Add 120ms frame ability to ANA

webrtc/media/BUILD.gn

 # Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 import("//build/config/linux/pkg_config.gni")
 import("../webrtc.gni")
@@ skipping 115 matching lines @@
     "engine/webrtcvideoencoderfactory.h",
     "engine/webrtcvideoengine2.cc",
     "engine/webrtcvideoengine2.h",
     "engine/webrtcvideoframe.h",
     "engine/webrtcvoe.h",
     "engine/webrtcvoiceengine.cc",
     "engine/webrtcvoiceengine.h",
     "sctp/sctptransportinternal.h",
   ]
 
+  defines = []

[minyue-webrtc, 2017/02/02 10:42:08]

rebased needed here

[michaelt, 2017/02/02 11:17:18]

Done.

+  if (rtc_opus_support_120ms_ptime) {
+    defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
+  } else {
+    defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
+  }
+
   if (rtc_enable_sctp) {
     sources += [
       "sctp/sctptransport.cc",
       "sctp/sctptransport.h",
     ]
   }
 
   configs += [ ":rtc_media_warnings_config" ]
 
   if (!build_with_chromium && is_clang) {
@@ skipping 227 matching lines @@
       # TODO(kjellander): Move as part of work in bugs.webrtc.org/4243.
       ":rtc_media",
       ":rtc_unittest_main",
       "../audio",
       "../base:rtc_base_tests_utils",
       "../modules/audio_device:mock_audio_device",
       "../system_wrappers:metrics_default",
     ]
   }
 }