| /* |
| * Copyright (c) 2017 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_RTP_FORMAT_VIDEO_STEREO_H_ |
| #define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_STEREO_H_ |
| |
| #include <string> |
| |
| #include "modules/rtp_rtcp/source/rtp_format.h" |
| #include "modules/rtp_rtcp/source/rtp_format_video_generic.h" |
| #include "rtc_base/constructormagic.h" |
| |
| namespace webrtc { |
| |
| class RtpPacketizerStereo : public RtpPacketizer { |
| public: |
| RtpPacketizerStereo(const RTPVideoHeaderStereo& header, |
| FrameType frame_type, |
| size_t max_payload_len, |
| size_t last_packet_reduction_len); |
| |
| ~RtpPacketizerStereo() override; |
| |
| size_t SetPayloadData(const uint8_t* payload_data, |
| size_t payload_size, |
| const RTPFragmentationHeader* fragmentation) override; |
| |
| // Get the next payload with generic payload header. |
| // Write payload and set marker bit of the |packet|. |
| // Returns true on success, false otherwise. |
| bool NextPacket(RtpPacketToSend* packet) override; |
| |
| std::string ToString() override; |
| |
| private: |
| const RTPVideoHeaderStereo header_; |
| const size_t max_payload_len_; |
| const size_t last_packet_reduction_len_; |
| size_t num_packets_remaining_ = 0; |
| // TODO(emircan): Use codec specific packetizers. If not possible, refactor |
| // this class to have similar logic to generic packetizer. |
| RtpPacketizerGeneric packetizer_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerStereo); |
| }; |
| |
| class RtpDepacketizerStereo : public RtpDepacketizer { |
| public: |
| ~RtpDepacketizerStereo() override; |
| |
| bool Parse(ParsedPayload* parsed_payload, |
| const uint8_t* payload_data, |
| size_t payload_data_length) override; |
| |
| private: |
| RtpDepacketizerGeneric depacketizer_; |
| }; |
| } // namespace webrtc |
| #endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_STEREO_H_ |