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Issue 2424173003: Move functionality out from AudioFrame and into AudioFrameOperations. (Closed)
Patch Set: Updated deprecation notice, minimized diff to original unittest. Created 4 years ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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 <algorithm>
12
13 #include "webrtc/audio/utility/audio_frame_operations.h"
11 #include "webrtc/modules/include/module_common_types.h" 14 #include "webrtc/modules/include/module_common_types.h"
12 #include "webrtc/modules/utility/include/audio_frame_operations.h"
13 #include "webrtc/base/checks.h" 15 #include "webrtc/base/checks.h"
14 16
15 namespace webrtc { 17 namespace webrtc {
16 namespace { 18 namespace {
17 19
18 // 2.7ms @ 48kHz, 4ms @ 32kHz, 8ms @ 16kHz. 20 // 2.7ms @ 48kHz, 4ms @ 32kHz, 8ms @ 16kHz.
19 const size_t kMuteFadeFrames = 128; 21 const size_t kMuteFadeFrames = 128;
20 const float kMuteFadeInc = 1.0f / kMuteFadeFrames; 22 const float kMuteFadeInc = 1.0f / kMuteFadeFrames;
21 23
22 } // namespace { 24 } // namespace
25
26 void AudioFrameOperations::Add(const AudioFrame& frame_to_add,
27 AudioFrame* result_frame) {
28 // Sanity check
29 RTC_DCHECK_GT(result_frame->num_channels_, 0u);
hlundin-webrtc 2016/11/30 11:24:36 I know we usually don't allow other than plain mon
aleloi 2016/11/30 11:33:39 Not really. If there is a chance that poly-channel
kwiberg-webrtc 2016/11/30 11:36:04 But there's no good reason to not remove the unnec
aleloi 2016/11/30 11:48:34 I think methods should in general only check their
hlundin-webrtc 2016/11/30 11:51:05 [Logic evaluation stalled due to double negation]
kwiberg-webrtc 2016/11/30 11:55:04 Yes, that's what I meant. I (mis-)interpreted Alex
30 RTC_DCHECK_LT(result_frame->num_channels_, 3u);
kwiberg-webrtc 2016/11/30 09:38:34 You don't need the "u"s anymore!
aleloi 2016/11/30 10:11:48 Done. Has the implementation of the DCHECK macros
kwiberg-webrtc 2016/11/30 11:32:44 Yes. (I sent a mail about it yesterday to discuss-
31 if ((result_frame->num_channels_ > 2) || (result_frame->num_channels_ < 1))
32 return;
33 if (result_frame->num_channels_ != frame_to_add.num_channels_)
34 return;
35
36 bool noPrevData = false;
hlundin-webrtc 2016/11/30 11:24:36 no_previous_data
37 if (result_frame->samples_per_channel_ != frame_to_add.samples_per_channel_) {
38 if (result_frame->samples_per_channel_ == 0) {
39 // special case we have no data to start with
hlundin-webrtc 2016/11/30 11:24:36 Capital start and end with '.'
40 result_frame->samples_per_channel_ = frame_to_add.samples_per_channel_;
41 noPrevData = true;
42 } else {
43 return;
44 }
45 }
46
47 if ((result_frame->vad_activity_ == AudioFrame::kVadActive) ||
48 frame_to_add.vad_activity_ == result_frame->kVadActive) {
kwiberg-webrtc 2016/11/30 09:38:34 Remove one layer of parentheses, since the relativ
aleloi 2016/11/30 10:11:48 Done.
49 result_frame->vad_activity_ = AudioFrame::kVadActive;
50 } else if (result_frame->vad_activity_ == AudioFrame::kVadUnknown ||
51 frame_to_add.vad_activity_ == AudioFrame::kVadUnknown) {
52 result_frame->vad_activity_ = AudioFrame::kVadUnknown;
53 }
54
55 if (result_frame->speech_type_ != frame_to_add.speech_type_)
56 result_frame->speech_type_ = AudioFrame::kUndefined;
57
58 if (noPrevData) {
59 std::copy(frame_to_add.data_, frame_to_add.data_ +
60 frame_to_add.samples_per_channel_ *
61 result_frame->num_channels_,
62 result_frame->data_);
63 } else {
64 for (size_t i = 0;
65 i < result_frame->samples_per_channel_ * result_frame->num_channels_;
66 i++) {
67 int32_t wrap_guard = static_cast<int32_t>(result_frame->data_[i]) +
kwiberg-webrtc 2016/11/30 09:38:34 const
aleloi 2016/11/30 10:11:48 Done.
68 static_cast<int32_t>(frame_to_add.data_[i]);
69 result_frame->data_[i] = ClampToInt16(wrap_guard);
70 }
71 }
72 return;
73 }
23 74
24 void AudioFrameOperations::MonoToStereo(const int16_t* src_audio, 75 void AudioFrameOperations::MonoToStereo(const int16_t* src_audio,
25 size_t samples_per_channel, 76 size_t samples_per_channel,
26 int16_t* dst_audio) { 77 int16_t* dst_audio) {
27 for (size_t i = 0; i < samples_per_channel; i++) { 78 for (size_t i = 0; i < samples_per_channel; i++) {
28 dst_audio[2 * i] = src_audio[i]; 79 dst_audio[2 * i] = src_audio[i];
29 dst_audio[2 * i + 1] = src_audio[i]; 80 dst_audio[2 * i + 1] = src_audio[i];
30 } 81 }
31 } 82 }
32 83
(...skipping 28 matching lines...) Expand all
61 return -1; 112 return -1;
62 } 113 }
63 114
64 StereoToMono(frame->data_, frame->samples_per_channel_, frame->data_); 115 StereoToMono(frame->data_, frame->samples_per_channel_, frame->data_);
65 frame->num_channels_ = 1; 116 frame->num_channels_ = 1;
66 117
67 return 0; 118 return 0;
68 } 119 }
69 120
70 void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) { 121 void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
71 if (frame->num_channels_ != 2) return; 122 if (frame->num_channels_ != 2)
123 return;
72 124
73 for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) { 125 for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
74 int16_t temp_data = frame->data_[i]; 126 int16_t temp_data = frame->data_[i];
75 frame->data_[i] = frame->data_[i + 1]; 127 frame->data_[i] = frame->data_[i + 1];
76 frame->data_[i + 1] = temp_data; 128 frame->data_[i + 1] = temp_data;
77 } 129 }
78 } 130 }
79 131
80 void AudioFrameOperations::Mute(AudioFrame* frame, bool previous_frame_muted, 132 void AudioFrameOperations::Mute(AudioFrame* frame,
133 bool previous_frame_muted,
81 bool current_frame_muted) { 134 bool current_frame_muted) {
82 RTC_DCHECK(frame); 135 RTC_DCHECK(frame);
83 if (!previous_frame_muted && !current_frame_muted) { 136 if (!previous_frame_muted && !current_frame_muted) {
84 // Not muted, don't touch. 137 // Not muted, don't touch.
85 } else if (previous_frame_muted && current_frame_muted) { 138 } else if (previous_frame_muted && current_frame_muted) {
86 // Frame fully muted. 139 // Frame fully muted.
87 size_t total_samples = frame->samples_per_channel_ * frame->num_channels_; 140 size_t total_samples = frame->samples_per_channel_ * frame->num_channels_;
88 RTC_DCHECK_GE(AudioFrame::kMaxDataSizeSamples, total_samples); 141 RTC_DCHECK_GE(AudioFrame::kMaxDataSizeSamples, total_samples);
89 memset(frame->data_, 0, sizeof(frame->data_[0]) * total_samples); 142 memset(frame->data_, 0, sizeof(frame->data_[0]) * total_samples);
90 } else { 143 } else {
(...skipping 27 matching lines...) Expand all
118 for (size_t j = 0; j < channels; ++j) { 171 for (size_t j = 0; j < channels; ++j) {
119 float g = start_g; 172 float g = start_g;
120 for (size_t i = start * channels; i < end * channels; i += channels) { 173 for (size_t i = start * channels; i < end * channels; i += channels) {
121 g += inc; 174 g += inc;
122 frame->data_[i + j] *= g; 175 frame->data_[i + j] *= g;
123 } 176 }
124 } 177 }
125 } 178 }
126 } 179 }
127 180
181 void AudioFrameOperations::Mute(AudioFrame* frame) {
182 Mute(frame, true, true);
183 }
184
185 void AudioFrameOperations::ApplyHalfGain(AudioFrame* frame) {
186 RTC_DCHECK_GT(frame->num_channels_, 0u);
187 RTC_DCHECK_LT(frame->num_channels_, 3u);
kwiberg-webrtc 2016/11/30 09:38:34 Remove "u"s.
aleloi 2016/11/30 10:11:48 Done.
188 if ((frame->num_channels_ > 2) || (frame->num_channels_ < 1))
kwiberg-webrtc 2016/11/30 09:38:34 Remove the extra parentheses.
aleloi 2016/11/30 10:11:48 Done.
hlundin-webrtc 2016/11/30 11:24:36 Again, does this function really need the restrict
aleloi 2016/11/30 11:33:38 No, there is no such assumption here either.
189 return;
190
191 for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
192 i++) {
193 frame->data_[i] = static_cast<int16_t>(frame->data_[i] >> 1);
194 }
195 }
196
128 int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) { 197 int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) {
129 if (frame.num_channels_ != 2) { 198 if (frame.num_channels_ != 2) {
130 return -1; 199 return -1;
131 } 200 }
132 201
133 for (size_t i = 0; i < frame.samples_per_channel_; i++) { 202 for (size_t i = 0; i < frame.samples_per_channel_; i++) {
134 frame.data_[2 * i] = 203 frame.data_[2 * i] = static_cast<int16_t>(left * frame.data_[2 * i]);
135 static_cast<int16_t>(left * frame.data_[2 * i]);
136 frame.data_[2 * i + 1] = 204 frame.data_[2 * i + 1] =
137 static_cast<int16_t>(right * frame.data_[2 * i + 1]); 205 static_cast<int16_t>(right * frame.data_[2 * i + 1]);
138 } 206 }
139 return 0; 207 return 0;
140 } 208 }
141 209
142 int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame& frame) { 210 int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame& frame) {
143 int32_t temp_data = 0; 211 int32_t temp_data = 0;
144 212
145 // Ensure that the output result is saturated [-32768, +32767]. 213 // Ensure that the output result is saturated [-32768, +32767].
146 for (size_t i = 0; i < frame.samples_per_channel_ * frame.num_channels_; 214 for (size_t i = 0; i < frame.samples_per_channel_ * frame.num_channels_;
147 i++) { 215 i++) {
148 temp_data = static_cast<int32_t>(scale * frame.data_[i]); 216 temp_data = static_cast<int32_t>(scale * frame.data_[i]);
149 if (temp_data < -32768) { 217 if (temp_data < -32768) {
150 frame.data_[i] = -32768; 218 frame.data_[i] = -32768;
151 } else if (temp_data > 32767) { 219 } else if (temp_data > 32767) {
152 frame.data_[i] = 32767; 220 frame.data_[i] = 32767;
153 } else { 221 } else {
154 frame.data_[i] = static_cast<int16_t>(temp_data); 222 frame.data_[i] = static_cast<int16_t>(temp_data);
155 } 223 }
156 } 224 }
157 return 0; 225 return 0;
158 } 226 }
159 227
228 int16_t ClampToInt16(int32_t input) {
229 if (input < -0x00008000) {
230 return -0x8000;
231 } else if (input > 0x00007FFF) {
232 return 0x7FFF;
233 } else {
234 return static_cast<int16_t>(input);
235 }
236 }
kwiberg-webrtc 2016/11/30 09:38:34 Instead of defining this function, use rtc::satura
aleloi 2016/11/30 10:11:48 Done. Great that we have saturated_cast! I missed
kwiberg-webrtc 2016/11/30 11:32:44 Except for the name---it should have been "saturat
237
160 } // namespace webrtc 238 } // namespace webrtc
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