Line data Source code
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 "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
12 :
13 : #include <limits>
14 : #include "webrtc/base/checks.h"
15 : #include "webrtc/common_types.h"
16 : #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
17 :
18 : namespace webrtc {
19 :
20 : namespace {
21 :
22 : const size_t kSampleRateHz = 16000;
23 :
24 0 : AudioEncoderG722::Config CreateConfig(const CodecInst& codec_inst) {
25 0 : AudioEncoderG722::Config config;
26 0 : config.num_channels = codec_inst.channels;
27 0 : config.frame_size_ms = codec_inst.pacsize / 16;
28 0 : config.payload_type = codec_inst.pltype;
29 0 : return config;
30 : }
31 :
32 : } // namespace
33 :
34 0 : bool AudioEncoderG722::Config::IsOk() const {
35 0 : return (frame_size_ms > 0) && (frame_size_ms % 10 == 0) &&
36 0 : (num_channels >= 1);
37 : }
38 :
39 0 : AudioEncoderG722::AudioEncoderG722(const Config& config)
40 0 : : num_channels_(config.num_channels),
41 0 : payload_type_(config.payload_type),
42 : num_10ms_frames_per_packet_(
43 0 : static_cast<size_t>(config.frame_size_ms / 10)),
44 : num_10ms_frames_buffered_(0),
45 : first_timestamp_in_buffer_(0),
46 0 : encoders_(new EncoderState[num_channels_]),
47 0 : interleave_buffer_(2 * num_channels_) {
48 0 : RTC_CHECK(config.IsOk());
49 : const size_t samples_per_channel =
50 0 : kSampleRateHz / 100 * num_10ms_frames_per_packet_;
51 0 : for (size_t i = 0; i < num_channels_; ++i) {
52 0 : encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
53 0 : encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
54 : }
55 0 : Reset();
56 0 : }
57 :
58 0 : AudioEncoderG722::AudioEncoderG722(const CodecInst& codec_inst)
59 0 : : AudioEncoderG722(CreateConfig(codec_inst)) {}
60 :
61 : AudioEncoderG722::~AudioEncoderG722() = default;
62 :
63 0 : int AudioEncoderG722::SampleRateHz() const {
64 0 : return kSampleRateHz;
65 : }
66 :
67 0 : size_t AudioEncoderG722::NumChannels() const {
68 0 : return num_channels_;
69 : }
70 :
71 0 : int AudioEncoderG722::RtpTimestampRateHz() const {
72 : // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz
73 : // codec.
74 0 : return kSampleRateHz / 2;
75 : }
76 :
77 0 : size_t AudioEncoderG722::Num10MsFramesInNextPacket() const {
78 0 : return num_10ms_frames_per_packet_;
79 : }
80 :
81 0 : size_t AudioEncoderG722::Max10MsFramesInAPacket() const {
82 0 : return num_10ms_frames_per_packet_;
83 : }
84 :
85 0 : int AudioEncoderG722::GetTargetBitrate() const {
86 : // 4 bits/sample, 16000 samples/s/channel.
87 0 : return static_cast<int>(64000 * NumChannels());
88 : }
89 :
90 0 : void AudioEncoderG722::Reset() {
91 0 : num_10ms_frames_buffered_ = 0;
92 0 : for (size_t i = 0; i < num_channels_; ++i)
93 0 : RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder));
94 0 : }
95 :
96 0 : AudioEncoder::EncodedInfo AudioEncoderG722::EncodeImpl(
97 : uint32_t rtp_timestamp,
98 : rtc::ArrayView<const int16_t> audio,
99 : rtc::Buffer* encoded) {
100 0 : if (num_10ms_frames_buffered_ == 0)
101 0 : first_timestamp_in_buffer_ = rtp_timestamp;
102 :
103 : // Deinterleave samples and save them in each channel's buffer.
104 0 : const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
105 0 : for (size_t i = 0; i < kSampleRateHz / 100; ++i)
106 0 : for (size_t j = 0; j < num_channels_; ++j)
107 0 : encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j];
108 :
109 : // If we don't yet have enough samples for a packet, we're done for now.
110 0 : if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
111 0 : return EncodedInfo();
112 : }
113 :
114 : // Encode each channel separately.
115 0 : RTC_CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
116 0 : num_10ms_frames_buffered_ = 0;
117 0 : const size_t samples_per_channel = SamplesPerChannel();
118 0 : for (size_t i = 0; i < num_channels_; ++i) {
119 0 : const size_t bytes_encoded = WebRtcG722_Encode(
120 0 : encoders_[i].encoder, encoders_[i].speech_buffer.get(),
121 0 : samples_per_channel, encoders_[i].encoded_buffer.data());
122 0 : RTC_CHECK_EQ(bytes_encoded, samples_per_channel / 2);
123 : }
124 :
125 0 : const size_t bytes_to_encode = samples_per_channel / 2 * num_channels_;
126 0 : EncodedInfo info;
127 0 : info.encoded_bytes = encoded->AppendData(
128 0 : bytes_to_encode, [&] (rtc::ArrayView<uint8_t> encoded) {
129 : // Interleave the encoded bytes of the different channels. Each separate
130 : // channel and the interleaved stream encodes two samples per byte, most
131 : // significant half first.
132 0 : for (size_t i = 0; i < samples_per_channel / 2; ++i) {
133 0 : for (size_t j = 0; j < num_channels_; ++j) {
134 0 : uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
135 0 : interleave_buffer_.data()[j] = two_samples >> 4;
136 0 : interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf;
137 : }
138 0 : for (size_t j = 0; j < num_channels_; ++j)
139 0 : encoded[i * num_channels_ + j] =
140 0 : interleave_buffer_.data()[2 * j] << 4 |
141 0 : interleave_buffer_.data()[2 * j + 1];
142 : }
143 :
144 0 : return bytes_to_encode;
145 : });
146 0 : info.encoded_timestamp = first_timestamp_in_buffer_;
147 0 : info.payload_type = payload_type_;
148 0 : info.encoder_type = CodecType::kG722;
149 0 : return info;
150 : }
151 :
152 0 : AudioEncoderG722::EncoderState::EncoderState() {
153 0 : RTC_CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
154 0 : }
155 :
156 0 : AudioEncoderG722::EncoderState::~EncoderState() {
157 0 : RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
158 0 : }
159 :
160 0 : size_t AudioEncoderG722::SamplesPerChannel() const {
161 0 : return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
162 : }
163 :
164 : } // namespace webrtc
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