105 lines
2.9 KiB
C++
105 lines
2.9 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 "modules/audio_processing/agc/agc.h"
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#include <cmath>
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#include <cstdlib>
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#include <vector>
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#include "modules/audio_processing/agc/loudness_histogram.h"
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#include "modules/audio_processing/agc/utility.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace {
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const int kDefaultLevelDbfs = -18;
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const int kNumAnalysisFrames = 100;
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const double kActivityThreshold = 0.3;
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} // namespace
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Agc::Agc()
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: target_level_loudness_(Dbfs2Loudness(kDefaultLevelDbfs)),
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target_level_dbfs_(kDefaultLevelDbfs),
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histogram_(LoudnessHistogram::Create(kNumAnalysisFrames)),
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inactive_histogram_(LoudnessHistogram::Create()) {}
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Agc::~Agc() {}
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float Agc::AnalyzePreproc(const int16_t* audio, size_t length) {
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RTC_DCHECK_GT(length, 0);
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size_t num_clipped = 0;
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for (size_t i = 0; i < length; ++i) {
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if (audio[i] == 32767 || audio[i] == -32768)
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++num_clipped;
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}
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return 1.0f * num_clipped / length;
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}
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void Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
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vad_.ProcessChunk(audio, length, sample_rate_hz);
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const std::vector<double>& rms = vad_.chunkwise_rms();
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const std::vector<double>& probabilities =
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vad_.chunkwise_voice_probabilities();
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RTC_DCHECK_EQ(rms.size(), probabilities.size());
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for (size_t i = 0; i < rms.size(); ++i) {
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histogram_->Update(rms[i], probabilities[i]);
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}
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}
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bool Agc::GetRmsErrorDb(int* error) {
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if (!error) {
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RTC_NOTREACHED();
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return false;
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}
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if (histogram_->num_updates() < kNumAnalysisFrames) {
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// We haven't yet received enough frames.
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return false;
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}
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if (histogram_->AudioContent() < kNumAnalysisFrames * kActivityThreshold) {
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// We are likely in an inactive segment.
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return false;
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}
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double loudness = Linear2Loudness(histogram_->CurrentRms());
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*error = std::floor(Loudness2Db(target_level_loudness_ - loudness) + 0.5);
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histogram_->Reset();
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return true;
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}
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void Agc::Reset() {
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histogram_->Reset();
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}
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int Agc::set_target_level_dbfs(int level) {
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// TODO(turajs): just some arbitrary sanity check. We can come up with better
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// limits. The upper limit should be chosen such that the risk of clipping is
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// low. The lower limit should not result in a too quiet signal.
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if (level >= 0 || level <= -100)
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return -1;
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target_level_dbfs_ = level;
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target_level_loudness_ = Dbfs2Loudness(level);
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return 0;
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}
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int Agc::target_level_dbfs() const {
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return target_level_dbfs_;
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}
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float Agc::voice_probability() const {
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return vad_.last_voice_probability();
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}
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} // namespace webrtc
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