| /* |
| * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef API_VIDEO_VIDEO_STREAM_ENCODER_INTERFACE_H_ |
| #define API_VIDEO_VIDEO_STREAM_ENCODER_INTERFACE_H_ |
| |
| #include <vector> |
| |
| #include "api/rtp_parameters.h" // For DegradationPreference. |
| #include "api/units/data_rate.h" |
| #include "api/video/video_bitrate_allocator.h" |
| #include "api/video/video_sink_interface.h" |
| #include "api/video/video_source_interface.h" |
| #include "api/video_codecs/video_encoder.h" |
| #include "api/video_codecs/video_encoder_config.h" |
| |
| namespace webrtc { |
| |
| // This interface represents a class responsible for creating and driving the |
| // encoder(s) for a single video stream. It is also responsible for adaptation |
| // decisions related to video quality, requesting reduced frame rate or |
| // resolution from the VideoSource when needed. |
| // TODO(bugs.webrtc.org/8830): This interface is under development. Changes |
| // under consideration include: |
| // |
| // 1. Taking out responsibility for adaptation decisions, instead only reporting |
| // per-frame measurements to the decision maker. |
| // |
| // 2. Moving responsibility for simulcast and for software fallback into this |
| // class. |
| class VideoStreamEncoderInterface : public rtc::VideoSinkInterface<VideoFrame> { |
| public: |
| // Interface for receiving encoded video frames and notifications about |
| // configuration changes. |
| class EncoderSink : public EncodedImageCallback { |
| public: |
| virtual void OnEncoderConfigurationChanged( |
| std::vector<VideoStream> streams, |
| VideoEncoderConfig::ContentType content_type, |
| int min_transmit_bitrate_bps) = 0; |
| }; |
| |
| // Sets the source that will provide video frames to the VideoStreamEncoder's |
| // OnFrame method. |degradation_preference| control whether or not resolution |
| // or frame rate may be reduced. The VideoStreamEncoder registers itself with |
| // |source|, and signals adaptation decisions to the source in the form of |
| // VideoSinkWants. |
| // TODO(nisse): When adaptation logic is extracted from this class, |
| // it no longer needs to know the source. |
| virtual void SetSource( |
| rtc::VideoSourceInterface<VideoFrame>* source, |
| const DegradationPreference& degradation_preference) = 0; |
| |
| // Sets the |sink| that gets the encoded frames. |rotation_applied| means |
| // that the source must support rotation. Only set |rotation_applied| if the |
| // remote side does not support the rotation extension. |
| virtual void SetSink(EncoderSink* sink, bool rotation_applied) = 0; |
| |
| // Sets an initial bitrate, later overriden by OnBitrateUpdated. Mainly |
| // affects the resolution of the initial key frame: If incoming frames are |
| // larger than reasonable for the start bitrate, and scaling is enabled, |
| // VideoStreamEncoder asks the source to scale down and drops a few initial |
| // frames. |
| // TODO(nisse): This is a poor interface, and mixes bandwidth estimation and |
| // codec configuration in an undesired way. For the actual send bandwidth, we |
| // should always be somewhat conservative, but we may nevertheless want to let |
| // the application configure a more optimistic quality for the initial |
| // resolution. Should be replaced by a construction time setting. |
| virtual void SetStartBitrate(int start_bitrate_bps) = 0; |
| |
| // Request a key frame. Used for signalling from the remote receiver. |
| virtual void SendKeyFrame() = 0; |
| |
| // Set the currently estimated network properties. A |target_bitrate| |
| // of zero pauses the encoder. |
| // |link_allocation| is the bandwidth available for this video stream on the |
| // network link. It is always at least |target_bitrate| but may be higher |
| // if we are not network constrained. |
| virtual void OnBitrateUpdated(DataRate target_bitrate, |
| DataRate link_allocation, |
| uint8_t fraction_lost, |
| int64_t round_trip_time_ms) = 0; |
| |
| // Register observer for the bitrate allocation between the temporal |
| // and spatial layers. |
| virtual void SetBitrateAllocationObserver( |
| VideoBitrateAllocationObserver* bitrate_observer) = 0; |
| |
| // Creates and configures an encoder with the given |config|. The |
| // |max_data_payload_length| is used to support single NAL unit |
| // packetization for H.264. |
| virtual void ConfigureEncoder(VideoEncoderConfig config, |
| size_t max_data_payload_length) = 0; |
| |
| // Permanently stop encoding. After this method has returned, it is |
| // guaranteed that no encoded frames will be delivered to the sink. |
| virtual void Stop() = 0; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // API_VIDEO_VIDEO_STREAM_ENCODER_INTERFACE_H_ |