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| 1 /* | 1 /* |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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 |
| 11 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h" | 11 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h" |
| 12 | 12 |
| 13 #include <assert.h> | 13 #include <assert.h> |
| 14 #include <string.h> // memmove | 14 #include <string.h> // memmove |
| 15 | 15 |
| 16 #include "webrtc/base/checks.h" | 16 #include "webrtc/base/checks.h" |
| 17 #include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h" | 17 #include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h" |
| 18 #include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h" | 18 #include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h" |
| 19 #ifdef WEBRTC_CODEC_G722 | 19 #ifdef WEBRTC_CODEC_G722 |
| 20 #include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h" | 20 #include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h" |
| 21 #endif | 21 #endif |
| 22 #ifdef WEBRTC_CODEC_ILBC | 22 #ifdef WEBRTC_CODEC_ILBC |
| 23 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h" | 23 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/audio_decoder_ilbc.h
" |
| 24 #endif | 24 #endif |
| 25 #ifdef WEBRTC_CODEC_ISACFX | 25 #ifdef WEBRTC_CODEC_ISACFX |
| 26 #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_is
acfix.h" | 26 #include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_is
acfix.h" |
| 27 #endif | 27 #endif |
| 28 #ifdef WEBRTC_CODEC_ISAC | 28 #ifdef WEBRTC_CODEC_ISAC |
| 29 #include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_i
sac.h" | 29 #include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_i
sac.h" |
| 30 #endif | 30 #endif |
| 31 #ifdef WEBRTC_CODEC_OPUS | 31 #ifdef WEBRTC_CODEC_OPUS |
| 32 #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h
" | 32 #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h
" |
| 33 #endif | 33 #endif |
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| 125 | 125 |
| 126 AudioDecoderPcm16BMultiCh::AudioDecoderPcm16BMultiCh(size_t num_channels) | 126 AudioDecoderPcm16BMultiCh::AudioDecoderPcm16BMultiCh(size_t num_channels) |
| 127 : channels_(num_channels) { | 127 : channels_(num_channels) { |
| 128 DCHECK(num_channels > 0); | 128 DCHECK(num_channels > 0); |
| 129 } | 129 } |
| 130 | 130 |
| 131 size_t AudioDecoderPcm16BMultiCh::Channels() const { | 131 size_t AudioDecoderPcm16BMultiCh::Channels() const { |
| 132 return channels_; | 132 return channels_; |
| 133 } | 133 } |
| 134 | 134 |
| 135 // iLBC | |
| 136 #ifdef WEBRTC_CODEC_ILBC | |
| 137 AudioDecoderIlbc::AudioDecoderIlbc() { | |
| 138 WebRtcIlbcfix_DecoderCreate(&dec_state_); | |
| 139 WebRtcIlbcfix_Decoderinit30Ms(dec_state_); | |
| 140 } | |
| 141 | |
| 142 AudioDecoderIlbc::~AudioDecoderIlbc() { | |
| 143 WebRtcIlbcfix_DecoderFree(dec_state_); | |
| 144 } | |
| 145 | |
| 146 bool AudioDecoderIlbc::HasDecodePlc() const { | |
| 147 return true; | |
| 148 } | |
| 149 | |
| 150 int AudioDecoderIlbc::DecodeInternal(const uint8_t* encoded, | |
| 151 size_t encoded_len, | |
| 152 int sample_rate_hz, | |
| 153 int16_t* decoded, | |
| 154 SpeechType* speech_type) { | |
| 155 DCHECK_EQ(sample_rate_hz, 8000); | |
| 156 int16_t temp_type = 1; // Default is speech. | |
| 157 int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded, | |
| 158 &temp_type); | |
| 159 *speech_type = ConvertSpeechType(temp_type); | |
| 160 return ret; | |
| 161 } | |
| 162 | |
| 163 size_t AudioDecoderIlbc::DecodePlc(size_t num_frames, int16_t* decoded) { | |
| 164 return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames); | |
| 165 } | |
| 166 | |
| 167 void AudioDecoderIlbc::Reset() { | |
| 168 WebRtcIlbcfix_Decoderinit30Ms(dec_state_); | |
| 169 } | |
| 170 | |
| 171 size_t AudioDecoderIlbc::Channels() const { | |
| 172 return 1; | |
| 173 } | |
| 174 #endif | |
| 175 | |
| 176 // G.722 | 135 // G.722 |
| 177 #ifdef WEBRTC_CODEC_G722 | 136 #ifdef WEBRTC_CODEC_G722 |
| 178 AudioDecoderG722::AudioDecoderG722() { | 137 AudioDecoderG722::AudioDecoderG722() { |
| 179 WebRtcG722_CreateDecoder(&dec_state_); | 138 WebRtcG722_CreateDecoder(&dec_state_); |
| 180 WebRtcG722_DecoderInit(dec_state_); | 139 WebRtcG722_DecoderInit(dec_state_); |
| 181 } | 140 } |
| 182 | 141 |
| 183 AudioDecoderG722::~AudioDecoderG722() { | 142 AudioDecoderG722::~AudioDecoderG722() { |
| 184 WebRtcG722_FreeDecoder(dec_state_); | 143 WebRtcG722_FreeDecoder(dec_state_); |
| 185 } | 144 } |
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| 494 case kDecoderRED: | 453 case kDecoderRED: |
| 495 case kDecoderAVT: | 454 case kDecoderAVT: |
| 496 case kDecoderArbitrary: | 455 case kDecoderArbitrary: |
| 497 default: { | 456 default: { |
| 498 return NULL; | 457 return NULL; |
| 499 } | 458 } |
| 500 } | 459 } |
| 501 } | 460 } |
| 502 | 461 |
| 503 } // namespace webrtc | 462 } // namespace webrtc |
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