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}
130 lines
4.5 KiB
C++
130 lines
4.5 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_SENDER_VIDEO_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
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#include <list>
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/common_types.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/bitrate.h"
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#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
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#include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
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#include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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struct RtpPacket;
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class RTPSenderVideo {
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public:
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RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender);
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virtual ~RTPSenderVideo();
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virtual RtpVideoCodecTypes VideoCodecType() const;
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size_t FECPacketOverhead() const;
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static RtpUtility::Payload* CreateVideoPayload(
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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const int8_t payloadType,
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const uint32_t maxBitRate);
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int32_t SendVideo(const RtpVideoCodecTypes videoType,
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const FrameType frameType,
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const int8_t payloadType,
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const uint32_t captureTimeStamp,
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int64_t capture_time_ms,
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const uint8_t* payloadData,
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const size_t payloadSize,
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const RTPFragmentationHeader* fragmentation,
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const RTPVideoHeader* rtpHdr);
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int32_t SendRTPIntraRequest();
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void SetVideoCodecType(RtpVideoCodecTypes type);
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void SetMaxConfiguredBitrateVideo(const uint32_t maxBitrate);
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uint32_t MaxConfiguredBitrateVideo() const;
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// FEC
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void SetGenericFECStatus(const bool enable,
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const uint8_t payloadTypeRED,
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const uint8_t payloadTypeFEC);
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void GenericFECStatus(bool& enable,
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uint8_t& payloadTypeRED,
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uint8_t& payloadTypeFEC) const;
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void SetFecParameters(const FecProtectionParams* delta_params,
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const FecProtectionParams* key_params);
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void ProcessBitrate();
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uint32_t VideoBitrateSent() const;
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uint32_t FecOverheadRate() const;
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int SelectiveRetransmissions() const;
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void SetSelectiveRetransmissions(uint8_t settings);
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private:
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void SendVideoPacket(uint8_t* dataBuffer,
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const size_t payloadLength,
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const size_t rtpHeaderLength,
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uint16_t seq_num,
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const uint32_t capture_timestamp,
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int64_t capture_time_ms,
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StorageType storage);
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void SendVideoPacketAsRed(uint8_t* dataBuffer,
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const size_t payloadLength,
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const size_t rtpHeaderLength,
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uint16_t video_seq_num,
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const uint32_t capture_timestamp,
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int64_t capture_time_ms,
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StorageType media_packet_storage,
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bool protect);
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RTPSenderInterface& _rtpSender;
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// Should never be held when calling out of this class.
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const rtc::scoped_ptr<CriticalSectionWrapper> crit_;
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RtpVideoCodecTypes _videoType;
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uint32_t _maxBitrate;
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int32_t _retransmissionSettings GUARDED_BY(crit_);
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// FEC
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ForwardErrorCorrection _fec;
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bool _fecEnabled GUARDED_BY(crit_);
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int8_t _payloadTypeRED GUARDED_BY(crit_);
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int8_t _payloadTypeFEC GUARDED_BY(crit_);
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FecProtectionParams delta_fec_params_ GUARDED_BY(crit_);
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FecProtectionParams key_fec_params_ GUARDED_BY(crit_);
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ProducerFec producer_fec_ GUARDED_BY(crit_);
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// Bitrate used for FEC payload, RED headers, RTP headers for FEC packets
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// and any padding overhead.
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Bitrate _fecOverheadRate;
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// Bitrate used for video payload and RTP headers
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Bitrate _videoBitrate;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
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