
Clean some logs and add asserts in the way. BUG=3153 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d
169 lines
5.4 KiB
C++
169 lines
5.4 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_INTERFACE_RTP_PAYLOAD_REGISTRY_H_
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#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_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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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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namespace webrtc {
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// This strategy deals with the audio/video-specific aspects
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// of payload handling.
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class RTPPayloadStrategy {
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public:
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virtual ~RTPPayloadStrategy() {}
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virtual bool CodecsMustBeUnique() const = 0;
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virtual bool PayloadIsCompatible(
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const ModuleRTPUtility::Payload& payload,
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const uint32_t frequency,
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const uint8_t channels,
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const uint32_t rate) const = 0;
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virtual void UpdatePayloadRate(
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ModuleRTPUtility::Payload* payload,
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const uint32_t rate) const = 0;
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virtual ModuleRTPUtility::Payload* CreatePayloadType(
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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const int8_t payloadType,
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const uint32_t frequency,
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const uint8_t channels,
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const uint32_t rate) const = 0;
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virtual int GetPayloadTypeFrequency(
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const ModuleRTPUtility::Payload& payload) const = 0;
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static RTPPayloadStrategy* CreateStrategy(const bool handling_audio);
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protected:
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RTPPayloadStrategy() {}
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};
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class RTPPayloadRegistry {
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public:
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// The registry takes ownership of the strategy.
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RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy);
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~RTPPayloadRegistry();
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int32_t RegisterReceivePayload(
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const int8_t payload_type,
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const uint32_t frequency,
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const uint8_t channels,
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const uint32_t rate,
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bool* created_new_payload_type);
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int32_t DeRegisterReceivePayload(
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const int8_t payload_type);
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int32_t ReceivePayloadType(
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const uint32_t frequency,
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const uint8_t channels,
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const uint32_t rate,
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int8_t* payload_type) const;
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void SetRtxStatus(bool enable, uint32_t ssrc);
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bool RtxEnabled() const;
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void SetRtxPayloadType(int 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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int* packet_length,
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uint32_t original_ssrc,
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const RTPHeader& header) const;
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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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bool PayloadTypeToPayload(
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const uint8_t payload_type,
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ModuleRTPUtility::Payload*& payload) const;
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void ResetLastReceivedPayloadTypes() {
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CriticalSectionScoped cs(crit_sect_.get());
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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 {
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CriticalSectionScoped cs(crit_sect_.get());
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return red_payload_type_;
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}
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int8_t ulpfec_payload_type() const {
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CriticalSectionScoped cs(crit_sect_.get());
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return ulpfec_payload_type_;
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}
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int8_t last_received_payload_type() const {
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CriticalSectionScoped cs(crit_sect_.get());
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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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CriticalSectionScoped cs(crit_sect_.get());
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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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CriticalSectionScoped cs(crit_sect_.get());
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return last_received_media_payload_type_;
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};
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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 char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const size_t payload_name_length,
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const uint32_t frequency,
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const uint8_t channels,
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const uint32_t rate);
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bool IsRtxInternal(const RTPHeader& header) const;
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scoped_ptr<CriticalSectionWrapper> crit_sect_;
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ModuleRTPUtility::PayloadTypeMap payload_type_map_;
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scoped_ptr<RTPPayloadStrategy> rtp_payload_strategy_;
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int8_t red_payload_type_;
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int8_t ulpfec_payload_type_;
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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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int8_t payload_type_rtx_;
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uint32_t ssrc_rtx_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_
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