f3feeffe03
Reason for revert: Downstream code has been updated. Original issue's description: > Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ ) > > Reason for revert: > Breaks downstream projects. > > Original issue's description: > > Remove RTPPayloadStrategy and simplify RTPPayloadRegistry > > > > This CL removes RTPPayloadStrategy that is currently used to handle > > audio/video specific aspects of payload handling. Instead, the audio and > > video specific aspects will now have different functions, with linear > > code flow. > > > > This CL does not contain any functional changes, and is just a > > preparation for future CL:s. > > > > The main purpose with this CL is to add this function: > > bool PayloadIsCompatible(const RtpUtility::Payload& payload, > > const webrtc::VideoCodec& video_codec); > > that can easily be extended in a future CL to look at video codec > > specific information. > > > > BUG=webrtc:6743 > > > > Committed: https://crrev.com/b881254dc86d2cc80a52e08155433458be002166 > > Cr-Commit-Position: refs/heads/master@{#15232} > > TBR=danilchap@webrtc.org,solenberg@webrtc.org,mflodman@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6743 > > Committed: https://crrev.com/33c81d05613f45f65ee17224ed381c6cdd1c6c6f > Cr-Commit-Position: refs/heads/master@{#15234} TBR=danilchap@webrtc.org,solenberg@webrtc.org,mflodman@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6743 Review-Url: https://codereview.webrtc.org/2531043002 Cr-Commit-Position: refs/heads/master@{#15245}
90 lines
3.1 KiB
C++
90 lines
3.1 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_SOURCE_RTP_RECEIVER_IMPL_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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#include <memory>
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
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#include "webrtc/typedefs.h"
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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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RtpFeedback* incoming_messages_callback,
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RTPPayloadRegistry* rtp_payload_registry,
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RTPReceiverStrategy* rtp_media_receiver);
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virtual ~RtpReceiverImpl();
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int32_t RegisterReceivePayload(const CodecInst& audio_codec) 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,
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bool in_order) override;
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// Returns the last received timestamp.
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bool Timestamp(uint32_t* timestamp) const override;
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bool LastReceivedTimeMs(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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private:
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bool HaveReceivedFrame() const;
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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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const int8_t first_payload_byte,
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bool* is_red,
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PayloadUnion* payload);
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Clock* clock_;
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RTPPayloadRegistry* rtp_payload_registry_;
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std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_;
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RtpFeedback* cb_rtp_feedback_;
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rtc::CriticalSection critical_section_rtp_receiver_;
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int64_t last_receive_time_;
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size_t last_received_payload_length_;
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// SSRCs.
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uint32_t ssrc_;
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uint8_t num_csrcs_;
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uint32_t current_remote_csrc_[kRtpCsrcSize];
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uint32_t last_received_timestamp_;
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int64_t last_received_frame_time_ms_;
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uint16_t last_received_sequence_number_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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