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}
100 lines
3.6 KiB
C++
100 lines
3.6 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_SOURCE_RTP_RECEIVER_IMPL_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.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_receiver_strategy.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 RtpReceiverImpl : public RtpReceiver {
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public:
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// Callbacks passed in here may not be NULL (use Null Object callbacks if you
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// want callbacks to do nothing). This class takes ownership of the media
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// receiver but nothing else.
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RtpReceiverImpl(Clock* clock,
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RtpAudioFeedback* incoming_audio_messages_callback,
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RtpFeedback* incoming_messages_callback,
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RTPPayloadRegistry* rtp_payload_registry,
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RTPReceiverStrategy* rtp_media_receiver);
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virtual ~RtpReceiverImpl();
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int32_t RegisterReceivePayload(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) override;
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int32_t DeRegisterReceivePayload(const int8_t payload_type) override;
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bool IncomingRtpPacket(const RTPHeader& rtp_header,
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const uint8_t* payload,
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size_t payload_length,
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PayloadUnion payload_specific,
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bool in_order) override;
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NACKMethod NACK() const override;
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// Turn negative acknowledgement requests on/off.
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void SetNACKStatus(const NACKMethod method) override;
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// Returns the last received timestamp.
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bool Timestamp(uint32_t* timestamp) const override;
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bool LastReceivedTimeMs(int64_t* receive_time_ms) const override;
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uint32_t SSRC() const override;
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int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const override;
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int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
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TelephoneEventHandler* GetTelephoneEventHandler() override;
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private:
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bool HaveReceivedFrame() const;
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void CheckSSRCChanged(const RTPHeader& rtp_header);
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void CheckCSRC(const WebRtcRTPHeader& rtp_header);
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int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
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const int8_t first_payload_byte,
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bool& is_red,
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PayloadUnion* payload);
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Clock* clock_;
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RTPPayloadRegistry* rtp_payload_registry_;
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rtc::scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_;
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RtpFeedback* cb_rtp_feedback_;
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rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtp_receiver_;
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int64_t last_receive_time_;
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size_t last_received_payload_length_;
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// SSRCs.
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uint32_t ssrc_;
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uint8_t num_csrcs_;
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uint32_t current_remote_csrc_[kRtpCsrcSize];
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uint32_t last_received_timestamp_;
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int64_t last_received_frame_time_ms_;
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uint16_t last_received_sequence_number_;
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NACKMethod nack_method_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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