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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/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 | |
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