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

Issue 1909333002: Switch voice transport to use Call and Stream instead of VoENetwork. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed coments on ps#6 Created 4 years, 7 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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196 return channels_[channel]->last_rtp_packet_time; 196 return channels_[channel]->last_rtp_packet_time;
197 } 197 }
198 int GetSendCNPayloadType(int channel, bool wideband) { 198 int GetSendCNPayloadType(int channel, bool wideband) {
199 return (wideband) ? 199 return (wideband) ?
200 channels_[channel]->cn16_type : 200 channels_[channel]->cn16_type :
201 channels_[channel]->cn8_type; 201 channels_[channel]->cn8_type;
202 } 202 }
203 int GetSendREDPayloadType(int channel) { 203 int GetSendREDPayloadType(int channel) {
204 return channels_[channel]->red_type; 204 return channels_[channel]->red_type;
205 } 205 }
206 bool CheckPacket(int channel, const void* data, size_t len) {
207 bool result = !CheckNoPacket(channel);
208 if (result) {
209 std::string packet = channels_[channel]->packets.front();
210 result = (packet == std::string(static_cast<const char*>(data), len));
211 channels_[channel]->packets.pop_front();
212 }
213 return result;
214 }
215 bool CheckNoPacket(int channel) {
216 return channels_[channel]->packets.empty();
217 }
218 void set_playout_fail_channel(int channel) { 206 void set_playout_fail_channel(int channel) {
219 playout_fail_channel_ = channel; 207 playout_fail_channel_ = channel;
220 } 208 }
221 void set_fail_create_channel(bool fail_create_channel) { 209 void set_fail_create_channel(bool fail_create_channel) {
222 fail_create_channel_ = fail_create_channel; 210 fail_create_channel_ = fail_create_channel;
223 } 211 }
224 int AddChannel(const webrtc::Config& config) { 212 int AddChannel(const webrtc::Config& config) {
225 if (fail_create_channel_) { 213 if (fail_create_channel_) {
226 return -1; 214 return -1;
227 } 215 }
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716 webrtc::NsModes ns_mode_ = webrtc::kNsDefault; 704 webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
717 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; 705 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
718 webrtc::AgcConfig agc_config_; 706 webrtc::AgcConfig agc_config_;
719 int playout_fail_channel_ = -1; 707 int playout_fail_channel_ = -1;
720 FakeAudioProcessing audio_processing_; 708 FakeAudioProcessing audio_processing_;
721 }; 709 };
722 710
723 } // namespace cricket 711 } // namespace cricket
724 712
725 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ 713 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
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