Move FilePlayer and FileRecorder to Voice Engine

Because Voice Engine was the only user.

(We have tried to land this many times before. I'm hoping that this
time all external dependencies on these files will really be gone.)

BUG=none

Review-Url: https://codereview.webrtc.org/2622493002
Cr-Commit-Position: refs/heads/master@{#15978}

.gn

@@ -42,6 +42,9 @@ check_targets = [
   "//webrtc/modules/remote_bitrate_estimator/*",
   "//webrtc/stats:rtc_stats",
   "//webrtc/voice_engine",
+  "//webrtc/voice_engine:audio_coder",
+  "//webrtc/voice_engine:file_player",
+  "//webrtc/voice_engine:file_recorder",
   "//webrtc/voice_engine:level_indicator",
 ]
 

webrtc/modules/BUILD.gn

@@ -231,8 +231,6 @@ 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",
@@ -483,7 +481,6 @@ if (rtc_include_tests) {
       "rtp_rtcp/test/testAPI/test_api_audio.cc",
       "rtp_rtcp/test/testAPI/test_api_rtcp.cc",
       "rtp_rtcp/test/testAPI/test_api_video.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",

webrtc/modules/utility/BUILD.gn

@@ -11,15 +11,9 @@ import("../../build/webrtc.gni")
 rtc_static_library("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/coder.cc",
-    "source/coder.h",
-    "source/file_player.cc",
-    "source/file_recorder.cc",
     "source/helpers_android.cc",
     "source/jvm_android.cc",
     "source/process_thread_impl.cc",

webrtc/voice_engine/BUILD.gn

@@ -8,6 +8,64 @@
 
 import("../build/webrtc.gni")
 
+rtc_static_library("audio_coder") {
+  sources = [
+    "coder.cc",
+    "coder.h",
+  ]
+  deps = [
+    "..:webrtc_common",
+    "../modules/audio_coding",
+    "../modules/audio_coding:audio_format_conversion",
+    "../modules/audio_coding:builtin_audio_decoder_factory",
+    "../modules/audio_coding:rent_a_codec",
+  ]
+
+  if (!build_with_chromium && is_clang) {
+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+  }
+}
+
+rtc_static_library("file_player") {
+  sources = [
+    "file_player.cc",
+    "file_player.h",
+  ]
+  deps = [
+    ":audio_coder",
+    "..:webrtc_common",
+    "../common_audio",
+    "../modules/media_file",
+    "../system_wrappers",
+  ]
+
+  if (!build_with_chromium && is_clang) {
+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+  }
+}
+
+rtc_static_library("file_recorder") {
+  sources = [
+    "file_recorder.cc",
+    "file_recorder.h",
+  ]
+  deps = [
+    ":audio_coder",
+    "..:webrtc_common",
+    "../base:rtc_base_approved",
+    "../common_audio",
+    "../modules/media_file:media_file",
+    "../system_wrappers",
+  ]
+
+  if (!build_with_chromium && is_clang) {
+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+  }
+}
+
 rtc_static_library("voice_engine") {
   sources = [
     "channel.cc",
@@ -85,6 +143,8 @@ rtc_static_library("voice_engine") {
     "../modules/audio_coding",
   ]
   deps = [
+    ":file_player",
+    ":file_recorder",
     ":level_indicator",
     "..:webrtc_common",
     "../api:audio_mixer_api",
@@ -191,6 +251,7 @@ 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",
@@ -210,6 +271,7 @@ if (rtc_include_tests) {
 
     sources = [
       "channel_unittest.cc",
+      "file_player_unittests.cc",
       "test/channel_transport/udp_socket_manager_unittest.cc",
       "test/channel_transport/udp_socket_wrapper_unittest.cc",
       "test/channel_transport/udp_transport_unittest.cc",
@@ -223,6 +285,11 @@ if (rtc_include_tests) {
       "voice_engine_fixture.h",
     ]
 
+    data = [
+      "//resources/utility/encapsulated_pcm16b_8khz.wav",
+      "//resources/utility/encapsulated_pcmu_8khz.wav",
+    ]
+
     if (is_win) {
       defines = [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ]
 

webrtc/voice_engine/channel.h

@@ -28,8 +28,8 @@
 #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_base.h"
 #include "webrtc/voice_engine/include/voe_network.h"

webrtc/voice_engine/coder.cc

@@ -8,11 +8,12 @@
  *  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/audio_format_conversion.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 {

webrtc/voice_engine/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_

webrtc/voice_engine/file_player.cc

@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/voice_engine/file_player.h"
 
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/media_file/media_file.h"
 #include "webrtc/modules/media_file/media_file_defines.h"
-#include "webrtc/modules/utility/source/coder.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/voice_engine/coder.h"
 
 namespace webrtc {
 

webrtc/voice_engine/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 <memory>
 
@@ -77,4 +77,4 @@ class FilePlayer {
   virtual int32_t SetAudioScaling(float scaleFactor) = 0;
 };
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+#endif  // WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_

webrtc/voice_engine/file_player_unittests.cc

@@ -10,7 +10,7 @@
 
 // Unit tests for FilePlayer.
 
-#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/voice_engine/file_player.h"
 
 #include <stdio.h>
 

webrtc/voice_engine/file_recorder.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/utility/include/file_recorder.h"
+#include "webrtc/voice_engine/file_recorder.h"
 
 #include <list>
 
@@ -18,10 +18,10 @@
 #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/source/coder.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/voice_engine/coder.h"
 
 namespace webrtc {
 

webrtc/voice_engine/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 <memory>
 
@@ -54,4 +54,4 @@ class FileRecorder {
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+#endif  // WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_

webrtc/voice_engine/output_mixer.h

@@ -18,7 +18,7 @@
 #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"
 

webrtc/voice_engine/transmit_mixer.h

@@ -18,8 +18,8 @@
 #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"