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1 /* | 1 /* |
2 * Copyright (c) 2010 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2010 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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121 class FakeWebRtcVoiceEngine | 121 class FakeWebRtcVoiceEngine |
122 : public webrtc::VoEAudioProcessing, | 122 : public webrtc::VoEAudioProcessing, |
123 public webrtc::VoEBase, public webrtc::VoECodec, | 123 public webrtc::VoEBase, public webrtc::VoECodec, |
124 public webrtc::VoEHardware, | 124 public webrtc::VoEHardware, |
125 public webrtc::VoEVolumeControl { | 125 public webrtc::VoEVolumeControl { |
126 public: | 126 public: |
127 struct Channel { | 127 struct Channel { |
128 Channel() { | 128 Channel() { |
129 memset(&send_codec, 0, sizeof(send_codec)); | 129 memset(&send_codec, 0, sizeof(send_codec)); |
130 } | 130 } |
131 bool playout = false; | |
132 bool vad = false; | 131 bool vad = false; |
133 bool codec_fec = false; | 132 bool codec_fec = false; |
134 int max_encoding_bandwidth = 0; | 133 int max_encoding_bandwidth = 0; |
135 bool opus_dtx = false; | 134 bool opus_dtx = false; |
136 int cn8_type = 13; | 135 int cn8_type = 13; |
137 int cn16_type = 105; | 136 int cn16_type = 105; |
138 int associate_send_channel = -1; | 137 int associate_send_channel = -1; |
139 std::vector<webrtc::CodecInst> recv_codecs; | 138 std::vector<webrtc::CodecInst> recv_codecs; |
140 webrtc::CodecInst send_codec; | 139 webrtc::CodecInst send_codec; |
141 int neteq_capacity = -1; | 140 int neteq_capacity = -1; |
142 bool neteq_fast_accelerate = false; | 141 bool neteq_fast_accelerate = false; |
143 }; | 142 }; |
144 | 143 |
145 FakeWebRtcVoiceEngine() { | 144 FakeWebRtcVoiceEngine() { |
146 memset(&agc_config_, 0, sizeof(agc_config_)); | 145 memset(&agc_config_, 0, sizeof(agc_config_)); |
147 } | 146 } |
148 ~FakeWebRtcVoiceEngine() override { | 147 ~FakeWebRtcVoiceEngine() override { |
149 RTC_CHECK(channels_.empty()); | 148 RTC_CHECK(channels_.empty()); |
150 } | 149 } |
151 | 150 |
152 bool ec_metrics_enabled() const { return ec_metrics_enabled_; } | 151 bool ec_metrics_enabled() const { return ec_metrics_enabled_; } |
153 | 152 |
154 bool IsInited() const { return inited_; } | 153 bool IsInited() const { return inited_; } |
155 int GetLastChannel() const { return last_channel_; } | 154 int GetLastChannel() const { return last_channel_; } |
156 int GetNumChannels() const { return static_cast<int>(channels_.size()); } | 155 int GetNumChannels() const { return static_cast<int>(channels_.size()); } |
157 bool GetPlayout(int channel) { | |
158 return channels_[channel]->playout; | |
159 } | |
160 bool GetVAD(int channel) { | 156 bool GetVAD(int channel) { |
161 return channels_[channel]->vad; | 157 return channels_[channel]->vad; |
162 } | 158 } |
163 bool GetOpusDtx(int channel) { | 159 bool GetOpusDtx(int channel) { |
164 return channels_[channel]->opus_dtx; | 160 return channels_[channel]->opus_dtx; |
165 } | 161 } |
166 bool GetCodecFEC(int channel) { | 162 bool GetCodecFEC(int channel) { |
167 return channels_[channel]->codec_fec; | 163 return channels_[channel]->codec_fec; |
168 } | 164 } |
169 int GetMaxEncodingBandwidth(int channel) { | 165 int GetMaxEncodingBandwidth(int channel) { |
170 return channels_[channel]->max_encoding_bandwidth; | 166 return channels_[channel]->max_encoding_bandwidth; |
171 } | 167 } |
172 int GetSendCNPayloadType(int channel, bool wideband) { | 168 int GetSendCNPayloadType(int channel, bool wideband) { |
173 return (wideband) ? | 169 return (wideband) ? |
174 channels_[channel]->cn16_type : | 170 channels_[channel]->cn16_type : |
175 channels_[channel]->cn8_type; | 171 channels_[channel]->cn8_type; |
176 } | 172 } |
177 void set_playout_fail_channel(int channel) { | |
178 playout_fail_channel_ = channel; | |
179 } | |
180 void set_fail_create_channel(bool fail_create_channel) { | 173 void set_fail_create_channel(bool fail_create_channel) { |
181 fail_create_channel_ = fail_create_channel; | 174 fail_create_channel_ = fail_create_channel; |
182 } | 175 } |
183 int AddChannel(const webrtc::Config& config) { | 176 int AddChannel(const webrtc::Config& config) { |
184 if (fail_create_channel_) { | 177 if (fail_create_channel_) { |
185 return -1; | 178 return -1; |
186 } | 179 } |
187 Channel* ch = new Channel(); | 180 Channel* ch = new Channel(); |
188 auto db = webrtc::acm2::RentACodec::Database(); | 181 auto db = webrtc::acm2::RentACodec::Database(); |
189 ch->recv_codecs.assign(db.begin(), db.end()); | 182 ch->recv_codecs.assign(db.begin(), db.end()); |
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238 for (const auto& ch : channels_) { | 231 for (const auto& ch : channels_) { |
239 if (ch.second->associate_send_channel == channel) { | 232 if (ch.second->associate_send_channel == channel) { |
240 ch.second->associate_send_channel = -1; | 233 ch.second->associate_send_channel = -1; |
241 } | 234 } |
242 } | 235 } |
243 delete channels_[channel]; | 236 delete channels_[channel]; |
244 channels_.erase(channel); | 237 channels_.erase(channel); |
245 return 0; | 238 return 0; |
246 } | 239 } |
247 WEBRTC_STUB(StartReceive, (int channel)); | 240 WEBRTC_STUB(StartReceive, (int channel)); |
248 WEBRTC_FUNC(StartPlayout, (int channel)) { | 241 WEBRTC_STUB(StartPlayout, (int channel)); |
249 if (playout_fail_channel_ != channel) { | |
250 WEBRTC_CHECK_CHANNEL(channel); | |
251 channels_[channel]->playout = true; | |
252 return 0; | |
253 } else { | |
254 // When playout_fail_channel_ == channel, fail the StartPlayout on this | |
255 // channel. | |
256 return -1; | |
257 } | |
258 } | |
259 WEBRTC_STUB(StartSend, (int channel)); | 242 WEBRTC_STUB(StartSend, (int channel)); |
260 WEBRTC_STUB(StopReceive, (int channel)); | 243 WEBRTC_STUB(StopReceive, (int channel)); |
261 WEBRTC_FUNC(StopPlayout, (int channel)) { | 244 WEBRTC_STUB(StopPlayout, (int channel)); |
262 WEBRTC_CHECK_CHANNEL(channel); | |
263 channels_[channel]->playout = false; | |
264 return 0; | |
265 } | |
266 WEBRTC_STUB(StopSend, (int channel)); | 245 WEBRTC_STUB(StopSend, (int channel)); |
267 WEBRTC_STUB(GetVersion, (char version[1024])); | 246 WEBRTC_STUB(GetVersion, (char version[1024])); |
268 WEBRTC_STUB(LastError, ()); | 247 WEBRTC_STUB(LastError, ()); |
269 WEBRTC_FUNC(AssociateSendChannel, (int channel, | 248 WEBRTC_FUNC(AssociateSendChannel, (int channel, |
270 int accociate_send_channel)) { | 249 int accociate_send_channel)) { |
271 WEBRTC_CHECK_CHANNEL(channel); | 250 WEBRTC_CHECK_CHANNEL(channel); |
272 channels_[channel]->associate_send_channel = accociate_send_channel; | 251 channels_[channel]->associate_send_channel = accociate_send_channel; |
273 return 0; | 252 return 0; |
274 } | 253 } |
275 | 254 |
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293 WEBRTC_CHECK_CHANNEL(channel); | 272 WEBRTC_CHECK_CHANNEL(channel); |
294 codec = channels_[channel]->send_codec; | 273 codec = channels_[channel]->send_codec; |
295 return 0; | 274 return 0; |
296 } | 275 } |
297 WEBRTC_STUB(SetBitRate, (int channel, int bitrate_bps)); | 276 WEBRTC_STUB(SetBitRate, (int channel, int bitrate_bps)); |
298 WEBRTC_STUB(GetRecCodec, (int channel, webrtc::CodecInst& codec)); | 277 WEBRTC_STUB(GetRecCodec, (int channel, webrtc::CodecInst& codec)); |
299 WEBRTC_FUNC(SetRecPayloadType, (int channel, | 278 WEBRTC_FUNC(SetRecPayloadType, (int channel, |
300 const webrtc::CodecInst& codec)) { | 279 const webrtc::CodecInst& codec)) { |
301 WEBRTC_CHECK_CHANNEL(channel); | 280 WEBRTC_CHECK_CHANNEL(channel); |
302 Channel* ch = channels_[channel]; | 281 Channel* ch = channels_[channel]; |
303 if (ch->playout) | |
304 return -1; // Channel is in use. | |
305 // Check if something else already has this slot. | 282 // Check if something else already has this slot. |
306 if (codec.pltype != -1) { | 283 if (codec.pltype != -1) { |
307 for (std::vector<webrtc::CodecInst>::iterator it = | 284 for (std::vector<webrtc::CodecInst>::iterator it = |
308 ch->recv_codecs.begin(); it != ch->recv_codecs.end(); ++it) { | 285 ch->recv_codecs.begin(); it != ch->recv_codecs.end(); ++it) { |
309 if (it->pltype == codec.pltype && | 286 if (it->pltype == codec.pltype && |
310 _stricmp(it->plname, codec.plname) != 0) { | 287 _stricmp(it->plname, codec.plname) != 0) { |
311 return -1; | 288 return -1; |
312 } | 289 } |
313 } | 290 } |
314 } | 291 } |
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576 bool ns_enabled_ = false; | 553 bool ns_enabled_ = false; |
577 bool agc_enabled_ = false; | 554 bool agc_enabled_ = false; |
578 bool highpass_filter_enabled_ = false; | 555 bool highpass_filter_enabled_ = false; |
579 bool stereo_swapping_enabled_ = false; | 556 bool stereo_swapping_enabled_ = false; |
580 bool typing_detection_enabled_ = false; | 557 bool typing_detection_enabled_ = false; |
581 webrtc::EcModes ec_mode_ = webrtc::kEcDefault; | 558 webrtc::EcModes ec_mode_ = webrtc::kEcDefault; |
582 webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone; | 559 webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone; |
583 webrtc::NsModes ns_mode_ = webrtc::kNsDefault; | 560 webrtc::NsModes ns_mode_ = webrtc::kNsDefault; |
584 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; | 561 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; |
585 webrtc::AgcConfig agc_config_; | 562 webrtc::AgcConfig agc_config_; |
586 int playout_fail_channel_ = -1; | |
587 FakeAudioProcessing audio_processing_; | 563 FakeAudioProcessing audio_processing_; |
588 }; | 564 }; |
589 | 565 |
590 } // namespace cricket | 566 } // namespace cricket |
591 | 567 |
592 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ | 568 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ |
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