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(); }

[tommi, 2015/10/16 11:08:43]

what about overloading operator -> instead?
looks like that would change call()->Foo() to call->Foo()

[the sun, 2015/10/16 11:30:01]

correct; silly moi

+
+ private:
+  rtc::scoped_ptr<webrtc::Call> call_;
+};

[stefan-webrtc, 2015/10/16 09:25:41]

I haven't seen this pattern used in many other unittests before. Typically
people use test fixtures instead. Would that not make sense here?

[The Sun (google.com), 2015/10/16 09:41:46]

gtest supports tests with/without fixtures.

Generally I think tests that don't use fixtures is more straightforward because
there is less happening behind the scenes. We should use less of them and
instead make it easy to set up the required state inline.

[stefan-webrtc, 2015/10/16 09:46:14]

Ok, I'm not going to block the CL on this, I was mostly curious as of why it was
done differently.

+}  // 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) {

[pbos-webrtc, 2015/10/16 10:48:29]

Is there any point to this pattern?

Also, can you test that you can re-add these streams again (e.g. that the SSRC
isn't "taken" permanently)?

[The Sun (google.com), 2015/10/16 11:01:57]

1. No. The intent is to add a bunch of streams and remove them in a different
order.

2. Ok.

+    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()) {

[pbos-webrtc, 2015/10/16 10:48:29]

for (AudioSendStream* stream : streams), clear after or skip pop_front()
completely.

[The Sun (google.com), 2015/10/16 11:01:57]

Done.

+    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()) {

[pbos-webrtc, 2015/10/16 10:48:29]

same

[The Sun (google.com), 2015/10/16 11:01:57]

Done.

+    call()->DestroyAudioReceiveStream(streams.front());
+    streams.pop_front();
+  }
+}
+}  // namespace webrtc