1 /*
2  *  Copyright (c) 2004 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 
11 #ifndef MEDIA_BASE_MEDIA_ENGINE_H_
12 #define MEDIA_BASE_MEDIA_ENGINE_H_
13 
14 #include <memory>
15 #include <string>
16 #include <vector>
17 
18 #include "api/audio_codecs/audio_decoder_factory.h"
19 #include "api/audio_codecs/audio_encoder_factory.h"
20 #include "api/crypto/crypto_options.h"
21 #include "api/rtp_parameters.h"
22 #include "api/transport/webrtc_key_value_config.h"
23 #include "api/video/video_bitrate_allocator_factory.h"
24 #include "call/audio_state.h"
25 #include "media/base/codec.h"
26 #include "media/base/media_channel.h"
27 #include "media/base/video_common.h"
28 #include "rtc_base/system/file_wrapper.h"
29 
30 namespace webrtc {
31 class AudioDeviceModule;
32 class AudioMixer;
33 class AudioProcessing;
34 class Call;
35 }  // namespace webrtc
36 
37 namespace cricket {
38 
39 webrtc::RTCError CheckRtpParametersValues(
40     const webrtc::RtpParameters& new_parameters);
41 
42 webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
43     const webrtc::RtpParameters& old_parameters,
44     const webrtc::RtpParameters& new_parameters);
45 
46 struct RtpCapabilities {
47   RtpCapabilities();
48   ~RtpCapabilities();
49   std::vector<webrtc::RtpExtension> header_extensions;
50 };
51 
52 class RtpHeaderExtensionQueryInterface {
53  public:
54   virtual ~RtpHeaderExtensionQueryInterface() = default;
55 
56   // Returns a vector of RtpHeaderExtensionCapability, whose direction is
57   // kStopped if the extension is stopped (not used) by default.
58   virtual std::vector<webrtc::RtpHeaderExtensionCapability>
59   GetRtpHeaderExtensions() const = 0;
60 };
61 
62 class VoiceEngineInterface : public RtpHeaderExtensionQueryInterface {
63  public:
64   VoiceEngineInterface() = default;
65   virtual ~VoiceEngineInterface() = default;
66   RTC_DISALLOW_COPY_AND_ASSIGN(VoiceEngineInterface);
67 
68   // Initialization
69   // Starts the engine.
70   virtual void Init() = 0;
71 
72   // TODO(solenberg): Remove once VoE API refactoring is done.
73   virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
74 
75   // MediaChannel creation
76   // Creates a voice media channel. Returns NULL on failure.
77   virtual VoiceMediaChannel* CreateMediaChannel(
78       webrtc::Call* call,
79       const MediaConfig& config,
80       const AudioOptions& options,
81       const webrtc::CryptoOptions& crypto_options) = 0;
82 
83   virtual const std::vector<AudioCodec>& send_codecs() const = 0;
84   virtual const std::vector<AudioCodec>& recv_codecs() const = 0;
85 
86   // Starts AEC dump using existing file, a maximum file size in bytes can be
87   // specified. Logging is stopped just before the size limit is exceeded.
88   // If max_size_bytes is set to a value <= 0, no limit will be used.
89   virtual bool StartAecDump(webrtc::FileWrapper file,
90                             int64_t max_size_bytes) = 0;
91 
92   // Stops recording AEC dump.
93   virtual void StopAecDump() = 0;
94 };
95 
96 class VideoEngineInterface : public RtpHeaderExtensionQueryInterface {
97  public:
98   VideoEngineInterface() = default;
99   virtual ~VideoEngineInterface() = default;
100   RTC_DISALLOW_COPY_AND_ASSIGN(VideoEngineInterface);
101 
102   // Creates a video media channel, paired with the specified voice channel.
103   // Returns NULL on failure.
104   virtual VideoMediaChannel* CreateMediaChannel(
105       webrtc::Call* call,
106       const MediaConfig& config,
107       const VideoOptions& options,
108       const webrtc::CryptoOptions& crypto_options,
109       webrtc::VideoBitrateAllocatorFactory*
110           video_bitrate_allocator_factory) = 0;
111 
112   virtual std::vector<VideoCodec> send_codecs() const = 0;
113   virtual std::vector<VideoCodec> recv_codecs() const = 0;
114 };
115 
116 // MediaEngineInterface is an abstraction of a media engine which can be
117 // subclassed to support different media componentry backends.
118 // It supports voice and video operations in the same class to facilitate
119 // proper synchronization between both media types.
120 class MediaEngineInterface {
121  public:
~MediaEngineInterface()122   virtual ~MediaEngineInterface() {}
123 
124   // Initialization
125   // Starts the engine.
126   virtual bool Init() = 0;
127   virtual VoiceEngineInterface& voice() = 0;
128   virtual VideoEngineInterface& video() = 0;
129   virtual const VoiceEngineInterface& voice() const = 0;
130   virtual const VideoEngineInterface& video() const = 0;
131 };
132 
133 // CompositeMediaEngine constructs a MediaEngine from separate
134 // voice and video engine classes.
135 // Optionally owns a WebRtcKeyValueConfig trials map.
136 class CompositeMediaEngine : public MediaEngineInterface {
137  public:
138   CompositeMediaEngine(std::unique_ptr<webrtc::WebRtcKeyValueConfig> trials,
139                        std::unique_ptr<VoiceEngineInterface> audio_engine,
140                        std::unique_ptr<VideoEngineInterface> video_engine);
141   CompositeMediaEngine(std::unique_ptr<VoiceEngineInterface> audio_engine,
142                        std::unique_ptr<VideoEngineInterface> video_engine);
143   ~CompositeMediaEngine() override;
144   bool Init() override;
145 
146   VoiceEngineInterface& voice() override;
147   VideoEngineInterface& video() override;
148   const VoiceEngineInterface& voice() const override;
149   const VideoEngineInterface& video() const override;
150 
151  private:
152   const std::unique_ptr<webrtc::WebRtcKeyValueConfig> trials_;
153   std::unique_ptr<VoiceEngineInterface> voice_engine_;
154   std::unique_ptr<VideoEngineInterface> video_engine_;
155 };
156 
157 enum DataChannelType {
158   DCT_NONE = 0,
159   DCT_RTP = 1,
160   DCT_SCTP = 2,
161 };
162 
163 class DataEngineInterface {
164  public:
~DataEngineInterface()165   virtual ~DataEngineInterface() {}
166   virtual DataMediaChannel* CreateChannel(const MediaConfig& config) = 0;
167   virtual const std::vector<DataCodec>& data_codecs() = 0;
168 };
169 
170 webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
171 webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp);
172 
173 // Returns a vector of RTP extensions as visible from RtpSender/Receiver
174 // GetCapabilities(). The returned vector only shows what will definitely be
175 // offered by default, i.e. the list of extensions returned from
176 // GetRtpHeaderExtensions() that are not kStopped.
177 std::vector<webrtc::RtpExtension> GetDefaultEnabledRtpHeaderExtensions(
178     const RtpHeaderExtensionQueryInterface& query_interface);
179 
180 }  // namespace cricket
181 
182 #endif  // MEDIA_BASE_MEDIA_ENGINE_H_
183