/* * Copyright (c) 2014 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 WEBRTC_VIDEO_ENCODER_H_ #define WEBRTC_VIDEO_ENCODER_H_ #include #include #include #include "webrtc/base/checks.h" #include "webrtc/common_types.h" #include "webrtc/typedefs.h" #include "webrtc/video_frame.h" #include "webrtc/base/optional.h" namespace webrtc { class RTPFragmentationHeader; // TODO(pbos): Expose these through a public (root) header or change these APIs. struct CodecSpecificInfo; class VideoCodec; class EncodedImageCallback { public: virtual ~EncodedImageCallback() {} struct Result { enum Error { OK, // Failed to send the packet. ERROR_SEND_FAILED, }; Result(Error error) : error(error) {} Result(Error error, uint32_t frame_id) : error(error), frame_id(frame_id) {} Error error; // Frame ID assigned to the frame. The frame ID should be the same as the ID // seen by the receiver for this frame. RTP timestamp of the frame is used // as frame ID when RTP is used to send video. Must be used only when // error=OK. uint32_t frame_id = 0; // Tells the encoder that the next frame is should be dropped. bool drop_next_frame = false; }; // Callback function which is called when an image has been encoded. virtual Result OnEncodedImage( const EncodedImage& encoded_image, const CodecSpecificInfo* codec_specific_info, const RTPFragmentationHeader* fragmentation) = 0; virtual void OnDroppedFrame() {} }; class VideoEncoder { public: enum EncoderType { kH264, kVp8, kVp9, kUnsupportedCodec, }; struct QpThresholds { QpThresholds(int l, int h) : low(l), high(h) {} QpThresholds() : low(-1), high(-1) {} int low; int high; }; struct ScalingSettings { ScalingSettings(bool on, int low, int high) : enabled(on), thresholds(rtc::Optional(QpThresholds(low, high))) {} explicit ScalingSettings(bool on) : enabled(on) {} const bool enabled; const rtc::Optional thresholds; }; static VideoEncoder* Create(EncoderType codec_type); // Returns true if this type of encoder can be created using // VideoEncoder::Create. static bool IsSupportedSoftware(EncoderType codec_type); static EncoderType CodecToEncoderType(VideoCodecType codec_type); static VideoCodecVP8 GetDefaultVp8Settings(); static VideoCodecVP9 GetDefaultVp9Settings(); static VideoCodecH264 GetDefaultH264Settings(); virtual ~VideoEncoder() {} // Initialize the encoder with the information from the codecSettings // // Input: // - codec_settings : Codec settings // - number_of_cores : Number of cores available for the encoder // - max_payload_size : The maximum size each payload is allowed // to have. Usually MTU - overhead. // // Return value : Set bit rate if OK // <0 - Errors: // WEBRTC_VIDEO_CODEC_ERR_PARAMETER // WEBRTC_VIDEO_CODEC_ERR_SIZE // WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED // WEBRTC_VIDEO_CODEC_MEMORY // WEBRTC_VIDEO_CODEC_ERROR virtual int32_t InitEncode(const VideoCodec* codec_settings, int32_t number_of_cores, size_t max_payload_size) = 0; // Register an encode complete callback object. // // Input: // - callback : Callback object which handles encoded images. // // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise. virtual int32_t RegisterEncodeCompleteCallback( EncodedImageCallback* callback) = 0; // Free encoder memory. // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise. virtual int32_t Release() = 0; // Encode an I420 image (as a part of a video stream). The encoded image // will be returned to the user through the encode complete callback. // // Input: // - frame : Image to be encoded // - frame_types : Frame type to be generated by the encoder. // // Return value : WEBRTC_VIDEO_CODEC_OK if OK // <0 - Errors: // WEBRTC_VIDEO_CODEC_ERR_PARAMETER // WEBRTC_VIDEO_CODEC_MEMORY // WEBRTC_VIDEO_CODEC_ERROR // WEBRTC_VIDEO_CODEC_TIMEOUT virtual int32_t Encode(const VideoFrame& frame, const CodecSpecificInfo* codec_specific_info, const std::vector* frame_types) = 0; // Inform the encoder of the new packet loss rate and the round-trip time of // the network. // // Input: // - packet_loss : Fraction lost // (loss rate in percent = 100 * packetLoss / 255) // - rtt : Round-trip time in milliseconds // Return value : WEBRTC_VIDEO_CODEC_OK if OK // <0 - Errors: WEBRTC_VIDEO_CODEC_ERROR virtual int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) = 0; // Inform the encoder about the new target bit rate. // // Input: // - bitrate : New target bit rate // - framerate : The target frame rate // // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise. virtual int32_t SetRates(uint32_t bitrate, uint32_t framerate) { RTC_NOTREACHED() << "SetRate(uint32_t, uint32_t) is deprecated."; return -1; } // Default fallback: Just use the sum of bitrates as the single target rate. // TODO(sprang): Remove this default implementation when we remove SetRates(). virtual int32_t SetRateAllocation(const BitrateAllocation& allocation, uint32_t framerate) { return SetRates(allocation.get_sum_kbps(), framerate); } // Any encoder implementation wishing to use the WebRTC provided // quality scaler must implement this method. virtual ScalingSettings GetScalingSettings() const { return ScalingSettings(false); } virtual int32_t SetPeriodicKeyFrames(bool enable) { return -1; } virtual bool SupportsNativeHandle() const { return false; } virtual const char* ImplementationName() const { return "unknown"; } }; } // namespace webrtc #endif // WEBRTC_VIDEO_ENCODER_H_