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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.h
parentInitial commit. (diff)
downloadfirefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz
firefox-26a029d407be480d791972afb5975cf62c9360a6.zip
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.h')
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1 files changed, 78 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.h b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.h
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+++ b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_replayer.h
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+/*
+ * Copyright (c) 2016 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_TEST_DEBUG_DUMP_REPLAYER_H_
+#define MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "common_audio/channel_buffer.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/ignore_wundef.h"
+
+RTC_PUSH_IGNORING_WUNDEF()
+#include "modules/audio_processing/debug.pb.h"
+RTC_POP_IGNORING_WUNDEF()
+
+namespace webrtc {
+namespace test {
+
+class DebugDumpReplayer {
+ public:
+ DebugDumpReplayer();
+ ~DebugDumpReplayer();
+
+ // Set dump file
+ bool SetDumpFile(absl::string_view filename);
+
+ // Return next event.
+ absl::optional<audioproc::Event> GetNextEvent() const;
+
+ // Run the next event. Returns true if succeeded.
+ bool RunNextEvent();
+
+ const ChannelBuffer<float>* GetOutput() const;
+ StreamConfig GetOutputConfig() const;
+
+ private:
+ // Following functions are facilities for replaying debug dumps.
+ void OnInitEvent(const audioproc::Init& msg);
+ void OnStreamEvent(const audioproc::Stream& msg);
+ void OnReverseStreamEvent(const audioproc::ReverseStream& msg);
+ void OnConfigEvent(const audioproc::Config& msg);
+ void OnRuntimeSettingEvent(const audioproc::RuntimeSetting& msg);
+
+ void MaybeRecreateApm(const audioproc::Config& msg);
+ void ConfigureApm(const audioproc::Config& msg);
+
+ void LoadNextMessage();
+
+ // Buffer for APM input/output.
+ std::unique_ptr<ChannelBuffer<float>> input_;
+ std::unique_ptr<ChannelBuffer<float>> reverse_;
+ std::unique_ptr<ChannelBuffer<float>> output_;
+
+ rtc::scoped_refptr<AudioProcessing> apm_;
+
+ FILE* debug_file_;
+
+ StreamConfig input_config_;
+ StreamConfig reverse_config_;
+ StreamConfig output_config_;
+
+ bool has_next_event_;
+ audioproc::Event next_event_;
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_