OLD | NEW |
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 |
(...skipping 166 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
177 // in fact it outputs 32000 Hz. This is the iSAC fullband mode. | 177 // in fact it outputs 32000 Hz. This is the iSAC fullband mode. |
178 if (sample_rate_hz == 48000) | 178 if (sample_rate_hz == 48000) |
179 sample_rate_hz = 32000; | 179 sample_rate_hz = 32000; |
180 CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000) | 180 CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000) |
181 << "Unsupported sample rate " << sample_rate_hz; | 181 << "Unsupported sample rate " << sample_rate_hz; |
182 if (sample_rate_hz != decoder_sample_rate_hz_) { | 182 if (sample_rate_hz != decoder_sample_rate_hz_) { |
183 CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz)); | 183 CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz)); |
184 decoder_sample_rate_hz_ = sample_rate_hz; | 184 decoder_sample_rate_hz_ = sample_rate_hz; |
185 } | 185 } |
186 int16_t temp_type = 1; // Default is speech. | 186 int16_t temp_type = 1; // Default is speech. |
187 int ret = | 187 int16_t ret = |
188 T::DecodeInternal(isac_state_, encoded, static_cast<int16_t>(encoded_len), | 188 T::DecodeInternal(isac_state_, encoded, static_cast<int16_t>(encoded_len), |
189 decoded, &temp_type); | 189 decoded, &temp_type); |
190 *speech_type = ConvertSpeechType(temp_type); | 190 *speech_type = ConvertSpeechType(temp_type); |
191 return ret; | 191 return ret; |
192 } | 192 } |
193 | 193 |
194 template <typename T> | 194 template <typename T> |
195 bool AudioEncoderDecoderIsacT<T>::HasDecodePlc() const { | 195 bool AudioEncoderDecoderIsacT<T>::HasDecodePlc() const { |
196 return false; | 196 return false; |
197 } | 197 } |
(...skipping 24 matching lines...) Expand all Loading... |
222 | 222 |
223 template <typename T> | 223 template <typename T> |
224 int AudioEncoderDecoderIsacT<T>::ErrorCode() { | 224 int AudioEncoderDecoderIsacT<T>::ErrorCode() { |
225 CriticalSectionScoped cs(state_lock_.get()); | 225 CriticalSectionScoped cs(state_lock_.get()); |
226 return T::GetErrorCode(isac_state_); | 226 return T::GetErrorCode(isac_state_); |
227 } | 227 } |
228 | 228 |
229 } // namespace webrtc | 229 } // namespace webrtc |
230 | 230 |
231 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_ | 231 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_ |
OLD | NEW |