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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117
webrtc/modules/audio_processing/agc/legacy/gain_control.h
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@ -8,46 +8,39 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
#ifndef MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
#define MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
#include "webrtc/typedefs.h"
namespace webrtc {
// Errors
#define AGC_UNSPECIFIED_ERROR 18000
#define AGC_UNSUPPORTED_FUNCTION_ERROR 18001
#define AGC_UNINITIALIZED_ERROR 18002
#define AGC_NULL_POINTER_ERROR 18003
#define AGC_BAD_PARAMETER_ERROR 18004
// Warnings
#define AGC_BAD_PARAMETER_WARNING 18050
enum
{
kAgcModeUnchanged,
kAgcModeAdaptiveAnalog,
kAgcModeAdaptiveDigital,
kAgcModeFixedDigital
enum {
kAgcModeUnchanged,
kAgcModeAdaptiveAnalog,
kAgcModeAdaptiveDigital,
kAgcModeFixedDigital
};
enum
{
kAgcFalse = 0,
kAgcTrue
};
enum { kAgcFalse = 0, kAgcTrue };
typedef struct
{
int16_t targetLevelDbfs; // default 3 (-3 dBOv)
int16_t compressionGaindB; // default 9 dB
uint8_t limiterEnable; // default kAgcTrue (on)
typedef struct {
int16_t targetLevelDbfs; // default 3 (-3 dBOv)
int16_t compressionGaindB; // default 9 dB
uint8_t limiterEnable; // default kAgcTrue (on)
} WebRtcAgcConfig;
#if defined(__cplusplus)
extern "C"
{
#endif
/*
* This function analyses the number of samples passed to
* farend and produces any error code that could arise.
*
* Input:
* - agcInst : AGC instance.
* - samples : Number of samples in input vector.
*
* Return value:
* : 0 - Normal operation.
* : -1 - Error.
*/
int WebRtcAgc_GetAddFarendError(void* state, size_t samples);
/*
* This function processes a 10 ms frame of far-end speech to determine
@ -64,9 +57,7 @@ extern "C"
* : 0 - Normal operation.
* : -1 - Error
*/
int WebRtcAgc_AddFarend(void* agcInst,
const int16_t* inFar,
size_t samples);
int WebRtcAgc_AddFarend(void* agcInst, const int16_t* inFar, size_t samples);
/*
* This function processes a 10 ms frame of microphone speech to determine
@ -124,12 +115,12 @@ int WebRtcAgc_VirtualMic(void* agcInst,
int32_t* micLevelOut);
/*
* This function processes a 10 ms frame and adjusts (normalizes) the gain both
* analog and digitally. The gain adjustments are done only during active
* periods of speech. The length of the speech vectors must be given in samples
* (80 when FS=8000, and 160 when FS=16000, FS=32000 or FS=48000). The echo
* parameter can be used to ensure the AGC will not adjust upward in the
* presence of echo.
* This function analyses a 10 ms frame and produces the analog and digital
* gains required to normalize the signal. The gain adjustments are done only
* during active periods of speech. The length of the speech vectors must be
* given in samples (80 when FS=8000, and 160 when FS=16000, FS=32000 or
* FS=48000). The echo parameter can be used to ensure the AGC will not adjust
* upward in the presence of echo.
*
* This function should be called after processing the near-end microphone
* signal, in any case after any echo cancellation.
@ -147,25 +138,47 @@ int WebRtcAgc_VirtualMic(void* agcInst,
*
* Output:
* - outMicLevel : Adjusted microphone volume level
* - out : Gain-adjusted near-end speech vector
* : May be the same vector as the input.
* - saturationWarning : A returned value of 1 indicates a saturation event
* has occurred and the volume cannot be further
* reduced. Otherwise will be set to 0.
* - gains : Vector of gains to apply for digital normalization
*
* Return value:
* : 0 - Normal operation.
* : -1 - Error
*/
int WebRtcAgc_Process(void* agcInst,
int WebRtcAgc_Analyze(void* agcInst,
const int16_t* const* inNear,
size_t num_bands,
size_t samples,
int16_t* const* out,
int32_t inMicLevel,
int32_t* outMicLevel,
int16_t echo,
uint8_t* saturationWarning);
uint8_t* saturationWarning,
int32_t gains[11]);
/*
* This function processes a 10 ms frame by applying precomputed digital gains.
*
* Input:
* - agcInst : AGC instance
* - gains : Vector of gains to apply for digital normalization
* - in_near : Near-end input speech vector for each band
* - num_bands : Number of bands in input/output vector
*
* Output:
* - out : Gain-adjusted near-end speech vector
* : May be the same vector as the input.
*
* Return value:
* : 0 - Normal operation.
* : -1 - Error
*/
int WebRtcAgc_Process(const void* agcInst,
const int32_t gains[11],
const int16_t* const* in_near,
size_t num_bands,
int16_t* const* out);
/*
* This function sets the config parameters (targetLevelDbfs,
@ -203,7 +216,7 @@ int WebRtcAgc_get_config(void* agcInst, WebRtcAgcConfig* config);
* This function creates and returns an AGC instance, which will contain the
* state information for one (duplex) channel.
*/
void* WebRtcAgc_Create();
void* WebRtcAgc_Create(void);
/*
* This function frees the AGC instance created at the beginning.
@ -229,14 +242,12 @@ void WebRtcAgc_Free(void* agcInst);
* Return value : 0 - Ok
* -1 - Error
*/
int WebRtcAgc_Init(void *agcInst,
int WebRtcAgc_Init(void* agcInst,
int32_t minLevel,
int32_t maxLevel,
int16_t agcMode,
uint32_t fs);
#if defined(__cplusplus)
}
#endif
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
#endif // MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_