "Generate reference files.");
diff --git a/voice_engine/file_recorder_impl.cc b/modules/utility/source/file_recorder_impl.cc
similarity index 98%
rename from voice_engine/file_recorder_impl.cc
rename to modules/utility/source/file_recorder_impl.cc
index bfdc01d..82b37f0 100644
--- a/voice_engine/file_recorder_impl.cc
+++ b/modules/utility/source/file_recorder_impl.cc
@@ -8,10 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/voice_engine/file_recorder_impl.h"
-
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/media_file/media_file.h"
+#include "webrtc/modules/utility/source/file_recorder_impl.h"
#include "webrtc/system_wrappers/include/logging.h"
namespace webrtc {
diff --git a/voice_engine/file_recorder_impl.h b/modules/utility/source/file_recorder_impl.h
similarity index 89%
rename from voice_engine/file_recorder_impl.h
rename to modules/utility/source/file_recorder_impl.h
index 67af742..a9dd3a8 100644
--- a/voice_engine/file_recorder_impl.h
+++ b/modules/utility/source/file_recorder_impl.h
@@ -12,8 +12,8 @@
// multiple file formats. The unencoded input data is written to file in the
// encoded format specified.
-#ifndef WEBRTC_VOICE_ENGINE_FILE_RECORDER_IMPL_H_
-#define WEBRTC_VOICE_ENGINE_FILE_RECORDER_IMPL_H_
+#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
+#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
#include <list>
@@ -24,10 +24,10 @@
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/media_file/media_file.h"
#include "webrtc/modules/media_file/media_file_defines.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
+#include "webrtc/modules/utility/source/coder.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/typedefs.h"
-#include "webrtc/voice_engine/coder.h"
-#include "webrtc/voice_engine/file_recorder.h"
namespace webrtc {
// The largest decoded frame size in samples (60ms with 32kHz sample rate).
@@ -76,5 +76,4 @@
Resampler _audioResampler;
};
} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_FILE_RECORDER_IMPL_H_
+#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
diff --git a/modules/utility/utility.gypi b/modules/utility/utility.gypi
index 2c4e20f..6e11f16 100644
--- a/modules/utility/utility.gypi
+++ b/modules/utility/utility.gypi
@@ -20,11 +20,19 @@
],
'sources': [
'include/audio_frame_operations.h',
+ 'include/file_player.h',
+ 'include/file_recorder.h',
'include/helpers_android.h',
'include/helpers_ios.h',
'include/jvm_android.h',
'include/process_thread.h',
'source/audio_frame_operations.cc',
+ 'source/coder.cc',
+ 'source/coder.h',
+ 'source/file_player_impl.cc',
+ 'source/file_player_impl.h',
+ 'source/file_recorder_impl.cc',
+ 'source/file_recorder_impl.h',
'source/helpers_android.cc',
'source/helpers_ios.mm',
'source/jvm_android.cc',
diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
index f174ab9..31b1d5b 100644
--- a/voice_engine/BUILD.gn
+++ b/voice_engine/BUILD.gn
@@ -16,14 +16,6 @@
"channel_manager.h",
"channel_proxy.cc",
"channel_proxy.h",
- "coder.cc",
- "coder.h",
- "file_player.h",
- "file_player_impl.cc",
- "file_player_impl.h",
- "file_recorder.h",
- "file_recorder_impl.cc",
- "file_recorder_impl.h",
"include/voe_audio_processing.h",
"include/voe_base.h",
"include/voe_codec.h",
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index fda8dcc..9b6a2a1 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -26,8 +26,8 @@
#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "webrtc/voice_engine/file_player.h"
-#include "webrtc/voice_engine/file_recorder.h"
+#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_network.h"
#include "webrtc/voice_engine/level_indicator.h"
diff --git a/voice_engine/output_mixer.h b/voice_engine/output_mixer.h
index 9bf3b35..ae2f53f 100644
--- a/voice_engine/output_mixer.h
+++ b/voice_engine/output_mixer.h
@@ -16,7 +16,7 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
-#include "webrtc/voice_engine/file_recorder.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/level_indicator.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
diff --git a/voice_engine/transmit_mixer.h b/voice_engine/transmit_mixer.h
index ebd90a7..483af05 100644
--- a/voice_engine/transmit_mixer.h
+++ b/voice_engine/transmit_mixer.h
@@ -16,8 +16,8 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_processing/typing_detection.h"
#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/voice_engine/file_player.h"
-#include "webrtc/voice_engine/file_recorder.h"
+#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/level_indicator.h"
#include "webrtc/voice_engine/monitor_module.h"
diff --git a/voice_engine/voice_engine.gyp b/voice_engine/voice_engine.gyp
index 5f26de1..17a64b5 100644
--- a/voice_engine/voice_engine.gyp
+++ b/voice_engine/voice_engine.gyp
@@ -52,14 +52,6 @@
'channel_manager.h',
'channel_proxy.cc',
'channel_proxy.h',
- 'coder.cc',
- 'coder.h',
- 'file_player.h',
- 'file_player_impl.cc',
- 'file_player_impl.h',
- 'file_recorder.h',
- 'file_recorder_impl.cc',
- 'file_recorder_impl.h',
'level_indicator.cc',
'level_indicator.h',
'monitor_module.cc',
@@ -117,7 +109,6 @@
'voice_engine',
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
- '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
# The rest are to satisfy the unittests' include chain.
# This would be unnecessary if we used qualified includes.
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
@@ -133,7 +124,6 @@
],
'sources': [
'channel_unittest.cc',
- 'file_player_unittests.cc',
'network_predictor_unittest.cc',
'transmit_mixer_unittest.cc',
'utility_unittest.cc',
