Add AudioSendStream to Call.

webrtc/audio/audio_send_stream_unittest.cc

Index: webrtc/audio/audio_send_stream_unittest.cc
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..61c8bbbd379453d4738dbbdd13d2275d12f1b89e
--- /dev/null
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -0,0 +1,34 @@
+/*
+ *  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 "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/audio/audio_send_stream.h"
+
+namespace webrtc {
+
+TEST(AudioSendStreamTest, ConfigToString) {
+  const int kAbsSendTimeId = 3;
+  AudioSendStream::Config config(nullptr);
+  config.rtp.ssrc = 1234;
+  config.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
+  config.voe_channel_id = 1;
+  config.cng_payload_type = 42;
+  config.red_payload_type = 17;
+  EXPECT_LT(0u, config.ToString().size());

[pbos-webrtc, 2015/10/15 15:32:12]

pref GT(size, 0u)

[the sun, 2015/10/16 08:47:05]

Done.

+}
+
+TEST(AudioSendStreamTest, ConstructDestruct) {
+  AudioSendStream::Config config(nullptr);
+  config.voe_channel_id = 1;
+  internal::AudioSendStream send_stream(config);
+}
+}  // namespace webrtc