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| 1 /* |
| 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
| 3 * |
| 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 |
| 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ |
| 10 |
| 11 #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_decoder_opus.h
" |
| 12 |
| 13 #include "webrtc/base/checks.h" |
| 14 |
| 15 namespace webrtc { |
| 16 |
| 17 AudioDecoderOpus::AudioDecoderOpus(size_t num_channels) |
| 18 : channels_(num_channels) { |
| 19 DCHECK(num_channels == 1 || num_channels == 2); |
| 20 WebRtcOpus_DecoderCreate(&dec_state_, static_cast<int>(channels_)); |
| 21 WebRtcOpus_DecoderInit(dec_state_); |
| 22 } |
| 23 |
| 24 AudioDecoderOpus::~AudioDecoderOpus() { |
| 25 WebRtcOpus_DecoderFree(dec_state_); |
| 26 } |
| 27 |
| 28 int AudioDecoderOpus::DecodeInternal(const uint8_t* encoded, |
| 29 size_t encoded_len, |
| 30 int sample_rate_hz, |
| 31 int16_t* decoded, |
| 32 SpeechType* speech_type) { |
| 33 DCHECK_EQ(sample_rate_hz, 48000); |
| 34 int16_t temp_type = 1; // Default is speech. |
| 35 int ret = |
| 36 WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type); |
| 37 if (ret > 0) |
| 38 ret *= static_cast<int>(channels_); // Return total number of samples. |
| 39 *speech_type = ConvertSpeechType(temp_type); |
| 40 return ret; |
| 41 } |
| 42 |
| 43 int AudioDecoderOpus::DecodeRedundantInternal(const uint8_t* encoded, |
| 44 size_t encoded_len, |
| 45 int sample_rate_hz, |
| 46 int16_t* decoded, |
| 47 SpeechType* speech_type) { |
| 48 if (!PacketHasFec(encoded, encoded_len)) { |
| 49 // This packet is a RED packet. |
| 50 return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, |
| 51 speech_type); |
| 52 } |
| 53 |
| 54 DCHECK_EQ(sample_rate_hz, 48000); |
| 55 int16_t temp_type = 1; // Default is speech. |
| 56 int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded, |
| 57 &temp_type); |
| 58 if (ret > 0) |
| 59 ret *= static_cast<int>(channels_); // Return total number of samples. |
| 60 *speech_type = ConvertSpeechType(temp_type); |
| 61 return ret; |
| 62 } |
| 63 |
| 64 void AudioDecoderOpus::Reset() { |
| 65 WebRtcOpus_DecoderInit(dec_state_); |
| 66 } |
| 67 |
| 68 int AudioDecoderOpus::PacketDuration(const uint8_t* encoded, |
| 69 size_t encoded_len) const { |
| 70 return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len); |
| 71 } |
| 72 |
| 73 int AudioDecoderOpus::PacketDurationRedundant(const uint8_t* encoded, |
| 74 size_t encoded_len) const { |
| 75 if (!PacketHasFec(encoded, encoded_len)) { |
| 76 // This packet is a RED packet. |
| 77 return PacketDuration(encoded, encoded_len); |
| 78 } |
| 79 |
| 80 return WebRtcOpus_FecDurationEst(encoded, encoded_len); |
| 81 } |
| 82 |
| 83 bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded, |
| 84 size_t encoded_len) const { |
| 85 int fec; |
| 86 fec = WebRtcOpus_PacketHasFec(encoded, encoded_len); |
| 87 return (fec == 1); |
| 88 } |
| 89 |
| 90 size_t AudioDecoderOpus::Channels() const { |
| 91 return channels_; |
| 92 } |
| 93 |
| 94 } // namespace webrtc |
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