mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 05:40:42 +01:00
bc959b61b3
This denies the ability to request RTP data channels to callers. Later CLs will rip out the actual code for creating these channels. Bug: chromium:928706 Change-Id: Ibb54197f192f567984a348f1539c26be120903f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177901 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33740}
73 lines
2.9 KiB
C++
73 lines
2.9 KiB
C++
/*
|
|
* Copyright 2016 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 "sdk/media_constraints.h"
|
|
|
|
#include "test/gtest.h"
|
|
|
|
namespace webrtc {
|
|
|
|
namespace {
|
|
|
|
// Checks all settings touched by CopyConstraintsIntoRtcConfiguration,
|
|
// plus audio_jitter_buffer_max_packets.
|
|
bool Matches(const PeerConnectionInterface::RTCConfiguration& a,
|
|
const PeerConnectionInterface::RTCConfiguration& b) {
|
|
return a.disable_ipv6 == b.disable_ipv6 &&
|
|
a.audio_jitter_buffer_max_packets ==
|
|
b.audio_jitter_buffer_max_packets &&
|
|
a.screencast_min_bitrate == b.screencast_min_bitrate &&
|
|
a.combined_audio_video_bwe == b.combined_audio_video_bwe &&
|
|
a.enable_dtls_srtp == b.enable_dtls_srtp &&
|
|
a.media_config == b.media_config;
|
|
}
|
|
|
|
TEST(MediaConstraints, CopyConstraintsIntoRtcConfiguration) {
|
|
const MediaConstraints constraints_empty;
|
|
PeerConnectionInterface::RTCConfiguration old_configuration;
|
|
PeerConnectionInterface::RTCConfiguration configuration;
|
|
|
|
CopyConstraintsIntoRtcConfiguration(&constraints_empty, &configuration);
|
|
EXPECT_TRUE(Matches(old_configuration, configuration));
|
|
|
|
const MediaConstraints constraits_enable_ipv6(
|
|
{MediaConstraints::Constraint(MediaConstraints::kEnableIPv6, "true")},
|
|
{});
|
|
CopyConstraintsIntoRtcConfiguration(&constraits_enable_ipv6, &configuration);
|
|
EXPECT_FALSE(configuration.disable_ipv6);
|
|
const MediaConstraints constraints_disable_ipv6(
|
|
{MediaConstraints::Constraint(MediaConstraints::kEnableIPv6, "false")},
|
|
{});
|
|
CopyConstraintsIntoRtcConfiguration(&constraints_disable_ipv6,
|
|
&configuration);
|
|
EXPECT_TRUE(configuration.disable_ipv6);
|
|
|
|
const MediaConstraints constraints_screencast(
|
|
{MediaConstraints::Constraint(MediaConstraints::kScreencastMinBitrate,
|
|
"27")},
|
|
{});
|
|
CopyConstraintsIntoRtcConfiguration(&constraints_screencast, &configuration);
|
|
EXPECT_TRUE(configuration.screencast_min_bitrate);
|
|
EXPECT_EQ(27, *(configuration.screencast_min_bitrate));
|
|
|
|
// An empty set of constraints will not overwrite
|
|
// values that are already present.
|
|
configuration = old_configuration;
|
|
configuration.enable_dtls_srtp = true;
|
|
configuration.audio_jitter_buffer_max_packets = 34;
|
|
CopyConstraintsIntoRtcConfiguration(&constraints_empty, &configuration);
|
|
EXPECT_EQ(34, configuration.audio_jitter_buffer_max_packets);
|
|
ASSERT_TRUE(configuration.enable_dtls_srtp);
|
|
EXPECT_TRUE(*(configuration.enable_dtls_srtp));
|
|
}
|
|
|
|
} // namespace
|
|
|
|
} // namespace webrtc
|