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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_DEBUG_DUMP_TEST_H_ | |
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_TEST_H_ | |
13 | |
14 #include <stddef.h> // size_t | |
15 #include <string> | |
16 #include <vector> | |
17 | |
18 #include "testing/gtest/include/gtest/gtest.h" | |
19 #include "webrtc/base/scoped_ptr.h" | |
20 #include "webrtc/common_audio/channel_buffer.h" | |
21 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h" | |
22 #include "webrtc/modules/audio_processing/include/audio_processing.h" | |
23 | |
24 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP | |
25 #include "webrtc/audio_processing/debug.pb.h" | |
26 #endif // WEBRTC_AUDIOPROC_DEBUG_DUMP | |
27 | |
28 namespace webrtc { | |
29 namespace test { | |
30 | |
31 class DebugDumpGenerator { | |
Andrew MacDonald
2015/10/20 01:22:09
Do you foresee anything else ever using this? If n
minyue-webrtc
2015/10/23 08:44:46
I thought that cc is too large and therefore put t
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32 public: | |
33 DebugDumpGenerator(std::string input_file_name, | |
34 int input_file_rate_hz, | |
35 size_t input_channels, | |
Andrew MacDonald
2015/10/20 01:22:09
Since these values are ints in the protobuf genera
minyue-webrtc
2015/10/23 08:44:46
Done.
| |
36 std::string reverse_file_name, | |
37 int reverse_file_rate_hz, | |
38 size_t reverse_channels, | |
39 const Config& config, | |
40 std::string dump_file_name); | |
41 | |
42 // Changes the sample rate of the input audio to the APM. | |
43 void SetInputRate(int rate_hz); | |
44 | |
45 // Sets if converts stereo input signal to mono by discarding other channels. | |
46 void ForceInputMono(bool mono); | |
47 | |
48 // Changes the sample rate of the reverse audio to the APM. | |
49 void SetReverseRate(int rate_hz); | |
50 | |
51 // Sets if converts stereo reverse signal to mono by discarding other | |
52 // channels. | |
53 void ForceReverseMono(bool mono); | |
54 | |
55 // Sets the required sample rate of the APM output. | |
56 void set_output_rate_hz(int rate_hz) { | |
57 output_rate_hz_ = rate_hz; | |
58 } | |
59 | |
60 // Sets the required channels of the APM output. | |
61 void set_output_channels(int channels) { | |
62 output_channels_ = channels; | |
63 } | |
64 | |
65 void StartRecording(); | |
66 void Process(size_t num_blocks); | |
67 void StopRecording(); | |
68 AudioProcessing* apm() const { return apm_.get(); } | |
69 | |
70 private: | |
71 void ReadAndDeinterleave(ResampleInputAudioFile* audio, size_t channels, | |
72 size_t frames_per_channel, bool force_mono, | |
73 float* const* buffer); | |
74 | |
75 // APM input/output settings. | |
76 int input_rate_hz_; | |
Andrew MacDonald
2015/10/20 01:22:09
Can any of these be const?
minyue-webrtc
2015/10/23 08:44:46
No. We want to change sample rate in the middle, t
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77 bool input_mono_; | |
78 int reverse_rate_hz_; | |
79 bool reverse_mono_; | |
80 int output_rate_hz_; | |
81 size_t output_channels_; | |
82 | |
83 // Input file format. | |
84 rtc::scoped_ptr<ResampleInputAudioFile> input_audio_; | |
Andrew MacDonald
2015/10/20 01:22:09
You initialize the ResampleInputAudioFiles in the
minyue-webrtc
2015/10/23 08:44:46
Done.
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85 size_t input_channels_; | |
Andrew MacDonald
2015/10/20 01:22:09
const?
minyue-webrtc
2015/10/23 08:44:46
Yes, this can be const
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86 | |
87 // Reverse file format. | |
88 rtc::scoped_ptr<ResampleInputAudioFile> reverse_audio_; | |
89 size_t reverse_channels_; | |
90 | |
91 // Buffer for APM input/output. | |
92 rtc::scoped_ptr<ChannelBuffer<float>> input_; | |
93 rtc::scoped_ptr<ChannelBuffer<float>> reverse_; | |
94 rtc::scoped_ptr<ChannelBuffer<float>> output_; | |
95 | |
96 rtc::scoped_ptr<AudioProcessing> apm_; | |
97 | |
98 std::string dump_file_name_; | |
Andrew MacDonald
2015/10/20 01:22:09
const
minyue-webrtc
2015/10/23 08:44:46
Done.
| |
99 | |
100 // Buffer for reading audio files. | |
101 std::vector<int16_t> signal_; | |
102 }; | |
103 | |
104 class DebugDumpTest : public ::testing::Test { | |
Andrew MacDonald
2015/10/20 01:22:09
This should definitely go in the cc file.
minyue-webrtc
2015/10/23 08:44:46
Done.
| |
105 public: | |
106 DebugDumpTest(); | |
107 | |
108 // VerifyDebugDump replays a debug dump using APM and verifies that the result | |
109 // is bit-exact identical to the output channel in the dump. This is only | |
110 // guaranteed if the debug dump is started on the first frame. | |
111 void VerifyDebugDump(std::string dump_file_name); | |
112 | |
113 private: | |
114 // Following functions are facilities for replaying debug dumps. | |
115 void OnInitEvent(const audioproc::Init& msg); | |
116 void OnStreamEvent(const audioproc::Stream& msg); | |
117 void OnReverseStreamEvent(const audioproc::ReverseStream& msg); | |
118 void OnConfigEvent(const audioproc::Config& msg); | |
119 void MaybeRecreateApm(const audioproc::Config& msg); | |
120 void ConfigurateApm(const audioproc::Config& msg); | |
121 | |
122 int input_rate_hz_; | |
123 size_t input_channels_; | |
124 | |
125 int output_rate_hz_; | |
126 size_t output_channels_; | |
127 | |
128 int reverse_rate_hz_; | |
129 size_t reverse_channels_; | |
130 | |
131 // Buffer for APM input/output. | |
132 rtc::scoped_ptr<ChannelBuffer<float>> input_; | |
133 rtc::scoped_ptr<ChannelBuffer<float>> reverse_; | |
134 rtc::scoped_ptr<ChannelBuffer<float>> output_; | |
135 | |
136 rtc::scoped_ptr<AudioProcessing> apm_; | |
137 }; | |
138 | |
139 } // namespace test | |
140 } // namespace webrtc | |
141 | |
142 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_TEST_H_ | |
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