From 81da488ab6c9c79e4a74e29d18c574285312241f Mon Sep 17 00:00:00 2001
From: aleloi <aleloi@webrtc.org>
Date: Tue, 8 Nov 2016 04:26:30 -0800
Subject: [PATCH] Added audio mixer and removed audio device module in
 AudioState::Config.

The audio_device_module field was currently unused. The audio_mixer
field is going to be used to pass an AudioMixer to AudioState.

In the hopefully-not-very-far future, the toplevel WebRTC API will allow passing
a custom AudioMixer, e.g. for spatialized audio (audio in space). If no
mixer is passed, a default mixer is created (the one in modules/audio_mixer).

The only object which will have a permanent reference to the mixer is AudioState.
AudioState is created in WebRTCVoiceEngine with a configuration object,
which already contains a VoiceEngine pointer. In this CL, we extend this
config object with a mixer pointer.

In summary: in an upcoming CL, a mixer will be either created in or passed to
WebRTCVoiceEngine. This mixer will be passed to the ctor of AudioState in a
config struct.

BUG=webrtc:6346
NOTRY=True

Review-Url: https://codereview.webrtc.org/2456363002
Cr-Commit-Position: refs/heads/master@{#14973}
---
 webrtc/api/BUILD.gn           | 1 +
 webrtc/api/api.gyp            | 1 +
 webrtc/api/call/audio_state.h | 6 ++++--
 3 files changed, 6 insertions(+), 2 deletions(-)

diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index 8a7ca03608..cf3073d500 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -36,6 +36,7 @@ rtc_source_set("call_api") {
 
   deps = [
     # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
+    ":audio_mixer_api",
     "..:webrtc_common",
     "../base:rtc_base_approved",
     "../modules/audio_coding:audio_encoder_interface",
diff --git a/webrtc/api/api.gyp b/webrtc/api/api.gyp
index b50dfd5683..70efb5a27e 100644
--- a/webrtc/api/api.gyp
+++ b/webrtc/api/api.gyp
@@ -99,6 +99,7 @@
       'type': 'static_library',
       'dependencies': [
         # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
+        ':audio_mixer_api',
         '<(webrtc_root)/base/base.gyp:rtc_base_approved',
         '<(webrtc_root)/common.gyp:webrtc_common',
         '<(webrtc_root)/modules/modules.gyp:audio_encoder_interface',
diff --git a/webrtc/api/call/audio_state.h b/webrtc/api/call/audio_state.h
index ac912773aa..1941cbf7bc 100644
--- a/webrtc/api/call/audio_state.h
+++ b/webrtc/api/call/audio_state.h
@@ -10,6 +10,7 @@
 #ifndef WEBRTC_API_CALL_AUDIO_STATE_H_
 #define WEBRTC_API_CALL_AUDIO_STATE_H_
 
+#include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 
@@ -33,8 +34,9 @@ class AudioState : public rtc::RefCountInterface {
     // the AudioState itself.
     VoiceEngine* voice_engine = nullptr;
 
-    // The AudioDeviceModule associated with the Calls.
-    AudioDeviceModule* audio_device_module = nullptr;
+    // The audio mixer connected to active receive streams. One per
+    // AudioState.
+    rtc::scoped_refptr<AudioMixer> audio_mixer;
   };
 
   // TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.