| /* |
| * Copyright (c) 2013 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_GENERIC_H_ |
| #define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_ |
| |
| #include <stdint.h> |
| |
| #include <vector> |
| |
| #include "api/array_view.h" |
| #include "modules/rtp_rtcp/source/rtp_format.h" |
| #include "rtc_base/constructor_magic.h" |
| |
| namespace webrtc { |
| |
| class RtpPacketToSend; |
| struct RTPVideoHeader; |
| |
| namespace RtpFormatVideoGeneric { |
| static const uint8_t kKeyFrameBit = 0x01; |
| static const uint8_t kFirstPacketBit = 0x02; |
| // If this bit is set, there will be an extended header contained in this |
| // packet. This was added later so old clients will not send this. |
| static const uint8_t kExtendedHeaderBit = 0x04; |
| } // namespace RtpFormatVideoGeneric |
| |
| class RtpPacketizerGeneric : public RtpPacketizer { |
| public: |
| // Initialize with payload from encoder. |
| // The payload_data must be exactly one encoded generic frame. |
| // Packets returned by `NextPacket` will contain the generic payload header. |
| RtpPacketizerGeneric(rtc::ArrayView<const uint8_t> payload, |
| PayloadSizeLimits limits, |
| const RTPVideoHeader& rtp_video_header); |
| // Initialize with payload from encoder. |
| // The payload_data must be exactly one encoded generic frame. |
| // Packets returned by `NextPacket` will contain raw payload without the |
| // generic payload header. |
| RtpPacketizerGeneric(rtc::ArrayView<const uint8_t> payload, |
| PayloadSizeLimits limits); |
| |
| ~RtpPacketizerGeneric() override; |
| |
| size_t NumPackets() const override; |
| |
| // Get the next payload. |
| // Write payload and set marker bit of the `packet`. |
| // Returns true on success, false otherwise. |
| bool NextPacket(RtpPacketToSend* packet) override; |
| |
| private: |
| // Fills header_ and header_size_ members. |
| void BuildHeader(const RTPVideoHeader& rtp_video_header); |
| |
| uint8_t header_[3]; |
| size_t header_size_; |
| rtc::ArrayView<const uint8_t> remaining_payload_; |
| std::vector<int> payload_sizes_; |
| std::vector<int>::const_iterator current_packet_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric); |
| }; |
| } // namespace webrtc |
| #endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_ |