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Side by Side Diff: webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc

Issue 1338833002: Fix the maximum native sample rate in AudioProcessing (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Format Created 5 years, 3 months ago
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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
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284 MaxNumChannels(&rampOutList))); 284 MaxNumChannels(&rampOutList)));
285 285
286 mixedAudio->UpdateFrame(-1, _timeStamp, NULL, 0, _outputFrequency, 286 mixedAudio->UpdateFrame(-1, _timeStamp, NULL, 0, _outputFrequency,
287 AudioFrame::kNormalSpeech, 287 AudioFrame::kNormalSpeech,
288 AudioFrame::kVadPassive, num_mixed_channels); 288 AudioFrame::kVadPassive, num_mixed_channels);
289 289
290 _timeStamp += static_cast<uint32_t>(_sampleSize); 290 _timeStamp += static_cast<uint32_t>(_sampleSize);
291 291
292 // We only use the limiter if it supports the output sample rate and 292 // We only use the limiter if it supports the output sample rate and
293 // we're actually mixing multiple streams. 293 // we're actually mixing multiple streams.
294 use_limiter_ = _numMixedParticipants > 1 && 294 use_limiter_ =
295 _outputFrequency <= kAudioProcMaxNativeSampleRateHz; 295 _numMixedParticipants > 1 &&
296 _outputFrequency <= AudioProcessing::kMaxNativeSampleRateHz;
296 297
297 MixFromList(*mixedAudio, &mixList); 298 MixFromList(*mixedAudio, &mixList);
298 MixAnonomouslyFromList(*mixedAudio, &additionalFramesList); 299 MixAnonomouslyFromList(*mixedAudio, &additionalFramesList);
299 MixAnonomouslyFromList(*mixedAudio, &rampOutList); 300 MixAnonomouslyFromList(*mixedAudio, &rampOutList);
300 301
301 if(mixedAudio->samples_per_channel_ == 0) { 302 if(mixedAudio->samples_per_channel_ == 0) {
302 // Nothing was mixed, set the audio samples to silence. 303 // Nothing was mixed, set the audio samples to silence.
303 mixedAudio->samples_per_channel_ = _sampleSize; 304 mixedAudio->samples_per_channel_ = _sampleSize;
304 mixedAudio->Mute(); 305 mixedAudio->Mute();
305 } else { 306 } else {
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923 924
924 if(error != _limiter->kNoError) { 925 if(error != _limiter->kNoError) {
925 WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id, 926 WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
926 "Error from AudioProcessing: %d", error); 927 "Error from AudioProcessing: %d", error);
927 assert(false); 928 assert(false);
928 return false; 929 return false;
929 } 930 }
930 return true; 931 return true;
931 } 932 }
932 } // namespace webrtc 933 } // namespace webrtc
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