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Side by Side Diff: webrtc/media/engine/fakewebrtcvoiceengine.h

Issue 2307533004: Moving/renaming webrtc/common.h. (Closed)
Patch Set: rebase Created 4 years, 3 months ago
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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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147 } 147 }
148 bool vad = false; 148 bool vad = false;
149 bool codec_fec = false; 149 bool codec_fec = false;
150 int max_encoding_bandwidth = 0; 150 int max_encoding_bandwidth = 0;
151 bool opus_dtx = false; 151 bool opus_dtx = false;
152 int cn8_type = 13; 152 int cn8_type = 13;
153 int cn16_type = 105; 153 int cn16_type = 105;
154 int associate_send_channel = -1; 154 int associate_send_channel = -1;
155 std::vector<webrtc::CodecInst> recv_codecs; 155 std::vector<webrtc::CodecInst> recv_codecs;
156 webrtc::CodecInst send_codec; 156 webrtc::CodecInst send_codec;
157 int neteq_capacity = -1; 157 size_t neteq_capacity = 0;
158 bool neteq_fast_accelerate = false; 158 bool neteq_fast_accelerate = false;
159 }; 159 };
160 160
161 FakeWebRtcVoiceEngine() { 161 FakeWebRtcVoiceEngine() {
162 memset(&agc_config_, 0, sizeof(agc_config_)); 162 memset(&agc_config_, 0, sizeof(agc_config_));
163 } 163 }
164 ~FakeWebRtcVoiceEngine() override { 164 ~FakeWebRtcVoiceEngine() override {
165 RTC_CHECK(channels_.empty()); 165 RTC_CHECK(channels_.empty());
166 } 166 }
167 167
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183 return channels_[channel]->max_encoding_bandwidth; 183 return channels_[channel]->max_encoding_bandwidth;
184 } 184 }
185 int GetSendCNPayloadType(int channel, bool wideband) { 185 int GetSendCNPayloadType(int channel, bool wideband) {
186 return (wideband) ? 186 return (wideband) ?
187 channels_[channel]->cn16_type : 187 channels_[channel]->cn16_type :
188 channels_[channel]->cn8_type; 188 channels_[channel]->cn8_type;
189 } 189 }
190 void set_fail_create_channel(bool fail_create_channel) { 190 void set_fail_create_channel(bool fail_create_channel) {
191 fail_create_channel_ = fail_create_channel; 191 fail_create_channel_ = fail_create_channel;
192 } 192 }
193 int AddChannel(const webrtc::Config& config) {
194 if (fail_create_channel_) {
195 return -1;
196 }
197 Channel* ch = new Channel();
198 auto db = webrtc::acm2::RentACodec::Database();
199 ch->recv_codecs.assign(db.begin(), db.end());
200 if (config.Get<webrtc::NetEqCapacityConfig>().enabled) {
201 ch->neteq_capacity = config.Get<webrtc::NetEqCapacityConfig>().capacity;
202 }
203 ch->neteq_fast_accelerate =
204 config.Get<webrtc::NetEqFastAccelerate>().enabled;
205 channels_[++last_channel_] = ch;
206 return last_channel_;
207 }
208 193
209 int GetNumSetSendCodecs() const { return num_set_send_codecs_; } 194 int GetNumSetSendCodecs() const { return num_set_send_codecs_; }
210 195
211 int GetAssociateSendChannel(int channel) { 196 int GetAssociateSendChannel(int channel) {
212 return channels_[channel]->associate_send_channel; 197 return channels_[channel]->associate_send_channel;
213 } 198 }
214 199
215 WEBRTC_STUB(Release, ()); 200 WEBRTC_STUB(Release, ());
216 201
217 // webrtc::VoEBase 202 // webrtc::VoEBase
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230 inited_ = false; 215 inited_ = false;
231 return 0; 216 return 0;
232 } 217 }
233 webrtc::AudioProcessing* audio_processing() override { 218 webrtc::AudioProcessing* audio_processing() override {
234 return &audio_processing_; 219 return &audio_processing_;
235 } 220 }
236 webrtc::AudioDeviceModule* audio_device_module() override { 221 webrtc::AudioDeviceModule* audio_device_module() override {
237 return nullptr; 222 return nullptr;
238 } 223 }
239 WEBRTC_FUNC(CreateChannel, ()) { 224 WEBRTC_FUNC(CreateChannel, ()) {
240 webrtc::Config empty_config; 225 return CreateChannel(webrtc::VoEBase::ChannelConfig());
241 return AddChannel(empty_config);
242 } 226 }
243 WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) { 227 WEBRTC_FUNC(CreateChannel, (const webrtc::VoEBase::ChannelConfig& config)) {
244 return AddChannel(config); 228 if (fail_create_channel_) {
229 return -1;
230 }
231 Channel* ch = new Channel();
232 auto db = webrtc::acm2::RentACodec::Database();
233 ch->recv_codecs.assign(db.begin(), db.end());
234 ch->neteq_capacity = config.acm_config.neteq_config.max_packets_in_buffer;
235 ch->neteq_fast_accelerate =
236 config.acm_config.neteq_config.enable_fast_accelerate;
237 channels_[++last_channel_] = ch;
238 return last_channel_;
245 } 239 }
246 WEBRTC_FUNC(DeleteChannel, (int channel)) { 240 WEBRTC_FUNC(DeleteChannel, (int channel)) {
247 WEBRTC_CHECK_CHANNEL(channel); 241 WEBRTC_CHECK_CHANNEL(channel);
248 for (const auto& ch : channels_) { 242 for (const auto& ch : channels_) {
249 if (ch.second->associate_send_channel == channel) { 243 if (ch.second->associate_send_channel == channel) {
250 ch.second->associate_send_channel = -1; 244 ch.second->associate_send_channel = -1;
251 } 245 }
252 } 246 }
253 delete channels_[channel]; 247 delete channels_[channel];
254 channels_.erase(channel); 248 channels_.erase(channel);
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540 } 534 }
541 bool IsHighPassFilterEnabled() override { 535 bool IsHighPassFilterEnabled() override {
542 return highpass_filter_enabled_; 536 return highpass_filter_enabled_;
543 } 537 }
544 bool IsStereoChannelSwappingEnabled() override { 538 bool IsStereoChannelSwappingEnabled() override {
545 return stereo_swapping_enabled_; 539 return stereo_swapping_enabled_;
546 } 540 }
547 void EnableStereoChannelSwapping(bool enable) override { 541 void EnableStereoChannelSwapping(bool enable) override {
548 stereo_swapping_enabled_ = enable; 542 stereo_swapping_enabled_ = enable;
549 } 543 }
550 int GetNetEqCapacity() const { 544 size_t GetNetEqCapacity() const {
551 auto ch = channels_.find(last_channel_); 545 auto ch = channels_.find(last_channel_);
552 ASSERT(ch != channels_.end()); 546 ASSERT(ch != channels_.end());
553 return ch->second->neteq_capacity; 547 return ch->second->neteq_capacity;
554 } 548 }
555 bool GetNetEqFastAccelerate() const { 549 bool GetNetEqFastAccelerate() const {
556 auto ch = channels_.find(last_channel_); 550 auto ch = channels_.find(last_channel_);
557 ASSERT(ch != channels_.end()); 551 ASSERT(ch != channels_.end());
558 return ch->second->neteq_fast_accelerate; 552 return ch->second->neteq_fast_accelerate;
559 } 553 }
560 554
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576 webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone; 570 webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone;
577 webrtc::NsModes ns_mode_ = webrtc::kNsDefault; 571 webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
578 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; 572 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
579 webrtc::AgcConfig agc_config_; 573 webrtc::AgcConfig agc_config_;
580 FakeAudioProcessing audio_processing_; 574 FakeAudioProcessing audio_processing_;
581 }; 575 };
582 576
583 } // namespace cricket 577 } // namespace cricket
584 578
585 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ 579 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
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