OLD | NEW |
---|---|
(Empty) | |
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/aec_dump/aec_dump.h" | |
12 | |
13 #include "webrtc/base/ignore_wundef.h" | |
14 // Files generated at build-time by the protobuf compiler. | |
15 RTC_PUSH_IGNORING_WUNDEF() | |
16 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD | |
17 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h" | |
18 #else | |
19 #include "webrtc/modules/audio_processing/debug.pb.h" | |
20 #endif | |
21 RTC_POP_IGNORING_WUNDEF() | |
22 | |
23 namespace webrtc { | |
24 AecDump::CaptureStreamInfo::CaptureStreamInfo() | |
25 : event_(new audioproc::Event()) { | |
peah-webrtc
2017/05/09 07:14:57
I think the separation of the capture stream infor
aleloi
2017/05/12 13:07:56
Changed after offline discussion.
To avoid creati
| |
26 RTC_DCHECK(event_); | |
27 event_->set_type(audioproc::Event::STREAM); | |
28 } | |
29 | |
30 AecDump::CaptureStreamInfo::~CaptureStreamInfo() { | |
31 // |event_| can't be in a unique_ptr, because of the no-protobuf | |
32 // implementation of AecDump::CaptureStreamInfo. In the no-protobuf | |
33 // case, a forward-declared audioproc::Event would have to be | |
34 // destroyed. | |
35 if (event_) { | |
36 delete event_; | |
37 } | |
38 } | |
39 | |
40 void AecDump::CaptureStreamInfo::AddInput(FloatAudioFrame src) { | |
41 auto* stream = event_->mutable_stream(); | |
42 | |
43 for (size_t i = 0; i < src.num_channels(); ++i) { | |
44 const auto& channel_view = src.channel(i); | |
45 stream->add_input_channel(channel_view.begin(), | |
46 sizeof(float) * channel_view.size()); | |
47 } | |
48 } | |
49 | |
50 void AecDump::CaptureStreamInfo::AddOutput(FloatAudioFrame src) { | |
51 auto* stream = event_->mutable_stream(); | |
52 | |
53 for (size_t i = 0; i < src.num_channels(); ++i) { | |
54 const auto& channel_view = src.channel(i); | |
55 stream->add_output_channel(channel_view.begin(), | |
56 sizeof(float) * channel_view.size()); | |
57 } | |
58 } | |
59 | |
60 void AecDump::CaptureStreamInfo::AddInput(const AudioFrame& frame) { | |
61 audioproc::Stream* stream = event_->mutable_stream(); | |
62 const size_t data_size = | |
63 sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_; | |
64 stream->set_input_data(frame.data_, data_size); | |
65 } | |
66 | |
67 void AecDump::CaptureStreamInfo::AddOutput(const AudioFrame& frame) { | |
68 audioproc::Stream* stream = event_->mutable_stream(); | |
69 const size_t data_size = | |
70 sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_; | |
71 stream->set_output_data(frame.data_, data_size); | |
72 } | |
73 | |
74 void AecDump::CaptureStreamInfo::set_delay(int delay) { | |
75 event_->mutable_stream()->set_delay(delay); | |
76 } | |
77 void AecDump::CaptureStreamInfo::set_drift(int drift) { | |
78 event_->mutable_stream()->set_drift(drift); | |
79 } | |
80 void AecDump::CaptureStreamInfo::set_level(int level) { | |
81 event_->mutable_stream()->set_level(level); | |
82 } | |
83 void AecDump::CaptureStreamInfo::set_keypress(bool keypress) { | |
84 event_->mutable_stream()->set_keypress(keypress); | |
85 } | |
86 | |
87 std::unique_ptr<audioproc::Event> AecDump::CaptureStreamInfo::GetEventMsg() { | |
peah-webrtc
2017/05/09 07:14:57
This part of the class is a bit error prone. It me
aleloi
2017/05/12 13:07:56
The code is rather different now after the Capture
| |
88 auto result = std::unique_ptr<audioproc::Event>(event_); | |
89 event_ = nullptr; | |
90 return result; | |
91 } | |
92 } // namespace webrtc | |
OLD | NEW |