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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2012 Google Inc. | 3 * Copyright 2012 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, | 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, |
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR | 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR |
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF | 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
26 */ | 26 */ |
27 | 27 |
28 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h" | 28 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h" |
29 | 29 |
30 #include <algorithm> | 30 #include <algorithm> |
31 | 31 |
| 32 #include "webrtc/base/criticalsection.h" |
32 #include "webrtc/base/gunit.h" | 33 #include "webrtc/base/gunit.h" |
33 #include "webrtc/base/scoped_ref_ptr.h" | 34 #include "webrtc/base/scoped_ref_ptr.h" |
34 #include "webrtc/base/thread.h" | 35 #include "webrtc/base/thread.h" |
35 | 36 |
36 using std::min; | 37 using std::min; |
37 | 38 |
38 class FakeAdmTest : public testing::Test, | 39 class FakeAdmTest : public testing::Test, |
39 public webrtc::AudioTransport { | 40 public webrtc::AudioTransport { |
40 protected: | 41 protected: |
41 static const int kMsInSecond = 1000; | 42 static const int kMsInSecond = 1000; |
42 | 43 |
43 FakeAdmTest() | 44 FakeAdmTest() |
44 : push_iterations_(0), | 45 : push_iterations_(0), |
45 pull_iterations_(0), | 46 pull_iterations_(0), |
46 rec_buffer_bytes_(0) { | 47 rec_buffer_bytes_(0) { |
47 memset(rec_buffer_, 0, sizeof(rec_buffer_)); | 48 memset(rec_buffer_, 0, sizeof(rec_buffer_)); |
48 } | 49 } |
49 | 50 |
50 virtual void SetUp() { | 51 virtual void SetUp() { |
51 fake_audio_capture_module_ = FakeAudioCaptureModule::Create( | 52 fake_audio_capture_module_ = FakeAudioCaptureModule::Create(); |
52 rtc::Thread::Current()); | |
53 EXPECT_TRUE(fake_audio_capture_module_.get() != NULL); | 53 EXPECT_TRUE(fake_audio_capture_module_.get() != NULL); |
54 } | 54 } |
55 | 55 |
56 // Callbacks inherited from webrtc::AudioTransport. | 56 // Callbacks inherited from webrtc::AudioTransport. |
57 // ADM is pushing data. | 57 // ADM is pushing data. |
58 int32_t RecordedDataIsAvailable(const void* audioSamples, | 58 int32_t RecordedDataIsAvailable(const void* audioSamples, |
59 const uint32_t nSamples, | 59 const uint32_t nSamples, |
60 const uint8_t nBytesPerSample, | 60 const uint8_t nBytesPerSample, |
61 const uint8_t nChannels, | 61 const uint8_t nChannels, |
62 const uint32_t samplesPerSec, | 62 const uint32_t samplesPerSec, |
63 const uint32_t totalDelayMS, | 63 const uint32_t totalDelayMS, |
64 const int32_t clockDrift, | 64 const int32_t clockDrift, |
65 const uint32_t currentMicLevel, | 65 const uint32_t currentMicLevel, |
66 const bool keyPressed, | 66 const bool keyPressed, |
67 uint32_t& newMicLevel) override { | 67 uint32_t& newMicLevel) override { |
| 68 rtc::CritScope cs(&crit_); |
68 rec_buffer_bytes_ = nSamples * nBytesPerSample; | 69 rec_buffer_bytes_ = nSamples * nBytesPerSample; |
69 if ((rec_buffer_bytes_ == 0) || | 70 if ((rec_buffer_bytes_ == 0) || |
70 (rec_buffer_bytes_ > FakeAudioCaptureModule::kNumberSamples * | 71 (rec_buffer_bytes_ > FakeAudioCaptureModule::kNumberSamples * |
71 FakeAudioCaptureModule::kNumberBytesPerSample)) { | 72 FakeAudioCaptureModule::kNumberBytesPerSample)) { |
72 ADD_FAILURE(); | 73 ADD_FAILURE(); |
73 return -1; | 74 return -1; |
74 } | 75 } |
75 memcpy(rec_buffer_, audioSamples, rec_buffer_bytes_); | 76 memcpy(rec_buffer_, audioSamples, rec_buffer_bytes_); |
76 ++push_iterations_; | 77 ++push_iterations_; |
77 newMicLevel = currentMicLevel; | 78 newMicLevel = currentMicLevel; |
78 return 0; | 79 return 0; |
79 } | 80 } |
80 | 81 |
81 // ADM is pulling data. | 82 // ADM is pulling data. |
82 int32_t NeedMorePlayData(const uint32_t nSamples, | 83 int32_t NeedMorePlayData(const uint32_t nSamples, |
83 const uint8_t nBytesPerSample, | 84 const uint8_t nBytesPerSample, |
84 const uint8_t nChannels, | 85 const uint8_t nChannels, |
85 const uint32_t samplesPerSec, | 86 const uint32_t samplesPerSec, |
86 void* audioSamples, | 87 void* audioSamples, |
87 uint32_t& nSamplesOut, | 88 uint32_t& nSamplesOut, |
88 int64_t* elapsed_time_ms, | 89 int64_t* elapsed_time_ms, |
89 int64_t* ntp_time_ms) override { | 90 int64_t* ntp_time_ms) override { |
| 91 rtc::CritScope cs(&crit_); |
90 ++pull_iterations_; | 92 ++pull_iterations_; |
91 const uint32_t audio_buffer_size = nSamples * nBytesPerSample; | 93 const uint32_t audio_buffer_size = nSamples * nBytesPerSample; |
92 const uint32_t bytes_out = RecordedDataReceived() ? | 94 const uint32_t bytes_out = RecordedDataReceived() ? |
93 CopyFromRecBuffer(audioSamples, audio_buffer_size): | 95 CopyFromRecBuffer(audioSamples, audio_buffer_size): |
94 GenerateZeroBuffer(audioSamples, audio_buffer_size); | 96 GenerateZeroBuffer(audioSamples, audio_buffer_size); |
95 nSamplesOut = bytes_out / nBytesPerSample; | 97 nSamplesOut = bytes_out / nBytesPerSample; |
96 *elapsed_time_ms = 0; | 98 *elapsed_time_ms = 0; |
97 *ntp_time_ms = 0; | 99 *ntp_time_ms = 0; |
98 return 0; | 100 return 0; |
99 } | 101 } |
100 | 102 |
101 int push_iterations() const { return push_iterations_; } | 103 int push_iterations() const { |
102 int pull_iterations() const { return pull_iterations_; } | 104 rtc::CritScope cs(&crit_); |
| 105 return push_iterations_; |
| 106 } |
| 107 int pull_iterations() const { |
| 108 rtc::CritScope cs(&crit_); |
| 109 return pull_iterations_; |
| 110 } |
103 | 111 |
104 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; | 112 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; |
105 | 113 |
106 private: | 114 private: |
107 bool RecordedDataReceived() const { | 115 bool RecordedDataReceived() const { |
108 return rec_buffer_bytes_ != 0; | 116 return rec_buffer_bytes_ != 0; |
109 } | 117 } |
110 int32_t GenerateZeroBuffer(void* audio_buffer, uint32_t audio_buffer_size) { | 118 int32_t GenerateZeroBuffer(void* audio_buffer, uint32_t audio_buffer_size) { |
111 memset(audio_buffer, 0, audio_buffer_size); | 119 memset(audio_buffer, 0, audio_buffer_size); |
112 return audio_buffer_size; | 120 return audio_buffer_size; |
113 } | 121 } |
114 int32_t CopyFromRecBuffer(void* audio_buffer, uint32_t audio_buffer_size) { | 122 int32_t CopyFromRecBuffer(void* audio_buffer, uint32_t audio_buffer_size) { |
115 EXPECT_EQ(audio_buffer_size, rec_buffer_bytes_); | 123 EXPECT_EQ(audio_buffer_size, rec_buffer_bytes_); |
116 const uint32_t min_buffer_size = min(audio_buffer_size, rec_buffer_bytes_); | 124 const uint32_t min_buffer_size = min(audio_buffer_size, rec_buffer_bytes_); |
117 memcpy(audio_buffer, rec_buffer_, min_buffer_size); | 125 memcpy(audio_buffer, rec_buffer_, min_buffer_size); |
118 return min_buffer_size; | 126 return min_buffer_size; |
119 } | 127 } |
120 | 128 |
| 129 mutable rtc::CriticalSection crit_; |
| 130 |
121 int push_iterations_; | 131 int push_iterations_; |
122 int pull_iterations_; | 132 int pull_iterations_; |
123 | 133 |
124 char rec_buffer_[FakeAudioCaptureModule::kNumberSamples * | 134 char rec_buffer_[FakeAudioCaptureModule::kNumberSamples * |
125 FakeAudioCaptureModule::kNumberBytesPerSample]; | 135 FakeAudioCaptureModule::kNumberBytesPerSample]; |
126 uint32_t rec_buffer_bytes_; | 136 uint32_t rec_buffer_bytes_; |
127 }; | 137 }; |
128 | 138 |
129 TEST_F(FakeAdmTest, TestProccess) { | 139 TEST_F(FakeAdmTest, TestProccess) { |
130 // Next process call must be some time in the future (or now). | 140 // Next process call must be some time in the future (or now). |
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197 | 207 |
198 EXPECT_EQ(0, fake_audio_capture_module_->InitRecording()); | 208 EXPECT_EQ(0, fake_audio_capture_module_->InitRecording()); |
199 EXPECT_EQ(0, fake_audio_capture_module_->StartRecording()); | 209 EXPECT_EQ(0, fake_audio_capture_module_->StartRecording()); |
200 | 210 |
201 EXPECT_TRUE_WAIT(push_iterations() > 0, kMsInSecond); | 211 EXPECT_TRUE_WAIT(push_iterations() > 0, kMsInSecond); |
202 EXPECT_TRUE_WAIT(pull_iterations() > 0, kMsInSecond); | 212 EXPECT_TRUE_WAIT(pull_iterations() > 0, kMsInSecond); |
203 | 213 |
204 EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout()); | 214 EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout()); |
205 EXPECT_EQ(0, fake_audio_capture_module_->StopRecording()); | 215 EXPECT_EQ(0, fake_audio_capture_module_->StopRecording()); |
206 } | 216 } |
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