Files
platform-external-webrtc/webrtc/modules/audio_processing/test/debug_dump_replayer.h
kjellander c8fa692ec4 Update includes for webrtc/{base => rtc_base} rename (1/3)
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
2017-06-30 21:02:00 +00:00

78 lines
2.2 KiB
C++

/*
* 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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_
#include <memory>
#include <string>
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/rtc_base/ignore_wundef.h"
RTC_PUSH_IGNORING_WUNDEF()
#include "webrtc/modules/audio_processing/debug.pb.h"
RTC_POP_IGNORING_WUNDEF()
namespace webrtc {
namespace test {
class DebugDumpReplayer {
public:
DebugDumpReplayer();
~DebugDumpReplayer();
// Set dump file
bool SetDumpFile(const std::string& filename);
// Return next event.
rtc::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 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_;
std::unique_ptr<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 // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_