1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
|
/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
#include <string.h>
#include <cstdint>
#include <utility>
#include <vector>
#include "api/audio_codecs/audio_decoder.h"
#include "modules/audio_coding/codecs/g722/g722_interface.h"
#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
namespace webrtc {
AudioDecoderG722Impl::AudioDecoderG722Impl() {
WebRtcG722_CreateDecoder(&dec_state_);
WebRtcG722_DecoderInit(dec_state_);
}
AudioDecoderG722Impl::~AudioDecoderG722Impl() {
WebRtcG722_FreeDecoder(dec_state_);
}
bool AudioDecoderG722Impl::HasDecodePlc() const {
return false;
}
int AudioDecoderG722Impl::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
int16_t temp_type = 1; // Default is speech.
size_t ret =
WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
*speech_type = ConvertSpeechType(temp_type);
return static_cast<int>(ret);
}
void AudioDecoderG722Impl::Reset() {
WebRtcG722_DecoderInit(dec_state_);
}
std::vector<AudioDecoder::ParseResult> AudioDecoderG722Impl::ParsePayload(
Buffer&& payload,
uint32_t timestamp) {
return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload),
timestamp, 8, 16);
}
int AudioDecoderG722Impl::PacketDuration(const uint8_t* /* encoded */,
size_t encoded_len) const {
// 1/2 encoded byte per sample per channel.
return static_cast<int>(2 * encoded_len / Channels());
}
int AudioDecoderG722Impl::PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const {
return PacketDuration(encoded, encoded_len);
}
int AudioDecoderG722Impl::SampleRateHz() const {
return 16000;
}
size_t AudioDecoderG722Impl::Channels() const {
return 1;
}
AudioDecoderG722StereoImpl::AudioDecoderG722StereoImpl() {
WebRtcG722_CreateDecoder(&dec_state_left_);
WebRtcG722_CreateDecoder(&dec_state_right_);
WebRtcG722_DecoderInit(dec_state_left_);
WebRtcG722_DecoderInit(dec_state_right_);
}
AudioDecoderG722StereoImpl::~AudioDecoderG722StereoImpl() {
WebRtcG722_FreeDecoder(dec_state_left_);
WebRtcG722_FreeDecoder(dec_state_right_);
}
int AudioDecoderG722StereoImpl::DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) {
RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
// Adjust the encoded length down to ensure the same number of samples in each
// channel.
const size_t encoded_len_adjusted = PacketDuration(encoded, encoded_len) *
Channels() /
2; // 1/2 byte per sample per channel
int16_t temp_type = 1; // Default is speech.
// De-interleave the bit-stream into two separate payloads.
uint8_t* encoded_deinterleaved = new uint8_t[encoded_len_adjusted];
SplitStereoPacket(encoded, encoded_len_adjusted, encoded_deinterleaved);
// Decode left and right.
size_t decoded_len =
WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
encoded_len_adjusted / 2, decoded, &temp_type);
size_t ret = WebRtcG722_Decode(
dec_state_right_, &encoded_deinterleaved[encoded_len_adjusted / 2],
encoded_len_adjusted / 2, &decoded[decoded_len], &temp_type);
if (ret == decoded_len) {
ret += decoded_len; // Return total number of samples.
// Interleave output.
for (size_t k = ret / 2; k < ret; k++) {
int16_t temp = decoded[k];
memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1],
(ret - k - 1) * sizeof(int16_t));
decoded[2 * k - ret + 1] = temp;
}
}
*speech_type = ConvertSpeechType(temp_type);
delete[] encoded_deinterleaved;
return static_cast<int>(ret);
}
int AudioDecoderG722StereoImpl::PacketDuration(const uint8_t* /* encoded */,
size_t encoded_len) const {
// 1/2 encoded byte per sample per channel. Make sure the length represents
// an equal number of bytes per channel. Otherwise, we cannot de-interleave
// the encoded data later.
return static_cast<int>(2 * (encoded_len / Channels()));
}
int AudioDecoderG722StereoImpl::SampleRateHz() const {
return 16000;
}
size_t AudioDecoderG722StereoImpl::Channels() const {
return 2;
}
void AudioDecoderG722StereoImpl::Reset() {
WebRtcG722_DecoderInit(dec_state_left_);
WebRtcG722_DecoderInit(dec_state_right_);
}
std::vector<AudioDecoder::ParseResult> AudioDecoderG722StereoImpl::ParsePayload(
Buffer&& payload,
uint32_t timestamp) {
return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload),
timestamp, 2 * 8, 16);
}
// Split the stereo packet and place left and right channel after each other
// in the output array.
void AudioDecoderG722StereoImpl::SplitStereoPacket(
const uint8_t* encoded,
size_t encoded_len,
uint8_t* encoded_deinterleaved) {
// Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ...,
// where "lx" is 4 bits representing left sample number x, and "rx" right
// sample. Two samples fit in one byte, represented with |...|.
for (size_t i = 0; i + 1 < encoded_len; i += 2) {
uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F);
encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4);
encoded_deinterleaved[i + 1] = right_byte;
}
// Move one byte representing right channel each loop, and place it at the
// end of the bytestream vector. After looping the data is reordered to:
// |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|,
// where N is the total number of samples.
for (size_t i = 0; i < encoded_len / 2; i++) {
uint8_t right_byte = encoded_deinterleaved[i + 1];
memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2],
encoded_len - i - 2);
encoded_deinterleaved[encoded_len - 1] = right_byte;
}
}
} // namespace webrtc
|
