| /* |
| * 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 <utility> |
| |
| #include "absl/types/optional.h" |
| #include "api/array_view.h" |
| #include "api/ref_counted_base.h" |
| #include "api/scoped_refptr.h" |
| #include "api/units/time_delta.h" |
| #include "api/units/timestamp.h" |
| #include "api/video/video_timing.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.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. |
| // The metadata is not send over the wire, but packet sender may use it to |
| // create rtp header extensions or other data that is sent over the wire. |
| class RtpPacketToSend : public RtpPacket { |
| public: |
| // RtpPacketToSend::Type is deprecated. Use RtpPacketMediaType directly. |
| using Type = RtpPacketMediaType; |
| |
| 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. |
| webrtc::Timestamp capture_time() const { return capture_time_; } |
| void set_capture_time(webrtc::Timestamp time) { capture_time_ = time; } |
| |
| void set_packet_type(RtpPacketMediaType type); |
| |
| absl::optional<RtpPacketMediaType> packet_type() const { |
| return packet_type_; |
| } |
| |
| enum class OriginalType { kAudio, kVideo }; |
| // Original type does not change if packet type is changed to kRetransmission. |
| absl::optional<OriginalType> original_packet_type() const { |
| return original_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() const { |
| return retransmitted_sequence_number_; |
| } |
| |
| void set_allow_retransmission(bool allow_retransmission) { |
| allow_retransmission_ = allow_retransmission; |
| } |
| bool allow_retransmission() const { return allow_retransmission_; } |
| |
| // An application can attach arbitrary data to an RTP packet using |
| // `additional_data`. The additional data does not affect WebRTC processing. |
| rtc::scoped_refptr<rtc::RefCountedBase> additional_data() const { |
| return additional_data_; |
| } |
| void set_additional_data(rtc::scoped_refptr<rtc::RefCountedBase> data) { |
| additional_data_ = std::move(data); |
| } |
| |
| void set_packetization_finish_time(webrtc::Timestamp time) { |
| SetExtension<VideoTimingExtension>( |
| VideoSendTiming::GetDeltaCappedMs(time - capture_time_), |
| VideoTimingExtension::kPacketizationFinishDeltaOffset); |
| } |
| |
| void set_pacer_exit_time(webrtc::Timestamp time) { |
| SetExtension<VideoTimingExtension>( |
| VideoSendTiming::GetDeltaCappedMs(time - capture_time_), |
| VideoTimingExtension::kPacerExitDeltaOffset); |
| } |
| |
| void set_network_time(webrtc::Timestamp time) { |
| SetExtension<VideoTimingExtension>( |
| VideoSendTiming::GetDeltaCappedMs(time - capture_time_), |
| VideoTimingExtension::kNetworkTimestampDeltaOffset); |
| } |
| |
| void set_network2_time(webrtc::Timestamp time) { |
| SetExtension<VideoTimingExtension>( |
| VideoSendTiming::GetDeltaCappedMs(time - capture_time_), |
| VideoTimingExtension::kNetwork2TimestampDeltaOffset); |
| } |
| |
| // Indicates if packet is the first packet of a video frame. |
| void set_first_packet_of_frame(bool is_first_packet) { |
| is_first_packet_of_frame_ = is_first_packet; |
| } |
| bool is_first_packet_of_frame() const { return is_first_packet_of_frame_; } |
| |
| // Indicates if packet contains payload for a video key-frame. |
| void set_is_key_frame(bool is_key_frame) { is_key_frame_ = is_key_frame; } |
| bool is_key_frame() const { return is_key_frame_; } |
| |
| // Indicates if packets should be protected by FEC (Forward Error Correction). |
| void set_fec_protect_packet(bool protect) { fec_protect_packet_ = protect; } |
| bool fec_protect_packet() const { return fec_protect_packet_; } |
| |
| // Indicates if packet is using RED encapsulation, in accordance with |
| // https://tools.ietf.org/html/rfc2198 |
| void set_is_red(bool is_red) { is_red_ = is_red; } |
| bool is_red() const { return is_red_; } |
| |
| // The amount of time spent in the send queue, used for totalPacketSendDelay. |
| // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay |
| void set_time_in_send_queue(TimeDelta time_in_send_queue) { |
| time_in_send_queue_ = time_in_send_queue; |
| } |
| absl::optional<TimeDelta> time_in_send_queue() const { |
| return time_in_send_queue_; |
| } |
| |
| private: |
| webrtc::Timestamp capture_time_ = webrtc::Timestamp::Zero(); |
| absl::optional<RtpPacketMediaType> packet_type_; |
| absl::optional<OriginalType> original_packet_type_; |
| bool allow_retransmission_ = false; |
| absl::optional<uint16_t> retransmitted_sequence_number_; |
| rtc::scoped_refptr<rtc::RefCountedBase> additional_data_; |
| bool is_first_packet_of_frame_ = false; |
| bool is_key_frame_ = false; |
| bool fec_protect_packet_ = false; |
| bool is_red_ = false; |
| absl::optional<TimeDelta> time_in_send_queue_; |
| }; |
| |
| } // namespace webrtc |
| #endif // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_ |