| /* |
| * 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_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ |
| #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ |
| |
| #include <deque> |
| #include <memory> |
| #include <queue> |
| #include <string> |
| |
| #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" |
| #include "webrtc/rtc_base/buffer.h" |
| #include "webrtc/rtc_base/constructormagic.h" |
| |
| namespace webrtc { |
| |
| class RtpPacketizerH264 : public RtpPacketizer { |
| public: |
| // Initialize with payload from encoder. |
| // The payload_data must be exactly one encoded H264 frame. |
| RtpPacketizerH264(size_t max_payload_len, |
| size_t last_packet_reduction_len, |
| H264PacketizationMode packetization_mode); |
| |
| virtual ~RtpPacketizerH264(); |
| |
| size_t SetPayloadData(const uint8_t* payload_data, |
| size_t payload_size, |
| const RTPFragmentationHeader* fragmentation) override; |
| |
| // Get the next payload with H264 payload header. |
| // Write payload and set marker bit of the |packet|. |
| // Returns true on success, false otherwise. |
| bool NextPacket(RtpPacketToSend* rtp_packet) override; |
| |
| std::string ToString() override; |
| |
| private: |
| // Input fragments (NAL units), with an optionally owned temporary buffer, |
| // used in case the fragment gets modified. |
| struct Fragment { |
| Fragment(const uint8_t* buffer, size_t length); |
| explicit Fragment(const Fragment& fragment); |
| const uint8_t* buffer = nullptr; |
| size_t length = 0; |
| std::unique_ptr<rtc::Buffer> tmp_buffer; |
| }; |
| |
| // A packet unit (H264 packet), to be put into an RTP packet: |
| // If a NAL unit is too large for an RTP packet, this packet unit will |
| // represent a FU-A packet of a single fragment of the NAL unit. |
| // If a NAL unit is small enough to fit within a single RTP packet, this |
| // packet unit may represent a single NAL unit or a STAP-A packet, of which |
| // there may be multiple in a single RTP packet (if so, aggregated = true). |
| struct PacketUnit { |
| PacketUnit(const Fragment& source_fragment, |
| bool first_fragment, |
| bool last_fragment, |
| bool aggregated, |
| uint8_t header) |
| : source_fragment(source_fragment), |
| first_fragment(first_fragment), |
| last_fragment(last_fragment), |
| aggregated(aggregated), |
| header(header) {} |
| |
| const Fragment source_fragment; |
| bool first_fragment; |
| bool last_fragment; |
| bool aggregated; |
| uint8_t header; |
| }; |
| |
| void GeneratePackets(); |
| void PacketizeFuA(size_t fragment_index); |
| size_t PacketizeStapA(size_t fragment_index); |
| void PacketizeSingleNalu(size_t fragment_index); |
| void NextAggregatePacket(RtpPacketToSend* rtp_packet, bool last); |
| void NextFragmentPacket(RtpPacketToSend* rtp_packet); |
| |
| const size_t max_payload_len_; |
| const size_t last_packet_reduction_len_; |
| size_t num_packets_left_; |
| const H264PacketizationMode packetization_mode_; |
| std::deque<Fragment> input_fragments_; |
| std::queue<PacketUnit> packets_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264); |
| }; |
| |
| // Depacketizer for H264. |
| class RtpDepacketizerH264 : public RtpDepacketizer { |
| public: |
| RtpDepacketizerH264(); |
| virtual ~RtpDepacketizerH264(); |
| |
| bool Parse(ParsedPayload* parsed_payload, |
| const uint8_t* payload_data, |
| size_t payload_data_length) override; |
| |
| private: |
| bool ParseFuaNalu(RtpDepacketizer::ParsedPayload* parsed_payload, |
| const uint8_t* payload_data); |
| bool ProcessStapAOrSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload, |
| const uint8_t* payload_data); |
| |
| size_t offset_; |
| size_t length_; |
| std::unique_ptr<rtc::Buffer> modified_buffer_; |
| }; |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ |