Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(17)

Side by Side Diff: webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc

Issue 2069723003: Move AudioCodingModuleImpl to anonymous namespace in audio_coding_module.cc (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 6 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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/acm2/acm_receiver.h" 11 #include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
12 12
13 #include <algorithm> // std::min 13 #include <algorithm> // std::min
14 #include <memory> 14 #include <memory>
15 15
16 #include "testing/gtest/include/gtest/gtest.h" 16 #include "testing/gtest/include/gtest/gtest.h"
17 #include "webrtc/base/checks.h" 17 #include "webrtc/base/checks.h"
18 #include "webrtc/base/safe_conversions.h" 18 #include "webrtc/base/safe_conversions.h"
19 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" 19 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
20 #include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
21 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h" 20 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
22 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" 21 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
23 #include "webrtc/system_wrappers/include/clock.h" 22 #include "webrtc/system_wrappers/include/clock.h"
24 #include "webrtc/test/test_suite.h" 23 #include "webrtc/test/test_suite.h"
25 #include "webrtc/test/testsupport/fileutils.h" 24 #include "webrtc/test/testsupport/fileutils.h"
26 25
27 namespace webrtc { 26 namespace webrtc {
28 27
29 namespace acm2 { 28 namespace acm2 {
30 namespace { 29 namespace {
(...skipping 29 matching lines...) Expand all
60 : timestamp_(0), 59 : timestamp_(0),
61 packet_sent_(false), 60 packet_sent_(false),
62 last_packet_send_timestamp_(timestamp_), 61 last_packet_send_timestamp_(timestamp_),
63 last_frame_type_(kEmptyFrame) { 62 last_frame_type_(kEmptyFrame) {
64 config_.decoder_factory = CreateBuiltinAudioDecoderFactory(); 63 config_.decoder_factory = CreateBuiltinAudioDecoderFactory();
65 } 64 }
66 65
67 ~AcmReceiverTestOldApi() {} 66 ~AcmReceiverTestOldApi() {}
68 67
69 void SetUp() override { 68 void SetUp() override {
70 acm_.reset(new AudioCodingModuleImpl(config_)); 69 acm_.reset(AudioCodingModule::Create(config_));
71 receiver_.reset(new AcmReceiver(config_)); 70 receiver_.reset(new AcmReceiver(config_));
72 ASSERT_TRUE(receiver_.get() != NULL); 71 ASSERT_TRUE(receiver_.get() != NULL);
73 ASSERT_TRUE(acm_.get() != NULL); 72 ASSERT_TRUE(acm_.get() != NULL);
74 codecs_ = RentACodec::Database(); 73 codecs_ = RentACodec::Database();
75 74
76 acm_->InitializeReceiver(); 75 acm_->InitializeReceiver();
77 acm_->RegisterTransportCallback(this); 76 acm_->RegisterTransportCallback(this);
78 77
79 rtp_header_.header.sequenceNumber = 0; 78 rtp_header_.header.sequenceNumber = 0;
80 rtp_header_.header.timestamp = 0; 79 rtp_header_.header.timestamp = 0;
(...skipping 419 matching lines...) Expand 10 before | Expand all | Expand 10 after
500 receiver_->last_packet_sample_rate_hz()); 499 receiver_->last_packet_sample_rate_hz());
501 EXPECT_EQ(0, receiver_->LastAudioCodec(&codec)); 500 EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
502 EXPECT_TRUE(CodecsEqual(c.inst, codec)); 501 EXPECT_TRUE(CodecsEqual(c.inst, codec));
503 } 502 }
504 } 503 }
505 #endif 504 #endif
506 505
507 } // namespace acm2 506 } // namespace acm2
508 507
509 } // namespace webrtc 508 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/audio_coding/BUILD.gn ('k') | webrtc/modules/audio_coding/acm2/audio_coding_module.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698