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