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Side by Side Diff: webrtc/modules/audio_processing/gain_control_impl.cc

Issue 2535593002: RTC_[D]CHECK_op: Remove "u" suffix on integer constants (Closed)
Patch Set: Created 4 years ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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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116 116
117 for (auto& gain_controller : gain_controllers_) { 117 for (auto& gain_controller : gain_controllers_) {
118 WebRtcAgc_AddFarend(gain_controller->state(), packed_render_audio.data(), 118 WebRtcAgc_AddFarend(gain_controller->state(), packed_render_audio.data(),
119 packed_render_audio.size()); 119 packed_render_audio.size());
120 } 120 }
121 } 121 }
122 122
123 void GainControlImpl::PackRenderAudioBuffer( 123 void GainControlImpl::PackRenderAudioBuffer(
124 AudioBuffer* audio, 124 AudioBuffer* audio,
125 std::vector<int16_t>* packed_buffer) { 125 std::vector<int16_t>* packed_buffer) {
126 RTC_DCHECK_GE(160u, audio->num_frames_per_band()); 126 RTC_DCHECK_GE(160, audio->num_frames_per_band());
127 127
128 packed_buffer->clear(); 128 packed_buffer->clear();
129 packed_buffer->insert( 129 packed_buffer->insert(
130 packed_buffer->end(), audio->mixed_low_pass_data(), 130 packed_buffer->end(), audio->mixed_low_pass_data(),
131 (audio->mixed_low_pass_data() + audio->num_frames_per_band())); 131 (audio->mixed_low_pass_data() + audio->num_frames_per_band()));
132 } 132 }
133 133
134 int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { 134 int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
135 rtc::CritScope cs(crit_capture_); 135 rtc::CritScope cs(crit_capture_);
136 136
137 if (!enabled_) { 137 if (!enabled_) {
138 return AudioProcessing::kNoError; 138 return AudioProcessing::kNoError;
139 } 139 }
140 140
141 RTC_DCHECK(num_proc_channels_); 141 RTC_DCHECK(num_proc_channels_);
142 RTC_DCHECK_GE(160u, audio->num_frames_per_band()); 142 RTC_DCHECK_GE(160, audio->num_frames_per_band());
143 RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_); 143 RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_);
144 RTC_DCHECK_LE(*num_proc_channels_, gain_controllers_.size()); 144 RTC_DCHECK_LE(*num_proc_channels_, gain_controllers_.size());
145 145
146 if (mode_ == kAdaptiveAnalog) { 146 if (mode_ == kAdaptiveAnalog) {
147 int capture_channel = 0; 147 int capture_channel = 0;
148 for (auto& gain_controller : gain_controllers_) { 148 for (auto& gain_controller : gain_controllers_) {
149 gain_controller->set_capture_level(analog_capture_level_); 149 gain_controller->set_capture_level(analog_capture_level_);
150 int err = WebRtcAgc_AddMic( 150 int err = WebRtcAgc_AddMic(
151 gain_controller->state(), audio->split_bands(capture_channel), 151 gain_controller->state(), audio->split_bands(capture_channel),
152 audio->num_bands(), audio->num_frames_per_band()); 152 audio->num_bands(), audio->num_frames_per_band());
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183 183
184 if (!enabled_) { 184 if (!enabled_) {
185 return AudioProcessing::kNoError; 185 return AudioProcessing::kNoError;
186 } 186 }
187 187
188 if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) { 188 if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
189 return AudioProcessing::kStreamParameterNotSetError; 189 return AudioProcessing::kStreamParameterNotSetError;
190 } 190 }
191 191
192 RTC_DCHECK(num_proc_channels_); 192 RTC_DCHECK(num_proc_channels_);
193 RTC_DCHECK_GE(160u, audio->num_frames_per_band()); 193 RTC_DCHECK_GE(160, audio->num_frames_per_band());
194 RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_); 194 RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_);
195 195
196 stream_is_saturated_ = false; 196 stream_is_saturated_ = false;
197 int capture_channel = 0; 197 int capture_channel = 0;
198 for (auto& gain_controller : gain_controllers_) { 198 for (auto& gain_controller : gain_controllers_) {
199 int32_t capture_level_out = 0; 199 int32_t capture_level_out = 0;
200 uint8_t saturation_warning = 0; 200 uint8_t saturation_warning = 0;
201 201
202 // The call to stream_has_echo() is ok from a deadlock perspective 202 // The call to stream_has_echo() is ok from a deadlock perspective
203 // as the capture lock is allready held. 203 // as the capture lock is allready held.
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430 for (auto& gain_controller : gain_controllers_) { 430 for (auto& gain_controller : gain_controllers_) {
431 const int handle_error = 431 const int handle_error =
432 WebRtcAgc_set_config(gain_controller->state(), config); 432 WebRtcAgc_set_config(gain_controller->state(), config);
433 if (handle_error != AudioProcessing::kNoError) { 433 if (handle_error != AudioProcessing::kNoError) {
434 error = handle_error; 434 error = handle_error;
435 } 435 }
436 } 436 }
437 return error; 437 return error;
438 } 438 }
439 } // namespace webrtc 439 } // namespace webrtc
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