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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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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310 EXPECT_GT(audio_frame.num_channels_, 0u); | 310 EXPECT_GT(audio_frame.num_channels_, 0u); |
311 EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100), | 311 EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100), |
312 audio_frame.samples_per_channel_); | 312 audio_frame.samples_per_channel_); |
313 EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_); | 313 EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_); |
314 } | 314 } |
315 | 315 |
316 // The below test is temporarily disabled on Windows due to problems | 316 // The below test is temporarily disabled on Windows due to problems |
317 // with clang debug builds. | 317 // with clang debug builds. |
318 // TODO(tommi): Re-enable when we've figured out what the problem is. | 318 // TODO(tommi): Re-enable when we've figured out what the problem is. |
319 // http://crbug.com/615050 | 319 // http://crbug.com/615050 |
320 #if !defined(WEBRTC_WIN) && defined(__clang__) && !defined(NDEBUG) | 320 #if !defined(WEBRTC_WIN) && defined(__clang__) && RTC_DCHECK_IS_ON && \ |
321 #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) | 321 GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) |
322 TEST_F(AudioCodingModuleTestOldApi, FailOnZeroDesiredFrequency) { | 322 TEST_F(AudioCodingModuleTestOldApi, FailOnZeroDesiredFrequency) { |
323 AudioFrame audio_frame; | 323 AudioFrame audio_frame; |
324 bool muted; | 324 bool muted; |
325 EXPECT_DEATH(acm_->PlayoutData10Ms(0, &audio_frame, &muted), | 325 EXPECT_DEATH(acm_->PlayoutData10Ms(0, &audio_frame, &muted), |
326 "dst_sample_rate_hz"); | 326 "dst_sample_rate_hz"); |
327 } | 327 } |
328 #endif | 328 #endif |
329 #endif | |
330 | 329 |
331 // Checks that the transport callback is invoked once for each speech packet. | 330 // Checks that the transport callback is invoked once for each speech packet. |
332 // Also checks that the frame type is kAudioFrameSpeech. | 331 // Also checks that the frame type is kAudioFrameSpeech. |
333 TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) { | 332 TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) { |
334 const int k10MsBlocksPerPacket = 3; | 333 const int k10MsBlocksPerPacket = 3; |
335 codec_.pacsize = k10MsBlocksPerPacket * kSampleRateHz / 100; | 334 codec_.pacsize = k10MsBlocksPerPacket * kSampleRateHz / 100; |
336 RegisterCodec(); | 335 RegisterCodec(); |
337 const int kLoops = 10; | 336 const int kLoops = 10; |
338 for (int i = 0; i < kLoops; ++i) { | 337 for (int i = 0; i < kLoops; ++i) { |
339 EXPECT_EQ(i / k10MsBlocksPerPacket, packet_cb_.num_calls()); | 338 EXPECT_EQ(i / k10MsBlocksPerPacket, packet_cb_.num_calls()); |
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1805 Run(16000, 8000, 1000); | 1804 Run(16000, 8000, 1000); |
1806 } | 1805 } |
1807 | 1806 |
1808 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) { | 1807 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) { |
1809 Run(8000, 16000, 1000); | 1808 Run(8000, 16000, 1000); |
1810 } | 1809 } |
1811 | 1810 |
1812 #endif | 1811 #endif |
1813 | 1812 |
1814 } // namespace webrtc | 1813 } // namespace webrtc |
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