Updating XTBs based on .GRDs from branch master
[chromium-blink-merge.git] / media / cast / test / utility / audio_utility.h
blob36ef858da104b89e525bc5a0d780586762381282
1 // Copyright 2014 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file.
5 #ifndef MEDIA_CAST_TEST_UTILITY_AUDIO_UTILITY_H_
6 #define MEDIA_CAST_TEST_UTILITY_AUDIO_UTILITY_H_
8 #include "media/audio/simple_sources.h"
10 namespace base {
11 class TimeDelta;
14 namespace media {
15 class AudioBus;
18 namespace media {
19 namespace cast {
21 // Produces AudioBuses of varying duration where each successive output contains
22 // the continuation of a single sine wave.
23 class TestAudioBusFactory {
24 public:
25 TestAudioBusFactory(int num_channels,
26 int sample_rate,
27 float sine_wave_frequency,
28 float volume);
29 ~TestAudioBusFactory();
31 // Creates a new AudioBus of the given |duration|, filled with the next batch
32 // of sine wave samples.
33 scoped_ptr<AudioBus> NextAudioBus(const base::TimeDelta& duration);
35 // A reasonable test tone.
36 static const int kMiddleANoteFreq = 440;
38 private:
39 const int num_channels_;
40 const int sample_rate_;
41 const float volume_;
42 SineWaveAudioSource source_;
44 DISALLOW_COPY_AND_ASSIGN(TestAudioBusFactory);
47 // Assuming |samples| contains a single-frequency sine wave (and maybe some
48 // low-amplitude noise), count the number of times the sine wave crosses
49 // zero.
51 // Example use case: When expecting a 440 Hz tone, this can be checked using the
52 // following expression:
54 // abs((CountZeroCrossings(...) / seconds_per_frame / 2) - 440) <= 1
56 // ...where seconds_per_frame is the number of samples divided by the sampling
57 // rate. The divide by two accounts for the fact that a sine wave crosses zero
58 // twice per cycle (first downwards, then upwards). The absolute maximum
59 // difference of 1 accounts for the sine wave being out of perfect phase.
60 int CountZeroCrossings(const float* samples, int length);
62 // Encode |timestamp| into the samples pointed to by 'samples' in a way
63 // that should be decodable even after compressing/decompressing the audio.
64 // Assumes 48Khz sampling rate and needs at least 240 samples. Returns
65 // false if |length| of |samples| is too small. If more than 240 samples are
66 // available, then the timestamp will be repeated. |sample_offset| should
67 // contain how many samples has been encoded so far, so that we can make smooth
68 // transitions between encoded chunks.
69 // See audio_utility.cc for details on how the encoding is done.
70 bool EncodeTimestamp(uint16 timestamp,
71 size_t sample_offset,
72 size_t length,
73 float* samples);
75 // Decode a timestamp encoded with EncodeTimestamp. Returns true if a
76 // timestamp was found in |samples|.
77 bool DecodeTimestamp(const float* samples, size_t length, uint16* timestamp);
79 } // namespace cast
80 } // namespace media
82 #endif // MEDIA_CAST_TEST_UTILITY_AUDIO_UTILITY_H_