1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10 #ifndef CALL_CALL_H_
11 #define CALL_CALL_H_
12
13 #include <algorithm>
14 #include <memory>
15 #include <string>
16 #include <vector>
17
18 #include "api/rtcerror.h"
19 #include "call/audio_receive_stream.h"
20 #include "call/audio_send_stream.h"
21 #include "call/audio_state.h"
22 #include "call/flexfec_receive_stream.h"
23 #include "call/rtp_transport_controller_send_interface.h"
24 #include "call/video_receive_stream.h"
25 #include "call/video_send_stream.h"
26 #include "common_types.h" // NOLINT(build/include)
27 #include "rtc_base/bitrateallocationstrategy.h"
28 #include "rtc_base/networkroute.h"
29 #include "rtc_base/platform_file.h"
30 #include "rtc_base/socket.h"
31
32 namespace webrtc {
33
34 class AudioProcessing;
35 class RtcEventLog;
36
37 enum class MediaType {
38 ANY,
39 AUDIO,
40 VIDEO,
41 DATA
42 };
43
44 // Like std::min, but considers non-positive values to be unset.
45 // TODO(zstein): Remove once all callers use rtc::Optional.
46 template <typename T>
MinPositive(T a,T b)47 static T MinPositive(T a, T b) {
48 if (a <= 0) {
49 return b;
50 }
51 if (b <= 0) {
52 return a;
53 }
54 return std::min(a, b);
55 }
56
57 class PacketReceiver {
58 public:
59 enum DeliveryStatus {
60 DELIVERY_OK,
61 DELIVERY_UNKNOWN_SSRC,
62 DELIVERY_PACKET_ERROR,
63 };
64
65 virtual DeliveryStatus DeliverPacket(MediaType media_type,
66 const uint8_t* packet,
67 size_t length,
68 const PacketTime& packet_time) = 0;
69
70 protected:
~PacketReceiver()71 virtual ~PacketReceiver() {}
72 };
73
74 // A Call instance can contain several send and/or receive streams. All streams
75 // are assumed to have the same remote endpoint and will share bitrate estimates
76 // etc.
77 class Call {
78 public:
79 struct Config {
ConfigConfig80 explicit Config(RtcEventLog* event_log) : event_log(event_log) {
81 RTC_DCHECK(event_log);
82 }
83
84 static constexpr int kDefaultStartBitrateBps = 300000;
85
86 // Bitrate config used until valid bitrate estimates are calculated. Also
87 // used to cap total bitrate used. This comes from the remote connection.
88 struct BitrateConfig {
89 int min_bitrate_bps = 0;
90 int start_bitrate_bps = kDefaultStartBitrateBps;
91 int max_bitrate_bps = -1;
92 } bitrate_config;
93
94 // The local client's bitrate preferences. The actual configuration used
95 // is a combination of this and |bitrate_config|. The combination is
96 // currently more complicated than a simple mask operation (see
97 // SetBitrateConfig and SetBitrateConfigMask). Assumes that 0 <= min <=
98 // start <= max holds for set parameters.
99 struct BitrateConfigMask {
100 rtc::Optional<int> min_bitrate_bps;
101 rtc::Optional<int> start_bitrate_bps;
102 rtc::Optional<int> max_bitrate_bps;
103 };
104
105 // AudioState which is possibly shared between multiple calls.
106 // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
107 rtc::scoped_refptr<AudioState> audio_state;
108
109 // Audio Processing Module to be used in this call.
110 // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
111 AudioProcessing* audio_processing = nullptr;
112
113 // RtcEventLog to use for this call. Required.
114 // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
115 RtcEventLog* event_log = nullptr;
116 };
117
118 struct Stats {
119 std::string ToString(int64_t time_ms) const;
120
121 int send_bandwidth_bps = 0; // Estimated available send bandwidth.
122 int max_padding_bitrate_bps = 0; // Cumulative configured max padding.
123 int recv_bandwidth_bps = 0; // Estimated available receive bandwidth.
124 int64_t pacer_delay_ms = 0;
125 int64_t rtt_ms = -1;
126 };
127
128 static Call* Create(const Call::Config& config);
129
130 // Allows mocking |transport_send| for testing.
131 static Call* Create(
132 const Call::Config& config,
133 std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
134
135 virtual AudioSendStream* CreateAudioSendStream(
136 const AudioSendStream::Config& config) = 0;
137 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
138
139 virtual AudioReceiveStream* CreateAudioReceiveStream(
140 const AudioReceiveStream::Config& config) = 0;
141 virtual void DestroyAudioReceiveStream(
142 AudioReceiveStream* receive_stream) = 0;
143
144 virtual VideoSendStream* CreateVideoSendStream(
145 VideoSendStream::Config config,
146 VideoEncoderConfig encoder_config) = 0;
147 virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
148
149 virtual VideoReceiveStream* CreateVideoReceiveStream(
150 VideoReceiveStream::Config configuration) = 0;
151 virtual void DestroyVideoReceiveStream(
152 VideoReceiveStream* receive_stream) = 0;
153
154 // In order for a created VideoReceiveStream to be aware that it is
155 // protected by a FlexfecReceiveStream, the latter should be created before
156 // the former.
157 virtual FlexfecReceiveStream* CreateFlexfecReceiveStream(
158 const FlexfecReceiveStream::Config& config) = 0;
159 virtual void DestroyFlexfecReceiveStream(
160 FlexfecReceiveStream* receive_stream) = 0;
161
162 // All received RTP and RTCP packets for the call should be inserted to this
163 // PacketReceiver. The PacketReceiver pointer is valid as long as the
164 // Call instance exists.
165 virtual PacketReceiver* Receiver() = 0;
166
167 // Returns the call statistics, such as estimated send and receive bandwidth,
168 // pacing delay, etc.
169 virtual Stats GetStats() const = 0;
170
171 // The greater min and smaller max set by this and SetBitrateConfigMask will
172 // be used. The latest non-negative start value from either call will be used.
173 // Specifying a start bitrate (>0) will reset the current bitrate estimate.
174 // This is due to how the 'x-google-start-bitrate' flag is currently
175 // implemented. Passing -1 leaves the start bitrate unchanged. Behavior is not
176 // guaranteed for other negative values or 0.
177 virtual void SetBitrateConfig(
178 const Config::BitrateConfig& bitrate_config) = 0;
179
180 // The greater min and smaller max set by this and SetBitrateConfig will be
181 // used. The latest non-negative start value form either call will be used.
182 // Specifying a start bitrate will reset the current bitrate estimate.
183 // Assumes 0 <= min <= start <= max holds for set parameters.
184 virtual void SetBitrateConfigMask(
185 const Config::BitrateConfigMask& bitrate_mask) = 0;
186
187 virtual void SetBitrateAllocationStrategy(
188 std::unique_ptr<rtc::BitrateAllocationStrategy>
189 bitrate_allocation_strategy) = 0;
190
191 // TODO(skvlad): When the unbundled case with multiple streams for the same
192 // media type going over different networks is supported, track the state
193 // for each stream separately. Right now it's global per media type.
194 virtual void SignalChannelNetworkState(MediaType media,
195 NetworkState state) = 0;
196
197 virtual void OnTransportOverheadChanged(
198 MediaType media,
199 int transport_overhead_per_packet) = 0;
200
201 virtual void OnNetworkRouteChanged(
202 const std::string& transport_name,
203 const rtc::NetworkRoute& network_route) = 0;
204
205 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
206
207 virtual VoiceEngine* voice_engine() = 0;
208
~Call()209 virtual ~Call() {}
210 };
211
212 } // namespace webrtc
213
214 #endif // CALL_CALL_H_
215