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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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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; | 131 bool playout = false; |
132 float volume_scale = 1.0f; | |
133 bool vad = false; | 132 bool vad = false; |
134 bool codec_fec = false; | 133 bool codec_fec = false; |
135 int max_encoding_bandwidth = 0; | 134 int max_encoding_bandwidth = 0; |
136 bool opus_dtx = false; | 135 bool opus_dtx = false; |
137 int cn8_type = 13; | 136 int cn8_type = 13; |
138 int cn16_type = 105; | 137 int cn16_type = 105; |
139 int associate_send_channel = -1; | 138 int associate_send_channel = -1; |
140 std::vector<webrtc::CodecInst> recv_codecs; | 139 std::vector<webrtc::CodecInst> recv_codecs; |
141 webrtc::CodecInst send_codec; | 140 webrtc::CodecInst send_codec; |
142 int neteq_capacity = -1; | 141 int neteq_capacity = -1; |
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434 WEBRTC_STUB(SetSpeakerVolume, (unsigned int)); | 433 WEBRTC_STUB(SetSpeakerVolume, (unsigned int)); |
435 WEBRTC_STUB(GetSpeakerVolume, (unsigned int&)); | 434 WEBRTC_STUB(GetSpeakerVolume, (unsigned int&)); |
436 WEBRTC_STUB(SetMicVolume, (unsigned int)); | 435 WEBRTC_STUB(SetMicVolume, (unsigned int)); |
437 WEBRTC_STUB(GetMicVolume, (unsigned int&)); | 436 WEBRTC_STUB(GetMicVolume, (unsigned int&)); |
438 WEBRTC_STUB(SetInputMute, (int, bool)); | 437 WEBRTC_STUB(SetInputMute, (int, bool)); |
439 WEBRTC_STUB(GetInputMute, (int, bool&)); | 438 WEBRTC_STUB(GetInputMute, (int, bool&)); |
440 WEBRTC_STUB(GetSpeechInputLevel, (unsigned int&)); | 439 WEBRTC_STUB(GetSpeechInputLevel, (unsigned int&)); |
441 WEBRTC_STUB(GetSpeechOutputLevel, (int, unsigned int&)); | 440 WEBRTC_STUB(GetSpeechOutputLevel, (int, unsigned int&)); |
442 WEBRTC_STUB(GetSpeechInputLevelFullRange, (unsigned int&)); | 441 WEBRTC_STUB(GetSpeechInputLevelFullRange, (unsigned int&)); |
443 WEBRTC_STUB(GetSpeechOutputLevelFullRange, (int, unsigned int&)); | 442 WEBRTC_STUB(GetSpeechOutputLevelFullRange, (int, unsigned int&)); |
444 WEBRTC_FUNC(SetChannelOutputVolumeScaling, (int channel, float scale)) { | 443 WEBRTC_STUB(SetChannelOutputVolumeScaling, (int channel, float scale)); |
445 WEBRTC_CHECK_CHANNEL(channel); | 444 WEBRTC_STUB(GetChannelOutputVolumeScaling, (int channel, float& scale)); |
446 channels_[channel]->volume_scale= scale; | |
447 return 0; | |
448 } | |
449 WEBRTC_FUNC(GetChannelOutputVolumeScaling, (int channel, float& scale)) { | |
450 WEBRTC_CHECK_CHANNEL(channel); | |
451 scale = channels_[channel]->volume_scale; | |
452 return 0; | |
453 } | |
454 WEBRTC_STUB(SetOutputVolumePan, (int channel, float left, float right)); | 445 WEBRTC_STUB(SetOutputVolumePan, (int channel, float left, float right)); |
455 WEBRTC_STUB(GetOutputVolumePan, (int channel, float& left, float& right)); | 446 WEBRTC_STUB(GetOutputVolumePan, (int channel, float& left, float& right)); |
456 | 447 |
457 // webrtc::VoEAudioProcessing | 448 // webrtc::VoEAudioProcessing |
458 WEBRTC_FUNC(SetNsStatus, (bool enable, webrtc::NsModes mode)) { | 449 WEBRTC_FUNC(SetNsStatus, (bool enable, webrtc::NsModes mode)) { |
459 ns_enabled_ = enable; | 450 ns_enabled_ = enable; |
460 ns_mode_ = mode; | 451 ns_mode_ = mode; |
461 return 0; | 452 return 0; |
462 } | 453 } |
463 WEBRTC_FUNC(GetNsStatus, (bool& enabled, webrtc::NsModes& mode)) { | 454 WEBRTC_FUNC(GetNsStatus, (bool& enabled, webrtc::NsModes& mode)) { |
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593 webrtc::NsModes ns_mode_ = webrtc::kNsDefault; | 584 webrtc::NsModes ns_mode_ = webrtc::kNsDefault; |
594 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; | 585 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; |
595 webrtc::AgcConfig agc_config_; | 586 webrtc::AgcConfig agc_config_; |
596 int playout_fail_channel_ = -1; | 587 int playout_fail_channel_ = -1; |
597 FakeAudioProcessing audio_processing_; | 588 FakeAudioProcessing audio_processing_; |
598 }; | 589 }; |
599 | 590 |
600 } // namespace cricket | 591 } // namespace cricket |
601 | 592 |
602 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ | 593 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ |
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