| /* |
| * Copyright (c) 2020 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 MODULES_RTP_RTCP_SOURCE_TRANSFORMABLE_ENCODED_FRAME_H_ |
| #define MODULES_RTP_RTCP_SOURCE_TRANSFORMABLE_ENCODED_FRAME_H_ |
| |
| #include <memory> |
| |
| #include "absl/types/optional.h" |
| #include "api/video/encoded_frame.h" |
| #include "modules/include/module_common_types.h" |
| #include "modules/rtp_rtcp/source/rtp_video_header.h" |
| |
| namespace webrtc { |
| |
| class TransformableEncodedFrame : public video_coding::EncodedFrame { |
| public: |
| TransformableEncodedFrame( |
| rtc::scoped_refptr<EncodedImageBufferInterface> encoded_data, |
| const RTPVideoHeader& video_header, |
| int payload_type, |
| absl::optional<VideoCodecType> codec_type, |
| uint32_t rtp_timestamp, |
| int64_t capture_time_ms, |
| const RTPFragmentationHeader* fragmentation, |
| absl::optional<int64_t> expected_retransmission_time_ms); |
| ~TransformableEncodedFrame() override; |
| |
| const RTPVideoHeader& video_header() const; |
| absl::optional<VideoCodecType> codec_type() const; |
| int64_t capture_time_ms() const { return capture_time_ms_; } |
| RTPFragmentationHeader* fragmentation_header() const { |
| return fragmentation_header_.get(); |
| } |
| const absl::optional<int64_t>& expected_retransmission_time_ms() const { |
| return expected_retransmission_time_ms_; |
| } |
| |
| // Implements EncodedFrame. |
| int64_t ReceivedTime() const override; |
| int64_t RenderTime() const override; |
| |
| private: |
| RTPVideoHeader video_header_; |
| absl::optional<VideoCodecType> codec_type_ = absl::nullopt; |
| std::unique_ptr<RTPFragmentationHeader> fragmentation_header_; |
| absl::optional<int64_t> expected_retransmission_time_ms_ = absl::nullopt; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_RTP_RTCP_SOURCE_TRANSFORMABLE_ENCODED_FRAME_H_ |