Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(258)

Side by Side Diff: webrtc/media/engine/fakewebrtcvoiceengine.h

Issue 1583773006: Remove fakewebrtccommon.h. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@wvoe_apm
Patch Set: removed include Created 4 years, 3 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
« no previous file with comments | « webrtc/media/engine/fakewebrtccommon.h ('k') | webrtc/media/media.gyp » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2010 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2010 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 #ifndef WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ 11 #ifndef WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
12 #define WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ 12 #define WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
13 13
14 #include <list> 14 #include <list>
15 #include <map> 15 #include <map>
16 #include <vector> 16 #include <vector>
17 17
18 #include "webrtc/base/basictypes.h" 18 #include "webrtc/base/basictypes.h"
19 #include "webrtc/base/checks.h" 19 #include "webrtc/base/checks.h"
20 #include "webrtc/base/gunit.h"
21 #include "webrtc/base/stringutils.h" 20 #include "webrtc/base/stringutils.h"
22 #include "webrtc/config.h" 21 #include "webrtc/config.h"
23 #include "webrtc/media/base/codec.h" 22 #include "webrtc/media/base/codec.h"
24 #include "webrtc/media/base/rtputils.h" 23 #include "webrtc/media/base/rtputils.h"
25 #include "webrtc/media/engine/fakewebrtccommon.h"
26 #include "webrtc/media/engine/webrtcvoe.h" 24 #include "webrtc/media/engine/webrtcvoe.h"
27 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" 25 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
28 #include "webrtc/modules/audio_processing/include/audio_processing.h" 26 #include "webrtc/modules/audio_processing/include/audio_processing.h"
29 27
30 namespace cricket { 28 namespace cricket {
31 29
32 static const int kOpusBandwidthNb = 4000; 30 static const int kOpusBandwidthNb = 4000;
33 static const int kOpusBandwidthMb = 6000; 31 static const int kOpusBandwidthMb = 6000;
34 static const int kOpusBandwidthWb = 8000; 32 static const int kOpusBandwidthWb = 8000;
35 static const int kOpusBandwidthSwb = 12000; 33 static const int kOpusBandwidthSwb = 12000;
36 static const int kOpusBandwidthFb = 20000; 34 static const int kOpusBandwidthFb = 20000;
37 35
38 #define WEBRTC_CHECK_CHANNEL(channel) \ 36 #define WEBRTC_CHECK_CHANNEL(channel) \
39 if (channels_.find(channel) == channels_.end()) return -1; 37 if (channels_.find(channel) == channels_.end()) return -1;
40 38
39 #define WEBRTC_STUB(method, args) \
40 int method args override { return 0; }
41
42 #define WEBRTC_STUB_CONST(method, args) \
43 int method args const override { return 0; }
44
45 #define WEBRTC_BOOL_STUB(method, args) \
46 bool method args override { return true; }
47
48 #define WEBRTC_BOOL_STUB_CONST(method, args) \
49 bool method args const override { return true; }
50
51 #define WEBRTC_VOID_STUB(method, args) \
52 void method args override {}
53
54 #define WEBRTC_FUNC(method, args) int method args override
55
56 #define WEBRTC_VOID_FUNC(method, args) void method args override
57
41 class FakeAudioProcessing : public webrtc::AudioProcessing { 58 class FakeAudioProcessing : public webrtc::AudioProcessing {
42 public: 59 public:
43 FakeAudioProcessing() : experimental_ns_enabled_(false) {} 60 FakeAudioProcessing() : experimental_ns_enabled_(false) {}
44 61
45 WEBRTC_STUB(Initialize, ()) 62 WEBRTC_STUB(Initialize, ())
46 WEBRTC_STUB(Initialize, ( 63 WEBRTC_STUB(Initialize, (
47 int input_sample_rate_hz, 64 int input_sample_rate_hz,
48 int output_sample_rate_hz, 65 int output_sample_rate_hz,
49 int reverse_sample_rate_hz, 66 int reverse_sample_rate_hz,
50 webrtc::AudioProcessing::ChannelLayout input_layout, 67 webrtc::AudioProcessing::ChannelLayout input_layout,
(...skipping 508 matching lines...) Expand 10 before | Expand all | Expand 10 after
559 webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone; 576 webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone;
560 webrtc::NsModes ns_mode_ = webrtc::kNsDefault; 577 webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
561 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; 578 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
562 webrtc::AgcConfig agc_config_; 579 webrtc::AgcConfig agc_config_;
563 FakeAudioProcessing audio_processing_; 580 FakeAudioProcessing audio_processing_;
564 }; 581 };
565 582
566 } // namespace cricket 583 } // namespace cricket
567 584
568 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ 585 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
OLDNEW
« no previous file with comments | « webrtc/media/engine/fakewebrtccommon.h ('k') | webrtc/media/media.gyp » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698