83d3ec177c
I need to replace the audio part of PayloadUnion with SdpAudioFormat, but that's a non-trivially-deletable class and those just don't work well in unions, especially unions that don't have a discriminator that says which member is currently active. This CL converts the union to a class, adds a discriminator, and provides accessor functions. CL #2 in the series will change all outsiders to use the accessors instead of the public member variables directly, and CL #3 will remove the public member variables. (It needs to be done in separate steps like this because PayloadUnion is unfortunately part of the API, and just changing it all in one go would break users.) BUG=webrtc:8159 Change-Id: I38c44bbb21a2d38600cff59bf37d8d47dfdbce21 Reviewed-on: https://webrtc-review.googlesource.com/4340 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20025}
98 lines
4.0 KiB
C++
98 lines
4.0 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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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) = 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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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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