OLD | NEW |
(Empty) | |
| 1 /* |
| 2 * Copyright 2017 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 #include "webrtc/pc/rtpreceivershim.h" |
| 12 |
| 13 #include "webrtc/base/checks.h" |
| 14 #include "webrtc/base/helpers.h" // For "CreateRandomX". |
| 15 |
| 16 namespace { |
| 17 |
| 18 static const int kDefaultVideoClockrate = 90000; |
| 19 |
| 20 void FillAudioReceiverParameters(webrtc::RtpParameters* parameters) { |
| 21 for (webrtc::RtpCodecParameters& codec : parameters->codecs) { |
| 22 if (!codec.num_channels) { |
| 23 codec.num_channels = rtc::Optional<int>(1); |
| 24 } |
| 25 } |
| 26 } |
| 27 |
| 28 void FillVideoReceiverParameters(webrtc::RtpParameters* parameters) { |
| 29 for (webrtc::RtpCodecParameters& codec : parameters->codecs) { |
| 30 if (!codec.clock_rate) { |
| 31 codec.clock_rate = rtc::Optional<int>(kDefaultVideoClockrate); |
| 32 } |
| 33 } |
| 34 } |
| 35 |
| 36 } // namespace |
| 37 |
| 38 namespace webrtc { |
| 39 |
| 40 BEGIN_OWNED_PROXY_MAP(OrtcRtpReceiver) |
| 41 PROXY_SIGNALING_THREAD_DESTRUCTOR() |
| 42 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, GetTrack) |
| 43 PROXY_METHOD1(RTCError, SetTransport, RtpTransportInterface*) |
| 44 PROXY_CONSTMETHOD0(RtpTransportInterface*, GetTransport) |
| 45 PROXY_METHOD1(RTCError, SetParameters, const RtpParameters&) |
| 46 PROXY_CONSTMETHOD0(RtpParameters, GetParameters) |
| 47 PROXY_CONSTMETHOD0(cricket::MediaType, GetKind) |
| 48 END_PROXY_MAP() |
| 49 |
| 50 // static |
| 51 RTCErrorOr<std::unique_ptr<OrtcRtpReceiverInterface>> |
| 52 RtpReceiverShim::CreateProxied(cricket::MediaType kind, |
| 53 const RtpParameters& parameters, |
| 54 RtpTransportShim* transport) { |
| 55 RTC_DCHECK(transport); |
| 56 RtpTransportControllerShim* rtp_transport_controller = |
| 57 transport->rtp_transport_controller(); |
| 58 // Call "attach" method to ensure more than one receiver of the same type |
| 59 // isn't attached to the same transport. |
| 60 RTCError err; |
| 61 switch (kind) { |
| 62 case cricket::MEDIA_TYPE_AUDIO: |
| 63 err = rtp_transport_controller->AttachAudioReceiver(transport); |
| 64 break; |
| 65 case cricket::MEDIA_TYPE_VIDEO: |
| 66 err = rtp_transport_controller->AttachVideoReceiver(transport); |
| 67 break; |
| 68 case cricket::MEDIA_TYPE_DATA: |
| 69 RTC_NOTREACHED(); |
| 70 } |
| 71 if (!err.ok()) { |
| 72 return err; |
| 73 } |
| 74 |
| 75 // Attempt to set parameters. |
| 76 std::unique_ptr<RtpReceiverShim> receiver_shim( |
| 77 new RtpReceiverShim(kind, transport, rtp_transport_controller)); |
| 78 err = receiver_shim->SetParameters(parameters); |
| 79 if (!err.ok()) { |
| 80 // Note: Destructor will automatically call "Detach" method. |
| 81 return err; |
| 82 } |
| 83 return OrtcRtpReceiverProxy::Create( |
| 84 rtp_transport_controller->signaling_thread(), |
| 85 rtp_transport_controller->worker_thread(), receiver_shim.release()); |
| 86 } |
| 87 |
| 88 RtpReceiverShim::~RtpReceiverShim() { |
| 89 internal_receiver_ = nullptr; |
| 90 // Need to detach from transport (was attached in Create method). |
| 91 switch (kind_) { |
| 92 case cricket::MEDIA_TYPE_AUDIO: |
| 93 rtp_transport_controller_->DetachAudioReceiver(); |
| 94 break; |
| 95 case cricket::MEDIA_TYPE_VIDEO: |
| 96 rtp_transport_controller_->DetachVideoReceiver(); |
| 97 break; |
| 98 case cricket::MEDIA_TYPE_DATA: |
| 99 RTC_NOTREACHED(); |
| 100 } |
| 101 } |
| 102 |
| 103 rtc::scoped_refptr<MediaStreamTrackInterface> RtpReceiverShim::GetTrack() |
| 104 const { |
| 105 return internal_receiver_->track(); |
| 106 } |
| 107 |
| 108 RTCError RtpReceiverShim::SetTransport(RtpTransportInterface* transport) { |
| 109 LOG(LS_ERROR) << "Changing the transport of an RtpReceiver is not yet " |
| 110 << "supported."; |
| 111 return RTCError(RTCErrorType::UNSUPPORTED_PARAMETER); |
| 112 } |
| 113 |
| 114 RtpTransportInterface* RtpReceiverShim::GetTransport() const { |
| 115 return transport_; |
| 116 } |
| 117 |
| 118 RTCError RtpReceiverShim::SetParameters(const RtpParameters& parameters) { |
| 119 RtpParameters filled_parameters = parameters; |
| 120 RTCError err; |
| 121 switch (kind_) { |
| 122 case cricket::MEDIA_TYPE_AUDIO: |
| 123 FillAudioReceiverParameters(&filled_parameters); |
| 124 err = rtp_transport_controller_->ValidateAndApplyAudioReceiverParameters( |
| 125 filled_parameters); |
| 126 if (!err.ok()) { |
| 127 return err; |
| 128 } |
| 129 break; |
| 130 case cricket::MEDIA_TYPE_VIDEO: |
| 131 FillVideoReceiverParameters(&filled_parameters); |
| 132 err = rtp_transport_controller_->ValidateAndApplyVideoReceiverParameters( |
| 133 filled_parameters); |
| 134 if (!err.ok()) { |
| 135 return err; |
| 136 } |
| 137 break; |
| 138 case cricket::MEDIA_TYPE_DATA: |
| 139 RTC_NOTREACHED(); |
| 140 return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR); |
| 141 } |
| 142 last_applied_parameters_ = filled_parameters; |
| 143 |
| 144 // Now that parameters were applied, can create (or recreate) the internal |
| 145 // receiver. |
| 146 // |
| 147 // This is analogous to a PeerConnection creating a receiver after |
| 148 // SetRemoteDescription is successful. |
| 149 MaybeRecreateInternalReceiver(); |
| 150 return RTCError(); |
| 151 } |
| 152 |
| 153 RtpParameters RtpReceiverShim::GetParameters() const { |
| 154 return last_applied_parameters_; |
| 155 } |
| 156 |
| 157 cricket::MediaType RtpReceiverShim::GetKind() const { |
| 158 return cricket::MediaTypeFromString(GetTrack()->kind()); |
| 159 } |
| 160 |
| 161 RtpReceiverShim::RtpReceiverShim( |
| 162 cricket::MediaType kind, |
| 163 RtpTransportShim* transport, |
| 164 RtpTransportControllerShim* rtp_transport_controller) |
| 165 : kind_(kind), |
| 166 transport_(transport), |
| 167 rtp_transport_controller_(rtp_transport_controller) {} |
| 168 |
| 169 void RtpReceiverShim::MaybeRecreateInternalReceiver() { |
| 170 if (last_applied_parameters_.encodings.empty()) { |
| 171 internal_receiver_ = nullptr; |
| 172 return; |
| 173 } |
| 174 // An SSRC of 0 is valid; this is used to identify "the default SSRC" (which |
| 175 // is the first one seen by the underlying media engine). |
| 176 uint32_t ssrc = 0; |
| 177 if (last_applied_parameters_.encodings[0].ssrc) { |
| 178 ssrc = *last_applied_parameters_.encodings[0].ssrc; |
| 179 } |
| 180 if (internal_receiver_ && ssrc == internal_receiver_->ssrc()) { |
| 181 // SSRC not changing; nothing to do. |
| 182 return; |
| 183 } |
| 184 internal_receiver_ = nullptr; |
| 185 switch (kind_) { |
| 186 case cricket::MEDIA_TYPE_AUDIO: |
| 187 internal_receiver_ = |
| 188 new AudioRtpReceiver(rtc::CreateRandomUuid(), ssrc, |
| 189 rtp_transport_controller_->voice_channel()); |
| 190 break; |
| 191 case cricket::MEDIA_TYPE_VIDEO: |
| 192 internal_receiver_ = new VideoRtpReceiver( |
| 193 rtc::CreateRandomUuid(), rtp_transport_controller_->worker_thread(), |
| 194 ssrc, rtp_transport_controller_->video_channel()); |
| 195 break; |
| 196 case cricket::MEDIA_TYPE_DATA: |
| 197 RTC_NOTREACHED(); |
| 198 } |
| 199 } |
| 200 |
| 201 } // namespace webrtc |
OLD | NEW |