Chromium Code Reviews

Side by Side Diff: webrtc/modules/audio_mixer/audio_mixer_impl.cc

Issue 2386383003: AudioMixer interface cleanup suggestions (Closed)
Patch Set: misc Created 4 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments.
Jump to:
View unified diff |
« no previous file with comments | « webrtc/modules/audio_mixer/audio_mixer_defines.h ('k') | no next file » | 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) 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_mixer/audio_mixer_impl.h" 11 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
12 12
13 #include <algorithm> 13 #include <algorithm>
14 #include <functional> 14 #include <functional>
15 #include <utility> 15 #include <utility>
16 16
17 #include "webrtc/modules/audio_mixer/audio_frame_manipulator.h" 17 #include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
18 #include "webrtc/modules/utility/include/audio_frame_operations.h" 18 #include "webrtc/modules/utility/include/audio_frame_operations.h"
19 #include "webrtc/system_wrappers/include/trace.h" 19 #include "webrtc/system_wrappers/include/trace.h"
20 20
21 namespace webrtc { 21 namespace webrtc {
22 namespace { 22 namespace {
23 23
24 class SourceFrame { 24 class SourceFrame {
25 public: 25 public:
26 // !!!: Style guide is against abbreviations. Plus these don't make sense. 'p'
27 // for MixerAudioSource?
26 SourceFrame(MixerAudioSource* p, AudioFrame* a, bool m, bool was_mixed_before) 28 SourceFrame(MixerAudioSource* p, AudioFrame* a, bool m, bool was_mixed_before)
27 : audio_source_(p), 29 : audio_source_(p),
28 audio_frame_(a), 30 audio_frame_(a),
29 muted_(m), 31 muted_(m),
30 was_mixed_before_(was_mixed_before) { 32 was_mixed_before_(was_mixed_before) {
33 // !!!: Here would be a good place to check pointers aren't null.
31 if (!muted_) { 34 if (!muted_) {
32 energy_ = NewMixerCalculateEnergy(*a); 35 energy_ = NewMixerCalculateEnergy(*a);
33 } 36 }
34 } 37 }
35 38
36 SourceFrame(MixerAudioSource* p, 39 SourceFrame(MixerAudioSource* p,
37 AudioFrame* a, 40 AudioFrame* a,
38 bool m, 41 bool m,
39 bool was_mixed_before, 42 bool was_mixed_before,
40 uint32_t energy) 43 uint32_t energy)
41 : audio_source_(p), 44 : audio_source_(p),
42 audio_frame_(a), 45 audio_frame_(a),
43 muted_(m), 46 muted_(m),
44 energy_(energy), 47 energy_(energy),
45 was_mixed_before_(was_mixed_before) {} 48 was_mixed_before_(was_mixed_before) {}
46 49
50 // !!!: Capital S here.
47 // a.shouldMixBefore(b) is used to select mixer participants. 51 // a.shouldMixBefore(b) is used to select mixer participants.
48 bool shouldMixBefore(const SourceFrame& other) const { 52 bool shouldMixBefore(const SourceFrame& other) const {
49 if (muted_ != other.muted_) { 53 if (muted_ != other.muted_) {
50 return other.muted_; 54 return other.muted_;
51 } 55 }
52 56
53 const auto our_activity = audio_frame_->vad_activity_; 57 const auto our_activity = audio_frame_->vad_activity_;
54 const auto other_activity = other.audio_frame_->vad_activity_; 58 const auto other_activity = other.audio_frame_->vad_activity_;
55 59
56 if (our_activity != other_activity) { 60 if (our_activity != other_activity) {
57 return our_activity == AudioFrame::kVadActive; 61 return our_activity == AudioFrame::kVadActive;
58 } 62 }
59 63
60 return energy_ > other.energy_; 64 return energy_ > other.energy_;
61 } 65 }
62 66
67 // !!!: You should have accessors for these, so that you can e.g. check that
63 MixerAudioSource* audio_source_; 68 MixerAudioSource* audio_source_;
64 AudioFrame* audio_frame_; 69 AudioFrame* audio_frame_;
65 bool muted_; 70 bool muted_;
66 uint32_t energy_; 71 uint32_t energy_;
67 bool was_mixed_before_; 72 bool was_mixed_before_;
68 }; 73 };
69 74
75 // !!!: The term is usually "Downmix".
70 // Remixes a frame between stereo and mono. 76 // Remixes a frame between stereo and mono.
71 void RemixFrame(AudioFrame* frame, size_t number_of_channels) { 77 void RemixFrame(AudioFrame* frame, size_t number_of_channels) {
72 RTC_DCHECK(number_of_channels == 1 || number_of_channels == 2); 78 RTC_DCHECK(number_of_channels == 1 || number_of_channels == 2);
73 if (frame->num_channels_ == 1 && number_of_channels == 2) { 79 if (frame->num_channels_ == 1 && number_of_channels == 2) {
74 AudioFrameOperations::MonoToStereo(frame); 80 AudioFrameOperations::MonoToStereo(frame);
75 } else if (frame->num_channels_ == 2 && number_of_channels == 1) { 81 } else if (frame->num_channels_ == 2 && number_of_channels == 1) {
76 AudioFrameOperations::StereoToMono(frame); 82 AudioFrameOperations::StereoToMono(frame);
77 } 83 }
78 } 84 }
79 85
(...skipping 10 matching lines...)
90 NewMixerRampOut(source_frame.audio_frame_); 96 NewMixerRampOut(source_frame.audio_frame_);
91 } 97 }
92 } 98 }
93 } 99 }
94 100
95 // Mix the AudioFrames stored in audioFrameList into mixed_audio. 101 // Mix the AudioFrames stored in audioFrameList into mixed_audio.
96 int32_t MixFromList(AudioFrame* mixed_audio, 102 int32_t MixFromList(AudioFrame* mixed_audio,
97 const AudioFrameList& audio_frame_list, 103 const AudioFrameList& audio_frame_list,
98 int32_t id, 104 int32_t id,
99 bool use_limiter) { 105 bool use_limiter) {
106 // !!!: Don't use WEBRTC_TRACE in new code
100 WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id, 107 WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, id,
101 "MixFromList(mixed_audio, audio_frame_list)"); 108 "MixFromList(mixed_audio, audio_frame_list)");
109 // !!!: Consistently use {} for one-liners, or don't.
102 if (audio_frame_list.empty()) 110 if (audio_frame_list.empty())
103 return 0; 111 return 0;
104 112
105 if (audio_frame_list.size() == 1) { 113 if (audio_frame_list.size() == 1) {
106 mixed_audio->timestamp_ = audio_frame_list.front()->timestamp_; 114 mixed_audio->timestamp_ = audio_frame_list.front()->timestamp_;
107 mixed_audio->elapsed_time_ms_ = audio_frame_list.front()->elapsed_time_ms_; 115 mixed_audio->elapsed_time_ms_ = audio_frame_list.front()->elapsed_time_ms_;
108 } else { 116 } else {
109 // TODO(wu): Issue 3390. 117 // TODO(wu): Issue 3390.
110 // Audio frame timestamp is only supported in one channel case. 118 // Audio frame timestamp is only supported in one channel case.
111 mixed_audio->timestamp_ = 0; 119 mixed_audio->timestamp_ = 0;
(...skipping 79 matching lines...)
191 199
192 if (sample_rate != kNbInHz && sample_rate != kWbInHz && 200 if (sample_rate != kNbInHz && sample_rate != kWbInHz &&
193 sample_rate != kSwbInHz && sample_rate != kFbInHz) { 201 sample_rate != kSwbInHz && sample_rate != kFbInHz) {
194 WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_, 202 WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, id_,
195 "Invalid frequency: %d", sample_rate); 203 "Invalid frequency: %d", sample_rate);
196 RTC_NOTREACHED(); 204 RTC_NOTREACHED();
197 return; 205 return;
198 } 206 }
199 207
200 if (OutputFrequency() != sample_rate) { 208 if (OutputFrequency() != sample_rate) {
209 // !!!: Dangerous cast!
201 SetOutputFrequency(static_cast<Frequency>(sample_rate)); 210 SetOutputFrequency(static_cast<Frequency>(sample_rate));
202 } 211 }
203 212
204 AudioFrameList mix_list; 213 AudioFrameList mix_list;
205 AudioFrameList anonymous_mix_list; 214 AudioFrameList anonymous_mix_list;
206 size_t num_mixed_audio_sources; 215 size_t num_mixed_audio_sources;
207 { 216 {
208 rtc::CritScope lock(&crit_); 217 rtc::CritScope lock(&crit_);
209 mix_list = GetNonAnonymousAudio(); 218 mix_list = GetNonAnonymousAudio();
210 anonymous_mix_list = GetAnonymousAudio(); 219 anonymous_mix_list = GetAnonymousAudio();
(...skipping 302 matching lines...)
513 } 522 }
514 523
515 int AudioMixerImpl::GetOutputAudioLevelFullRange() { 524 int AudioMixerImpl::GetOutputAudioLevelFullRange() {
516 RTC_DCHECK_RUN_ON(&thread_checker_); 525 RTC_DCHECK_RUN_ON(&thread_checker_);
517 const int level = audio_level_.LevelFullRange(); 526 const int level = audio_level_.LevelFullRange();
518 WEBRTC_TRACE(kTraceStateInfo, kTraceAudioMixerServer, id_, 527 WEBRTC_TRACE(kTraceStateInfo, kTraceAudioMixerServer, id_,
519 "GetAudioOutputLevelFullRange() => level=%d", level); 528 "GetAudioOutputLevelFullRange() => level=%d", level);
520 return level; 529 return level;
521 } 530 }
522 } // namespace webrtc 531 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/audio_mixer/audio_mixer_defines.h ('k') | no next file » | no next file with comments »

Powered by Google App Engine