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https://github.com/mollyim/webrtc.git
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b9b146c9fe
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
55 lines
1.9 KiB
C++
55 lines
1.9 KiB
C++
/*
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* Copyright (c) 2017 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 "call/rtp_rtcp_demuxer_helper.h"
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#include "modules/rtp_rtcp/source/byte_io.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
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namespace webrtc {
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absl::optional<uint32_t> ParseRtcpPacketSenderSsrc(
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rtc::ArrayView<const uint8_t> packet) {
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rtcp::CommonHeader header;
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for (const uint8_t* next_packet = packet.begin(); next_packet < packet.end();
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next_packet = header.NextPacket()) {
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if (!header.Parse(next_packet, packet.end() - next_packet)) {
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return absl::nullopt;
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}
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switch (header.type()) {
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case rtcp::Bye::kPacketType:
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case rtcp::ExtendedReports::kPacketType:
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case rtcp::Psfb::kPacketType:
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case rtcp::ReceiverReport::kPacketType:
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case rtcp::Rtpfb::kPacketType:
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case rtcp::SenderReport::kPacketType: {
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// Sender SSRC at the beginning of the RTCP payload.
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if (header.payload_size_bytes() >= sizeof(uint32_t)) {
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const uint32_t ssrc_sender =
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ByteReader<uint32_t>::ReadBigEndian(header.payload());
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return ssrc_sender;
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} else {
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return absl::nullopt;
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}
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}
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}
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}
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return absl::nullopt;
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}
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} // namespace webrtc
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