webrtc/api/rtp_transceiver_interface.cc
Harald Alvestrand 6060df5948 Reland "Implement transceiver.stop()"
This is a reland of 11dc6571cb

One fix that makes Web Platform Tests pass in debug mode is applied.

Original change's description:
> Implement transceiver.stop()
>
> This adds RtpTransceiver.StopStandard(), which behaves according to
> the specification at
> https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
>
> It modifies RTCPeerConnection.getTransceivers() to return only
> transceivers that have not been stopped.
>
> Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
>
> Bug: chromium:980879
> Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31893}

Bug: chromium:980879
Change-Id: Ide31d929ac5ea118d83fdf6a35a592af23f7dfa7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181263
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31907}
2020-08-11 10:46:23 +00:00

80 lines
2.2 KiB
C++

/*
* Copyright 2018 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 "api/rtp_transceiver_interface.h"
#include "rtc_base/checks.h"
namespace webrtc {
RtpTransceiverInit::RtpTransceiverInit() = default;
RtpTransceiverInit::RtpTransceiverInit(const RtpTransceiverInit& rhs) = default;
RtpTransceiverInit::~RtpTransceiverInit() = default;
absl::optional<RtpTransceiverDirection>
RtpTransceiverInterface::fired_direction() const {
return absl::nullopt;
}
bool RtpTransceiverInterface::stopping() const {
return false;
}
void RtpTransceiverInterface::Stop() {
StopInternal();
}
RTCError RtpTransceiverInterface::StopStandard() {
RTC_NOTREACHED() << "DEBUG: RtpTransceiverInterface::StopStandard called";
return RTCError::OK();
}
void RtpTransceiverInterface::StopInternal() {
RTC_NOTREACHED() << "DEBUG: RtpTransceiverInterface::StopInternal called";
}
RTCError RtpTransceiverInterface::SetCodecPreferences(
rtc::ArrayView<RtpCodecCapability>) {
RTC_NOTREACHED() << "Not implemented";
return {};
}
std::vector<RtpCodecCapability> RtpTransceiverInterface::codec_preferences()
const {
return {};
}
std::vector<RtpHeaderExtensionCapability>
RtpTransceiverInterface::HeaderExtensionsToOffer() const {
return {};
}
webrtc::RTCError RtpTransceiverInterface::SetOfferedRtpHeaderExtensions(
rtc::ArrayView<const RtpHeaderExtensionCapability>
header_extensions_to_offer) {
return webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_OPERATION);
}
// TODO(bugs.webrtc.org/11839) Remove default implementations when clients
// are updated.
void RtpTransceiverInterface::SetDirection(
RtpTransceiverDirection new_direction) {
SetDirectionWithError(new_direction);
}
RTCError RtpTransceiverInterface::SetDirectionWithError(
RtpTransceiverDirection new_direction) {
RTC_NOTREACHED() << "Default implementation called";
return RTCError::OK();
}
} // namespace webrtc