ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
107 lines
4.4 KiB
C++
107 lines
4.4 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/scoped_ptr.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/system_wrappers/include/critical_section_wrapper.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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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(
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RtpData* data_callback,
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RtpAudioFeedback* incoming_messages_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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// Retrieves the last known applicable frequency.
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virtual int GetPayloadTypeFrequency() const = 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 payload type in
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// the payload registry.
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virtual int32_t OnNewPayloadTypeCreated(
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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int8_t payloadType,
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uint32_t frequency) = 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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RTPReceiverStrategy(RtpData* data_callback);
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rtc::scoped_ptr<CriticalSectionWrapper> 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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