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

Issue 1392173004: Delete full-band mode from the iSAC codec (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@remove-isac-fb-neteq
Patch Set: Created 5 years, 2 months ago
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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);
(...skipping 38 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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