60730cfe3c
Instead have ProcessStream transparently handle changes to the stream audio parameters (sample rate and channels). This removes two locks per 10 ms ProcessStream call taken by VoiceEngine (four total with the audio level indicator.) Also, prepare future improvements by having the splitting filter take a length parameter. This will allow it to work at different sample rates. Remove the useless splitting_filter wrapper. TESTED=voe_cmd_test with audio processing enabled and switching between codecs; unit tests. R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3949004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
22 lines
867 B
C
22 lines
867 B
C
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/interface/module_common_types.h"
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static const int kChunkSizeMs = 10;
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static const webrtc::AudioProcessing::Error kNoErr =
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webrtc::AudioProcessing::kNoError;
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static void SetFrameSampleRate(webrtc::AudioFrame* frame, int sample_rate_hz) {
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frame->sample_rate_hz_ = sample_rate_hz;
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frame->samples_per_channel_ = kChunkSizeMs * sample_rate_hz / 1000;
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}
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