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9cfb18c5b3
Bug: None Change-Id: I55e01e5ff1c54c76c43b378414a31fc43c9aa444 Reviewed-on: https://webrtc-review.googlesource.com/62142 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22561}
91 lines
3.7 KiB
C++
91 lines
3.7 KiB
C++
/*
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* Copyright (c) 2012 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_STRATEGY_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_utility.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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struct CodecInst;
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class TelephoneEventHandler;
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// This strategy deals with media-specific RTP packet processing.
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// This class is not thread-safe and must be protected by its caller.
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class RTPReceiverStrategy {
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public:
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static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback);
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static RTPReceiverStrategy* CreateAudioStrategy(RtpData* data_callback);
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virtual ~RTPReceiverStrategy();
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// Parses the RTP packet and calls the data callback with the payload data.
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// Implementations are encouraged to use the provided packet buffer and RTP
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// header as arguments to the callback; implementations are also allowed to
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// make changes in the data as necessary. The specific_payload argument
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// provides audio or video-specific data.
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virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
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const PayloadUnion& specific_payload,
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const uint8_t* payload,
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size_t payload_length,
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int64_t timestamp_ms) = 0;
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virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
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// Computes the current dead-or-alive state.
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virtual RTPAliveType ProcessDeadOrAlive(
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uint16_t last_payload_length) const = 0;
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// Returns true if we should report CSRC changes for this payload type.
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// TODO(phoglund): should move out of here along with other payload stuff.
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virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const = 0;
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// Notifies the strategy that we have created a new non-RED audio payload type
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// in the payload registry.
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virtual int32_t OnNewPayloadTypeCreated(
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int payload_type,
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const SdpAudioFormat& audio_format) = 0;
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// Checks if the payload type has changed, and returns whether we should
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// reset statistics and/or discard this packet.
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virtual void CheckPayloadChanged(int8_t payload_type,
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PayloadUnion* specific_payload,
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bool* should_discard_changes);
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virtual int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const;
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// Stores / retrieves the last media specific payload for later reference.
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void GetLastMediaSpecificPayload(PayloadUnion* payload) const;
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void SetLastMediaSpecificPayload(const PayloadUnion& payload);
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protected:
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// The data callback is where we should send received payload data.
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// See ParseRtpPacket. This class does not claim ownership of the callback.
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// Implementations must NOT hold any critical sections while calling the
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// callback.
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//
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// Note: Implementations may call the callback for other reasons than calls
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// to ParseRtpPacket, for instance if the implementation somehow recovers a
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// packet.
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explicit RTPReceiverStrategy(RtpData* data_callback);
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rtc::CriticalSection crit_sect_;
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rtc::Optional<PayloadUnion> last_payload_;
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RtpData* data_callback_;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
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