AecDump implementation.

webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc

Index: webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc
diff --git a/webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc b/webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc
new file mode 100644
index 0000000000000000000000000000000000000000..5450440886fdcd453d1d3870356f4ef9722f8af6
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/aec_dump/capture_stream_info.h"
+
+namespace webrtc {
+CaptureStreamInfo::CaptureStreamInfo(std::unique_ptr<WriteToFileTask> task)
+    : task_(std::move(task)) {
+  RTC_DCHECK(task_);
+  task_->GetEvent()->set_type(audioproc::Event::STREAM);
+}
+
+CaptureStreamInfo::~CaptureStreamInfo() = default;
+
+void CaptureStreamInfo::AddInput(const FloatAudioFrame& src) {
+  RTC_DCHECK(task_);
+  auto* stream = task_->GetEvent()->mutable_stream();
+
+  for (size_t i = 0; i < src.num_channels(); ++i) {
+    const auto& channel_view = src.channel(i);
+    stream->add_input_channel(channel_view.begin(),
+                              sizeof(float) * channel_view.size());
+  }
+}
+
+void CaptureStreamInfo::AddOutput(const FloatAudioFrame& src) {
+  RTC_DCHECK(task_);
+  auto* stream = task_->GetEvent()->mutable_stream();
+
+  for (size_t i = 0; i < src.num_channels(); ++i) {
+    const auto& channel_view = src.channel(i);
+    stream->add_output_channel(channel_view.begin(),
+                               sizeof(float) * channel_view.size());
+  }
+}
+
+void CaptureStreamInfo::AddInput(const AudioFrame& frame) {
+  RTC_DCHECK(task_);
+  audioproc::Stream* stream = task_->GetEvent()->mutable_stream();

[peah-webrtc, 2017/05/16 06:30:38]

auto* to conform to 24 and 35?

[aleloi, 2017/05/16 20:10:17]

Done.

+  const size_t data_size =
+      sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_;
+  stream->set_input_data(frame.data_, data_size);
+}
+
+void CaptureStreamInfo::AddOutput(const AudioFrame& frame) {
+  RTC_DCHECK(task_);
+  audioproc::Stream* stream = task_->GetEvent()->mutable_stream();

[peah-webrtc, 2017/05/16 06:30:38]

auto* to conform to 24 and 35?

[aleloi, 2017/05/16 20:10:17]

Done.

+  const size_t data_size =
+      sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_;
+  stream->set_output_data(frame.data_, data_size);
+}
+
+void CaptureStreamInfo::AddAudioProcessingState(
+    const AudioProcessingState& state) {
+  RTC_DCHECK(task_);
+  auto* stream = task_->GetEvent()->mutable_stream();
+  stream->set_delay(state.delay);
+  stream->set_drift(state.drift);
+  stream->set_level(state.level);
+  stream->set_keypress(state.keypress);
+}
+}  // namespace webrtc