blob: 4ab53c0c45ccdff9f59b95e9f1812d5616b37066 [file] [log] [blame]
* Copyright (c) 2012 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 <memory>
#include <string>
#include <vector>
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/crypto/crypto_options.h"
#include "api/media_transport_interface.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "rtc_base/function_view.h"
#include "rtc_base/task_queue.h"
namespace webrtc {
class FrameEncryptorInterface;
class ProcessThread;
class RtcEventLog;
class RtpRtcp;
class RtpTransportControllerSendInterface;
struct CallSendStatistics {
int64_t rttMs;
size_t bytesSent;
int packetsSent;
// See section 6.4.2 in for details.
struct ReportBlock {
uint32_t sender_SSRC; // SSRC of sender
uint32_t source_SSRC;
uint8_t fraction_lost;
int32_t cumulative_num_packets_lost;
uint32_t extended_highest_sequence_number;
uint32_t interarrival_jitter;
uint32_t last_SR_timestamp;
uint32_t delay_since_last_SR;
namespace voe {
class ChannelSendInterface {
virtual ~ChannelSendInterface() = default;
virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length) = 0;
virtual CallSendStatistics GetRTCPStatistics() const = 0;
virtual bool SetEncoder(int payload_type,
std::unique_ptr<AudioEncoder> encoder) = 0;
virtual void ModifyEncoder(
rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) = 0;
virtual void CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) = 0;
virtual void SetLocalSSRC(uint32_t ssrc) = 0;
// Use 0 to indicate that the extension should not be registered.
virtual void SetRid(const std::string& rid,
int extension_id,
int repaired_extension_id) = 0;
virtual void SetMid(const std::string& mid, int extension_id) = 0;
virtual void SetRTCP_CNAME(absl::string_view c_name) = 0;
virtual void SetExtmapAllowMixed(bool extmap_allow_mixed) = 0;
virtual void SetSendAudioLevelIndicationStatus(bool enable, int id) = 0;
virtual void EnableSendTransportSequenceNumber(int id) = 0;
virtual void RegisterSenderCongestionControlObjects(
RtpTransportControllerSendInterface* transport,
RtcpBandwidthObserver* bandwidth_observer) = 0;
virtual void ResetSenderCongestionControlObjects() = 0;
virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const = 0;
virtual ANAStats GetANAStatistics() const = 0;
virtual bool SetSendTelephoneEventPayloadType(int payload_type,
int payload_frequency) = 0;
virtual bool SendTelephoneEventOutband(int event, int duration_ms) = 0;
virtual void OnBitrateAllocation(BitrateAllocationUpdate update) = 0;
virtual int GetBitrate() const = 0;
virtual void SetInputMute(bool muted) = 0;
virtual void ProcessAndEncodeAudio(
std::unique_ptr<AudioFrame> audio_frame) = 0;
virtual RtpRtcp* GetRtpRtcp() const = 0;
virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) = 0;
virtual void OnRecoverableUplinkPacketLossRate(
float recoverable_packet_loss_rate) = 0;
// In RTP we currently rely on RTCP packets (|ReceivedRTCPPacket|) to inform
// about RTT.
// In media transport we rely on the TargetTransferRateObserver instead.
// In other words, if you are using RTP, you should expect
// |ReceivedRTCPPacket| to be called, if you are using media transport,
// |OnTargetTransferRate| will be called.
// In future, RTP media will move to the media transport implementation and
// these conditions will be removed.
// Returns the RTT in milliseconds.
virtual int64_t GetRTT() const = 0;
virtual void StartSend() = 0;
virtual void StopSend() = 0;
// E2EE Custom Audio Frame Encryption (Optional)
virtual void SetFrameEncryptor(
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) = 0;
std::unique_ptr<ChannelSendInterface> CreateChannelSend(
Clock* clock,
rtc::TaskQueue* encoder_queue,
ProcessThread* module_process_thread,
MediaTransportInterface* media_transport,
OverheadObserver* overhead_observer,
Transport* rtp_transport,
RtcpRttStats* rtcp_rtt_stats,
RtcEventLog* rtc_event_log,
FrameEncryptorInterface* frame_encryptor,
const webrtc::CryptoOptions& crypto_options,
bool extmap_allow_mixed,
int rtcp_report_interval_ms);
} // namespace voe
} // namespace webrtc