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/audio_decoder_opus.h"
12
13 #include "webrtc/base/checks.h"
14
15 namespace webrtc {
16
AudioDecoderOpus(size_t num_channels)17 AudioDecoderOpus::AudioDecoderOpus(size_t num_channels)
18 : channels_(num_channels) {
19 RTC_DCHECK(num_channels == 1 || num_channels == 2);
20 WebRtcOpus_DecoderCreate(&dec_state_, channels_);
21 WebRtcOpus_DecoderInit(dec_state_);
22 }
23
~AudioDecoderOpus()24 AudioDecoderOpus::~AudioDecoderOpus() {
25 WebRtcOpus_DecoderFree(dec_state_);
26 }
27
DecodeInternal(const uint8_t * encoded,size_t encoded_len,int sample_rate_hz,int16_t * decoded,SpeechType * speech_type)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 RTC_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
DecodeRedundantInternal(const uint8_t * encoded,size_t encoded_len,int sample_rate_hz,int16_t * decoded,SpeechType * speech_type)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 RTC_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
Reset()64 void AudioDecoderOpus::Reset() {
65 WebRtcOpus_DecoderInit(dec_state_);
66 }
67
PacketDuration(const uint8_t * encoded,size_t encoded_len) const68 int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
69 size_t encoded_len) const {
70 return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
71 }
72
PacketDurationRedundant(const uint8_t * encoded,size_t encoded_len) const73 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
PacketHasFec(const uint8_t * encoded,size_t encoded_len) const83 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
Channels() const90 size_t AudioDecoderOpus::Channels() const {
91 return channels_;
92 }
93
94 } // namespace webrtc
95