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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55
webrtc/modules/audio_processing/vad/vad_audio_proc.h
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@ -8,16 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
#ifndef MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
#define MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_processing/vad/common.h"
#include "webrtc/typedefs.h"
#include <stddef.h>
#include <stdint.h>
#include <memory>
#include "modules/audio_processing/vad/common.h" // AudioFeatures, kSampleR...
namespace webrtc {
class AudioFrame;
class PoleZeroFilter;
class VadAudioProc {
@ -49,25 +51,28 @@ class VadAudioProc {
// For every 30 ms we compute 3 spectral peak there for 3 LPC analysis.
// LPC is computed over 15 ms of windowed audio. For every 10 ms sub-frame
// we need 5 ms of past signal to create the input of LPC analysis.
static const size_t kNumPastSignalSamples =
static_cast<size_t>(kSampleRateHz / 200);
enum : size_t {
kNumPastSignalSamples = static_cast<size_t>(kSampleRateHz / 200)
};
// TODO(turajs): maybe defining this at a higher level (maybe enum) so that
// all the code recognize it as "no-error."
static const int kNoError = 0;
enum : int { kNoError = 0 };
static const size_t kNum10msSubframes = 3;
static const size_t kNumSubframeSamples =
static_cast<size_t>(kSampleRateHz / 100);
static const size_t kNumSamplesToProcess =
kNum10msSubframes *
kNumSubframeSamples; // Samples in 30 ms @ given sampling rate.
static const size_t kBufferLength =
kNumPastSignalSamples + kNumSamplesToProcess;
static const size_t kIpLength = kDftSize >> 1;
static const size_t kWLength = kDftSize >> 1;
static const size_t kLpcOrder = 16;
enum : size_t { kNum10msSubframes = 3 };
enum : size_t {
kNumSubframeSamples = static_cast<size_t>(kSampleRateHz / 100)
};
enum : size_t {
// Samples in 30 ms @ given sampling rate.
kNumSamplesToProcess = kNum10msSubframes * kNumSubframeSamples
};
enum : size_t {
kBufferLength = kNumPastSignalSamples + kNumSamplesToProcess
};
enum : size_t { kIpLength = kDftSize >> 1 };
enum : size_t { kWLength = kDftSize >> 1 };
enum : size_t { kLpcOrder = 16 };
size_t ip_[kIpLength];
float w_fft_[kWLength];
@ -79,11 +84,11 @@ class VadAudioProc {
double log_old_gain_;
double old_lag_;
rtc::scoped_ptr<PitchAnalysisStruct> pitch_analysis_handle_;
rtc::scoped_ptr<PreFiltBankstr> pre_filter_handle_;
rtc::scoped_ptr<PoleZeroFilter> high_pass_filter_;
std::unique_ptr<PitchAnalysisStruct> pitch_analysis_handle_;
std::unique_ptr<PreFiltBankstr> pre_filter_handle_;
std::unique_ptr<PoleZeroFilter> high_pass_filter_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
#endif // MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_