Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 66 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h b/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h
new file mode 100644
index 0000000000..0819bbcb23
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/capture_stream_info.h
@@ -0,0 +1,66 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
+#define MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
+
+#include <memory>
+#include <utility>
+
+#include "modules/audio_processing/include/aec_dump.h"
+#include "rtc_base/ignore_wundef.h"
+
+// Files generated at build-time by the protobuf compiler.
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
+#else
+#include "modules/audio_processing/debug.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+
+namespace webrtc {
+
+class CaptureStreamInfo {
+ public:
+  CaptureStreamInfo() { CreateNewEvent(); }
+  CaptureStreamInfo(const CaptureStreamInfo&) = delete;
+  CaptureStreamInfo& operator=(const CaptureStreamInfo&) = delete;
+  ~CaptureStreamInfo() = default;
+
+  void AddInput(const AudioFrameView<const float>& src);
+  void AddOutput(const AudioFrameView<const float>& src);
+
+  void AddInput(const int16_t* const data,
+                int num_channels,
+                int samples_per_channel);
+  void AddOutput(const int16_t* const data,
+                 int num_channels,
+                 int samples_per_channel);
+
+  void AddAudioProcessingState(const AecDump::AudioProcessingState& state);
+
+  std::unique_ptr<audioproc::Event> FetchEvent() {
+    std::unique_ptr<audioproc::Event> result = std::move(event_);
+    CreateNewEvent();
+    return result;
+  }
+
+ private:
+  void CreateNewEvent() {
+    event_ = std::make_unique<audioproc::Event>();
+    event_->set_type(audioproc::Event::STREAM);
+  }
+  std::unique_ptr<audioproc::Event> event_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_