56124bd158
The purpose with this CL is to be able to send video codec specific information down to RTPPayloadRegistry. We already do this for audio with explicit arguments for e.g. number of channels. Instead of extracting the arguments from webrtc::CodecInst (audio) and webrtc::VideoCodec, this CL sends the types unmodified all the way down to RTPPayloadRegistry. This CL does not contain any functional changes, and is just a preparation for future CL:s. In the dependent CL https://codereview.webrtc.org/2524923002/, RTPPayloadStrategy is removed. RTPPayloadStrategy previously handled audio/video specific aspects of payload handling. After this CL, we will know if we get audio or video codecs without any dependency injection, since we have different functions with different signatures for audio vs video. BUG=webrtc:6743 TBR=mflodman Review-Url: https://codereview.webrtc.org/2523843002 Cr-Commit-Position: refs/heads/master@{#15231}
100 lines
4.1 KiB
C++
100 lines
4.1 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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.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_utility.h"
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#include "webrtc/typedefs.h"
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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. The is_first_packet argument is true
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// if this packet is either the first packet ever or the first in its frame.
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virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
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const PayloadUnion& specific_payload,
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bool is_red,
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const uint8_t* payload,
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size_t payload_length,
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int64_t timestamp_ms,
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bool is_first_packet) = 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(const CodecInst& audio_codec) = 0;
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// Invokes the OnInitializeDecoder callback in a media-specific way.
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virtual int32_t InvokeOnInitializeDecoder(
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RtpFeedback* callback,
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int8_t payload_type,
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const PayloadUnion& specific_payload) const = 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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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 // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
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