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d264df587f
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated using script:
#!/bin/bash
dir=modules/rtp_rtcp
find $dir -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $dir -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: Ife720849709959046329c1c9faa3f31aa13274dc
Reviewed-on: https://webrtc-review.googlesource.com/83584
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23624}
108 lines
3.9 KiB
C++
108 lines
3.9 KiB
C++
/*
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* Copyright (c) 2013 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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#ifndef MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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#include <list>
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#include <memory>
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#include <unordered_map>
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#include <vector>
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#include "absl/types/optional.h"
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#include "modules/rtp_rtcp/include/rtp_receiver.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
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#include "rtc_base/criticalsection.h"
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#include "typedefs.h" // NOLINT(build/include)
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namespace webrtc {
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class RtpReceiverImpl : public RtpReceiver {
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public:
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// Callbacks passed in here may not be NULL (use Null Object callbacks if you
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// want callbacks to do nothing). This class takes ownership of the media
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// receiver but nothing else.
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RtpReceiverImpl(Clock* clock,
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RTPPayloadRegistry* rtp_payload_registry,
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RTPReceiverStrategy* rtp_media_receiver);
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~RtpReceiverImpl() override;
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int32_t RegisterReceivePayload(int payload_type,
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const SdpAudioFormat& audio_format) override;
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int32_t RegisterReceivePayload(const VideoCodec& video_codec) override;
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int32_t DeRegisterReceivePayload(const int8_t payload_type) override;
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bool IncomingRtpPacket(const RTPHeader& rtp_header,
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const uint8_t* payload,
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size_t payload_length,
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PayloadUnion payload_specific) override;
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bool GetLatestTimestamps(uint32_t* timestamp,
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int64_t* receive_time_ms) const override;
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uint32_t SSRC() const override;
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int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const override;
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int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
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TelephoneEventHandler* GetTelephoneEventHandler() override;
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std::vector<RtpSource> GetSources() const override;
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const std::vector<RtpSource>& ssrc_sources_for_testing() const {
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return ssrc_sources_;
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}
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const std::list<RtpSource>& csrc_sources_for_testing() const {
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return csrc_sources_;
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}
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private:
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void CheckSSRCChanged(const RTPHeader& rtp_header);
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void CheckCSRC(const WebRtcRTPHeader& rtp_header);
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int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
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PayloadUnion* payload);
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void UpdateSources(const absl::optional<uint8_t>& ssrc_audio_level);
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void RemoveOutdatedSources(int64_t now_ms);
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Clock* clock_;
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rtc::CriticalSection critical_section_rtp_receiver_;
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RTPPayloadRegistry* const rtp_payload_registry_
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RTC_PT_GUARDED_BY(critical_section_rtp_receiver_);
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const std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_;
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// SSRCs.
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uint32_t ssrc_ RTC_GUARDED_BY(critical_section_rtp_receiver_);
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uint8_t num_csrcs_ RTC_GUARDED_BY(critical_section_rtp_receiver_);
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uint32_t current_remote_csrc_[kRtpCsrcSize] RTC_GUARDED_BY(
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critical_section_rtp_receiver_);
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// Sequence number and timestamps for the latest in-order packet.
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absl::optional<uint16_t> last_received_sequence_number_
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RTC_GUARDED_BY(critical_section_rtp_receiver_);
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uint32_t last_received_timestamp_
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RTC_GUARDED_BY(critical_section_rtp_receiver_);
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int64_t last_received_frame_time_ms_
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RTC_GUARDED_BY(critical_section_rtp_receiver_);
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std::unordered_map<uint32_t, std::list<RtpSource>::iterator>
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iterator_by_csrc_;
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// The RtpSource objects are sorted chronologically.
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std::list<RtpSource> csrc_sources_;
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std::vector<RtpSource> ssrc_sources_;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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