Replace AudioConferenceMixer with AudioMixer.

webrtc/audio/BUILD.gn

Index: webrtc/audio/BUILD.gn
diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
index e89350e89ae25e805f6de0944d59ef5488e2b538..bbfc2279eca4f1b600430d360663fcb0f07ed743 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -32,6 +32,7 @@ rtc_static_library("audio") {
     "../api:audio_mixer_api",
     "../api:call_api",
     "../base:rtc_base_approved",
+    "../common_audio",
     "../modules/audio_device",
     "../modules/audio_processing",
     "../system_wrappers",
@@ -44,12 +45,14 @@ if (rtc_include_tests) {
     sources = [
       "audio_receive_stream_unittest.cc",
       "audio_send_stream_unittest.cc",
+      "audio_state_audio_path_unittest.cc",
       "audio_state_unittest.cc",
     ]
     deps = [
       ":audio",
       "../base:rtc_base_approved",
       "../modules/audio_device:mock_audio_device",
+      "../modules/audio_mixer:audio_mixer_impl",
       "../test:test_common",
       "//testing/gmock",
       "//testing/gtest",