1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
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_
|
