#include <time_stretch.h>
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enum | ReturnCodes {
kSuccess = 0,
kSuccessLowEnergy = 1,
kNoStretch = 2,
kError = -1,
kSuccess = 0,
kSuccessLowEnergy = 1,
kNoStretch = 2,
kError = -1
} |
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enum | ReturnCodes {
kSuccess = 0,
kSuccessLowEnergy = 1,
kNoStretch = 2,
kError = -1,
kSuccess = 0,
kSuccessLowEnergy = 1,
kNoStretch = 2,
kError = -1
} |
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| TimeStretch (int sample_rate_hz, size_t num_channels, const BackgroundNoise &background_noise) |
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virtual | ~TimeStretch () |
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ReturnCodes | Process (const int16_t *input, size_t input_len, bool fast_mode, AudioMultiVector *output, size_t *length_change_samples) |
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| TimeStretch (int sample_rate_hz, size_t num_channels, const BackgroundNoise &background_noise) |
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virtual | ~TimeStretch () |
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ReturnCodes | Process (const int16_t *input, size_t input_len, bool fast_mode, AudioMultiVector *output, size_t *length_change_samples) |
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virtual void | SetParametersForPassiveSpeech (size_t input_length, int16_t *best_correlation, size_t *peak_index) const =0 |
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virtual ReturnCodes | CheckCriteriaAndStretch (const int16_t *input, size_t input_length, size_t peak_index, int16_t best_correlation, bool active_speech, bool fast_mode, AudioMultiVector *output) const =0 |
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virtual void | SetParametersForPassiveSpeech (size_t input_length, int16_t *best_correlation, size_t *peak_index) const =0 |
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virtual ReturnCodes | CheckCriteriaAndStretch (const int16_t *input, size_t input_length, size_t peak_index, int16_t best_correlation, bool active_speech, bool fast_mode, AudioMultiVector *output) const =0 |
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◆ ReturnCodes [1/2]
Enumerator |
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kSuccess | |
kSuccessLowEnergy | |
kNoStretch | |
kError | |
kSuccess | |
kSuccessLowEnergy | |
kNoStretch | |
kError | |
◆ ReturnCodes [2/2]
Enumerator |
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kSuccess | |
kSuccessLowEnergy | |
kNoStretch | |
kError | |
kSuccess | |
kSuccessLowEnergy | |
kNoStretch | |
kError | |
◆ TimeStretch() [1/2]
webrtc::TimeStretch::TimeStretch |
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int |
sample_rate_hz, |
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size_t |
num_channels, |
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const BackgroundNoise & |
background_noise |
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) |
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inline |
◆ ~TimeStretch() [1/2]
virtual webrtc::TimeStretch::~TimeStretch |
( |
| ) |
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inlinevirtual |
◆ TimeStretch() [2/2]
webrtc::TimeStretch::TimeStretch |
( |
int |
sample_rate_hz, |
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size_t |
num_channels, |
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const BackgroundNoise & |
background_noise |
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) |
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inline |
◆ ~TimeStretch() [2/2]
virtual webrtc::TimeStretch::~TimeStretch |
( |
| ) |
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inlinevirtual |
◆ CheckCriteriaAndStretch() [1/2]
◆ CheckCriteriaAndStretch() [2/2]
◆ Process() [1/2]
◆ Process() [2/2]
◆ SetParametersForPassiveSpeech() [1/2]
virtual void webrtc::TimeStretch::SetParametersForPassiveSpeech |
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size_t |
input_length, |
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int16_t * |
best_correlation, |
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size_t * |
peak_index |
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) |
| const |
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protectedpure virtual |
◆ SetParametersForPassiveSpeech() [2/2]
virtual void webrtc::TimeStretch::SetParametersForPassiveSpeech |
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size_t |
input_length, |
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int16_t * |
best_correlation, |
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size_t * |
peak_index |
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) |
| const |
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protectedpure virtual |
◆ auto_correlation_
int16_t webrtc::TimeStretch::auto_correlation_ |
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protected |
◆ background_noise_
◆ downsampled_input_
int16_t webrtc::TimeStretch::downsampled_input_ |
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protected |
◆ fs_mult_
const int webrtc::TimeStretch::fs_mult_ |
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protected |
◆ kCorrelationLen
static const size_t webrtc::TimeStretch::kCorrelationLen = 50 |
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staticprotected |
◆ kCorrelationThreshold
static const int webrtc::TimeStretch::kCorrelationThreshold = 14746 |
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staticprotected |
◆ kDownsampledLen
◆ kLogCorrelationLen
static const size_t webrtc::TimeStretch::kLogCorrelationLen = 6 |
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staticprotected |
◆ kMaxLag
static const size_t webrtc::TimeStretch::kMaxLag = 60 |
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staticprotected |
◆ kMinLag
static const size_t webrtc::TimeStretch::kMinLag = 10 |
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staticprotected |
◆ master_channel_
const size_t webrtc::TimeStretch::master_channel_ |
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protected |
◆ max_input_value_
int16_t webrtc::TimeStretch::max_input_value_ |
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protected |
◆ num_channels_
const size_t webrtc::TimeStretch::num_channels_ |
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protected |
◆ sample_rate_hz_
const int webrtc::TimeStretch::sample_rate_hz_ |
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protected |
The documentation for this class was generated from the following files:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/audio_coding/neteq/time_stretch.h
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/time_stretch.cc