Add aecdump support to audioproc_f.

webrtc/modules/audio_processing/test/audio_file_processor.h

+/*
+ *  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_