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1 /* | |
2 * Copyright (c) 2017 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_processing/test/fake_recording_device.h" | |
12 | |
13 #include "webrtc/base/logging.h" | |
14 | |
15 namespace webrtc { | |
16 | |
17 FakeRecordingDevice::FakeRecordingDevice(LevelToScalingMappingKind mapping_kind) | |
18 : mapping_kind_(mapping_kind) {} | |
19 | |
20 FakeRecordingDevice::~FakeRecordingDevice() = default; | |
21 | |
22 void FakeRecordingDevice::SimulateAnalogGain( | |
23 std::vector<rtc::ArrayView<const float>> src, | |
24 std::vector<rtc::ArrayView<float>> dest, | |
25 int level, | |
26 int real_device_level) { | |
27 RTC_DCHECK_EQ(src.size(), dest.size()); | |
28 const float scaling_factor = ComputeCompoundScalingFactor( | |
29 level, real_device_level); | |
30 | |
31 for (size_t i = 0; i < src.size(); ++i) { | |
32 RTC_DCHECK_EQ(src[i].size(), dest[i].size()); | |
33 std::transform(src[i].begin(), src[i].end(), dest[i].begin(), | |
34 [scaling_factor](float x) { return scaling_factor * x; }); | |
peah-webrtc
2017/05/05 20:25:21
This scaling could create absolute sample values a
AleBzk
2017/05/16 08:53:03
Good point. And we should apply saturation twice:
peah-webrtc
2017/05/16 12:19:35
I don't see why we need the hard-clipping in step
| |
35 } | |
36 } | |
37 | |
38 void FakeRecordingDevice::SimulateAnalogGain(const AudioFrame* src, | |
39 AudioFrame* dest, | |
40 int level, | |
41 int real_device_level) { | |
42 RTC_DCHECK_EQ(src->num_channels_, dest->num_channels_); | |
43 RTC_DCHECK_EQ(src->samples_per_channel_, dest->samples_per_channel_); | |
44 const size_t number_of_samples = | |
45 src->samples_per_channel_ * src->num_channels_; | |
46 RTC_DCHECK_LE(number_of_samples, AudioFrame::kMaxDataSizeSamples); | |
47 const float scaling_factor = ComputeCompoundScalingFactor( | |
48 level, real_device_level); | |
49 | |
50 std::transform(src->data_, src->data_ + number_of_samples, dest->data_, | |
51 [scaling_factor](int16_t x) { | |
52 return rtc::saturated_cast<int16_t>(scaling_factor * x); | |
peah-webrtc
2017/05/05 20:25:21
This simulation of the microphone scaling is too s
AleBzk
2017/05/16 08:53:03
Great point! Thanks.
I redesigned what goes on be
| |
53 }); | |
54 } | |
55 | |
56 float FakeRecordingDevice::ComputeCompoundScalingFactor( | |
57 int level, int real_device_level) const { | |
58 const float scaling_factor = GetScalingFactor(level); | |
59 const float undo_factor = (real_device_level == kRealDeviceLevelUnknown) ? | |
60 1.0f : GetScalingFactor(real_device_level); | |
61 | |
62 if (undo_factor == 0) { | |
63 LOG(LS_VERBOSE) << "[fake rec device] Cannot undo analog level of 0."; | |
peah-webrtc
2017/05/05 20:25:21
As this is only simulation/testing code, you shoul
aleloi
2017/05/08 10:15:23
I think a level of 0 can appear in real recordings
peah-webrtc
2017/05/08 11:41:33
No, nothing that occurs in real recordings should
AleBzk
2017/05/16 08:53:03
I'm a bit confused here.
We shouldn't include erro
peah-webrtc
2017/05/16 12:19:35
The action to take there is totally dependent on t
| |
64 return scaling_factor; | |
65 } else { | |
66 return scaling_factor / undo_factor; | |
67 } | |
68 } | |
69 | |
70 float FakeRecordingDevice::GetScalingFactor(int level) const { | |
71 RTC_DCHECK_LE(0, level); | |
72 RTC_DCHECK_LE(level, 255); | |
73 | |
74 switch (mapping_kind_) { | |
75 case LevelToScalingMappingKind::kLinear: { | |
76 return static_cast<float>(level) / 255.0f; | |
77 } | |
78 case LevelToScalingMappingKind::kIdentity: { | |
79 return 1.0f; | |
80 } | |
81 default: { RTC_NOTREACHED(); } | |
82 } | |
83 } | |
84 | |
85 } // namespace webrtc | |
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