Reason for revert:
Makes perf and Chromium FYI bots unhappy.
Original issue's description:
> WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
>
> This removes one more place where we were unable to handle codecs not
> in the built-in set.
>
> BUG=webrtc:5805
>
> Review-Url: https://codereview.webrtc.org/2686043006
> Cr-Commit-Position: refs/heads/master@{#17370}
> Committed: 1724cfbdba
TBR=ossu@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2772043002
Cr-Commit-Position: refs/heads/master@{#17374}
153 lines
5.3 KiB
C++
153 lines
5.3 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 WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
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#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
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#include <map>
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#include <memory>
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#include <set>
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/deprecation.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
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namespace webrtc {
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struct CodecInst;
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class VideoCodec;
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// TODO(magjed): Remove once external code is updated.
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class RTPPayloadStrategy {
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public:
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static RTPPayloadStrategy* CreateStrategy(bool handling_audio) {
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return nullptr;
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}
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};
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class RTPPayloadRegistry {
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public:
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RTPPayloadRegistry();
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~RTPPayloadRegistry();
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// TODO(magjed): Remove once external code is updated.
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explicit RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy)
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: RTPPayloadRegistry() {}
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// TODO(magjed): Split RTPPayloadRegistry into separate Audio and Video class
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// and simplify the code. http://crbug/webrtc/6743.
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int32_t RegisterReceivePayload(const CodecInst& audio_codec,
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bool* created_new_payload_type);
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int32_t RegisterReceivePayload(const VideoCodec& video_codec);
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int32_t DeRegisterReceivePayload(int8_t payload_type);
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int32_t ReceivePayloadType(const CodecInst& audio_codec,
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int8_t* payload_type) const;
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int32_t ReceivePayloadType(const VideoCodec& video_codec,
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int8_t* payload_type) const;
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bool RtxEnabled() const;
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void SetRtxSsrc(uint32_t ssrc);
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bool GetRtxSsrc(uint32_t* ssrc) const;
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void SetRtxPayloadType(int payload_type, int associated_payload_type);
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bool IsRtx(const RTPHeader& header) const;
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bool RestoreOriginalPacket(uint8_t* restored_packet,
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const uint8_t* packet,
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size_t* packet_length,
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uint32_t original_ssrc,
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const RTPHeader& header);
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bool IsRed(const RTPHeader& header) const;
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// Returns true if the media of this RTP packet is encapsulated within an
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// extra header, such as RTX or RED.
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bool IsEncapsulated(const RTPHeader& header) const;
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bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const;
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int GetPayloadTypeFrequency(uint8_t payload_type) const;
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const RtpUtility::Payload* PayloadTypeToPayload(uint8_t payload_type) const;
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void ResetLastReceivedPayloadTypes() {
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rtc::CritScope cs(&crit_sect_);
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last_received_payload_type_ = -1;
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last_received_media_payload_type_ = -1;
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}
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// This sets the payload type of the packets being received from the network
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// on the media SSRC. For instance if packets are encapsulated with RED, this
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// payload type will be the RED payload type.
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void SetIncomingPayloadType(const RTPHeader& header);
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// Returns true if the new media payload type has not changed.
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bool ReportMediaPayloadType(uint8_t media_payload_type);
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int8_t red_payload_type() const { return GetPayloadTypeWithName("red"); }
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int8_t ulpfec_payload_type() const {
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return GetPayloadTypeWithName("ulpfec");
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}
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int8_t last_received_payload_type() const {
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rtc::CritScope cs(&crit_sect_);
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return last_received_payload_type_;
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}
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void set_last_received_payload_type(int8_t last_received_payload_type) {
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rtc::CritScope cs(&crit_sect_);
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last_received_payload_type_ = last_received_payload_type;
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}
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int8_t last_received_media_payload_type() const {
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rtc::CritScope cs(&crit_sect_);
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return last_received_media_payload_type_;
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}
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RTC_DEPRECATED void set_use_rtx_payload_mapping_on_restore(bool val) {}
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private:
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// Prunes the payload type map of the specific payload type, if it exists.
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void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
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const CodecInst& audio_codec);
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bool IsRtxInternal(const RTPHeader& header) const;
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// Returns the payload type for the payload with name |payload_name|, or -1 if
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// no such payload is registered.
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int8_t GetPayloadTypeWithName(const char* payload_name) const;
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rtc::CriticalSection crit_sect_;
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std::map<int, RtpUtility::Payload> payload_type_map_;
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int8_t incoming_payload_type_;
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int8_t last_received_payload_type_;
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int8_t last_received_media_payload_type_;
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bool rtx_;
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// Mapping rtx_payload_type_map_[rtx] = associated.
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std::map<int, int> rtx_payload_type_map_;
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uint32_t ssrc_rtx_;
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// Only warn once per payload type, if an RTX packet is received but
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// no associated payload type found in |rtx_payload_type_map_|.
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std::set<int> payload_types_with_suppressed_warnings_ GUARDED_BY(crit_sect_);
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// As a first step in splitting this class up in separate cases for audio and
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// video, DCHECK that no instance is used for both audio and video.
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#if RTC_DCHECK_IS_ON
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bool used_for_audio_ GUARDED_BY(crit_sect_) = false;
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bool used_for_video_ GUARDED_BY(crit_sect_) = false;
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#endif
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
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