Add AudioSendStream to Call.

webrtc/call/call_unittest.cc

Index: webrtc/call/call_unittest.cc
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..dc4d9098fce449c1907b80e9fa409ecc4afbb9d8
--- /dev/null
+++ b/webrtc/call/call_unittest.cc
@@ -0,0 +1,100 @@
+/*
+ *  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 (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);
+    }
+  }
+  while (!streams.empty()) {
+    call()->DestroyAudioSendStream(streams.front());
+    streams.pop_front();
+  }
+}
+
+TEST(CallTest, CreateDestroy_AudioReceiveStreams) {
+  CallHelper call;
+  AudioReceiveStream::Config config;
+  config.voe_channel_id = 123;
+  std::list<AudioReceiveStream*> streams;
+  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);
+    }
+  }
+  while (!streams.empty()) {
+    call()->DestroyAudioReceiveStream(streams.front());
+    streams.pop_front();
+  }
+}
+}  // namespace webrtc

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

Do you want to add some of these for Video equivalents?

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

Makes more sense if you do that.