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741daaf039
Bug: webrtc:10138 Change-Id: Icc25a2a277a9608701aaddd546882366739991ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127898 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27227}
137 lines
5 KiB
C++
137 lines
5 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef AUDIO_CHANNEL_SEND_H_
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#define AUDIO_CHANNEL_SEND_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/audio/audio_frame.h"
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#include "api/audio_codecs/audio_encoder.h"
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#include "api/crypto/crypto_options.h"
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#include "api/function_view.h"
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#include "api/media_transport_interface.h"
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#include "api/task_queue/task_queue_factory.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
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namespace webrtc {
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class FrameEncryptorInterface;
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class ProcessThread;
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class RtcEventLog;
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class RtpRtcp;
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class RtpTransportControllerSendInterface;
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struct CallSendStatistics {
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int64_t rttMs;
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size_t bytesSent;
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int packetsSent;
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};
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// See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details.
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struct ReportBlock {
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uint32_t sender_SSRC; // SSRC of sender
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uint32_t source_SSRC;
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uint8_t fraction_lost;
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int32_t cumulative_num_packets_lost;
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uint32_t extended_highest_sequence_number;
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uint32_t interarrival_jitter;
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uint32_t last_SR_timestamp;
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uint32_t delay_since_last_SR;
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};
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namespace voe {
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class ChannelSendInterface {
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public:
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virtual ~ChannelSendInterface() = default;
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virtual void ReceivedRTCPPacket(const uint8_t* packet, size_t length) = 0;
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virtual CallSendStatistics GetRTCPStatistics() const = 0;
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virtual void SetEncoder(int payload_type,
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std::unique_ptr<AudioEncoder> encoder) = 0;
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virtual void ModifyEncoder(
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rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) = 0;
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virtual void CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) = 0;
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virtual void SetLocalSSRC(uint32_t ssrc) = 0;
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// Use 0 to indicate that the extension should not be registered.
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virtual void SetRid(const std::string& rid,
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int extension_id,
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int repaired_extension_id) = 0;
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virtual void SetMid(const std::string& mid, int extension_id) = 0;
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virtual void SetRTCP_CNAME(absl::string_view c_name) = 0;
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virtual void SetExtmapAllowMixed(bool extmap_allow_mixed) = 0;
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virtual void SetSendAudioLevelIndicationStatus(bool enable, int id) = 0;
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virtual void EnableSendTransportSequenceNumber(int id) = 0;
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virtual void RegisterSenderCongestionControlObjects(
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RtpTransportControllerSendInterface* transport,
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RtcpBandwidthObserver* bandwidth_observer) = 0;
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virtual void ResetSenderCongestionControlObjects() = 0;
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virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const = 0;
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virtual ANAStats GetANAStatistics() const = 0;
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virtual void RegisterCngPayloadType(int payload_type,
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int payload_frequency) = 0;
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virtual void SetSendTelephoneEventPayloadType(int payload_type,
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int payload_frequency) = 0;
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virtual bool SendTelephoneEventOutband(int event, int duration_ms) = 0;
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virtual void OnBitrateAllocation(BitrateAllocationUpdate update) = 0;
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virtual int GetBitrate() const = 0;
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virtual void SetInputMute(bool muted) = 0;
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virtual void ProcessAndEncodeAudio(
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std::unique_ptr<AudioFrame> audio_frame) = 0;
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virtual RtpRtcp* GetRtpRtcp() const = 0;
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virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) = 0;
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virtual void OnRecoverableUplinkPacketLossRate(
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float recoverable_packet_loss_rate) = 0;
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// In RTP we currently rely on RTCP packets (|ReceivedRTCPPacket|) to inform
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// about RTT.
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// In media transport we rely on the TargetTransferRateObserver instead.
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// In other words, if you are using RTP, you should expect
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// |ReceivedRTCPPacket| to be called, if you are using media transport,
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// |OnTargetTransferRate| will be called.
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//
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// In future, RTP media will move to the media transport implementation and
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// these conditions will be removed.
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// Returns the RTT in milliseconds.
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virtual int64_t GetRTT() const = 0;
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virtual void StartSend() = 0;
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virtual void StopSend() = 0;
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// E2EE Custom Audio Frame Encryption (Optional)
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virtual void SetFrameEncryptor(
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rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) = 0;
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};
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std::unique_ptr<ChannelSendInterface> CreateChannelSend(
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Clock* clock,
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TaskQueueFactory* task_queue_factory,
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ProcessThread* module_process_thread,
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MediaTransportInterface* media_transport,
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OverheadObserver* overhead_observer,
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Transport* rtp_transport,
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RtcpRttStats* rtcp_rtt_stats,
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RtcEventLog* rtc_event_log,
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FrameEncryptorInterface* frame_encryptor,
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const webrtc::CryptoOptions& crypto_options,
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bool extmap_allow_mixed,
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int rtcp_report_interval_ms);
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} // namespace voe
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} // namespace webrtc
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#endif // AUDIO_CHANNEL_SEND_H_
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