Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.

AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1397123003

Cr-Commit-Position: refs/heads/master@{#10307}

webrtc/call/call_unittest.cc

@@ -0,0 +1,104 @@
+/*
+ *  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 <list>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/call.h"
+
+namespace {
+
+struct CallHelper {
+  CallHelper() {
+    webrtc::Call::Config config;
+    // TODO(solenberg): Fill in with VoiceEngine* etc.
+    call_.reset(webrtc::Call::Create(config));
+  }
+
+  webrtc::Call* operator->() { return call_.get(); }
+
+ private:
+  rtc::scoped_ptr<webrtc::Call> call_;
+};
+}  // namespace
+
+namespace webrtc {
+
+TEST(CallTest, ConstructDestruct) {
+  CallHelper call;
+}
+
+TEST(CallTest, CreateDestroy_AudioSendStream) {
+  CallHelper call;
+  AudioSendStream::Config config(nullptr);
+  config.rtp.ssrc = 42;
+  config.voe_channel_id = 123;
+  AudioSendStream* stream = call->CreateAudioSendStream(config);
+  EXPECT_NE(stream, nullptr);
+  call->DestroyAudioSendStream(stream);
+}
+
+TEST(CallTest, CreateDestroy_AudioReceiveStream) {
+  CallHelper call;
+  AudioReceiveStream::Config config;
+  config.rtp.remote_ssrc = 42;
+  config.voe_channel_id = 123;
+  AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
+  EXPECT_NE(stream, nullptr);
+  call->DestroyAudioReceiveStream(stream);
+}
+
+TEST(CallTest, CreateDestroy_AudioSendStreams) {
+  CallHelper call;
+  AudioSendStream::Config config(nullptr);
+  config.voe_channel_id = 123;
+  std::list<AudioSendStream*> streams;
+  for (int i = 0; i < 2; ++i) {
+    for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
+      config.rtp.ssrc = ssrc;
+      AudioSendStream* stream = call->CreateAudioSendStream(config);
+      EXPECT_NE(stream, nullptr);
+      if (ssrc & 1) {
+        streams.push_back(stream);
+      } else {
+        streams.push_front(stream);
+      }
+    }
+    for (auto s : streams) {
+      call->DestroyAudioSendStream(s);
+    }
+    streams.clear();
+  }
+}
+
+TEST(CallTest, CreateDestroy_AudioReceiveStreams) {
+  CallHelper call;
+  AudioReceiveStream::Config config;
+  config.voe_channel_id = 123;
+  std::list<AudioReceiveStream*> streams;
+  for (int i = 0; i < 2; ++i) {
+    for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
+      config.rtp.remote_ssrc = ssrc;
+      AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
+      EXPECT_NE(stream, nullptr);
+      if (ssrc & 1) {
+        streams.push_back(stream);
+      } else {
+        streams.push_front(stream);
+      }
+    }
+    for (auto s : streams) {
+      call->DestroyAudioReceiveStream(s);
+    }
+    streams.clear();
+  }
+}
+}  // namespace webrtc