AnotherFoxGuy/webrtc-audio-processing
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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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41
webrtc/modules/audio_processing/agc/agc.cc
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@ -8,18 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_processing/agc/agc.h"
#include "modules/audio_processing/agc/agc.h"
#include <cmath>
#include <cstdlib>
#include <algorithm>
#include <vector>
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_processing/agc/histogram.h"
#include "webrtc/modules/audio_processing/agc/utility.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "modules/audio_processing/agc/loudness_histogram.h"
#include "modules/audio_processing/agc/utility.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
@ -33,23 +30,12 @@ const double kActivityThreshold = 0.3;
Agc::Agc()
: target_level_loudness_(Dbfs2Loudness(kDefaultLevelDbfs)),
target_level_dbfs_(kDefaultLevelDbfs),
histogram_(Histogram::Create(kNumAnalysisFrames)),
inactive_histogram_(Histogram::Create()) {
}
histogram_(LoudnessHistogram::Create(kNumAnalysisFrames)),
inactive_histogram_(LoudnessHistogram::Create()) {}
Agc::~Agc() {}
Agc::~Agc() = default;
float Agc::AnalyzePreproc(const int16_t* audio, size_t length) {
assert(length > 0);
size_t num_clipped = 0;
for (size_t i = 0; i < length; ++i) {
if (audio[i] == 32767 || audio[i] == -32768)
++num_clipped;
}
return 1.0f * num_clipped / length;
}
int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
void Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
vad_.ProcessChunk(audio, length, sample_rate_hz);
const std::vector<double>& rms = vad_.chunkwise_rms();
const std::vector<double>& probabilities =
@ -58,12 +44,11 @@ int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
for (size_t i = 0; i < rms.size(); ++i) {
histogram_->Update(rms[i], probabilities[i]);
}
return 0;
}
bool Agc::GetRmsErrorDb(int* error) {
if (!error) {
assert(false);
RTC_NOTREACHED();
return false;
}
@ -98,4 +83,12 @@ int Agc::set_target_level_dbfs(int level) {
return 0;
}
int Agc::target_level_dbfs() const {
return target_level_dbfs_;
}
float Agc::voice_probability() const {
return vad_.last_voice_probability();
}
} // namespace webrtc