blob: 0501b9af7f526e22af7267c79b11a2f717beba7f [file] [log] [blame]
* Copyright (c) 2017 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.
#include <string>
#include "api/rtp_headers.h"
#include "api/task_queue/task_queue_base.h"
#include "api/video/video_bitrate_allocation.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/ntp_time.h"
namespace webrtc {
class ReceiveStatisticsProvider;
class Transport;
// Interface to watch incoming rtcp packets by media (rtp) receiver.
class MediaReceiverRtcpObserver {
virtual ~MediaReceiverRtcpObserver() = default;
// All message handlers have default empty implementation. This way users only
// need to implement the ones they are interested in.
virtual void OnSenderReport(uint32_t sender_ssrc,
NtpTime ntp_time,
uint32_t rtp_time) {}
virtual void OnBye(uint32_t sender_ssrc) {}
virtual void OnBitrateAllocation(uint32_t sender_ssrc,
const VideoBitrateAllocation& allocation) {}
struct RtcpTransceiverConfig {
RtcpTransceiverConfig(const RtcpTransceiverConfig&);
RtcpTransceiverConfig& operator=(const RtcpTransceiverConfig&);
// Logs the error and returns false if configuration miss key objects or
// is inconsistant. May log warnings.
bool Validate() const;
// Used to prepend all log messages. Can be empty.
std::string debug_id;
// Ssrc to use as default sender ssrc, e.g. for transport-wide feedbacks.
uint32_t feedback_ssrc = 1;
// Canonical End-Point Identifier of the local particiapnt.
// Defined in rfc3550 section 6 note 2 and section 6.5.1.
std::string cname;
// Maximum packet size outgoing transport accepts.
size_t max_packet_size = 1200;
// The clock to use when querying for the NTP time. Should be set.
Clock* clock = nullptr;
// Transport to send rtcp packets to. Should be set.
Transport* outgoing_transport = nullptr;
// Queue for scheduling delayed tasks, e.g. sending periodic compound packets.
TaskQueueBase* task_queue = nullptr;
// Rtcp report block generator for outgoing receiver reports.
ReceiveStatisticsProvider* receive_statistics = nullptr;
// Callback to pass result of rtt calculation. Should outlive RtcpTransceiver.
// Callbacks will be invoked on the task_queue.
RtcpRttStats* rtt_observer = nullptr;
// Configures if sending should
// enforce compound packets:
// or allow reduced size packets:
// Receiving accepts both compound and reduced-size packets.
RtcpMode rtcp_mode = RtcpMode::kCompound;
// Tuning parameters.
// Initial state if |outgoing_transport| ready to accept packets.
bool initial_ready_to_send = true;
// Delay before 1st periodic compound packet.
int initial_report_delay_ms = 500;
// Period between periodic compound packets.
int report_period_ms = 1000;
// Flags for features and experiments.
bool schedule_periodic_compound_packets = true;
// Estimate RTT as non-sender as described in
// and #section-4.5
bool non_sender_rtt_measurement = false;
// Allows a REMB message to be sent immediately when SetRemb is called without
// having to wait for the next compount message to be sent.
bool send_remb_on_change = false;
} // namespace webrtc