cd349d9743
Reason for revert:
Upstream fixes in place, should be OK now.
Original issue's description:
> Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ )
>
> Reason for revert:
> Breaks upstream code.
>
> Original issue's description:
> > Refactor NACK bitrate allocation
> >
> > Nack bitrate allocation should not be done on a per-rtp-module basis,
> > but rather shared bitrate pool per call. This CL moves allocation to the
> > pacer and cleans up a bunch if bitrate stats handling.
> >
> > BUG=
> > R=danilchap@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
> >
> > Committed: 5fc59e810b
>
> TBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/e5dd44101eca485f5ad12e5f7ce6f6b0d204116b
> Cr-Commit-Position: refs/heads/master@{#13417}
TBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=
Review-Url: https://codereview.webrtc.org/2146013002
Cr-Commit-Position: refs/heads/master@{#13465}
124 lines
4.4 KiB
C++
124 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_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/criticalsection.h"
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#include "webrtc/base/onetimeevent.h"
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#include "webrtc/base/rate_statistics.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/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 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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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* video_header);
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int32_t SendRTPIntraRequest();
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void SetVideoCodecType(RtpVideoCodecTypes type);
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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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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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Clock* const clock_;
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// Should never be held when calling out of this class.
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rtc::CriticalSection crit_;
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RtpVideoCodecTypes _videoType;
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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 fec_enabled_ GUARDED_BY(crit_);
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int8_t red_payload_type_ GUARDED_BY(crit_);
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int8_t fec_payload_type_ 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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rtc::CriticalSection stats_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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RateStatistics fec_bitrate_ GUARDED_BY(stats_crit_);
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// Bitrate used for video payload and RTP headers.
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RateStatistics video_bitrate_ GUARDED_BY(stats_crit_);
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OneTimeEvent first_frame_sent_;
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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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