Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(89)

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

Issue 2221443002: Changed mixing api and removed resampler (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Described Mix args in comments. Created 4 years, 4 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) 2011 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2011 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
(...skipping 31 matching lines...) Expand 10 before | Expand all | Expand 10 after
42 42
43 // Inform the mixer that the audio source should always be mixed and not 43 // Inform the mixer that the audio source should always be mixed and not
44 // count toward the number of mixed audio sources. Note that an audio source 44 // count toward the number of mixed audio sources. Note that an audio source
45 // must have been added to the mixer (by calling SetMixabilityStatus()) 45 // must have been added to the mixer (by calling SetMixabilityStatus())
46 // before this function can be successfully called. 46 // before this function can be successfully called.
47 virtual int32_t SetAnonymousMixabilityStatus(MixerAudioSource* audio_source, 47 virtual int32_t SetAnonymousMixabilityStatus(MixerAudioSource* audio_source,
48 bool mixable) = 0; 48 bool mixable) = 0;
49 49
50 // Performs mixing by asking registered audio sources for audio. The 50 // Performs mixing by asking registered audio sources for audio. The
51 // mixed result is placed in the provided AudioFrame. Can only be 51 // mixed result is placed in the provided AudioFrame. Can only be
52 // called from a single thread. 52 // called from a single thread. The rate and channels arguments
53 virtual void Mix(AudioFrame* audio_frame_for_mixing) = 0; 53 // specify the rate and number of channels of the mix result.
54 54 virtual void Mix(int sample_rate,
55 // Set the minimum sampling frequency at which to mix. The mixing algorithm 55 size_t number_of_channels,
56 // may still choose to mix at a higher samling frequency to avoid 56 AudioFrame* audio_frame_for_mixing) = 0;
57 // downsampling of audio contributing to the mixed audio.
58 virtual int32_t SetMinimumMixingFrequency(Frequency freq) = 0;
59 57
60 // Returns true if the audio source is mixed anonymously. 58 // Returns true if the audio source is mixed anonymously.
61 virtual bool AnonymousMixabilityStatus( 59 virtual bool AnonymousMixabilityStatus(
62 const MixerAudioSource& audio_source) const = 0; 60 const MixerAudioSource& audio_source) const = 0;
63 61
64 protected: 62 protected:
65 NewAudioConferenceMixer() {} 63 NewAudioConferenceMixer() {}
66 }; 64 };
67 } // namespace webrtc 65 } // namespace webrtc
68 66
69 #endif // WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_NEW_AUDIO_CONFERENCE_MIXER_H_ 67 #endif // WEBRTC_MODULES_AUDIO_MIXER_INCLUDE_NEW_AUDIO_CONFERENCE_MIXER_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698