| /* |
| * Copyright (c) 2016 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_PACKET_TO_SEND_H_ |
| #define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ |
| |
| #include <stddef.h> |
| #include <stdint.h> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/array_view.h" |
| #include "api/video/video_timing.h" |
| #include "modules/rtp_rtcp/source/rtp_header_extensions.h" |
| #include "modules/rtp_rtcp/source/rtp_packet.h" |
| |
| namespace webrtc { |
| // Class to hold rtp packet with metadata for sender side. |
| class RtpPacketToSend : public RtpPacket { |
| public: |
| enum class Type { |
| kAudio, // Audio media packets. |
| kVideo, // Video media packets. |
| kRetransmission, // RTX (usually) packets send as response to NACK. |
| kForwardErrorCorrection, // FEC packets. |
| kPadding // RTX or plain padding sent to maintain BWE. |
| }; |
| |
| explicit RtpPacketToSend(const ExtensionManager* extensions); |
| RtpPacketToSend(const ExtensionManager* extensions, size_t capacity); |
| RtpPacketToSend(const RtpPacketToSend& packet); |
| RtpPacketToSend(RtpPacketToSend&& packet); |
| |
| RtpPacketToSend& operator=(const RtpPacketToSend& packet); |
| RtpPacketToSend& operator=(RtpPacketToSend&& packet); |
| |
| ~RtpPacketToSend(); |
| |
| // Time in local time base as close as it can to frame capture time. |
| int64_t capture_time_ms() const { return capture_time_ms_; } |
| |
| void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; } |
| |
| void set_packet_type(Type type) { packet_type_ = type; } |
| absl::optional<Type> packet_type() const { return packet_type_; } |
| |
| // If this is a retransmission, indicates the sequence number of the original |
| // media packet that this packet represents. If RTX is used this will likely |
| // be different from SequenceNumber(). |
| void set_retransmitted_sequence_number(uint16_t sequence_number) { |
| retransmitted_sequence_number_ = sequence_number; |
| } |
| absl::optional<uint16_t> retransmitted_sequence_number() { |
| return retransmitted_sequence_number_; |
| } |
| |
| void set_allow_retransmission(bool allow_retransmission) { |
| allow_retransmission_ = allow_retransmission; |
| } |
| bool allow_retransmission() { return allow_retransmission_; } |
| |
| // Additional data bound to the RTP packet for use in application code, |
| // outside of WebRTC. |
| rtc::ArrayView<const uint8_t> application_data() const { |
| return application_data_; |
| } |
| |
| void set_application_data(rtc::ArrayView<const uint8_t> data) { |
| application_data_.assign(data.begin(), data.end()); |
| } |
| |
| void set_packetization_finish_time_ms(int64_t time) { |
| SetExtension<VideoTimingExtension>( |
| VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time), |
| VideoSendTiming::kPacketizationFinishDeltaOffset); |
| } |
| |
| void set_pacer_exit_time_ms(int64_t time) { |
| SetExtension<VideoTimingExtension>( |
| VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time), |
| VideoSendTiming::kPacerExitDeltaOffset); |
| } |
| |
| void set_network_time_ms(int64_t time) { |
| SetExtension<VideoTimingExtension>( |
| VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time), |
| VideoSendTiming::kNetworkTimestampDeltaOffset); |
| } |
| |
| void set_network2_time_ms(int64_t time) { |
| SetExtension<VideoTimingExtension>( |
| VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time), |
| VideoSendTiming::kNetwork2TimestampDeltaOffset); |
| } |
| |
| private: |
| int64_t capture_time_ms_ = 0; |
| absl::optional<Type> packet_type_; |
| bool allow_retransmission_ = false; |
| absl::optional<uint16_t> retransmitted_sequence_number_; |
| std::vector<uint8_t> application_data_; |
| }; |
| |
| } // namespace webrtc |
| #endif // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ |