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Side by Side Diff: webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h

Issue 1311533010: Remove AudioEncoder methods SetMaxBitrate and SetMaxPayloadSize (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@ifc-merge-2
Patch Set: Created 5 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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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151 info.payload_type = config_.payload_type; 151 info.payload_type = config_.payload_type;
152 return info; 152 return info;
153 } 153 }
154 154
155 template <typename T> 155 template <typename T>
156 void AudioEncoderIsacT<T>::Reset() { 156 void AudioEncoderIsacT<T>::Reset() {
157 RecreateEncoderInstance(config_); 157 RecreateEncoderInstance(config_);
158 } 158 }
159 159
160 template <typename T> 160 template <typename T>
161 void AudioEncoderIsacT<T>::SetMaxPayloadSize(int max_payload_size_bytes) {
162 auto conf = config_;
163 conf.max_payload_size_bytes = max_payload_size_bytes;
164 RecreateEncoderInstance(conf);
165 }
166
167 template <typename T>
168 void AudioEncoderIsacT<T>::SetMaxBitrate(int max_rate_bps) {
169 auto conf = config_;
170 conf.max_bit_rate = max_rate_bps;
171 RecreateEncoderInstance(conf);
172 }
173
174 template <typename T>
175 void AudioEncoderIsacT<T>::RecreateEncoderInstance(const Config& config) { 161 void AudioEncoderIsacT<T>::RecreateEncoderInstance(const Config& config) {
176 CHECK(config.IsOk()); 162 CHECK(config.IsOk());
177 packet_in_progress_ = false; 163 packet_in_progress_ = false;
178 bwinfo_ = config.bwinfo; 164 bwinfo_ = config.bwinfo;
179 if (isac_state_) 165 if (isac_state_)
180 CHECK_EQ(0, T::Free(isac_state_)); 166 CHECK_EQ(0, T::Free(isac_state_));
181 CHECK_EQ(0, T::Create(&isac_state_)); 167 CHECK_EQ(0, T::Create(&isac_state_));
182 CHECK_EQ(0, T::EncoderInit(isac_state_, config.adaptive_mode ? 0 : 1)); 168 CHECK_EQ(0, T::EncoderInit(isac_state_, config.adaptive_mode ? 0 : 1));
183 CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz)); 169 CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
184 const int bit_rate = config.bit_rate == 0 ? kDefaultBitRate : config.bit_rate; 170 const int bit_rate = config.bit_rate == 0 ? kDefaultBitRate : config.bit_rate;
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289 } 275 }
290 276
291 template <typename T> 277 template <typename T>
292 size_t AudioDecoderIsacT<T>::Channels() const { 278 size_t AudioDecoderIsacT<T>::Channels() const {
293 return 1; 279 return 1;
294 } 280 }
295 281
296 } // namespace webrtc 282 } // namespace webrtc
297 283
298 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_ 284 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
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