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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/audio/utility/audio_frame_operations.h" |
11 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d
efines.h" | 12 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d
efines.h" |
12 #include "webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_im
pl.h" | 13 #include "webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_im
pl.h" |
13 #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h
" | 14 #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h
" |
14 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 15 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
15 #include "webrtc/modules/utility/include/audio_frame_operations.h" | |
16 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" | 16 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
17 #include "webrtc/system_wrappers/include/trace.h" | 17 #include "webrtc/system_wrappers/include/trace.h" |
18 | 18 |
19 namespace webrtc { | 19 namespace webrtc { |
20 namespace { | 20 namespace { |
21 | 21 |
22 struct ParticipantFrameStruct { | 22 struct ParticipantFrameStruct { |
23 ParticipantFrameStruct(MixerParticipant* p, AudioFrame* a, bool m) | 23 ParticipantFrameStruct(MixerParticipant* p, AudioFrame* a, bool m) |
24 : participant(p), audioFrame(a), muted(m) {} | 24 : participant(p), audioFrame(a), muted(m) {} |
25 MixerParticipant* participant; | 25 MixerParticipant* participant; |
26 AudioFrame* audioFrame; | 26 AudioFrame* audioFrame; |
27 bool muted; | 27 bool muted; |
28 }; | 28 }; |
29 | 29 |
30 typedef std::list<ParticipantFrameStruct*> ParticipantFrameStructList; | 30 typedef std::list<ParticipantFrameStruct*> ParticipantFrameStructList; |
31 | 31 |
32 // Mix |frame| into |mixed_frame|, with saturation protection and upmixing. | 32 // Mix |frame| into |mixed_frame|, with saturation protection and upmixing. |
33 // These effects are applied to |frame| itself prior to mixing. Assumes that | 33 // These effects are applied to |frame| itself prior to mixing. Assumes that |
34 // |mixed_frame| always has at least as many channels as |frame|. Supports | 34 // |mixed_frame| always has at least as many channels as |frame|. Supports |
35 // stereo at most. | 35 // stereo at most. |
36 // | 36 // |
37 // TODO(andrew): consider not modifying |frame| here. | 37 // TODO(andrew): consider not modifying |frame| here. |
38 void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) { | 38 void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) { |
39 assert(mixed_frame->num_channels_ >= frame->num_channels_); | 39 assert(mixed_frame->num_channels_ >= frame->num_channels_); |
40 if (use_limiter) { | 40 if (use_limiter) { |
41 // Divide by two to avoid saturation in the mixing. | 41 // This is to avoid saturation in the mixing. It is only |
42 // This is only meaningful if the limiter will be used. | 42 // meaningful if the limiter will be used. |
43 *frame >>= 1; | 43 AudioFrameOperations::ApplyHalfGain(frame); |
44 } | 44 } |
45 if (mixed_frame->num_channels_ > frame->num_channels_) { | 45 if (mixed_frame->num_channels_ > frame->num_channels_) { |
46 // We only support mono-to-stereo. | 46 // We only support mono-to-stereo. |
47 assert(mixed_frame->num_channels_ == 2 && | 47 assert(mixed_frame->num_channels_ == 2 && |
48 frame->num_channels_ == 1); | 48 frame->num_channels_ == 1); |
49 AudioFrameOperations::MonoToStereo(frame); | 49 AudioFrameOperations::MonoToStereo(frame); |
50 } | 50 } |
51 | 51 |
52 *mixed_frame += *frame; | 52 AudioFrameOperations::Add(*frame, mixed_frame); |
53 } | 53 } |
54 | 54 |
55 // Return the max number of channels from a |list| composed of AudioFrames. | 55 // Return the max number of channels from a |list| composed of AudioFrames. |
56 size_t MaxNumChannels(const AudioFrameList* list) { | 56 size_t MaxNumChannels(const AudioFrameList* list) { |
57 size_t max_num_channels = 1; | 57 size_t max_num_channels = 1; |
58 for (AudioFrameList::const_iterator iter = list->begin(); | 58 for (AudioFrameList::const_iterator iter = list->begin(); |
59 iter != list->end(); | 59 iter != list->end(); |
60 ++iter) { | 60 ++iter) { |
61 max_num_channels = std::max(max_num_channels, (*iter).frame->num_channels_); | 61 max_num_channels = std::max(max_num_channels, (*iter).frame->num_channels_); |
62 } | 62 } |
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296 _numMixedParticipants > 1 && | 296 _numMixedParticipants > 1 && |
297 _outputFrequency <= AudioProcessing::kMaxNativeSampleRateHz; | 297 _outputFrequency <= AudioProcessing::kMaxNativeSampleRateHz; |
298 | 298 |
299 MixFromList(mixedAudio, mixList); | 299 MixFromList(mixedAudio, mixList); |
300 MixAnonomouslyFromList(mixedAudio, additionalFramesList); | 300 MixAnonomouslyFromList(mixedAudio, additionalFramesList); |
301 MixAnonomouslyFromList(mixedAudio, rampOutList); | 301 MixAnonomouslyFromList(mixedAudio, rampOutList); |
302 | 302 |
303 if(mixedAudio->samples_per_channel_ == 0) { | 303 if(mixedAudio->samples_per_channel_ == 0) { |
304 // Nothing was mixed, set the audio samples to silence. | 304 // Nothing was mixed, set the audio samples to silence. |
305 mixedAudio->samples_per_channel_ = _sampleSize; | 305 mixedAudio->samples_per_channel_ = _sampleSize; |
306 mixedAudio->Mute(); | 306 AudioFrameOperations::Mute(mixedAudio); |
307 } else { | 307 } else { |
308 // Only call the limiter if we have something to mix. | 308 // Only call the limiter if we have something to mix. |
309 LimitMixedAudio(mixedAudio); | 309 LimitMixedAudio(mixedAudio); |
310 } | 310 } |
311 } | 311 } |
312 | 312 |
313 { | 313 { |
314 CriticalSectionScoped cs(_cbCrit.get()); | 314 CriticalSectionScoped cs(_cbCrit.get()); |
315 if(_mixReceiver != NULL) { | 315 if(_mixReceiver != NULL) { |
316 const AudioFrame** dummy = NULL; | 316 const AudioFrame** dummy = NULL; |
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915 // And now we can safely restore the level. This procedure results in | 915 // And now we can safely restore the level. This procedure results in |
916 // some loss of resolution, deemed acceptable. | 916 // some loss of resolution, deemed acceptable. |
917 // | 917 // |
918 // It's possible to apply the gain in the AGC (with a target level of 0 dbFS | 918 // It's possible to apply the gain in the AGC (with a target level of 0 dbFS |
919 // and compression gain of 6 dB). However, in the transition frame when this | 919 // and compression gain of 6 dB). However, in the transition frame when this |
920 // is enabled (moving from one to two participants) it has the potential to | 920 // is enabled (moving from one to two participants) it has the potential to |
921 // create discontinuities in the mixed frame. | 921 // create discontinuities in the mixed frame. |
922 // | 922 // |
923 // Instead we double the frame (with addition since left-shifting a | 923 // Instead we double the frame (with addition since left-shifting a |
924 // negative value is undefined). | 924 // negative value is undefined). |
925 *mixedAudio += *mixedAudio; | 925 AudioFrameOperations::Add(*mixedAudio, mixedAudio); |
926 | 926 |
927 if(error != _limiter->kNoError) { | 927 if(error != _limiter->kNoError) { |
928 WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, | 928 WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, |
929 "Error from AudioProcessing: %d", error); | 929 "Error from AudioProcessing: %d", error); |
930 assert(false); | 930 assert(false); |
931 return false; | 931 return false; |
932 } | 932 } |
933 return true; | 933 return true; |
934 } | 934 } |
935 } // namespace webrtc | 935 } // namespace webrtc |
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