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https://github.com/oxen-io/session-android.git
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d83a3d71bc
Merge in RedPhone // FREEBIE
72 lines
2.3 KiB
C++
72 lines
2.3 KiB
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_coding/neteq/audio_classifier.h"
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#include <assert.h>
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#include <string.h>
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namespace webrtc {
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static const int kDefaultSampleRateHz = 48000;
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static const int kDefaultFrameRateHz = 50;
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static const int kDefaultFrameSizeSamples =
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kDefaultSampleRateHz / kDefaultFrameRateHz;
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static const float kDefaultThreshold = 0.5f;
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AudioClassifier::AudioClassifier()
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: analysis_info_(),
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is_music_(false),
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music_probability_(0),
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// This actually assigns the pointer to a static constant struct
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// rather than creates a struct and |celt_mode_| does not need
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// to be deleted.
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celt_mode_(opus_custom_mode_create(kDefaultSampleRateHz,
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kDefaultFrameSizeSamples,
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NULL)),
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analysis_state_() {
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assert(celt_mode_);
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}
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AudioClassifier::~AudioClassifier() {}
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bool AudioClassifier::Analysis(const int16_t* input,
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int input_length,
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int channels) {
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// Must be 20 ms frames at 48 kHz sampling.
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assert((input_length / channels) == kDefaultFrameSizeSamples);
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// Only mono or stereo are allowed.
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assert(channels == 1 || channels == 2);
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// Call Opus' classifier, defined in
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// "third_party/opus/src/src/analysis.h", with lsb_depth = 16.
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// Also uses a down-mixing function downmix_int, defined in
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// "third_party/opus/src/src/opus_private.h", with
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// constants c1 = 0, and c2 = -2.
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run_analysis(&analysis_state_,
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celt_mode_,
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input,
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kDefaultFrameSizeSamples,
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kDefaultFrameSizeSamples,
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0,
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-2,
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channels,
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kDefaultSampleRateHz,
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16,
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downmix_int,
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&analysis_info_);
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music_probability_ = analysis_info_.music_prob;
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is_music_ = music_probability_ > kDefaultThreshold;
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return is_music_;
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}
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} // namespace webrtc
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