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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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| 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 } |
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| 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_ |
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