Align new VoE API with design.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1376153003

Cr-Commit-Position: refs/heads/master@{#10136}

webrtc/audio_receive_stream.h

@@ -17,6 +17,7 @@
 
 #include "webrtc/config.h"
 #include "webrtc/stream.h"
+#include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -44,9 +45,13 @@ class AudioReceiveStream : public ReceiveStream {
       std::vector<RtpExtension> extensions;
     } rtp;
 
-    // Underlying VoiceEngine handle, used to map AudioReceiveStream to
-    // lower-level components. Temporarily used while VoiceEngine channels are
-    // created outside of Call.
+    Transport* receive_transport = nullptr;
+    Transport* rtcp_send_transport = nullptr;
+
+    // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
+    // level components.
+    // TODO(solenberg): Remove when VoiceEngine channels are created outside
+    // of Call.
     int voe_channel_id = -1;
 
     // Identifier for an A/V synchronization group. Empty string to disable.

webrtc/audio_send_stream.h

@@ -48,6 +48,12 @@ class AudioSendStream : public SendStream {
     // Transport for outgoing packets.
     Transport* send_transport = nullptr;
 
+    // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level
+    // components.
+    // TODO(solenberg): Remove when VoiceEngine channels are created outside
+    // of Call.
+    int voe_channel_id = -1;
+
     rtc::scoped_ptr<AudioEncoder> encoder;
     int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
     int red_payload_type = -1;  // pt, or -1 to disable REDundant coding.

webrtc/call.h

@@ -85,7 +85,7 @@ class Call {
     } bitrate_config;
 
     struct AudioConfig {
-      AudioDeviceModule* audio_device_manager = nullptr;
+      AudioDeviceModule* audio_device_module = nullptr;
       AudioProcessing* audio_processing = nullptr;
       VoiceEngineObserver* voice_engine_observer = nullptr;
     } audio_config;