6f727da62b
This reverts commit 71db9acc4019b8c9c13b14e6a022cbb3b4255b09. Reason for revert: breaks downstream project. Reason for force push: win bot broken. Original change's description: > RtpTransceiverInterface: introduce SetOfferedRtpHeaderExtensions. > > This change adds exposure of a new transceiver method for > modifying the extensions offered in the next SDP negotiation, > following spec details in https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface. > > Features: > - The interface allows to control the negotiated direction as > per https://tools.ietf.org/html/rfc5285#page-7. > - The interface allows to remove an extension from SDP > negotiation by modifying the direction to > RtpTransceiverDirection::kStopped. > > Note: support for signalling directionality of header extensions > in the SDP isn't implemented yet. > > https://chromestatus.com/feature/5680189201711104. > Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk > > Bug: chromium:1051821 > Change-Id: Iaabc34446f038c46d93c442e90c2a77f77d542d4 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176408 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Markus Handell <handellm@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31487} TBR=hta@webrtc.org,handellm@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. No-Try: true Bug: chromium:1051821 Change-Id: I70e1a07225d7eeec7480fa5577d8ff647eba6902 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177103 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31516}
105 lines
3.7 KiB
C++
105 lines
3.7 KiB
C++
/*
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* Copyright 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// This file contains tests for |RtpTransceiver|.
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#include "pc/rtp_transceiver.h"
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#include <memory>
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#include "media/base/fake_media_engine.h"
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#include "pc/test/mock_channel_interface.h"
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#include "pc/test/mock_rtp_receiver_internal.h"
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#include "pc/test/mock_rtp_sender_internal.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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using ::testing::ElementsAre;
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using ::testing::Eq;
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using ::testing::Field;
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using ::testing::Not;
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using ::testing::Return;
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using ::testing::ReturnRef;
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namespace webrtc {
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// Checks that a channel cannot be set on a stopped |RtpTransceiver|.
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TEST(RtpTransceiverTest, CannotSetChannelOnStoppedTransceiver) {
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RtpTransceiver transceiver(cricket::MediaType::MEDIA_TYPE_AUDIO);
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cricket::MockChannelInterface channel1;
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sigslot::signal1<cricket::ChannelInterface*> signal;
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EXPECT_CALL(channel1, media_type())
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.WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
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EXPECT_CALL(channel1, SignalFirstPacketReceived())
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.WillRepeatedly(ReturnRef(signal));
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transceiver.SetChannel(&channel1);
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EXPECT_EQ(&channel1, transceiver.channel());
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// Stop the transceiver.
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transceiver.Stop();
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EXPECT_EQ(&channel1, transceiver.channel());
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cricket::MockChannelInterface channel2;
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EXPECT_CALL(channel2, media_type())
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.WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
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// Channel can no longer be set, so this call should be a no-op.
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transceiver.SetChannel(&channel2);
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EXPECT_EQ(&channel1, transceiver.channel());
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}
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// Checks that a channel can be unset on a stopped |RtpTransceiver|
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TEST(RtpTransceiverTest, CanUnsetChannelOnStoppedTransceiver) {
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RtpTransceiver transceiver(cricket::MediaType::MEDIA_TYPE_VIDEO);
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cricket::MockChannelInterface channel;
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sigslot::signal1<cricket::ChannelInterface*> signal;
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EXPECT_CALL(channel, media_type())
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.WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_VIDEO));
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EXPECT_CALL(channel, SignalFirstPacketReceived())
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.WillRepeatedly(ReturnRef(signal));
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transceiver.SetChannel(&channel);
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EXPECT_EQ(&channel, transceiver.channel());
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// Stop the transceiver.
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transceiver.Stop();
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EXPECT_EQ(&channel, transceiver.channel());
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// Set the channel to |nullptr|.
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transceiver.SetChannel(nullptr);
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EXPECT_EQ(nullptr, transceiver.channel());
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}
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TEST(RtpTransceiverTest,
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InitsWithChannelManagerRtpHeaderExtensionCapabilities) {
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cricket::ChannelManager channel_manager(
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std::make_unique<cricket::FakeMediaEngine>(),
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std::make_unique<cricket::FakeDataEngine>(), rtc::Thread::Current(),
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rtc::Thread::Current());
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std::vector<RtpHeaderExtensionCapability> extensions({
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RtpHeaderExtensionCapability("uri1", 1,
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RtpTransceiverDirection::kSendRecv),
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RtpHeaderExtensionCapability("uri2", 2,
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RtpTransceiverDirection::kRecvOnly),
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});
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RtpTransceiver transceiver(
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RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
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rtc::Thread::Current(),
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new rtc::RefCountedObject<MockRtpSenderInternal>()),
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RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
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rtc::Thread::Current(),
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new rtc::RefCountedObject<MockRtpReceiverInternal>()),
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&channel_manager, extensions);
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EXPECT_EQ(transceiver.HeaderExtensionsToOffer(), extensions);
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}
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} // namespace webrtc
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