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Side by Side Diff: webrtc/api/test/fakeaudiocapturemodule_unittest.cc

Issue 2219653004: Remove old methods in AudioTransport, make it pass a gn build (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix compile. Created 4 years, 4 months ago
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1 /* 1 /*
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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55 FakeAudioCaptureModule::kNumberBytesPerSample)) { 55 FakeAudioCaptureModule::kNumberBytesPerSample)) {
56 ADD_FAILURE(); 56 ADD_FAILURE();
57 return -1; 57 return -1;
58 } 58 }
59 memcpy(rec_buffer_, audioSamples, rec_buffer_bytes_); 59 memcpy(rec_buffer_, audioSamples, rec_buffer_bytes_);
60 ++push_iterations_; 60 ++push_iterations_;
61 newMicLevel = currentMicLevel; 61 newMicLevel = currentMicLevel;
62 return 0; 62 return 0;
63 } 63 }
64 64
65 void PushCaptureData(int voe_channel,
66 const void* audio_data,
67 int bits_per_sample,
68 int sample_rate,
69 size_t number_of_channels,
70 size_t number_of_frames) override {}
71
72 void PullRenderData(int bits_per_sample,
73 int sample_rate,
74 size_t number_of_channels,
75 size_t number_of_frames,
76 void* audio_data,
77 int64_t* elapsed_time_ms,
78 int64_t* ntp_time_ms) override {}
79
65 // ADM is pulling data. 80 // ADM is pulling data.
66 int32_t NeedMorePlayData(const size_t nSamples, 81 int32_t NeedMorePlayData(const size_t nSamples,
67 const size_t nBytesPerSample, 82 const size_t nBytesPerSample,
68 const size_t nChannels, 83 const size_t nChannels,
69 const uint32_t samplesPerSec, 84 const uint32_t samplesPerSec,
70 void* audioSamples, 85 void* audioSamples,
71 size_t& nSamplesOut, 86 size_t& nSamplesOut,
72 int64_t* elapsed_time_ms, 87 int64_t* elapsed_time_ms,
73 int64_t* ntp_time_ms) override { 88 int64_t* ntp_time_ms) override {
74 rtc::CritScope cs(&crit_); 89 rtc::CritScope cs(&crit_);
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190 205
191 EXPECT_EQ(0, fake_audio_capture_module_->InitRecording()); 206 EXPECT_EQ(0, fake_audio_capture_module_->InitRecording());
192 EXPECT_EQ(0, fake_audio_capture_module_->StartRecording()); 207 EXPECT_EQ(0, fake_audio_capture_module_->StartRecording());
193 208
194 EXPECT_TRUE_WAIT(push_iterations() > 0, kMsInSecond); 209 EXPECT_TRUE_WAIT(push_iterations() > 0, kMsInSecond);
195 EXPECT_TRUE_WAIT(pull_iterations() > 0, kMsInSecond); 210 EXPECT_TRUE_WAIT(pull_iterations() > 0, kMsInSecond);
196 211
197 EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout()); 212 EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout());
198 EXPECT_EQ(0, fake_audio_capture_module_->StopRecording()); 213 EXPECT_EQ(0, fake_audio_capture_module_->StopRecording());
199 } 214 }
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