Move FilePlayer and FileRecorder to Voice Engine

Because Voice Engine was the only user. R=perkj@webrtc.org, solenberg@webrtc.org Review URL: https://codereview.webrtc.org/2037623002 . Cr-Commit-Position: refs/heads/master@{#13261}
2016-06-22 23:47:42 +02:00 · 2016-06-22 23:47:42 +02:00 · 65874b163e
commit 65874b163e
parent 821942d8b2
18 changed files with 53 additions and 47 deletions
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@ -223,7 +223,6 @@ 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",
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@ -351,7 +351,6 @@
            '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',
--- a/webrtc/modules/utility/BUILD.gn
+++ b/webrtc/modules/utility/BUILD.gn
@ -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",
--- a/webrtc/modules/utility/utility.gypi
+++ b/webrtc/modules/utility/utility.gypi
@ -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',
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@ -16,6 +16,14 @@ source_set("voice_engine") {
    "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",
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/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/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"
--- a/webrtc/modules/utility/source/coder.cc
+++ b/webrtc/modules/utility/source/coder.cc
@ -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 {
--- a/webrtc/modules/utility/source/coder.h
+++ b/webrtc/modules/utility/source/coder.h
@ -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_
--- a/webrtc/modules/utility/include/file_player.h
+++ b/webrtc/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_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_
--- a/webrtc/modules/utility/source/file_player_impl.cc
+++ b/webrtc/modules/utility/source/file_player_impl.cc
@ -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 {
--- a/webrtc/modules/utility/source/file_player_impl.h
+++ b/webrtc/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_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_
--- a/webrtc/modules/utility/source/file_player_unittests.cc
+++ b/webrtc/modules/utility/source/file_player_unittests.cc
@ -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.");

--- a/webrtc/modules/utility/include/file_recorder.h
+++ b/webrtc/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_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_
--- a/webrtc/modules/utility/source/file_recorder_impl.cc
+++ b/webrtc/modules/utility/source/file_recorder_impl.cc
@ -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 {
--- a/webrtc/modules/utility/source/file_recorder_impl.h
+++ b/webrtc/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_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_
--- a/webrtc/voice_engine/output_mixer.h
+++ b/webrtc/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/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"

--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/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/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"
--- a/webrtc/voice_engine/voice_engine.gyp
+++ b/webrtc/voice_engine/voice_engine.gyp
@ -52,6 +52,14 @@
        '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',
@ -109,6 +117,7 @@
            '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',
@ -124,6 +133,7 @@
          ],
          'sources': [
            'channel_unittest.cc',
+            'file_player_unittests.cc',
            'network_predictor_unittest.cc',
            'transmit_mixer_unittest.cc',
            'utility_unittest.cc',