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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h" | 11 #include "webrtc/modules/audio_coding/codecs/g722/include/audio_encoder_g722.h" |
12 | 12 |
13 #include <limits> | 13 #include <limits> |
14 #include "webrtc/base/checks.h" | 14 #include "webrtc/base/checks.h" |
15 #include "webrtc/common_types.h" | 15 #include "webrtc/common_types.h" |
16 #include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h" | 16 #include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h" |
17 | 17 |
18 namespace webrtc { | 18 namespace webrtc { |
19 | 19 |
20 namespace { | 20 namespace { |
21 | 21 |
22 const int kSampleRateHz = 16000; | 22 const size_t kSampleRateHz = 16000; |
23 | 23 |
24 } // namespace | 24 } // namespace |
25 | 25 |
26 bool AudioEncoderG722::Config::IsOk() const { | 26 bool AudioEncoderG722::Config::IsOk() const { |
27 return (frame_size_ms > 0) && (frame_size_ms % 10 == 0) && | 27 return (frame_size_ms > 0) && (frame_size_ms % 10 == 0) && |
28 (num_channels >= 1); | 28 (num_channels >= 1); |
29 } | 29 } |
30 | 30 |
31 AudioEncoderG722::EncoderState::EncoderState() { | 31 AudioEncoderG722::EncoderState::EncoderState() { |
32 CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder)); | 32 CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder)); |
33 CHECK_EQ(0, WebRtcG722_EncoderInit(encoder)); | 33 CHECK_EQ(0, WebRtcG722_EncoderInit(encoder)); |
34 } | 34 } |
35 | 35 |
36 AudioEncoderG722::EncoderState::~EncoderState() { | 36 AudioEncoderG722::EncoderState::~EncoderState() { |
37 CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder)); | 37 CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder)); |
38 } | 38 } |
39 | 39 |
40 AudioEncoderG722::AudioEncoderG722(const Config& config) | 40 AudioEncoderG722::AudioEncoderG722(const Config& config) |
41 : num_channels_(config.num_channels), | 41 : num_channels_(config.num_channels), |
42 payload_type_(config.payload_type), | 42 payload_type_(config.payload_type), |
43 num_10ms_frames_per_packet_(config.frame_size_ms / 10), | 43 num_10ms_frames_per_packet_( |
| 44 static_cast<size_t>(config.frame_size_ms / 10)), |
44 num_10ms_frames_buffered_(0), | 45 num_10ms_frames_buffered_(0), |
45 first_timestamp_in_buffer_(0), | 46 first_timestamp_in_buffer_(0), |
46 encoders_(new EncoderState[num_channels_]), | 47 encoders_(new EncoderState[num_channels_]), |
47 interleave_buffer_(2 * num_channels_) { | 48 interleave_buffer_(2 * num_channels_) { |
48 CHECK(config.IsOk()); | 49 CHECK(config.IsOk()); |
49 const int samples_per_channel = | 50 const size_t samples_per_channel = |
50 kSampleRateHz / 100 * num_10ms_frames_per_packet_; | 51 kSampleRateHz / 100 * num_10ms_frames_per_packet_; |
51 for (int i = 0; i < num_channels_; ++i) { | 52 for (int i = 0; i < num_channels_; ++i) { |
52 encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]); | 53 encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]); |
53 encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2); | 54 encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2); |
54 } | 55 } |
55 } | 56 } |
56 | 57 |
57 AudioEncoderG722::~AudioEncoderG722() {} | 58 AudioEncoderG722::~AudioEncoderG722() {} |
58 | 59 |
59 int AudioEncoderG722::SampleRateHz() const { | 60 int AudioEncoderG722::SampleRateHz() const { |
60 return kSampleRateHz; | 61 return kSampleRateHz; |
61 } | 62 } |
62 | 63 |
63 int AudioEncoderG722::RtpTimestampRateHz() const { | 64 int AudioEncoderG722::RtpTimestampRateHz() const { |
64 // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz | 65 // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz |
65 // codec. | 66 // codec. |
66 return kSampleRateHz / 2; | 67 return kSampleRateHz / 2; |
67 } | 68 } |
68 | 69 |
69 int AudioEncoderG722::NumChannels() const { | 70 int AudioEncoderG722::NumChannels() const { |
70 return num_channels_; | 71 return num_channels_; |
71 } | 72 } |
72 | 73 |
73 size_t AudioEncoderG722::MaxEncodedBytes() const { | 74 size_t AudioEncoderG722::MaxEncodedBytes() const { |
74 return static_cast<size_t>(SamplesPerChannel() / 2 * num_channels_); | 75 return SamplesPerChannel() / 2 * num_channels_; |
75 } | 76 } |
76 | 77 |
77 int AudioEncoderG722::Num10MsFramesInNextPacket() const { | 78 size_t AudioEncoderG722::Num10MsFramesInNextPacket() const { |
78 return num_10ms_frames_per_packet_; | 79 return num_10ms_frames_per_packet_; |
79 } | 80 } |
80 | 81 |
81 int AudioEncoderG722::Max10MsFramesInAPacket() const { | 82 size_t AudioEncoderG722::Max10MsFramesInAPacket() const { |
82 return num_10ms_frames_per_packet_; | 83 return num_10ms_frames_per_packet_; |
83 } | 84 } |
84 | 85 |
85 int AudioEncoderG722::GetTargetBitrate() const { | 86 int AudioEncoderG722::GetTargetBitrate() const { |
86 // 4 bits/sample, 16000 samples/s/channel. | 87 // 4 bits/sample, 16000 samples/s/channel. |
87 return 64000 * NumChannels(); | 88 return 64000 * NumChannels(); |
88 } | 89 } |
89 | 90 |
90 AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal( | 91 AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal( |
91 uint32_t rtp_timestamp, | 92 uint32_t rtp_timestamp, |
92 const int16_t* audio, | 93 const int16_t* audio, |
93 size_t max_encoded_bytes, | 94 size_t max_encoded_bytes, |
94 uint8_t* encoded) { | 95 uint8_t* encoded) { |
95 CHECK_GE(max_encoded_bytes, MaxEncodedBytes()); | 96 CHECK_GE(max_encoded_bytes, MaxEncodedBytes()); |
96 | 97 |
97 if (num_10ms_frames_buffered_ == 0) | 98 if (num_10ms_frames_buffered_ == 0) |
98 first_timestamp_in_buffer_ = rtp_timestamp; | 99 first_timestamp_in_buffer_ = rtp_timestamp; |
99 | 100 |
100 // Deinterleave samples and save them in each channel's buffer. | 101 // Deinterleave samples and save them in each channel's buffer. |
101 const int start = kSampleRateHz / 100 * num_10ms_frames_buffered_; | 102 const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_; |
102 for (int i = 0; i < kSampleRateHz / 100; ++i) | 103 for (size_t i = 0; i < kSampleRateHz / 100; ++i) |
103 for (int j = 0; j < num_channels_; ++j) | 104 for (int j = 0; j < num_channels_; ++j) |
104 encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j]; | 105 encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j]; |
105 | 106 |
106 // If we don't yet have enough samples for a packet, we're done for now. | 107 // If we don't yet have enough samples for a packet, we're done for now. |
107 if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) { | 108 if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) { |
108 return EncodedInfo(); | 109 return EncodedInfo(); |
109 } | 110 } |
110 | 111 |
111 // Encode each channel separately. | 112 // Encode each channel separately. |
112 CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_); | 113 CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_); |
113 num_10ms_frames_buffered_ = 0; | 114 num_10ms_frames_buffered_ = 0; |
114 const int samples_per_channel = SamplesPerChannel(); | 115 const size_t samples_per_channel = SamplesPerChannel(); |
115 for (int i = 0; i < num_channels_; ++i) { | 116 for (int i = 0; i < num_channels_; ++i) { |
116 const int encoded = WebRtcG722_Encode( | 117 const size_t encoded = WebRtcG722_Encode( |
117 encoders_[i].encoder, encoders_[i].speech_buffer.get(), | 118 encoders_[i].encoder, encoders_[i].speech_buffer.get(), |
118 samples_per_channel, encoders_[i].encoded_buffer.data<uint8_t>()); | 119 samples_per_channel, encoders_[i].encoded_buffer.data<uint8_t>()); |
119 CHECK_GE(encoded, 0); | |
120 CHECK_EQ(encoded, samples_per_channel / 2); | 120 CHECK_EQ(encoded, samples_per_channel / 2); |
121 } | 121 } |
122 | 122 |
123 // Interleave the encoded bytes of the different channels. Each separate | 123 // Interleave the encoded bytes of the different channels. Each separate |
124 // channel and the interleaved stream encodes two samples per byte, most | 124 // channel and the interleaved stream encodes two samples per byte, most |
125 // significant half first. | 125 // significant half first. |
126 for (int i = 0; i < samples_per_channel / 2; ++i) { | 126 for (size_t i = 0; i < samples_per_channel / 2; ++i) { |
127 for (int j = 0; j < num_channels_; ++j) { | 127 for (int j = 0; j < num_channels_; ++j) { |
128 uint8_t two_samples = encoders_[j].encoded_buffer.data()[i]; | 128 uint8_t two_samples = encoders_[j].encoded_buffer.data()[i]; |
129 interleave_buffer_.data()[j] = two_samples >> 4; | 129 interleave_buffer_.data()[j] = two_samples >> 4; |
130 interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf; | 130 interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf; |
131 } | 131 } |
132 for (int j = 0; j < num_channels_; ++j) | 132 for (int j = 0; j < num_channels_; ++j) |
133 encoded[i * num_channels_ + j] = interleave_buffer_.data()[2 * j] << 4 | | 133 encoded[i * num_channels_ + j] = interleave_buffer_.data()[2 * j] << 4 | |
134 interleave_buffer_.data()[2 * j + 1]; | 134 interleave_buffer_.data()[2 * j + 1]; |
135 } | 135 } |
136 EncodedInfo info; | 136 EncodedInfo info; |
137 info.encoded_bytes = samples_per_channel / 2 * num_channels_; | 137 info.encoded_bytes = samples_per_channel / 2 * num_channels_; |
138 info.encoded_timestamp = first_timestamp_in_buffer_; | 138 info.encoded_timestamp = first_timestamp_in_buffer_; |
139 info.payload_type = payload_type_; | 139 info.payload_type = payload_type_; |
140 return info; | 140 return info; |
141 } | 141 } |
142 | 142 |
143 int AudioEncoderG722::SamplesPerChannel() const { | 143 size_t AudioEncoderG722::SamplesPerChannel() const { |
144 return kSampleRateHz / 100 * num_10ms_frames_per_packet_; | 144 return kSampleRateHz / 100 * num_10ms_frames_per_packet_; |
145 } | 145 } |
146 | 146 |
147 namespace { | 147 namespace { |
148 AudioEncoderG722::Config CreateConfig(const CodecInst& codec_inst) { | 148 AudioEncoderG722::Config CreateConfig(const CodecInst& codec_inst) { |
149 AudioEncoderG722::Config config; | 149 AudioEncoderG722::Config config; |
150 config.num_channels = codec_inst.channels; | 150 config.num_channels = codec_inst.channels; |
151 config.frame_size_ms = codec_inst.pacsize / 16; | 151 config.frame_size_ms = codec_inst.pacsize / 16; |
152 config.payload_type = codec_inst.pltype; | 152 config.payload_type = codec_inst.pltype; |
153 return config; | 153 return config; |
154 } | 154 } |
155 } // namespace | 155 } // namespace |
156 | 156 |
157 AudioEncoderMutableG722::AudioEncoderMutableG722(const CodecInst& codec_inst) | 157 AudioEncoderMutableG722::AudioEncoderMutableG722(const CodecInst& codec_inst) |
158 : AudioEncoderMutableImpl<AudioEncoderG722>(CreateConfig(codec_inst)) { | 158 : AudioEncoderMutableImpl<AudioEncoderG722>(CreateConfig(codec_inst)) { |
159 } | 159 } |
160 | 160 |
161 } // namespace webrtc | 161 } // namespace webrtc |
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