Move FilePlayer and FileRecorder to Voice Engine

webrtc/voice_engine/BUILD.gn

Index: webrtc/voice_engine/BUILD.gn
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
index 6413c1842a27c172c71418078ebefdda3c93050a..0591999def5d20ab6bf3e9acde633515b2c1e58b 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -8,6 +8,64 @@
 
 import("../build/webrtc.gni")
 
+rtc_static_library("audio_coder") {

[the sun, 2017/01/09 12:44:43]

Do we really benefit from having separate targets for these? can we just add the
files to the voice_engine static lib? It seems to pull in the same deps, and
also, "file_player_unittest.cc" is included in voice_engine_unittests.

[kwiberg-webrtc, 2017/01/09 12:53:55]

Grouping them by themselves like this makes it easier to see what pieces there
are, instead of just seeing it as one big Voice Engine pile. In particular, it's
now obvious at a glance which source files are responsible for which
dependencies.

There are currently two more advantages that will eventually go away: I was able
to easily mark the new targets as being subject to "gn check", and will easily
be able to fix the find_bad_constructs errors and remove the suppressions. Being
able to do this for several small targets, one by one, rather than having to do
it for giant targets increases the chance that it gets done.

[kjellander_webrtc, 2017/01/09 15:17:21]

Nice, I like small targets. There are many benefits with that and you already
mentioned a few of them.

+  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" ]