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Side by Side Diff: webrtc/modules/audio_coding/test/TestAllCodecs.cc

Issue 1513223002: Reduce the runtime of some ACM tests in modules_tests (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years ago
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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
11 #include "webrtc/modules/audio_coding/test/TestAllCodecs.h" 11 #include "webrtc/modules/audio_coding/test/TestAllCodecs.h"
12 12
13 #include <cstdio> 13 #include <cstdio>
14 #include <limits> 14 #include <limits>
15 #include <string> 15 #include <string>
16 16
17 #include "testing/gtest/include/gtest/gtest.h" 17 #include "testing/gtest/include/gtest/gtest.h"
18 18
19 #include "webrtc/base/checks.h"
19 #include "webrtc/common_types.h" 20 #include "webrtc/common_types.h"
20 #include "webrtc/engine_configurations.h" 21 #include "webrtc/engine_configurations.h"
21 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" 22 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
22 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" 23 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
23 #include "webrtc/modules/audio_coding/test/utility.h" 24 #include "webrtc/modules/audio_coding/test/utility.h"
24 #include "webrtc/system_wrappers/include/trace.h" 25 #include "webrtc/system_wrappers/include/trace.h"
25 #include "webrtc/test/testsupport/fileutils.h" 26 #include "webrtc/test/testsupport/fileutils.h"
26 #include "webrtc/typedefs.h" 27 #include "webrtc/typedefs.h"
27 28
28 // Description of the test: 29 // Description of the test:
(...skipping 386 matching lines...)
415 } 416 }
416 417
417 void TestAllCodecs::Run(TestPack* channel) { 418 void TestAllCodecs::Run(TestPack* channel) {
418 AudioFrame audio_frame; 419 AudioFrame audio_frame;
419 420
420 int32_t out_freq_hz = outfile_b_.SamplingFrequency(); 421 int32_t out_freq_hz = outfile_b_.SamplingFrequency();
421 size_t receive_size; 422 size_t receive_size;
422 uint32_t timestamp_diff; 423 uint32_t timestamp_diff;
423 channel->reset_payload_size(); 424 channel->reset_payload_size();
424 int error_count = 0; 425 int error_count = 0;
426 int counter = 0;
427 static const int kTestLengthMs = 500;
428 const int test_length_blocks = rtc::CheckedDivExact(kTestLengthMs, 10);
429 infile_a_.SetNum10MsBlocksToRead(test_length_blocks);
ivoc 2015/12/10 12:23:57 You could consider hardcoding the number of blocks
hlundin-webrtc 2015/12/10 12:49:59 Done.
430 // Fast-forward 1 second (100 blocks) since the file starts with silence.
431 infile_a_.FastForward(100);
425 432
426 int counter = 0;
427 while (!infile_a_.EndOfFile()) { 433 while (!infile_a_.EndOfFile()) {
428 // Add 10 msec to ACM. 434 // Add 10 msec to ACM.
429 infile_a_.Read10MsData(audio_frame); 435 infile_a_.Read10MsData(audio_frame);
430 CHECK_ERROR(acm_a_->Add10MsData(audio_frame)); 436 CHECK_ERROR(acm_a_->Add10MsData(audio_frame));
431 437
432 // Verify that the received packet size matches the settings. 438 // Verify that the received packet size matches the settings.
433 receive_size = channel->payload_size(); 439 receive_size = channel->payload_size();
434 if (receive_size) { 440 if (receive_size) {
435 if ((receive_size != packet_size_bytes_) && 441 if ((receive_size != packet_size_bytes_) &&
436 (packet_size_bytes_ != kVariableSize)) { 442 (packet_size_bytes_ != kVariableSize)) {
(...skipping 39 matching lines...)
476 } 482 }
477 483
478 void TestAllCodecs::DisplaySendReceiveCodec() { 484 void TestAllCodecs::DisplaySendReceiveCodec() {
479 CodecInst my_codec_param; 485 CodecInst my_codec_param;
480 printf("%s -> ", acm_a_->SendCodec()->plname); 486 printf("%s -> ", acm_a_->SendCodec()->plname);
481 acm_b_->ReceiveCodec(&my_codec_param); 487 acm_b_->ReceiveCodec(&my_codec_param);
482 printf("%s\n", my_codec_param.plname); 488 printf("%s\n", my_codec_param.plname);
483 } 489 }
484 490
485 } // namespace webrtc 491 } // namespace webrtc
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