commit d2845f82461dbde7deffe58ca5cae7aad51a649f
author Sebastian Jansson <srte@webrtc.org> Tue Aug 06 17:20:14 2019 +0200
committer Commit Bot <commit-bot@chromium.org> Wed Aug 07 09:36:42 2019 +0000
tree 35e98b199e6789cef58a5ce636e6fa7add66e890
parent d23f67e6beb55b030cdb564b5980f8b8b1f7883f

Removes unused AudioAllocationSettings from voice engine.

Bug: webrtc:9883
Change-Id: Ie322a1cae1f9682f64a05767f3933cba13b70ae0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148281
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28787}

media/BUILD.gn

diff --git a/media/BUILD.gn b/media/BUILD.gn
index ab2f8f2..dd19656 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -291,7 +291,6 @@
     "../rtc_base:checks",
     "../rtc_base:rtc_task_queue",
     "../rtc_base:stringutils",
-    "../rtc_base/experiments:audio_allocation_settings",
     "../rtc_base/experiments:experimental_screenshare_settings",
     "../rtc_base/experiments:field_trial_parser",
     "../rtc_base/experiments:normalize_simulcast_size_experiment",

media/engine/webrtc_voice_engine.cc

diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 9fe6f79..189d7a6 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -1024,7 +1024,6 @@
 
   rtc::ThreadChecker worker_thread_checker_;
   rtc::RaceChecker audio_capture_race_checker_;
-  const webrtc::AudioAllocationSettings allocation_settings_;
   webrtc::Call* call_ = nullptr;
   webrtc::AudioSendStream::Config config_;
   // The stream is owned by WebRtcAudioSendStream and may be reallocated if

media/engine/webrtc_voice_engine.h

diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index a4c8baa..5ef2fde 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -27,7 +27,6 @@
 #include "media/engine/apm_helpers.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/constructor_magic.h"
-#include "rtc_base/experiments/audio_allocation_settings.h"
 #include "rtc_base/network_route.h"
 #include "rtc_base/task_queue.h"
 #include "rtc_base/thread_checker.h"
@@ -104,8 +103,6 @@
   rtc::ThreadChecker signal_thread_checker_;
   rtc::ThreadChecker worker_thread_checker_;
 
-  const webrtc::AudioAllocationSettings allocation_settings_;
-
   // The audio device module.
   rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_;
   rtc::scoped_refptr<webrtc::AudioEncoderFactory> encoder_factory_;