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| 1 /* | 1 /* |
| 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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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| 37 AudioDecoderIsacT<T>::~AudioDecoderIsacT() { | 37 AudioDecoderIsacT<T>::~AudioDecoderIsacT() { |
| 38 RTC_CHECK_EQ(0, T::Free(isac_state_)); | 38 RTC_CHECK_EQ(0, T::Free(isac_state_)); |
| 39 } | 39 } |
| 40 | 40 |
| 41 template <typename T> | 41 template <typename T> |
| 42 int AudioDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded, | 42 int AudioDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded, |
| 43 size_t encoded_len, | 43 size_t encoded_len, |
| 44 int sample_rate_hz, | 44 int sample_rate_hz, |
| 45 int16_t* decoded, | 45 int16_t* decoded, |
| 46 SpeechType* speech_type) { | 46 SpeechType* speech_type) { |
| 47 // We want to crate the illusion that iSAC supports 48000 Hz decoding, while | |
| 48 // in fact it outputs 32000 Hz. This is the iSAC fullband mode. | |
| 49 if (sample_rate_hz == 48000) | |
| 50 sample_rate_hz = 32000; | |
| 51 RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000) | 47 RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000) |
| 52 << "Unsupported sample rate " << sample_rate_hz; | 48 << "Unsupported sample rate " << sample_rate_hz; |
| 53 if (sample_rate_hz != decoder_sample_rate_hz_) { | 49 if (sample_rate_hz != decoder_sample_rate_hz_) { |
| 54 RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz)); | 50 RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz)); |
| 55 decoder_sample_rate_hz_ = sample_rate_hz; | 51 decoder_sample_rate_hz_ = sample_rate_hz; |
| 56 } | 52 } |
| 57 int16_t temp_type = 1; // Default is speech. | 53 int16_t temp_type = 1; // Default is speech. |
| 58 int ret = | 54 int ret = |
| 59 T::DecodeInternal(isac_state_, encoded, encoded_len, decoded, &temp_type); | 55 T::DecodeInternal(isac_state_, encoded, encoded_len, decoded, &temp_type); |
| 60 *speech_type = ConvertSpeechType(temp_type); | 56 *speech_type = ConvertSpeechType(temp_type); |
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| 99 } | 95 } |
| 100 | 96 |
| 101 template <typename T> | 97 template <typename T> |
| 102 size_t AudioDecoderIsacT<T>::Channels() const { | 98 size_t AudioDecoderIsacT<T>::Channels() const { |
| 103 return 1; | 99 return 1; |
| 104 } | 100 } |
| 105 | 101 |
| 106 } // namespace webrtc | 102 } // namespace webrtc |
| 107 | 103 |
| 108 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_ | 104 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_ |
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