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db9f7ab9f9
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'modules/audio_processing'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Id29f8de59dba704787c2c38a3d05c60827c181b0
Reviewed-on: https://webrtc-review.googlesource.com/83982
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23653}
77 lines
2.1 KiB
C++
77 lines
2.1 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_
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#define MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_
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#include <memory>
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#include <string>
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#include "common_audio/channel_buffer.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "rtc_base/ignore_wundef.h"
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RTC_PUSH_IGNORING_WUNDEF()
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#include "modules/audio_processing/debug.pb.h"
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RTC_POP_IGNORING_WUNDEF()
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namespace webrtc {
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namespace test {
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class DebugDumpReplayer {
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public:
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DebugDumpReplayer();
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~DebugDumpReplayer();
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// Set dump file
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bool SetDumpFile(const std::string& filename);
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// Return next event.
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absl::optional<audioproc::Event> GetNextEvent() const;
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// Run the next event. Returns true if succeeded.
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bool RunNextEvent();
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const ChannelBuffer<float>* GetOutput() const;
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StreamConfig GetOutputConfig() const;
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private:
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// Following functions are facilities for replaying debug dumps.
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void OnInitEvent(const audioproc::Init& msg);
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void OnStreamEvent(const audioproc::Stream& msg);
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void OnReverseStreamEvent(const audioproc::ReverseStream& msg);
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void OnConfigEvent(const audioproc::Config& msg);
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void MaybeRecreateApm(const audioproc::Config& msg);
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void ConfigureApm(const audioproc::Config& msg);
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void LoadNextMessage();
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// Buffer for APM input/output.
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std::unique_ptr<ChannelBuffer<float>> input_;
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std::unique_ptr<ChannelBuffer<float>> reverse_;
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std::unique_ptr<ChannelBuffer<float>> output_;
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std::unique_ptr<AudioProcessing> apm_;
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FILE* debug_file_;
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StreamConfig input_config_;
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StreamConfig reverse_config_;
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StreamConfig output_config_;
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bool has_next_event_;
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audioproc::Event next_event_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_
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