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Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #6 id:100001 of https://codereview.webrtc.org/2240163002/ )

Browse Source 
Reason for revert:
Breaks downstream code, so revert again. Yay.

Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> (This is a re-land of https://codereview.webrtc.org/2037623002, which
> had to be reverted.)
>
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/dc65ea29b3270ad418050658ad962ddd33ee70c1
> Cr-Commit-Position: refs/heads/master@{#13757}

TBR=perkj@webrtc.org,kjellander@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/2245153002
Cr-Commit-Position: refs/heads/master@{#13758}
This commit is contained in:
kwiberg
2016-08-15 11:43:51 -07:00
committed by Commit bot
parent dc65ea29b3
commit c8c71f484e
22 changed files with 55 additions and 168 deletions
Download Patch File Download Diff File Expand all files Collapse all files

7
.gn
View File

@ -19,12 +19,7 @@ secondary_source = "//build/secondary/"
# their includes checked for proper dependencies when you run either
# "gn check" or "gn gen --check".
# TODO(kjellander): Keep adding paths to this list as work in webrtc:5589 is done.
check_targets = [
"//webrtc/voice_engine:audio_coder",
"//webrtc/voice_engine:file_player",
"//webrtc/voice_engine:file_recorder",
"//webrtc/voice_engine:level_indicator",
]
check_targets = [ "//webrtc/voice_engine:level_indicator" ]
# These are the list of GN files that run exec_script. This whitelist exists
# to force additional review for new uses of exec_script, which is strongly

3
webrtc/modules/BUILD.gn
View File

@ -306,6 +306,7 @@ if (rtc_include_tests) {
"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",
@ -595,6 +596,8 @@ if (rtc_include_tests) {
"//resources/synthetic-trace.rx",
"//resources/tmobile-downlink.rx",
"//resources/tmobile-uplink.rx",
"//resources/utility/encapsulated_pcm16b_8khz.wav",
"//resources/utility/encapsulated_pcmu_8khz.wav",
"//resources/verizon3g-downlink.rx",
"//resources/verizon3g-uplink.rx",
"//resources/verizon4g-downlink.rx",

2
webrtc/modules/audio_mixer/audio_mixer.h
View File

@ -16,7 +16,7 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_mixer/new_audio_conference_mixer.h"
#include "webrtc/modules/audio_mixer/audio_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"

3
webrtc/modules/modules.gyp
View File

@ -358,6 +358,7 @@
'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',
@ -598,6 +599,8 @@
'<(DEPTH)/resources/synthetic-trace.rx',
'<(DEPTH)/resources/tmobile-downlink.rx',
'<(DEPTH)/resources/tmobile-uplink.rx',
'<(DEPTH)/resources/utility/encapsulated_pcm16b_8khz.wav',
'<(DEPTH)/resources/utility/encapsulated_pcmu_8khz.wav',
'<(DEPTH)/resources/verizon3g-downlink.rx',
'<(DEPTH)/resources/verizon3g-uplink.rx',
'<(DEPTH)/resources/verizon4g-downlink.rx',

2
webrtc/modules/modules_unittests.isolate
View File

@ -110,6 +110,8 @@
'<(DEPTH)/resources/synthetic-trace.rx',
'<(DEPTH)/resources/tmobile-downlink.rx',
'<(DEPTH)/resources/tmobile-uplink.rx',
'<(DEPTH)/resources/utility/encapsulated_pcm16b_8khz.wav',
'<(DEPTH)/resources/utility/encapsulated_pcmu_8khz.wav',
'<(DEPTH)/resources/verizon3g-downlink.rx',
'<(DEPTH)/resources/verizon3g-uplink.rx',
'<(DEPTH)/resources/verizon4g-downlink.rx',

8
webrtc/modules/utility/BUILD.gn
View File

@ -11,10 +11,18 @@ import("../../build/webrtc.gni")
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",

7
webrtc/voice_engine/file_player.h → webrtc/modules/utility/include/file_player.h
View File

@ -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 @@ protected:
};
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_

7
webrtc/voice_engine/file_recorder.h → webrtc/modules/utility/include/file_recorder.h
View File

@ -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 @@ protected:
};
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_
#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_

3
webrtc/voice_engine/coder.cc → webrtc/modules/utility/source/coder.cc
View File

@ -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 {

6
webrtc/voice_engine/coder.h → webrtc/modules/utility/source/coder.h
View File

@ -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 @@ class AudioCoder : public AudioPacketizationCallback {
};
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_CODER_H_
#endif // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_

3
webrtc/voice_engine/file_player_impl.cc → webrtc/modules/utility/source/file_player_impl.cc
View File

@ -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 {

11
webrtc/voice_engine/file_player_impl.h → webrtc/modules/utility/source/file_player_impl.h
View File

@ -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 @@ private:
float _scaling;
};
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_IMPL_H_
#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_

3
webrtc/voice_engine/file_player_unittests.cc → webrtc/modules/utility/source/file_player_unittests.cc
View File

@ -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.");

3
webrtc/voice_engine/file_recorder_impl.cc → webrtc/modules/utility/source/file_recorder_impl.cc
View File

@ -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 {

11
webrtc/voice_engine/file_recorder_impl.h → webrtc/modules/utility/source/file_recorder_impl.h
View File

@ -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 @@ private:
Resampler _audioResampler;
};
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_FILE_RECORDER_IMPL_H_
#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_

8
webrtc/modules/utility/utility.gypi
View File

@ -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',

76
webrtc/voice_engine/BUILD.gn
View File

@ -9,74 +9,6 @@
import("../build/webrtc.gni")
import("//testing/test.gni")
source_set("audio_coder") {
sources = [
"coder.cc",
"coder.h",
]
configs += [ "..:common_config" ]
public_configs = [ "..:common_inherited_config" ]
deps = [
"../modules/audio_coding:audio_coding",
"../modules/audio_coding:builtin_audio_decoder_factory",
"../modules/audio_coding:rent_a_codec",
"..:webrtc_common",
]
if (is_clang) {
# Suppress warnings from Chrome's Clang plugins.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
configs -= [ "//build/config/clang:find_bad_constructs" ]
}
}
source_set("file_player") {
sources = [
"file_player.h",
"file_player_impl.cc",
"file_player_impl.h",
]
configs += [ "..:common_config" ]
public_configs = [ "..:common_inherited_config" ]
deps = [
"../common_audio:common_audio",
"../modules/media_file:media_file",
"../system_wrappers:system_wrappers",
"..:webrtc_common",
":audio_coder",
]
if (is_clang) {
# Suppress warnings from Chrome's Clang plugins.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
configs -= [ "//build/config/clang:find_bad_constructs" ]
}
}
source_set("file_recorder") {
sources = [
"file_recorder.h",
"file_recorder_impl.cc",
"file_recorder_impl.h",
]
configs += [ "..:common_config" ]
public_configs = [ "..:common_inherited_config" ]
deps = [
"../base:rtc_base_approved",
"../common_audio:common_audio",
"../modules/media_file:media_file",
"../system_wrappers:system_wrappers",
"..:webrtc_common",
":audio_coder",
]
if (is_clang) {
# Suppress warnings from Chrome's Clang plugins.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
configs -= [ "//build/config/clang:find_bad_constructs" ]
}
}
source_set("voice_engine") {
sources = [
"channel.cc",
@ -157,8 +89,6 @@ source_set("voice_engine") {
}
deps = [
":file_player",
":file_recorder",
":level_indicator",
"..:rtc_event_log",
"..:webrtc_common",
@ -199,7 +129,6 @@ if (rtc_include_tests) {
":voice_engine",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
"//webrtc/common_audio",
"//webrtc/modules/audio_coding",
"//webrtc/modules/audio_conference_mixer",
@ -215,15 +144,10 @@ if (rtc_include_tests) {
if (is_android) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
shard_timeout = 900
data = [
"//resources/utility/encapsulated_pcm16b_8khz.wav",
"//resources/utility/encapsulated_pcmu_8khz.wav",
]
}
sources = [
"channel_unittest.cc",
"file_player_unittests.cc",
"network_predictor_unittest.cc",
"transmit_mixer_unittest.cc",
"utility_unittest.cc",

4
webrtc/voice_engine/channel.h
View File

@ -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"

2
webrtc/voice_engine/output_mixer.h
View File

@ -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"

4
webrtc/voice_engine/transmit_mixer.h
View File

@ -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"

42
webrtc/voice_engine/voice_engine.gyp
View File

@ -29,8 +29,6 @@
'<(webrtc_root)/modules/modules.gyp:webrtc_utility',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/webrtc.gyp:rtc_event_log',
'file_player',
'file_recorder',
'level_indicator',
],
'export_dependent_settings': [
@ -96,38 +94,6 @@
'voice_engine_impl.h',
],
},
{
'target_name': 'audio_coder',
'type': 'static_library',
'sources': [
'coder.cc',
'coder.h',
],
},
{
'target_name': 'file_player',
'type': 'static_library',
'sources': [
'file_player.h',
'file_player_impl.cc',
'file_player_impl.h',
],
'dependencies': [
'audio_coder',
],
},
{
'target_name': 'file_recorder',
'type': 'static_library',
'sources': [
'file_recorder.h',
'file_recorder_impl.cc',
'file_recorder_impl.h',
],
'dependencies': [
'audio_coder',
],
},
{
'target_name': 'level_indicator',
'type': 'static_library',
@ -155,7 +121,6 @@
'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',
@ -171,7 +136,6 @@
],
'sources': [
'channel_unittest.cc',
'file_player_unittests.cc',
'network_predictor_unittest.cc',
'transmit_mixer_unittest.cc',
'utility_unittest.cc',
@ -188,12 +152,6 @@
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
],
}],
['OS=="ios"', {
'mac_bundle_resources': [
'<(DEPTH)/resources/utility/encapsulated_pcm16b_8khz.wav',
'<(DEPTH)/resources/utility/encapsulated_pcmu_8khz.wav',
],
}],
],
},
{

8
webrtc/voice_engine/voice_engine_unittests.isolate
View File

@ -19,13 +19,5 @@
],
},
}],
['OS=="linux" or OS=="mac" or OS=="win" or OS=="android"', {
'variables': {
'files': [
'<(DEPTH)/resources/utility/encapsulated_pcm16b_8khz.wav',
'<(DEPTH)/resources/utility/encapsulated_pcmu_8khz.wav',
],
},
}],
],
}