diff options
Diffstat (limited to 'modules/rtp_rtcp/source/rtcp_sender.h')
| -rw-r--r-- | modules/rtp_rtcp/source/rtcp_sender.h | 72 |
1 files changed, 65 insertions, 7 deletions
diff --git a/modules/rtp_rtcp/source/rtcp_sender.h b/modules/rtp_rtcp/source/rtcp_sender.h index 463666a22a..e50ce44e13 100644 --- a/modules/rtp_rtcp/source/rtcp_sender.h +++ b/modules/rtp_rtcp/source/rtcp_sender.h @@ -19,6 +19,8 @@ #include "absl/types/optional.h" #include "api/call/transport.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" #include "api/video/video_bitrate_allocation.h" #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "modules/rtp_rtcp/include/receive_statistics.h" @@ -42,6 +44,38 @@ class RtcEventLog; class RTCPSender final { public: + struct Configuration { + // TODO(bugs.webrtc.org/11581): Remove this temporary conversion utility + // once rtc_rtcp_impl.cc/h are gone. + static Configuration FromRtpRtcpConfiguration( + const RtpRtcpInterface::Configuration& config); + + // True for a audio version of the RTP/RTCP module object false will create + // a video version. + bool audio = false; + // SSRCs for media and retransmission, respectively. + // FlexFec SSRC is fetched from |flexfec_sender|. + uint32_t local_media_ssrc = 0; + // The clock to use to read time. If nullptr then system clock will be used. + Clock* clock = nullptr; + // Transport object that will be called when packets are ready to be sent + // out on the network. + Transport* outgoing_transport = nullptr; + // Estimate RTT as non-sender as described in + // https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5 + bool non_sender_rtt_measurement = false; + // Optional callback which, if specified, is used by RTCPSender to schedule + // the next time to evaluate if RTCP should be sent by means of + // TimeToSendRTCPReport/SendRTCP. + // The RTCPSender client still needs to call TimeToSendRTCPReport/SendRTCP + // to actually get RTCP sent. + std::function<void(TimeDelta)> schedule_next_rtcp_send_evaluation_function; + + RtcEventLog* event_log = nullptr; + absl::optional<TimeDelta> rtcp_report_interval; + ReceiveStatisticsProvider* receive_statistics = nullptr; + RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr; + }; struct FeedbackState { FeedbackState(); FeedbackState(const FeedbackState&); @@ -63,6 +97,9 @@ class RTCPSender final { RTCPReceiver* receiver; }; + explicit RTCPSender(Configuration config); + // TODO(bugs.webrtc.org/11581): delete this temporary compatibility helper + // once downstream dependencies migrates. explicit RTCPSender(const RtpRtcpInterface::Configuration& config); RTCPSender() = delete; @@ -75,8 +112,8 @@ class RTCPSender final { void SetRTCPStatus(RtcpMode method) RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_); bool Sending() const RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_); - int32_t SetSendingStatus(const FeedbackState& feedback_state, - bool enabled) + void SetSendingStatus(const FeedbackState& feedback_state, + bool enabled) RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_); // combine the functions int32_t SetNackStatus(bool enable) RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_); @@ -84,8 +121,14 @@ class RTCPSender final { void SetTimestampOffset(uint32_t timestamp_offset) RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_); + void SetLastRtpTime(uint32_t rtp_timestamp, + absl::optional<Timestamp> capture_time, + absl::optional<int8_t> payload_type) + RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_); // TODO(bugs.webrtc.org/6458): Remove default parameter value when all the // depending projects are updated to correctly set payload type. + // TODO(bugs.webrtc.org/12873): Remove once downstream consumers migrates to + // the new version of SetLastRtpTime declared above. void SetLastRtpTime(uint32_t rtp_timestamp, int64_t capture_time_ms, int8_t payload_type = -1) @@ -94,7 +137,8 @@ class RTCPSender final { void SetRtpClockRate(int8_t payload_type, int rtp_clock_rate_hz) RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_); - uint32_t SSRC() const { return ssrc_; } + uint32_t SSRC() const; + void SetSsrc(uint32_t ssrc); void SetRemoteSSRC(uint32_t ssrc) RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_); @@ -186,8 +230,16 @@ class RTCPSender final { void BuildNACK(const RtcpContext& context, PacketSender& sender) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_); + // |duration| being TimeDelta::Zero() means schedule immediately. + void SetNextRtcpSendEvaluationDuration(TimeDelta duration) + RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_); + const bool audio_; - const uint32_t ssrc_; + // TODO(bugs.webrtc.org/11581): `mutex_rtcp_sender_` shouldn't be required if + // we consistently run network related operations on the network thread. + // This is currently not possible due to callbacks from the process thread in + // ModuleRtpRtcpImpl2. + uint32_t ssrc_ RTC_GUARDED_BY(mutex_rtcp_sender_); Clock* const clock_; Random random_ RTC_GUARDED_BY(mutex_rtcp_sender_); RtcpMode method_ RTC_GUARDED_BY(mutex_rtcp_sender_); @@ -195,16 +247,22 @@ class RTCPSender final { RtcEventLog* const event_log_; Transport* const transport_; - const int report_interval_ms_; + const TimeDelta report_interval_; + // Set from + // RTCPSender::Configuration::schedule_next_rtcp_send_evaluation_function. + const std::function<void(TimeDelta)> + schedule_next_rtcp_send_evaluation_function_; mutable Mutex mutex_rtcp_sender_; bool sending_ RTC_GUARDED_BY(mutex_rtcp_sender_); - int64_t next_time_to_send_rtcp_ RTC_GUARDED_BY(mutex_rtcp_sender_); + absl::optional<Timestamp> next_time_to_send_rtcp_ + RTC_GUARDED_BY(mutex_rtcp_sender_); uint32_t timestamp_offset_ RTC_GUARDED_BY(mutex_rtcp_sender_); uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(mutex_rtcp_sender_); - int64_t last_frame_capture_time_ms_ RTC_GUARDED_BY(mutex_rtcp_sender_); + absl::optional<Timestamp> last_frame_capture_time_ + RTC_GUARDED_BY(mutex_rtcp_sender_); // SSRC that we receive on our RTP channel uint32_t remote_ssrc_ RTC_GUARDED_BY(mutex_rtcp_sender_); std::string cname_ RTC_GUARDED_BY(mutex_rtcp_sender_); |