diff options
Diffstat (limited to 'third_party/libwebrtc/audio/channel_send.h')
| -rw-r--r-- | third_party/libwebrtc/audio/channel_send.h | 148 |
1 files changed, 148 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/channel_send.h b/third_party/libwebrtc/audio/channel_send.h new file mode 100644 index 0000000000..9b3969161c --- /dev/null +++ b/third_party/libwebrtc/audio/channel_send.h @@ -0,0 +1,148 @@ +/* + * 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 <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/field_trials_view.h" +#include "api/frame_transformer_interface.h" +#include "api/function_view.h" +#include "api/task_queue/task_queue_factory.h" +#include "modules/rtp_rtcp/include/report_block_data.h" +#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" +#include "modules/rtp_rtcp/source/rtp_sender_audio.h" + +namespace webrtc { + +class FrameEncryptorInterface; +class RtcEventLog; +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; + RtcpPacketTypeCounter rtcp_packet_type_counts; + // 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_rcvd; +}; + +// See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt 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 { + 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, + 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, + RtcpBandwidthObserver* bandwidth_observer) = 0; + virtual void ResetSenderCongestionControlObjects() = 0; + virtual std::vector<ReportBlock> 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; + + // 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; + + // 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; +}; + +std::unique_ptr<ChannelSendInterface> CreateChannelSend( + Clock* clock, + TaskQueueFactory* task_queue_factory, + 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, + uint32_t ssrc, + rtc::scoped_refptr<FrameTransformerInterface> frame_transformer, + TransportFeedbackObserver* feedback_observer, + const FieldTrialsView& field_trials); + +} // namespace voe +} // namespace webrtc + +#endif // AUDIO_CHANNEL_SEND_H_ |