Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.

webrtc/api/call/audio_send_stream.h

Index: webrtc/api/call/audio_send_stream.h
diff --git a/webrtc/api/call/audio_send_stream.h b/webrtc/api/call/audio_send_stream.h
index b309f7a221c6c07298f0a13133dfbb2389a5b746..2c9ce6ea8ed78e8baee4551ad781f0933fbbfa1a 100644
--- a/webrtc/api/call/audio_send_stream.h
+++ b/webrtc/api/call/audio_send_stream.h
@@ -22,6 +22,28 @@
 
 namespace webrtc {
 
+// TODO(minyue): This is copied from cricket::SendCodecSpec. Find better place
+// for it.
+struct SendCodecSpec {
+  SendCodecSpec() {
+    webrtc::CodecInst empty_inst = {0};
+    codec_inst = empty_inst;
+    codec_inst.pltype = -1;
+  }
+  bool operator==(const SendCodecSpec& rhs) const;
+  bool operator!=(const SendCodecSpec& rhs) const;
+
+  bool nack_enabled = false;
+  bool transport_cc_enabled = false;
+  bool enable_codec_fec = false;
+  bool enable_opus_dtx = false;
+  int opus_max_playback_rate = 0;
+  int red_payload_type = -1;
+  int cng_payload_type = -1;
+  int cng_plfreq = -1;
+  webrtc::CodecInst codec_inst;
+};
+
 // WORK IN PROGRESS
 // This class is under development and is not yet intended for for use outside
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
@@ -94,6 +116,9 @@ class AudioSendStream {
     // Note: This is still an experimental feature and not ready for real usage.
     int min_bitrate_kbps = -1;
     int max_bitrate_kbps = -1;
+
+    int max_send_bitrate_bps = 0;
+    SendCodecSpec send_codec_spec;
   };
 
   // Starts stream activity.