382c0c209d
Break out the computation to a separate class, and call directly into this from channel.cc rather than going through AudioProcessing. This circumvents AudioProcessing's sample rate limitations. We now compute the RMS over all samples rather than downmixing to a single channel. This makes the call point in channel.cc easier, is more "correct" and should have similar (negligible) complexity. This caused slight changes in the RMS output, so the ApmTest.Process reference has been updated. Snippet of the failing output: [ RUN ] ApmTest.Process Running test 4 of 12... Value of: rms_level Actual: 27 Expected: test->rms_level() Which is: 28 Running test 5 of 12... Value of: rms_level Actual: 26 Expected: test->rms_level() Which is: 27 Running test 6 of 12... Value of: rms_level Actual: 26 Expected: test->rms_level() Which is: 27 Running test 10 of 12... Value of: rms_level Actual: 27 Expected: test->rms_level() Which is: 28 Running test 11 of 12... Value of: rms_level Actual: 26 Expected: test->rms_level() Which is: 27 Running test 12 of 12... Value of: rms_level Actual: 26 Expected: test->rms_level() Which is: 27 BUG=3290 TESTED=Chrome assert is avoided and both voe_cmd_test and apprtc produce reasonable printed out results from RMS(). R=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6056 4adac7df-926f-26a2-2b94-8c16560cd09d
90 lines
2.6 KiB
C++
90 lines
2.6 KiB
C++
/*
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* Copyright (c) 2012 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/level_estimator_impl.h"
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#include "webrtc/modules/audio_processing/audio_buffer.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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namespace webrtc {
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LevelEstimatorImpl::LevelEstimatorImpl(const AudioProcessing* apm,
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CriticalSectionWrapper* crit)
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: ProcessingComponent(),
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crit_(crit) {}
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LevelEstimatorImpl::~LevelEstimatorImpl() {}
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int LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
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if (!is_component_enabled()) {
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return AudioProcessing::kNoError;
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}
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RMSLevel* rms_level = static_cast<RMSLevel*>(handle(0));
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if (audio->is_muted()) {
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rms_level->ProcessMuted(audio->samples_per_channel() *
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audio->num_channels());
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} else {
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for (int i = 0; i < audio->num_channels(); ++i) {
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rms_level->Process(audio->data(i), audio->samples_per_channel());
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}
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}
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return AudioProcessing::kNoError;
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}
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int LevelEstimatorImpl::Enable(bool enable) {
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CriticalSectionScoped crit_scoped(crit_);
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return EnableComponent(enable);
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}
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bool LevelEstimatorImpl::is_enabled() const {
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return is_component_enabled();
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}
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int LevelEstimatorImpl::RMS() {
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if (!is_component_enabled()) {
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return AudioProcessing::kNotEnabledError;
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}
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RMSLevel* rms_level = static_cast<RMSLevel*>(handle(0));
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return rms_level->RMS();
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}
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// The ProcessingComponent implementation is pretty weird in this class since
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// we have only a single instance of the trivial underlying component.
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void* LevelEstimatorImpl::CreateHandle() const {
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return new RMSLevel;
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}
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void LevelEstimatorImpl::DestroyHandle(void* handle) const {
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delete static_cast<RMSLevel*>(handle);
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}
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int LevelEstimatorImpl::InitializeHandle(void* handle) const {
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static_cast<RMSLevel*>(handle)->Reset();
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return AudioProcessing::kNoError;
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}
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int LevelEstimatorImpl::ConfigureHandle(void* /*handle*/) const {
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return AudioProcessing::kNoError;
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}
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int LevelEstimatorImpl::num_handles_required() const {
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return 1;
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}
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int LevelEstimatorImpl::GetHandleError(void* /*handle*/) const {
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return AudioProcessing::kUnspecifiedError;
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}
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} // namespace webrtc
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