2017-04-05 03:02:20 -07:00
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/*
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* Copyright (c) 2014 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_API_VIDEO_CODECS_VIDEO_ENCODER_H_
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#define WEBRTC_API_VIDEO_CODECS_VIDEO_ENCODER_H_
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#include <memory>
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#include <string>
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#include <vector>
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Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ )
Reason for revert:
Downstream projects have been fixed.
Original issue's description:
> Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854873003/ )
>
> Reason for revert:
> More downstream breakage discovered.
>
> Original issue's description:
> > Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2854883002/ )
> >
> > Reason for revert:
> > Will make another attempt to track down and fix downstream projects.
> >
> > Original issue's description:
> > > Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2852303002/ )
> > >
> > > Reason for revert:
> > > Unfortunately, more downstream updates needed.
> > >
> > > Original issue's description:
> > > > Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2845333002/ )
> > > >
> > > > Reason for revert:
> > > > Downstream projects being updated.
> > > >
> > > > For Chrome, relanding depends on cl
> > > > https://codereview.chromium.org/2855783003/
> > > >
> > > > Original issue's description:
> > > > > Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ )
> > > > >
> > > > > Reason for revert:
> > > > > Broke Chrome build, see, e.g.,
> > > > > http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/16237
> > > > >
> > > > > Original issue's description:
> > > > > > Delete deprecated and transitional stuff related to video frame refactoring.
> > > > > >
> > > > > > BUG=webrtc:5880
> > > > > >
> > > > > > Review-Url: https://codereview.webrtc.org/2622263002
> > > > > > Cr-Commit-Position: refs/heads/master@{#17928}
> > > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/713a3bbcc74a4c1a1f54d6dbd0bdaa715f5c1ae1
> > > > >
> > > > > TBR=mflodman@webrtc.org,perkj@webrtc.org
> > > > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > > > NOPRESUBMIT=true
> > > > > NOTREECHECKS=true
> > > > > NOTRY=true
> > > > > BUG=webrtc:5880
> > > > >
> > > > > Review-Url: https://codereview.webrtc.org/2845333002
> > > > > Cr-Commit-Position: refs/heads/master@{#17929}
> > > > > Committed: https://chromium.googlesource.com/external/webrtc/+/aec49d2b49e1beceb286cbb29ad220aefd0ac648
> > > >
> > > > TBR=mflodman@webrtc.org,perkj@webrtc.org
> > > > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > > > BUG=webrtc:5880
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2852303002
> > > > Cr-Commit-Position: refs/heads/master@{#17974}
> > > > Committed: https://chromium.googlesource.com/external/webrtc/+/d71ebd70f6c40a37559a7ee2930002a804ad3f13
> > >
> > > TBR=mflodman@webrtc.org,perkj@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:5880
> > >
> > > Review-Url: https://codereview.webrtc.org/2854883002
> > > Cr-Commit-Position: refs/heads/master@{#17978}
> > > Committed: https://chromium.googlesource.com/external/webrtc/+/6e6a485a028799501ea4753ceefc811dd826f52f
> >
> > TBR=mflodman@webrtc.org,perkj@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> >
> > BUG=webrtc:5880
> >
> > Review-Url: https://codereview.webrtc.org/2854873003
> > Cr-Commit-Position: refs/heads/master@{#18006}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/3870a071c452dc2e782196bc5bba0375d019b653
>
> TBR=mflodman@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5880
>
> Review-Url: https://codereview.webrtc.org/2853383005
> Cr-Commit-Position: refs/heads/master@{#18008}
> Committed: https://chromium.googlesource.com/external/webrtc/+/61b22dde133aff91a0db6a628d87b2ea9134fa15
TBR=mflodman@webrtc.org,perkj@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5880
Review-Url: https://codereview.webrtc.org/2867703002
Cr-Commit-Position: refs/heads/master@{#18081}
2017-05-10 05:25:59 -07:00
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#include "webrtc/api/video/video_frame.h"
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2017-04-05 03:02:20 -07:00
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#include "webrtc/common_types.h"
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2017-05-15 02:42:11 -07:00
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#include "webrtc/common_video/include/video_frame.h"
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2017-07-06 19:44:34 +02:00
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#include "webrtc/rtc_base/checks.h"
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#include "webrtc/rtc_base/optional.h"
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2017-04-05 03:02:20 -07:00
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class RTPFragmentationHeader;
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// TODO(pbos): Expose these through a public (root) header or change these APIs.
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struct CodecSpecificInfo;
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class VideoCodec;
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class EncodedImageCallback {
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public:
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virtual ~EncodedImageCallback() {}
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struct Result {
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enum Error {
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OK,
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// Failed to send the packet.
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ERROR_SEND_FAILED,
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};
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Result(Error error) : error(error) {}
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Result(Error error, uint32_t frame_id) : error(error), frame_id(frame_id) {}
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Error error;
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// Frame ID assigned to the frame. The frame ID should be the same as the ID
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// seen by the receiver for this frame. RTP timestamp of the frame is used
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// as frame ID when RTP is used to send video. Must be used only when
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// error=OK.
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uint32_t frame_id = 0;
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// Tells the encoder that the next frame is should be dropped.
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bool drop_next_frame = false;
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};
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// Callback function which is called when an image has been encoded.
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virtual Result OnEncodedImage(
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const EncodedImage& encoded_image,
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const CodecSpecificInfo* codec_specific_info,
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const RTPFragmentationHeader* fragmentation) = 0;
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virtual void OnDroppedFrame() {}
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};
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class VideoEncoder {
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public:
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struct QpThresholds {
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QpThresholds(int l, int h) : low(l), high(h) {}
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QpThresholds() : low(-1), high(-1) {}
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int low;
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int high;
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};
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struct ScalingSettings {
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ScalingSettings(bool on, int low, int high)
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: enabled(on),
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thresholds(rtc::Optional<QpThresholds>(QpThresholds(low, high))) {}
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explicit ScalingSettings(bool on) : enabled(on) {}
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const bool enabled;
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const rtc::Optional<QpThresholds> thresholds;
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};
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static VideoCodecVP8 GetDefaultVp8Settings();
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static VideoCodecVP9 GetDefaultVp9Settings();
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static VideoCodecH264 GetDefaultH264Settings();
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virtual ~VideoEncoder() {}
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// Initialize the encoder with the information from the codecSettings
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//
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// Input:
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// - codec_settings : Codec settings
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// - number_of_cores : Number of cores available for the encoder
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// - max_payload_size : The maximum size each payload is allowed
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// to have. Usually MTU - overhead.
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//
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// Return value : Set bit rate if OK
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// <0 - Errors:
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// WEBRTC_VIDEO_CODEC_ERR_PARAMETER
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// WEBRTC_VIDEO_CODEC_ERR_SIZE
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// WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED
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// WEBRTC_VIDEO_CODEC_MEMORY
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// WEBRTC_VIDEO_CODEC_ERROR
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virtual int32_t InitEncode(const VideoCodec* codec_settings,
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int32_t number_of_cores,
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size_t max_payload_size) = 0;
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// Register an encode complete callback object.
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//
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// Input:
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// - callback : Callback object which handles encoded images.
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//
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// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
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virtual int32_t RegisterEncodeCompleteCallback(
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EncodedImageCallback* callback) = 0;
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// Free encoder memory.
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// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
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virtual int32_t Release() = 0;
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// Encode an I420 image (as a part of a video stream). The encoded image
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// will be returned to the user through the encode complete callback.
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//
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// Input:
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// - frame : Image to be encoded
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// - frame_types : Frame type to be generated by the encoder.
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//
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// Return value : WEBRTC_VIDEO_CODEC_OK if OK
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// <0 - Errors:
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// WEBRTC_VIDEO_CODEC_ERR_PARAMETER
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// WEBRTC_VIDEO_CODEC_MEMORY
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// WEBRTC_VIDEO_CODEC_ERROR
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// WEBRTC_VIDEO_CODEC_TIMEOUT
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virtual int32_t Encode(const VideoFrame& frame,
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const CodecSpecificInfo* codec_specific_info,
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const std::vector<FrameType>* frame_types) = 0;
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// Inform the encoder of the new packet loss rate and the round-trip time of
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// the network.
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//
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// Input:
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// - packet_loss : Fraction lost
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// (loss rate in percent = 100 * packetLoss / 255)
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// - rtt : Round-trip time in milliseconds
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// Return value : WEBRTC_VIDEO_CODEC_OK if OK
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// <0 - Errors: WEBRTC_VIDEO_CODEC_ERROR
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virtual int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) = 0;
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// Inform the encoder about the new target bit rate.
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//
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// Input:
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// - bitrate : New target bit rate
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// - framerate : The target frame rate
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//
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// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
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virtual int32_t SetRates(uint32_t bitrate, uint32_t framerate) {
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RTC_NOTREACHED() << "SetRate(uint32_t, uint32_t) is deprecated.";
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return -1;
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}
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// Default fallback: Just use the sum of bitrates as the single target rate.
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// TODO(sprang): Remove this default implementation when we remove SetRates().
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virtual int32_t SetRateAllocation(const BitrateAllocation& allocation,
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uint32_t framerate) {
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return SetRates(allocation.get_sum_kbps(), framerate);
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}
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// Any encoder implementation wishing to use the WebRTC provided
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// quality scaler must implement this method.
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virtual ScalingSettings GetScalingSettings() const {
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return ScalingSettings(false);
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}
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virtual int32_t SetPeriodicKeyFrames(bool enable) { return -1; }
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virtual bool SupportsNativeHandle() const { return false; }
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virtual const char* ImplementationName() const { return "unknown"; }
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};
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} // namespace webrtc
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#endif // WEBRTC_API_VIDEO_CODECS_VIDEO_ENCODER_H_
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