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_ |