| Index: webrtc/modules/audio_processing/test/audio_file_processor.h
|
| diff --git a/webrtc/modules/audio_processing/test/audio_file_processor.h b/webrtc/modules/audio_processing/test/audio_file_processor.h
|
| deleted file mode 100644
|
| index a3153b2244cb57b6edc67ad233ebc55501d135be..0000000000000000000000000000000000000000
|
| --- a/webrtc/modules/audio_processing/test/audio_file_processor.h
|
| +++ /dev/null
|
| @@ -1,139 +0,0 @@
|
| -/*
|
| - * Copyright (c) 2015 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_AUDIO_FILE_PROCESSOR_H_
|
| -#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
|
| -
|
| -#include <algorithm>
|
| -#include <limits>
|
| -#include <vector>
|
| -
|
| -#include "webrtc/base/scoped_ptr.h"
|
| -#include "webrtc/common_audio/channel_buffer.h"
|
| -#include "webrtc/common_audio/wav_file.h"
|
| -#include "webrtc/modules/audio_processing/include/audio_processing.h"
|
| -#include "webrtc/modules/audio_processing/test/test_utils.h"
|
| -#include "webrtc/system_wrappers/include/tick_util.h"
|
| -
|
| -#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
|
| -#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
|
| -#else
|
| -#include "webrtc/audio_processing/debug.pb.h"
|
| -#endif
|
| -
|
| -namespace webrtc {
|
| -
|
| -// Holds a few statistics about a series of TickIntervals.
|
| -struct TickIntervalStats {
|
| - TickIntervalStats() : min(std::numeric_limits<int64_t>::max()) {}
|
| - TickInterval sum;
|
| - TickInterval max;
|
| - TickInterval min;
|
| -};
|
| -
|
| -// Interface for processing an input file with an AudioProcessing instance and
|
| -// dumping the results to an output file.
|
| -class AudioFileProcessor {
|
| - public:
|
| - static const int kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs;
|
| -
|
| - virtual ~AudioFileProcessor() {}
|
| -
|
| - // Processes one AudioProcessing::kChunkSizeMs of data from the input file and
|
| - // writes to the output file.
|
| - virtual bool ProcessChunk() = 0;
|
| -
|
| - // Returns the execution time of all AudioProcessing calls.
|
| - const TickIntervalStats& proc_time() const { return proc_time_; }
|
| -
|
| - protected:
|
| - // RAII class for execution time measurement. Updates the provided
|
| - // TickIntervalStats based on the time between ScopedTimer creation and
|
| - // leaving the enclosing scope.
|
| - class ScopedTimer {
|
| - public:
|
| - explicit ScopedTimer(TickIntervalStats* proc_time)
|
| - : proc_time_(proc_time), start_time_(TickTime::Now()) {}
|
| -
|
| - ~ScopedTimer() {
|
| - TickInterval interval = TickTime::Now() - start_time_;
|
| - proc_time_->sum += interval;
|
| - proc_time_->max = std::max(proc_time_->max, interval);
|
| - proc_time_->min = std::min(proc_time_->min, interval);
|
| - }
|
| -
|
| - private:
|
| - TickIntervalStats* const proc_time_;
|
| - TickTime start_time_;
|
| - };
|
| -
|
| - TickIntervalStats* mutable_proc_time() { return &proc_time_; }
|
| -
|
| - private:
|
| - TickIntervalStats proc_time_;
|
| -};
|
| -
|
| -// Used to read from and write to WavFile objects.
|
| -class WavFileProcessor final : public AudioFileProcessor {
|
| - public:
|
| - // Takes ownership of all parameters.
|
| - WavFileProcessor(rtc::scoped_ptr<AudioProcessing> ap,
|
| - rtc::scoped_ptr<WavReader> in_file,
|
| - rtc::scoped_ptr<WavWriter> out_file);
|
| - virtual ~WavFileProcessor() {}
|
| -
|
| - // Processes one chunk from the WAV input and writes to the WAV output.
|
| - bool ProcessChunk() override;
|
| -
|
| - private:
|
| - rtc::scoped_ptr<AudioProcessing> ap_;
|
| -
|
| - ChannelBuffer<float> in_buf_;
|
| - ChannelBuffer<float> out_buf_;
|
| - const StreamConfig input_config_;
|
| - const StreamConfig output_config_;
|
| - ChannelBufferWavReader buffer_reader_;
|
| - ChannelBufferWavWriter buffer_writer_;
|
| -};
|
| -
|
| -// Used to read from an aecdump file and write to a WavWriter.
|
| -class AecDumpFileProcessor final : public AudioFileProcessor {
|
| - public:
|
| - // Takes ownership of all parameters.
|
| - AecDumpFileProcessor(rtc::scoped_ptr<AudioProcessing> ap,
|
| - FILE* dump_file,
|
| - rtc::scoped_ptr<WavWriter> out_file);
|
| -
|
| - virtual ~AecDumpFileProcessor();
|
| -
|
| - // Processes messages from the aecdump file until the first Stream message is
|
| - // completed. Passes other data from the aecdump messages as appropriate.
|
| - bool ProcessChunk() override;
|
| -
|
| - private:
|
| - void HandleMessage(const webrtc::audioproc::Init& msg);
|
| - void HandleMessage(const webrtc::audioproc::Stream& msg);
|
| - void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
|
| -
|
| - rtc::scoped_ptr<AudioProcessing> ap_;
|
| - FILE* dump_file_;
|
| -
|
| - rtc::scoped_ptr<ChannelBuffer<float>> in_buf_;
|
| - rtc::scoped_ptr<ChannelBuffer<float>> reverse_buf_;
|
| - ChannelBuffer<float> out_buf_;
|
| - StreamConfig input_config_;
|
| - StreamConfig reverse_config_;
|
| - const StreamConfig output_config_;
|
| - ChannelBufferWavWriter buffer_writer_;
|
| -};
|
| -
|
| -} // namespace webrtc
|
| -
|
| -#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
|
|
|