| /* |
| * 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. |
| */ |
| |
| #ifndef AUDIO_CHANNEL_SEND_H_ |
| #define AUDIO_CHANNEL_SEND_H_ |
| |
| #include <cstddef> |
| #include <cstdint> |
| #include <memory> |
| #include <optional> |
| #include <vector> |
| |
| #include "absl/strings/string_view.h" |
| #include "api/audio/audio_frame.h" |
| #include "api/audio_codecs/audio_encoder.h" |
| #include "api/audio_codecs/audio_format.h" |
| #include "api/call/bitrate_allocation.h" |
| #include "api/crypto/crypto_options.h" |
| #include "api/environment/environment.h" |
| #include "api/frame_transformer_interface.h" |
| #include "api/function_view.h" |
| #include "api/scoped_refptr.h" |
| #include "api/units/data_rate.h" |
| #include "api/units/time_delta.h" |
| #include "modules/rtp_rtcp/include/report_block_data.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" |
| |
| namespace webrtc { |
| |
| class FrameEncryptorInterface; |
| class RtpTransportControllerSendInterface; |
| |
| struct CallSendStatistics { |
| int64_t rttMs; |
| int64_t payload_bytes_sent; |
| int64_t header_and_padding_bytes_sent; |
| // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent |
| uint64_t retransmitted_bytes_sent; |
| int packetsSent; |
| // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay |
| TimeDelta total_packet_send_delay = TimeDelta::Zero(); |
| // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent |
| uint64_t retransmitted_packets_sent; |
| // A snapshot of Report Blocks with additional data of interest to statistics. |
| // Within this list, the sender-source SSRC pair is unique and per-pair the |
| // ReportBlockData represents the latest Report Block that was received for |
| // that pair. |
| std::vector<ReportBlockData> report_block_datas; |
| uint32_t nacks_received; |
| }; |
| |
| namespace voe { |
| |
| class ChannelSendInterface { |
| public: |
| virtual ~ChannelSendInterface() = default; |
| |
| virtual void ReceivedRTCPPacket(const uint8_t* packet, size_t length) = 0; |
| |
| virtual CallSendStatistics GetRTCPStatistics() const = 0; |
| |
| virtual void SetEncoder(int payload_type, |
| const SdpAudioFormat& encoder_format, |
| 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; |
| |
| // Use 0 to indicate that the extension should not be registered. |
| virtual void SetRTCP_CNAME(absl::string_view c_name) = 0; |
| virtual void SetSendAudioLevelIndicationStatus(bool enable, int id) = 0; |
| virtual void RegisterSenderCongestionControlObjects( |
| RtpTransportControllerSendInterface* transport) = 0; |
| virtual void ResetSenderCongestionControlObjects() = 0; |
| virtual std::vector<ReportBlockData> GetRemoteRTCPReportBlocks() const = 0; |
| virtual ANAStats GetANAStatistics() const = 0; |
| virtual void RegisterCngPayloadType(int payload_type, |
| int payload_frequency) = 0; |
| virtual void 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 GetTargetBitrate() const = 0; |
| virtual void SetInputMute(bool muted) = 0; |
| |
| virtual void ProcessAndEncodeAudio( |
| std::unique_ptr<AudioFrame> audio_frame) = 0; |
| virtual RtpRtcpInterface* GetRtpRtcp() 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; |
| |
| // Sets a frame transformer between encoder and packetizer, to transform |
| // encoded frames before sending them out the network. |
| virtual void SetEncoderToPacketizerFrameTransformer( |
| rtc::scoped_refptr<webrtc::FrameTransformerInterface> |
| frame_transformer) = 0; |
| |
| // Returns payload bitrate actually used. |
| virtual std::optional<DataRate> GetUsedRate() const = 0; |
| |
| // Registers per packet byte overhead. |
| virtual void RegisterPacketOverhead(int packet_byte_overhead) = 0; |
| }; |
| |
| std::unique_ptr<ChannelSendInterface> CreateChannelSend( |
| const Environment& env, |
| Transport* rtp_transport, |
| RtcpRttStats* rtcp_rtt_stats, |
| FrameEncryptorInterface* frame_encryptor, |
| const webrtc::CryptoOptions& crypto_options, |
| bool extmap_allow_mixed, |
| int rtcp_report_interval_ms, |
| uint32_t ssrc, |
| rtc::scoped_refptr<FrameTransformerInterface> frame_transformer, |
| RtpTransportControllerSendInterface* transport_controller); |
| |
| } // namespace voe |
| } // namespace webrtc |
| |
| #endif // AUDIO_CHANNEL_SEND_H_ |