1 /*
2 * Copyright (c) 2014 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/ilbc/audio_encoder_ilbc.h"
12
13 #include <algorithm>
14 #include <limits>
15 #include "common_types.h" // NOLINT(build/include)
16 #include "modules/audio_coding/codecs/ilbc/ilbc.h"
17 #include "rtc_base/checks.h"
18 #include "rtc_base/numerics/safe_conversions.h"
19
20 namespace webrtc {
21
22 namespace {
23
24 const int kSampleRateHz = 8000;
25
CreateConfig(const CodecInst & codec_inst)26 AudioEncoderIlbcConfig CreateConfig(const CodecInst& codec_inst) {
27 AudioEncoderIlbcConfig config;
28 config.frame_size_ms = codec_inst.pacsize / 8;
29 return config;
30 }
31
GetIlbcBitrate(int ptime)32 int GetIlbcBitrate(int ptime) {
33 switch (ptime) {
34 case 20:
35 case 40:
36 // 38 bytes per frame of 20 ms => 15200 bits/s.
37 return 15200;
38 case 30:
39 case 60:
40 // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
41 return 13333;
42 default:
43 FATAL();
44 }
45 }
46
47 } // namespace
48
AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig & config,int payload_type)49 AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config,
50 int payload_type)
51 : frame_size_ms_(config.frame_size_ms),
52 payload_type_(payload_type),
53 num_10ms_frames_per_packet_(
54 static_cast<size_t>(config.frame_size_ms / 10)),
55 encoder_(nullptr) {
56 RTC_CHECK(config.IsOk());
57 Reset();
58 }
59
AudioEncoderIlbcImpl(const CodecInst & codec_inst)60 AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const CodecInst& codec_inst)
61 : AudioEncoderIlbcImpl(CreateConfig(codec_inst), codec_inst.pltype) {}
62
~AudioEncoderIlbcImpl()63 AudioEncoderIlbcImpl::~AudioEncoderIlbcImpl() {
64 RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
65 }
66
SampleRateHz() const67 int AudioEncoderIlbcImpl::SampleRateHz() const {
68 return kSampleRateHz;
69 }
70
NumChannels() const71 size_t AudioEncoderIlbcImpl::NumChannels() const {
72 return 1;
73 }
74
Num10MsFramesInNextPacket() const75 size_t AudioEncoderIlbcImpl::Num10MsFramesInNextPacket() const {
76 return num_10ms_frames_per_packet_;
77 }
78
Max10MsFramesInAPacket() const79 size_t AudioEncoderIlbcImpl::Max10MsFramesInAPacket() const {
80 return num_10ms_frames_per_packet_;
81 }
82
GetTargetBitrate() const83 int AudioEncoderIlbcImpl::GetTargetBitrate() const {
84 return GetIlbcBitrate(rtc::dchecked_cast<int>(num_10ms_frames_per_packet_) *
85 10);
86 }
87
EncodeImpl(uint32_t rtp_timestamp,rtc::ArrayView<const int16_t> audio,rtc::Buffer * encoded)88 AudioEncoder::EncodedInfo AudioEncoderIlbcImpl::EncodeImpl(
89 uint32_t rtp_timestamp,
90 rtc::ArrayView<const int16_t> audio,
91 rtc::Buffer* encoded) {
92
93 // Save timestamp if starting a new packet.
94 if (num_10ms_frames_buffered_ == 0)
95 first_timestamp_in_buffer_ = rtp_timestamp;
96
97 // Buffer input.
98 std::copy(audio.cbegin(), audio.cend(),
99 input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_);
100
101 // If we don't yet have enough buffered input for a whole packet, we're done
102 // for now.
103 if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
104 return EncodedInfo();
105 }
106
107 // Encode buffered input.
108 RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
109 num_10ms_frames_buffered_ = 0;
110 size_t encoded_bytes =
111 encoded->AppendData(
112 RequiredOutputSizeBytes(),
113 [&] (rtc::ArrayView<uint8_t> encoded) {
114 const int r = WebRtcIlbcfix_Encode(
115 encoder_,
116 input_buffer_,
117 kSampleRateHz / 100 * num_10ms_frames_per_packet_,
118 encoded.data());
119 RTC_CHECK_GE(r, 0);
120
121 return static_cast<size_t>(r);
122 });
123
124 RTC_DCHECK_EQ(encoded_bytes, RequiredOutputSizeBytes());
125
126 EncodedInfo info;
127 info.encoded_bytes = encoded_bytes;
128 info.encoded_timestamp = first_timestamp_in_buffer_;
129 info.payload_type = payload_type_;
130 info.encoder_type = CodecType::kIlbc;
131 return info;
132 }
133
Reset()134 void AudioEncoderIlbcImpl::Reset() {
135 if (encoder_)
136 RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
137 RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
138 const int encoder_frame_size_ms = frame_size_ms_ > 30
139 ? frame_size_ms_ / 2
140 : frame_size_ms_;
141 RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
142 num_10ms_frames_buffered_ = 0;
143 }
144
RequiredOutputSizeBytes() const145 size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
146 switch (num_10ms_frames_per_packet_) {
147 case 2: return 38;
148 case 3: return 50;
149 case 4: return 2 * 38;
150 case 6: return 2 * 50;
151 default: FATAL();
152 }
153 }
154
155 } // namespace webrtc
156