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https://github.com/mollyim/webrtc.git
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5932fe1392
This changes adds exposure of a new transceiver method for accessing header extensions that have been negotiated, following spec details in https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface. The change contains unit tests testing the functionality. 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: If963beed37e96eed2dff3a2822db4e30caaea4a0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198126 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32860}
85 lines
2.3 KiB
C++
85 lines
2.3 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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#include "api/rtp_transceiver_interface.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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RtpTransceiverInit::RtpTransceiverInit() = default;
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RtpTransceiverInit::RtpTransceiverInit(const RtpTransceiverInit& rhs) = default;
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RtpTransceiverInit::~RtpTransceiverInit() = default;
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absl::optional<RtpTransceiverDirection>
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RtpTransceiverInterface::fired_direction() const {
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return absl::nullopt;
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}
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bool RtpTransceiverInterface::stopping() const {
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return false;
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}
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void RtpTransceiverInterface::Stop() {
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StopInternal();
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}
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RTCError RtpTransceiverInterface::StopStandard() {
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RTC_NOTREACHED() << "DEBUG: RtpTransceiverInterface::StopStandard called";
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return RTCError::OK();
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}
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void RtpTransceiverInterface::StopInternal() {
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RTC_NOTREACHED() << "DEBUG: RtpTransceiverInterface::StopInternal called";
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}
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RTCError RtpTransceiverInterface::SetCodecPreferences(
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rtc::ArrayView<RtpCodecCapability>) {
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RTC_NOTREACHED() << "Not implemented";
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return {};
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}
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std::vector<RtpCodecCapability> RtpTransceiverInterface::codec_preferences()
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const {
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return {};
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}
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std::vector<RtpHeaderExtensionCapability>
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RtpTransceiverInterface::HeaderExtensionsToOffer() const {
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return {};
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}
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webrtc::RTCError RtpTransceiverInterface::SetOfferedRtpHeaderExtensions(
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rtc::ArrayView<const RtpHeaderExtensionCapability>
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header_extensions_to_offer) {
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return webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_OPERATION);
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}
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std::vector<RtpHeaderExtensionCapability>
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RtpTransceiverInterface::HeaderExtensionsNegotiated() const {
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return {};
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}
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// TODO(bugs.webrtc.org/11839) Remove default implementations when clients
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// are updated.
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void RtpTransceiverInterface::SetDirection(
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RtpTransceiverDirection new_direction) {
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SetDirectionWithError(new_direction);
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}
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RTCError RtpTransceiverInterface::SetDirectionWithError(
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RtpTransceiverDirection new_direction) {
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RTC_NOTREACHED() << "Default implementation called";
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return RTCError::OK();
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}
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} // namespace webrtc
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