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Move FilePlayer and FileRecorder to Voice Engine

Browse Source 
Because Voice Engine was the only user.

(This has been landed twice before, as
https://codereview.webrtc.org/2037623002 and
https://codereview.webrtc.org/2240163002. Third time's a charm!)

NOPRESUBMIT=True
TBR=kjellander@webrtc.org

Review-Url: https://codereview.webrtc.org/2247033003
Cr-Commit-Position: refs/heads/master@{#13777}
This commit is contained in:
kwiberg
2016-08-16 03:34:44 -07:00
committed by Commit bot
parent 2f69ce9498
commit 427ce3d86f
22 changed files with 180 additions and 54 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
.gn
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@ -21,6 +21,9 @@ secondary_source = "//build/secondary/"
# TODO(kjellander): Keep adding paths to this list as work in webrtc:5589 is done.
check_targets = [
"//webrtc/modules/audio_device/*",
"//webrtc/voice_engine:audio_coder",
"//webrtc/voice_engine:file_player",
"//webrtc/voice_engine:file_recorder",
"//webrtc/voice_engine:level_indicator",
"//webrtc/modules/audio_coding:isac_fix_test",
"//webrtc/modules/audio_mixer:audio_conference_mixer",

3
webrtc/modules/BUILD.gn
View File

@ -306,7 +306,6 @@ 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",
@ -596,8 +595,6 @@ 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/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/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

@ -358,7 +358,6 @@
'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',
@ -599,8 +598,6 @@
'<(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

@ -110,8 +110,6 @@
'<(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
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@ -11,18 +11,10 @@ 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",

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

@ -20,19 +20,11 @@
],
'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

@ -9,6 +9,74 @@
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 = [
"..:webrtc_common",
"../modules/audio_coding:audio_coding",
"../modules/audio_coding:builtin_audio_decoder_factory",
"../modules/audio_coding:rent_a_codec",
]
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 = [
":audio_coder",
"..:webrtc_common",
"../common_audio:common_audio",
"../modules/media_file:media_file",
"../system_wrappers:system_wrappers",
]
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 = [
":audio_coder",
"..:webrtc_common",
"../base:rtc_base_approved",
"../common_audio:common_audio",
"../modules/media_file:media_file",
"../system_wrappers:system_wrappers",
]
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",
@ -89,6 +157,8 @@ source_set("voice_engine") {
}
deps = [
":file_player",
":file_recorder",
":level_indicator",
"..:rtc_event_log",
"..:webrtc_common",
@ -129,6 +199,7 @@ 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",
@ -144,10 +215,15 @@ 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/modules/utility/include/file_player.h"
#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/file_player.h"
#include "webrtc/voice_engine/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"

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

@ -8,10 +8,11 @@
* 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/modules/utility/source/coder.h → webrtc/voice_engine/coder.h
View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
#define WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
#ifndef WEBRTC_VOICE_ENGINE_CODER_H_
#define WEBRTC_VOICE_ENGINE_CODER_H_
#include <memory>
@ -65,4 +65,4 @@ class AudioCoder : public AudioPacketizationCallback {
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
#endif // WEBRTC_VOICE_ENGINE_CODER_H_

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

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
#ifndef WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
#define WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
@ -83,4 +83,5 @@ protected:
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
#endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_

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

@ -8,7 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/utility/source/file_player_impl.h"
#include "webrtc/voice_engine/file_player_impl.h"
#include "webrtc/system_wrappers/include/logging.h"
namespace webrtc {

11
webrtc/modules/utility/source/file_player_impl.h → webrtc/voice_engine/file_player_impl.h
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@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
#ifndef WEBRTC_VOICE_ENGINE_FILE_PLAYER_IMPL_H_
#define WEBRTC_VOICE_ENGINE_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
@ -75,4 +75,5 @@ private:
float _scaling;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
#endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_IMPL_H_

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

@ -10,8 +10,6 @@
// Unit tests for FilePlayer.
#include "webrtc/modules/utility/include/file_player.h"
#include <stdio.h>
#include <string>
@ -20,6 +18,7 @@
#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.");

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

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
#ifndef WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_
#define WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
@ -61,4 +61,5 @@ protected:
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
#endif // WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_

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

@ -8,9 +8,10 @@
* 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/modules/utility/source/file_recorder_impl.h → webrtc/voice_engine/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_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
#ifndef WEBRTC_VOICE_ENGINE_FILE_RECORDER_IMPL_H_
#define WEBRTC_VOICE_ENGINE_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).
@ -76,4 +76,5 @@ private:
Resampler _audioResampler;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
#endif // WEBRTC_VOICE_ENGINE_FILE_RECORDER_IMPL_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/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/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/modules/utility/include/file_player.h"
#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/file_player.h"
#include "webrtc/voice_engine/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"

57
webrtc/voice_engine/voice_engine.gyp
View File

@ -29,6 +29,8 @@
'<(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': [
@ -94,6 +96,53 @@
'voice_engine_impl.h',
],
},
{
'target_name': 'audio_coder',
'type': 'static_library',
'sources': [
'coder.cc',
'coder.h',
],
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/modules/modules.gyp:audio_coding_module',
'<(webrtc_root)/modules/modules.gyp:builtin_audio_decoder_factory',
'<(webrtc_root)/modules/modules.gyp:rent_a_codec',
],
},
{
'target_name': 'file_player',
'type': 'static_library',
'sources': [
'file_player.h',
'file_player_impl.cc',
'file_player_impl.h',
],
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/modules/modules.gyp:media_file',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'audio_coder',
],
},
{
'target_name': 'file_recorder',
'type': 'static_library',
'sources': [
'file_recorder.h',
'file_recorder_impl.cc',
'file_recorder_impl.h',
],
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/modules/modules.gyp:media_file',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'audio_coder',
],
},
{
'target_name': 'level_indicator',
'type': 'static_library',
@ -121,6 +170,7 @@
'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',
@ -136,6 +186,7 @@
],
'sources': [
'channel_unittest.cc',
'file_player_unittests.cc',
'network_predictor_unittest.cc',
'transmit_mixer_unittest.cc',
'utility_unittest.cc',
@ -152,6 +203,12 @@
'<(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

@ -19,5 +19,13 @@
],
},
}],
['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',
],
},
}],
],
}