Add AudioSendStream to Call.

webrtc/audio/audio_send_stream.cc

+/*
+ *  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"
+
+namespace webrtc {
+std::string AudioSendStream::Config::Rtp::ToString() const {

[tommi, 2015/10/14 13:21:18]

is this needed for debugging purposes? If so, can we included only in such
builds?

[the sun, 2015/10/14 14:25:24]

I like that idea, but it runs deeper than just this function, or its siblings.
The use is usually something like this:
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/webrtc...

In other words, the function will be called in any build, so we can 
a) make it an empty function and hope that the compiler removes the
implementation for us, or
b) change the logging semantics so no cycles are wasted on manipulating strings
in release builds.

Thoughts?

[stefan-webrtc, 2015/10/14 14:29:22]

This string is used to LOG the stream configurations into the webrtc log in a
call, and is very useful for debugging real calls. Is it really worth getting
rid of?

+  std::stringstream ss;
+  ss << "{ssrc: " << ssrc;
+  ss << ", extensions: [";
+  for (size_t i = 0; i < extensions.size(); ++i) {

[tommi, 2015/10/14 13:21:18]

use range based loop?

[the sun, 2015/10/14 14:25:24]

With the conditional "ss << ", ";" below, I don't see much point. Need to keep
an iterator or an index anyway.

[tommi, 2015/10/14 14:49:28]

Acknowledged.

+    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) {
+  RTC_DCHECK(config.voe_channel_id != -1);
+}
+
+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