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 "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
12
13 #include <memory>
14 #include <utility>
15
16 #include "absl/types/optional.h"
17 #include "api/array_view.h"
18 #include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
19 #include "rtc_base/checks.h"
20
21 namespace webrtc {
22
AudioDecoderOpusImpl(size_t num_channels,int sample_rate_hz)23 AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels,
24 int sample_rate_hz)
25 : channels_{num_channels}, sample_rate_hz_{sample_rate_hz} {
26 RTC_DCHECK(num_channels == 1 || num_channels == 2);
27 RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 48000);
28 const int error =
29 WebRtcOpus_DecoderCreate(&dec_state_, channels_, sample_rate_hz_);
30 RTC_DCHECK(error == 0);
31 WebRtcOpus_DecoderInit(dec_state_);
32 }
33
~AudioDecoderOpusImpl()34 AudioDecoderOpusImpl::~AudioDecoderOpusImpl() {
35 WebRtcOpus_DecoderFree(dec_state_);
36 }
37
ParsePayload(rtc::Buffer && payload,uint32_t timestamp)38 std::vector<AudioDecoder::ParseResult> AudioDecoderOpusImpl::ParsePayload(
39 rtc::Buffer&& payload,
40 uint32_t timestamp) {
41 std::vector<ParseResult> results;
42
43 if (PacketHasFec(payload.data(), payload.size())) {
44 const int duration =
45 PacketDurationRedundant(payload.data(), payload.size());
46 RTC_DCHECK_GE(duration, 0);
47 rtc::Buffer payload_copy(payload.data(), payload.size());
48 std::unique_ptr<EncodedAudioFrame> fec_frame(
49 new OpusFrame(this, std::move(payload_copy), false));
50 results.emplace_back(timestamp - duration, 1, std::move(fec_frame));
51 }
52 std::unique_ptr<EncodedAudioFrame> frame(
53 new OpusFrame(this, std::move(payload), true));
54 results.emplace_back(timestamp, 0, std::move(frame));
55 return results;
56 }
57
DecodeInternal(const uint8_t * encoded,size_t encoded_len,int sample_rate_hz,int16_t * decoded,SpeechType * speech_type)58 int AudioDecoderOpusImpl::DecodeInternal(const uint8_t* encoded,
59 size_t encoded_len,
60 int sample_rate_hz,
61 int16_t* decoded,
62 SpeechType* speech_type) {
63 RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_);
64 int16_t temp_type = 1; // Default is speech.
65 int ret =
66 WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
67 if (ret > 0)
68 ret *= static_cast<int>(channels_); // Return total number of samples.
69 *speech_type = ConvertSpeechType(temp_type);
70 return ret;
71 }
72
DecodeRedundantInternal(const uint8_t * encoded,size_t encoded_len,int sample_rate_hz,int16_t * decoded,SpeechType * speech_type)73 int AudioDecoderOpusImpl::DecodeRedundantInternal(const uint8_t* encoded,
74 size_t encoded_len,
75 int sample_rate_hz,
76 int16_t* decoded,
77 SpeechType* speech_type) {
78 if (!PacketHasFec(encoded, encoded_len)) {
79 // This packet is a RED packet.
80 return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
81 speech_type);
82 }
83
84 RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_);
85 int16_t temp_type = 1; // Default is speech.
86 int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
87 &temp_type);
88 if (ret > 0)
89 ret *= static_cast<int>(channels_); // Return total number of samples.
90 *speech_type = ConvertSpeechType(temp_type);
91 return ret;
92 }
93
Reset()94 void AudioDecoderOpusImpl::Reset() {
95 WebRtcOpus_DecoderInit(dec_state_);
96 }
97
PacketDuration(const uint8_t * encoded,size_t encoded_len) const98 int AudioDecoderOpusImpl::PacketDuration(const uint8_t* encoded,
99 size_t encoded_len) const {
100 return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
101 }
102
PacketDurationRedundant(const uint8_t * encoded,size_t encoded_len) const103 int AudioDecoderOpusImpl::PacketDurationRedundant(const uint8_t* encoded,
104 size_t encoded_len) const {
105 if (!PacketHasFec(encoded, encoded_len)) {
106 // This packet is a RED packet.
107 return PacketDuration(encoded, encoded_len);
108 }
109
110 return WebRtcOpus_FecDurationEst(encoded, encoded_len, sample_rate_hz_);
111 }
112
PacketHasFec(const uint8_t * encoded,size_t encoded_len) const113 bool AudioDecoderOpusImpl::PacketHasFec(const uint8_t* encoded,
114 size_t encoded_len) const {
115 int fec;
116 fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
117 return (fec == 1);
118 }
119
SampleRateHz() const120 int AudioDecoderOpusImpl::SampleRateHz() const {
121 return sample_rate_hz_;
122 }
123
Channels() const124 size_t AudioDecoderOpusImpl::Channels() const {
125 return channels_;
126 }
127
128 } // namespace webrtc
129