Chromium Code Reviews

Side by Side Diff: webrtc/modules/audio_mixer/include/audio_mixer_defines.h

Issue 2111293003: Removed callback between old AudioConferenceMixer and OutputMixer. The audio frame with mixed audio… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@new_mixer_format
Patch Set: Removed asserts, decreased lock holding time & minor changes. Created 4 years, 5 months ago
Use n/p to move between diff chunks; N/P to move between comments.
Jump to:
View unified diff |
OLDNEW
(Empty)
1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 *
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
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #ifndef WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_AUDIO_MIXER_DEFINES_H_
12 #define WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_AUDIO_MIXER_DEFINES_H_
13
14 #include "webrtc/base/checks.h"
15 #include "webrtc/modules/include/module_common_types.h"
16 #include "webrtc/typedefs.h"
17
18 namespace webrtc {
19 class NewMixHistory;
20
21 // A callback class that all mixer participants must inherit from/implement.
22 class MixerAudioSource {
23 public:
24 // The implementation of this function should update audioFrame with new
25 // audio every time it's called.
26 //
27 // If it returns -1, the frame will not be added to the mix.
28 //
29 // NOTE: This function should not be called. It will remain for a short
30 // time so that subclasses can override it without getting warnings.
31 // TODO(henrik.lundin) Remove this function.
32 virtual int32_t GetAudioFrame(int32_t id, AudioFrame* audioFrame) {
33 RTC_CHECK(false);
34 return -1;
35 }
36
37 // The implementation of GetAudioFrameWithMuted should update audio_frame
38 // with new audio every time it's called. The return value will be
39 // interpreted as follows.
40 enum class AudioFrameInfo {
41 kNormal, // The samples in audio_frame are valid and should be used.
42 kMuted, // The samples in audio_frame should not be used, but should be
43 // implicitly interpreted as zero. Other fields in audio_frame
44 // may be read and should contain meaningful values.
45 kError // audio_frame will not be used.
46 };
47
48 virtual AudioFrameInfo GetAudioFrameWithMuted(int32_t id,
49 AudioFrame* audio_frame) {
50 return GetAudioFrame(id, audio_frame) == -1 ? AudioFrameInfo::kError
51 : AudioFrameInfo::kNormal;
52 }
53
54 // Returns true if the participant was mixed this mix iteration.
55 bool IsMixed() const;
56
57 // This function specifies the sampling frequency needed for the AudioFrame
58 // for future GetAudioFrame(..) calls.
59 virtual int32_t NeededFrequency(int32_t id) const = 0;
60
61 NewMixHistory* _mixHistory;
62
63 protected:
64 MixerAudioSource();
65 virtual ~MixerAudioSource();
66 };
67 } // namespace webrtc
68
69 #endif // WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_AUDIO_MIXER_DEFINES_H_
OLDNEW

Powered by Google App Engine