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Remove MediaTransport from Call.

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There aren't any tests for this and the code isn't currently
active except for the fact that it adds complexity to the Call
class, synchronization into the active code path and makes future
improvements to the class more complex or impossible.

Bug: webrtc:9719
Change-Id: Ia41af0b2186b8a36ca70a07858990b6af7f3a5c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148078
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28807}
This commit is contained in:
Tommi
2019-08-08 12:27:53 +02:00
committed by Commit Bot
parent 44327c33ed
commit 78a7138600
8 changed files with 39 additions and 195 deletions
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26
call/call_unittest.cc
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@ -295,30 +295,4 @@ TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) {
EXPECT_EQ(rtp_state1.media_has_been_sent, rtp_state2.media_has_been_sent);
}
TEST(CallTest, RegisterMediaTransportBitrateCallbacksInCreateStream) {
CallHelper call;
MediaTransportSettings settings;
webrtc::FakeMediaTransport fake_media_transport(settings);
EXPECT_EQ(0, fake_media_transport.target_rate_observers_size());
// TODO(solenberg): This test shouldn't require a Transport, but currently
// RTCPSender requires one.
MockTransport send_transport;
AudioSendStream::Config config(&send_transport,
MediaTransportConfig(&fake_media_transport));
call->MediaTransportChange(&fake_media_transport);
AudioSendStream* stream = call->CreateAudioSendStream(config);
// We get 2 subscribers: one subscriber from call.cc, and one from
// ChannelSend.
EXPECT_EQ(2, fake_media_transport.target_rate_observers_size());
call->DestroyAudioSendStream(stream);
EXPECT_EQ(1, fake_media_transport.target_rate_observers_size());
call->MediaTransportChange(nullptr);
EXPECT_EQ(0, fake_media_transport.target_rate_observers_size());
}
} // namespace webrtc