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

Issue 1397773002: Change SetOutputScaling to set a single level, not left/right levels. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase+rename Created 5 years, 2 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2010 Google Inc. 3 * Copyright 2010 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
(...skipping 182 matching lines...) Expand 10 before | Expand all | Expand 10 after
193 int dtmf_event_code; 193 int dtmf_event_code;
194 bool dtmf_out_of_band; 194 bool dtmf_out_of_band;
195 int dtmf_length_ms; 195 int dtmf_length_ms;
196 }; 196 };
197 struct Channel { 197 struct Channel {
198 explicit Channel() 198 explicit Channel()
199 : external_transport(false), 199 : external_transport(false),
200 send(false), 200 send(false),
201 playout(false), 201 playout(false),
202 volume_scale(1.0), 202 volume_scale(1.0),
203 volume_pan_left(1.0),
204 volume_pan_right(1.0),
205 vad(false), 203 vad(false),
206 codec_fec(false), 204 codec_fec(false),
207 max_encoding_bandwidth(0), 205 max_encoding_bandwidth(0),
208 opus_dtx(false), 206 opus_dtx(false),
209 red(false), 207 red(false),
210 nack(false), 208 nack(false),
211 rx_agc_enabled(false), 209 rx_agc_enabled(false),
212 rx_agc_mode(webrtc::kAgcDefault), 210 rx_agc_mode(webrtc::kAgcDefault),
213 cn8_type(13), 211 cn8_type(13),
214 cn16_type(105), 212 cn16_type(105),
215 dtmf_type(106), 213 dtmf_type(106),
216 red_type(117), 214 red_type(117),
217 nack_max_packets(0), 215 nack_max_packets(0),
218 send_ssrc(0), 216 send_ssrc(0),
219 send_audio_level_ext_(-1), 217 send_audio_level_ext_(-1),
220 receive_audio_level_ext_(-1), 218 receive_audio_level_ext_(-1),
221 send_absolute_sender_time_ext_(-1), 219 send_absolute_sender_time_ext_(-1),
222 receive_absolute_sender_time_ext_(-1), 220 receive_absolute_sender_time_ext_(-1),
223 associate_send_channel(-1), 221 associate_send_channel(-1),
224 neteq_capacity(-1), 222 neteq_capacity(-1),
225 neteq_fast_accelerate(false) { 223 neteq_fast_accelerate(false) {
226 memset(&send_codec, 0, sizeof(send_codec)); 224 memset(&send_codec, 0, sizeof(send_codec));
227 memset(&rx_agc_config, 0, sizeof(rx_agc_config)); 225 memset(&rx_agc_config, 0, sizeof(rx_agc_config));
228 } 226 }
229 bool external_transport; 227 bool external_transport;
230 bool send; 228 bool send;
231 bool playout; 229 bool playout;
232 float volume_scale; 230 float volume_scale;
233 float volume_pan_left;
234 float volume_pan_right;
235 bool vad; 231 bool vad;
236 bool codec_fec; 232 bool codec_fec;
237 int max_encoding_bandwidth; 233 int max_encoding_bandwidth;
238 bool opus_dtx; 234 bool opus_dtx;
239 bool red; 235 bool red;
240 bool nack; 236 bool nack;
241 bool rx_agc_enabled; 237 bool rx_agc_enabled;
242 webrtc::AgcModes rx_agc_mode; 238 webrtc::AgcModes rx_agc_mode;
243 webrtc::AgcConfig rx_agc_config; 239 webrtc::AgcConfig rx_agc_config;
244 int cn8_type; 240 int cn8_type;
(...skipping 678 matching lines...) Expand 10 before | Expand all | Expand 10 after
923 WEBRTC_FUNC(SetChannelOutputVolumeScaling, (int channel, float scale)) { 919 WEBRTC_FUNC(SetChannelOutputVolumeScaling, (int channel, float scale)) {
924 WEBRTC_CHECK_CHANNEL(channel); 920 WEBRTC_CHECK_CHANNEL(channel);
925 channels_[channel]->volume_scale= scale; 921 channels_[channel]->volume_scale= scale;
926 return 0; 922 return 0;
927 } 923 }
928 WEBRTC_FUNC(GetChannelOutputVolumeScaling, (int channel, float& scale)) { 924 WEBRTC_FUNC(GetChannelOutputVolumeScaling, (int channel, float& scale)) {
929 WEBRTC_CHECK_CHANNEL(channel); 925 WEBRTC_CHECK_CHANNEL(channel);
930 scale = channels_[channel]->volume_scale; 926 scale = channels_[channel]->volume_scale;
931 return 0; 927 return 0;
932 } 928 }
933 WEBRTC_FUNC(SetOutputVolumePan, (int channel, float left, float right)) { 929 WEBRTC_STUB(SetOutputVolumePan, (int channel, float left, float right));
934 WEBRTC_CHECK_CHANNEL(channel); 930 WEBRTC_STUB(GetOutputVolumePan, (int channel, float& left, float& right));
935 channels_[channel]->volume_pan_left = left;
936 channels_[channel]->volume_pan_right = right;
937 return 0;
938 }
939 WEBRTC_FUNC(GetOutputVolumePan, (int channel, float& left, float& right)) {
940 WEBRTC_CHECK_CHANNEL(channel);
941 left = channels_[channel]->volume_pan_left;
942 right = channels_[channel]->volume_pan_right;
943 return 0;
944 }
945 931
946 // webrtc::VoEAudioProcessing 932 // webrtc::VoEAudioProcessing
947 WEBRTC_FUNC(SetNsStatus, (bool enable, webrtc::NsModes mode)) { 933 WEBRTC_FUNC(SetNsStatus, (bool enable, webrtc::NsModes mode)) {
948 ns_enabled_ = enable; 934 ns_enabled_ = enable;
949 ns_mode_ = mode; 935 ns_mode_ = mode;
950 return 0; 936 return 0;
951 } 937 }
952 WEBRTC_FUNC(GetNsStatus, (bool& enabled, webrtc::NsModes& mode)) { 938 WEBRTC_FUNC(GetNsStatus, (bool& enabled, webrtc::NsModes& mode)) {
953 enabled = ns_enabled_; 939 enabled = ns_enabled_;
954 mode = ns_mode_; 940 mode = ns_mode_;
(...skipping 198 matching lines...) Expand 10 before | Expand all | Expand 10 after
1153 int playout_sample_rate_; 1139 int playout_sample_rate_;
1154 DtmfInfo dtmf_info_; 1140 DtmfInfo dtmf_info_;
1155 FakeAudioProcessing audio_processing_; 1141 FakeAudioProcessing audio_processing_;
1156 }; 1142 };
1157 1143
1158 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID 1144 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID
1159 1145
1160 } // namespace cricket 1146 } // namespace cricket
1161 1147
1162 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_ 1148 #endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_
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