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/*
* 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/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-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;
};
// 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 GetBitrate() 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);
} // namespace voe
} // namespace webrtc
#endif // AUDIO_CHANNEL_SEND_H_
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