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| 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 |
| 31 namespace webrtc { |
| 32 |
| 33 // Holds a few statistics about a series of TickIntervals. |
| 34 struct 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. |
| 43 class 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. |
| 84 class 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. |
| 107 class 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_ |
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