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In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
77 lines
2.2 KiB
C++
77 lines
2.2 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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_
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#define WEBRTC_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 "webrtc/common_audio/channel_buffer.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/rtc_base/ignore_wundef.h"
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RTC_PUSH_IGNORING_WUNDEF()
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#include "webrtc/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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rtc::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 // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_REPLAYER_H_
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