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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 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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384 void StreamParametersTest(Format format); | 384 void StreamParametersTest(Format format); |
385 int ProcessStreamChooser(Format format); | 385 int ProcessStreamChooser(Format format); |
386 int AnalyzeReverseStreamChooser(Format format); | 386 int AnalyzeReverseStreamChooser(Format format); |
387 void ProcessDebugDump(const std::string& in_filename, | 387 void ProcessDebugDump(const std::string& in_filename, |
388 const std::string& out_filename, | 388 const std::string& out_filename, |
389 Format format, | 389 Format format, |
390 int max_size_bytes); | 390 int max_size_bytes); |
391 void VerifyDebugDumpTest(Format format); | 391 void VerifyDebugDumpTest(Format format); |
392 | 392 |
393 const std::string output_path_; | 393 const std::string output_path_; |
394 const std::string ref_path_; | |
395 const std::string ref_filename_; | 394 const std::string ref_filename_; |
396 std::unique_ptr<AudioProcessing> apm_; | 395 std::unique_ptr<AudioProcessing> apm_; |
397 AudioFrame* frame_; | 396 AudioFrame* frame_; |
398 AudioFrame* revframe_; | 397 AudioFrame* revframe_; |
399 std::unique_ptr<ChannelBuffer<float> > float_cb_; | 398 std::unique_ptr<ChannelBuffer<float> > float_cb_; |
400 std::unique_ptr<ChannelBuffer<float> > revfloat_cb_; | 399 std::unique_ptr<ChannelBuffer<float> > revfloat_cb_; |
401 int output_sample_rate_hz_; | 400 int output_sample_rate_hz_; |
402 size_t num_output_channels_; | 401 size_t num_output_channels_; |
403 FILE* far_file_; | 402 FILE* far_file_; |
404 FILE* near_file_; | 403 FILE* near_file_; |
405 FILE* out_file_; | 404 FILE* out_file_; |
406 }; | 405 }; |
407 | 406 |
408 ApmTest::ApmTest() | 407 ApmTest::ApmTest() |
409 : output_path_(test::OutputPath()), | 408 : output_path_(test::OutputPath()), |
410 #ifndef WEBRTC_IOS | |
411 ref_path_(test::ProjectRootPath() + "data/audio_processing/"), | |
412 #else | |
413 // On iOS test data is flat in the project root dir | |
414 ref_path_(test::ProjectRootPath()), | |
415 #endif | |
416 #if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE) | 409 #if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE) |
417 ref_filename_(ref_path_ + "output_data_fixed.pb"), | 410 ref_filename_(test::ResourcePath("audio_processing/output_data_fixed", |
411 "pb")), | |
418 #elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE) | 412 #elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE) |
419 #if defined(WEBRTC_MAC) | 413 #if defined(WEBRTC_MAC) |
420 // A different file for Mac is needed because on this platform the AEC | 414 // A different file for Mac is needed because on this platform the AEC |
421 // constant |kFixedDelayMs| value is 20 and not 50 as it is on the rest. | 415 // constant |kFixedDelayMs| value is 20 and not 50 as it is on the rest. |
422 ref_filename_(ref_path_ + "output_data_mac.pb"), | 416 ref_filename_(test::ResourcePath("audio_processing/output_data_mac", |
417 "pb")), | |
423 #else | 418 #else |
424 ref_filename_(ref_path_ + "output_data_float.pb"), | 419 ref_filename_(test::ResourcePath("audio_processing/output_data_float", |
420 "pb")), | |
425 #endif | 421 #endif |
426 #endif | 422 #endif |
427 frame_(NULL), | 423 frame_(NULL), |
428 revframe_(NULL), | 424 revframe_(NULL), |
429 output_sample_rate_hz_(0), | 425 output_sample_rate_hz_(0), |
430 num_output_channels_(0), | 426 num_output_channels_(0), |
431 far_file_(NULL), | 427 far_file_(NULL), |
432 near_file_(NULL), | 428 near_file_(NULL), |
433 out_file_(NULL) { | 429 out_file_(NULL) { |
430 fprintf(stderr, "ref_filename: %s\n", ref_filename_.c_str()); | |
kjellander_webrtc
2016/11/17 16:40:31
Please remove this. I think it's very useful if it
ehmaldonado_webrtc
2016/11/17 16:46:25
Done.
| |
434 Config config; | 431 Config config; |
435 config.Set<ExperimentalAgc>(new ExperimentalAgc(false)); | 432 config.Set<ExperimentalAgc>(new ExperimentalAgc(false)); |
436 apm_.reset(AudioProcessing::Create(config)); | 433 apm_.reset(AudioProcessing::Create(config)); |
437 } | 434 } |
438 | 435 |
439 void ApmTest::SetUp() { | 436 void ApmTest::SetUp() { |
440 ASSERT_TRUE(apm_.get() != NULL); | 437 ASSERT_TRUE(apm_.get() != NULL); |
441 | 438 |
442 frame_ = new AudioFrame(); | 439 frame_ = new AudioFrame(); |
443 revframe_ = new AudioFrame(); | 440 revframe_ = new AudioFrame(); |
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2877 // TODO(peah): Remove the testing for | 2874 // TODO(peah): Remove the testing for |
2878 // apm->capture_nonlocked_.level_controller_enabled once the value in config_ | 2875 // apm->capture_nonlocked_.level_controller_enabled once the value in config_ |
2879 // is instead used to activate the level controller. | 2876 // is instead used to activate the level controller. |
2880 EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled); | 2877 EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled); |
2881 EXPECT_NEAR(kTargetLcPeakLeveldBFS, | 2878 EXPECT_NEAR(kTargetLcPeakLeveldBFS, |
2882 apm->config_.level_controller.initial_peak_level_dbfs, | 2879 apm->config_.level_controller.initial_peak_level_dbfs, |
2883 std::numeric_limits<float>::epsilon()); | 2880 std::numeric_limits<float>::epsilon()); |
2884 } | 2881 } |
2885 | 2882 |
2886 } // namespace webrtc | 2883 } // namespace webrtc |
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