1/*
2 *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10
11#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
12#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
13
14#include <algorithm>
15#include <limits>
16#include <vector>
17
18#include "webrtc/base/scoped_ptr.h"
19#include "webrtc/common_audio/channel_buffer.h"
20#include "webrtc/common_audio/wav_file.h"
21#include "webrtc/modules/audio_processing/include/audio_processing.h"
22#include "webrtc/modules/audio_processing/test/test_utils.h"
23#include "webrtc/system_wrappers/include/tick_util.h"
24
25#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
26#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
27#else
28#include "webrtc/audio_processing/debug.pb.h"
29#endif
30
31namespace webrtc {
32
33// Holds a few statistics about a series of TickIntervals.
34struct TickIntervalStats {
35  TickIntervalStats() : min(std::numeric_limits<int64_t>::max()) {}
36  TickInterval sum;
37  TickInterval max;
38  TickInterval min;
39};
40
41// Interface for processing an input file with an AudioProcessing instance and
42// dumping the results to an output file.
43class AudioFileProcessor {
44 public:
45  static const int kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs;
46
47  virtual ~AudioFileProcessor() {}
48
49  // Processes one AudioProcessing::kChunkSizeMs of data from the input file and
50  // writes to the output file.
51  virtual bool ProcessChunk() = 0;
52
53  // Returns the execution time of all AudioProcessing calls.
54  const TickIntervalStats& proc_time() const { return proc_time_; }
55
56 protected:
57  // RAII class for execution time measurement. Updates the provided
58  // TickIntervalStats based on the time between ScopedTimer creation and
59  // leaving the enclosing scope.
60  class ScopedTimer {
61   public:
62    explicit ScopedTimer(TickIntervalStats* proc_time)
63        : proc_time_(proc_time), start_time_(TickTime::Now()) {}
64
65    ~ScopedTimer() {
66      TickInterval interval = TickTime::Now() - start_time_;
67      proc_time_->sum += interval;
68      proc_time_->max = std::max(proc_time_->max, interval);
69      proc_time_->min = std::min(proc_time_->min, interval);
70    }
71
72   private:
73    TickIntervalStats* const proc_time_;
74    TickTime start_time_;
75  };
76
77  TickIntervalStats* mutable_proc_time() { return &proc_time_; }
78
79 private:
80  TickIntervalStats proc_time_;
81};
82
83// Used to read from and write to WavFile objects.
84class WavFileProcessor final : public AudioFileProcessor {
85 public:
86  // Takes ownership of all parameters.
87  WavFileProcessor(rtc::scoped_ptr<AudioProcessing> ap,
88                   rtc::scoped_ptr<WavReader> in_file,
89                   rtc::scoped_ptr<WavWriter> out_file);
90  virtual ~WavFileProcessor() {}
91
92  // Processes one chunk from the WAV input and writes to the WAV output.
93  bool ProcessChunk() override;
94
95 private:
96  rtc::scoped_ptr<AudioProcessing> ap_;
97
98  ChannelBuffer<float> in_buf_;
99  ChannelBuffer<float> out_buf_;
100  const StreamConfig input_config_;
101  const StreamConfig output_config_;
102  ChannelBufferWavReader buffer_reader_;
103  ChannelBufferWavWriter buffer_writer_;
104};
105
106// Used to read from an aecdump file and write to a WavWriter.
107class AecDumpFileProcessor final : public AudioFileProcessor {
108 public:
109  // Takes ownership of all parameters.
110  AecDumpFileProcessor(rtc::scoped_ptr<AudioProcessing> ap,
111                       FILE* dump_file,
112                       rtc::scoped_ptr<WavWriter> out_file);
113
114  virtual ~AecDumpFileProcessor();
115
116  // Processes messages from the aecdump file until the first Stream message is
117  // completed. Passes other data from the aecdump messages as appropriate.
118  bool ProcessChunk() override;
119
120 private:
121  void HandleMessage(const webrtc::audioproc::Init& msg);
122  void HandleMessage(const webrtc::audioproc::Stream& msg);
123  void HandleMessage(const webrtc::audioproc::ReverseStream& msg);
124
125  rtc::scoped_ptr<AudioProcessing> ap_;
126  FILE* dump_file_;
127
128  rtc::scoped_ptr<ChannelBuffer<float>> in_buf_;
129  rtc::scoped_ptr<ChannelBuffer<float>> reverse_buf_;
130  ChannelBuffer<float> out_buf_;
131  StreamConfig input_config_;
132  StreamConfig reverse_config_;
133  const StreamConfig output_config_;
134  ChannelBufferWavWriter buffer_writer_;
135};
136
137}  // namespace webrtc
138
139#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_FILE_PROCESSOR_H_
140