ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
102 lines
2.8 KiB
C++
102 lines
2.8 KiB
C++
/*
|
|
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#include "webrtc/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/include/module_common_types.h"
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
|
|
const int kDefaultLevelDbfs = -18;
|
|
const int kNumAnalysisFrames = 100;
|
|
const double kActivityThreshold = 0.3;
|
|
|
|
} // namespace
|
|
|
|
Agc::Agc()
|
|
: target_level_loudness_(Dbfs2Loudness(kDefaultLevelDbfs)),
|
|
target_level_dbfs_(kDefaultLevelDbfs),
|
|
histogram_(Histogram::Create(kNumAnalysisFrames)),
|
|
inactive_histogram_(Histogram::Create()) {
|
|
}
|
|
|
|
Agc::~Agc() {}
|
|
|
|
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) {
|
|
vad_.ProcessChunk(audio, length, sample_rate_hz);
|
|
const std::vector<double>& rms = vad_.chunkwise_rms();
|
|
const std::vector<double>& probabilities =
|
|
vad_.chunkwise_voice_probabilities();
|
|
RTC_DCHECK_EQ(rms.size(), probabilities.size());
|
|
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);
|
|
return false;
|
|
}
|
|
|
|
if (histogram_->num_updates() < kNumAnalysisFrames) {
|
|
// We haven't yet received enough frames.
|
|
return false;
|
|
}
|
|
|
|
if (histogram_->AudioContent() < kNumAnalysisFrames * kActivityThreshold) {
|
|
// We are likely in an inactive segment.
|
|
return false;
|
|
}
|
|
|
|
double loudness = Linear2Loudness(histogram_->CurrentRms());
|
|
*error = std::floor(Loudness2Db(target_level_loudness_ - loudness) + 0.5);
|
|
histogram_->Reset();
|
|
return true;
|
|
}
|
|
|
|
void Agc::Reset() {
|
|
histogram_->Reset();
|
|
}
|
|
|
|
int Agc::set_target_level_dbfs(int level) {
|
|
// TODO(turajs): just some arbitrary sanity check. We can come up with better
|
|
// limits. The upper limit should be chosen such that the risk of clipping is
|
|
// low. The lower limit should not result in a too quiet signal.
|
|
if (level >= 0 || level <= -100)
|
|
return -1;
|
|
target_level_dbfs_ = level;
|
|
target_level_loudness_ = Dbfs2Loudness(level);
|
|
return 0;
|
|
}
|
|
|
|
} // namespace webrtc
|