Add AudioSendStream to Call.

webrtc/audio/audio_send_stream.cc

Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
new file mode 100644
index 0000000000000000000000000000000000000000..0d0c072bf4a58fd94ac82639a439530afc5bea5a
--- /dev/null
+++ b/webrtc/audio/audio_send_stream.cc
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/audio/audio_send_stream.h"
+
+#include <string>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+
+namespace webrtc {
+std::string AudioSendStream::Config::Rtp::ToString() const {
+  std::stringstream ss;
+  ss << "{ssrc: " << ssrc;
+  ss << ", extensions: [";
+  for (size_t i = 0; i < extensions.size(); ++i) {
+    ss << extensions[i].ToString();
+    if (i != extensions.size() - 1)
+      ss << ", ";
+  }
+  ss << ']';
+  ss << '}';
+  return ss.str();
+}
+
+std::string AudioSendStream::Config::ToString() const {
+  std::stringstream ss;
+  ss << "{rtp: " << rtp.ToString();
+  ss << ", voe_channel_id: " << voe_channel_id;
+  // TODO(solenberg): Encoder config.
+  ss << ", cng_payload_type: " << cng_payload_type;
+  ss << ", red_payload_type: " << red_payload_type;
+  ss << '}';
+  return ss.str();
+}
+
+namespace internal {
+AudioSendStream::AudioSendStream(const webrtc::AudioSendStream::Config& config)
+    : config_(config) {
+  LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
+  RTC_DCHECK(config.voe_channel_id != -1);
+}
+
+AudioSendStream::~AudioSendStream() {
+  LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
+}
+
+webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
+  return webrtc::AudioSendStream::Stats();
+}
+
+void AudioSendStream::Start() {
+}
+
+void AudioSendStream::Stop() {
+}
+
+void AudioSendStream::SignalNetworkState(NetworkState state) {
+}
+
+bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  return false;
+}
+}  // namespace internal
+}  // namespace webrtc