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Remove 20ms support in AGC

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Today, 10ms is the standard chunk length used in whole AudioProcessing, so this was only adding unnecessary complexity and maintainance.
Removing it doesn't change the bahavior in any use case of today.

R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7904 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
aluebs@webrtc.org 2014-12-15 21:54:50 +00:00
parent 1f05c45976
commit 96a626262a
3 changed files with 86 additions and 248 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

270
webrtc/modules/audio_processing/agc/analog_agc.c
View File

@ -40,6 +40,7 @@ static const int16_t kOffset2[8] = {18432, 18379, 18290, 18177, 18052, 17920, 17
static const int16_t kMuteGuardTimeMs = 8000;
static const int16_t kInitCheck = 42;
static const int16_t kNumSubframes = 10;
/* Default settings if config is not used */
#define AGC_DEFAULT_TARGET_LEVEL 3
@ -116,72 +117,18 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H,
int32_t nrg, max_nrg, sample, tmp32;
int32_t *ptr;
uint16_t targetGainIdx, gain;
int16_t i, n, L, M, subFrames, tmp16, tmp_speech[16];
int16_t i, n, L, tmp16, tmp_speech[16];
Agc_t *stt;
stt = (Agc_t *)state;
//default/initial values corresponding to 10ms for wb and swb
M = 10;
L = 16;
subFrames = 160;
if (stt->fs == 8000)
{
if (samples == 80)
{
subFrames = 80;
M = 10;
L = 8;
} else if (samples == 160)
{
subFrames = 80;
M = 20;
L = 8;
} else
{
#ifdef WEBRTC_AGC_DEBUG_DUMP
fprintf(stt->fpt,
"AGC->add_mic, frame %d: Invalid number of samples\n\n",
stt->fcount + 1);
#endif
if (stt->fs == 8000) {
L = 8;
if (samples != 80) {
return -1;
}
} else if (stt->fs == 16000)
{
if (samples == 160)
{
subFrames = 160;
M = 10;
L = 16;
} else if (samples == 320)
{
subFrames = 160;
M = 20;
L = 16;
} else
{
#ifdef WEBRTC_AGC_DEBUG_DUMP
fprintf(stt->fpt,
"AGC->add_mic, frame %d: Invalid number of samples\n\n",
stt->fcount + 1);
#endif
return -1;
}
} else if (stt->fs == 32000)
{
/* SWB is processed as 160 sample for L and H bands */
if (samples == 160)
{
subFrames = 160;
M = 10;
L = 16;
} else
{
#ifdef WEBRTC_AGC_DEBUG_DUMP
fprintf(stt->fpt,
"AGC->add_mic, frame %d: Invalid sample rate\n\n",
stt->fcount + 1);
#endif
} else {
L = 16;
if (samples != 160) {
return -1;
}
}
@ -262,7 +209,7 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H,
}
/* compute envelope */
if ((M == 10) && (stt->inQueue > 0))
if (stt->inQueue > 0)
{
ptr = stt->env[1];
} else
@ -270,7 +217,7 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H,
ptr = stt->env[0];
}
for (i = 0; i < M; i++)
for (i = 0; i < kNumSubframes; i++)
{
/* iterate over samples */
max_nrg = 0;
@ -286,7 +233,7 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H,
}
/* compute energy */
if ((M == 10) && (stt->inQueue > 0))
if (stt->inQueue > 0)
{
ptr = stt->Rxx16w32_array[1];
} else
@ -294,7 +241,7 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H,
ptr = stt->Rxx16w32_array[0];
}
for (i = 0; i < M / 2; i++)
for (i = 0; i < kNumSubframes / 2; i++)
{
if (stt->fs == 16000)
{
@ -308,7 +255,7 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H,
}
/* update queue information */
if ((stt->inQueue == 0) && (M == 10))
if (stt->inQueue == 0)
{
stt->inQueue = 1;
} else
@ -317,18 +264,13 @@ int WebRtcAgc_AddMic(void *state, int16_t *in_mic, int16_t *in_mic_H,
}
/* call VAD (use low band only) */
for (i = 0; i < samples; i += subFrames)
{
WebRtcAgc_ProcessVad(&stt->vadMic, &in_mic[i], subFrames);
}
WebRtcAgc_ProcessVad(&stt->vadMic, in_mic, samples);
return 0;
}
int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, int16_t samples)
{
int32_t errHandle = 0;
int16_t i, subFrames;
Agc_t *stt;
stt = (Agc_t *)state;
@ -339,56 +281,22 @@ int WebRtcAgc_AddFarend(void *state, const int16_t *in_far, int16_t samples)
if (stt->fs == 8000)
{
if ((samples != 80) && (samples != 160))
if (samples != 80)
{
#ifdef WEBRTC_AGC_DEBUG_DUMP
fprintf(stt->fpt,
"AGC->add_far_end, frame %d: Invalid number of samples\n\n",
stt->fcount);
#endif
return -1;
}
subFrames = 80;
} else if (stt->fs == 16000)
} else if (stt->fs == 16000 || stt->fs == 32000)
{
if ((samples != 160) && (samples != 320))
if (samples != 160)
{
#ifdef WEBRTC_AGC_DEBUG_DUMP
fprintf(stt->fpt,
"AGC->add_far_end, frame %d: Invalid number of samples\n\n",
stt->fcount);
#endif
return -1;
}
subFrames = 160;
} else if (stt->fs == 32000)
{
if ((samples != 160) && (samples != 320))
{
#ifdef WEBRTC_AGC_DEBUG_DUMP
fprintf(stt->fpt,
"AGC->add_far_end, frame %d: Invalid number of samples\n\n",
stt->fcount);
#endif
return -1;
}
subFrames = 160;
} else
{
#ifdef WEBRTC_AGC_DEBUG_DUMP
fprintf(stt->fpt,
"AGC->add_far_end, frame %d: Invalid sample rate\n\n",
stt->fcount + 1);
#endif
return -1;
}
for (i = 0; i < samples; i += subFrames)
{
errHandle += WebRtcAgc_AddFarendToDigital(&stt->digitalAgc, &in_far[i], subFrames);
}
return errHandle;
return WebRtcAgc_AddFarendToDigital(&stt->digitalAgc, in_far, samples);
}
int WebRtcAgc_VirtualMic(void *agcInst, int16_t *in_near, int16_t *in_near_H,
@ -1256,9 +1164,6 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near,
uint8_t *saturationWarning)
{
Agc_t *stt;
int32_t inMicLevelTmp;
int16_t subFrames, i;
uint8_t satWarningTmp = 0;
stt = (Agc_t *)agcInst;
@ -1272,47 +1177,18 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near,
if (stt->fs == 8000)
{
if ((samples != 80) && (samples != 160))
if (samples != 80)
{
#ifdef WEBRTC_AGC_DEBUG_DUMP
fprintf(stt->fpt,
"AGC->Process, frame %d: Invalid number of samples\n\n",
stt->fcount);
#endif
return -1;
}
subFrames = 80;
} else if (stt->fs == 16000)
} else if (stt->fs == 16000 || stt->fs == 32000)
{
if ((samples != 160) && (samples != 320))
if (samples != 160)
{
#ifdef WEBRTC_AGC_DEBUG_DUMP
fprintf(stt->fpt,
"AGC->Process, frame %d: Invalid number of samples\n\n",
stt->fcount);
#endif
return -1;
}
subFrames = 160;
} else if (stt->fs == 32000)
{
if ((samples != 160) && (samples != 320))
{
#ifdef WEBRTC_AGC_DEBUG_DUMP
fprintf(stt->fpt,
"AGC->Process, frame %d: Invalid number of samples\n\n",
stt->fcount);
#endif
return -1;
}
subFrames = 160;
} else
{
#ifdef WEBRTC_AGC_DEBUG_DUMP
fprintf(stt->fpt,
"AGC->Process, frame %d: Invalid sample rate\n\n",
stt->fcount);
#endif
return -1;
}
@ -1330,84 +1206,62 @@ int WebRtcAgc_Process(void *agcInst, const int16_t *in_near,
*saturationWarning = 0;
//TODO: PUT IN RANGE CHECKING FOR INPUT LEVELS
*outMicLevel = inMicLevel;
inMicLevelTmp = inMicLevel;
// TODO(andrew): clearly we don't need input and output pointers...
// Change the interface to take a shared input/output.
if (in_near != out)
{
// Only needed if they don't already point to the same place.
memcpy(out, in_near, samples * sizeof(int16_t));
}
if (stt->fs == 32000)
{
if (in_near_H != out_H)
{
memcpy(out_H, in_near_H, samples * sizeof(int16_t));
}
}
#ifdef WEBRTC_AGC_DEBUG_DUMP
stt->fcount++;
#endif
for (i = 0; i < samples; i += subFrames)
if (WebRtcAgc_ProcessDigital(&stt->digitalAgc,
in_near,
in_near_H,
out,
out_H,
stt->fs,
stt->lowLevelSignal) == -1)
{
if (WebRtcAgc_ProcessDigital(&stt->digitalAgc, &in_near[i], &in_near_H[i], &out[i], &out_H[i],
stt->fs, stt->lowLevelSignal) == -1)
{
#ifdef WEBRTC_AGC_DEBUG_DUMP
fprintf(stt->fpt,
"AGC->Process, frame %d: Error from DigAGC\n\n",
stt->fcount);
fprintf(stt->fpt,
"AGC->Process, frame %d: Error from DigAGC\n\n",
stt->fcount);
#endif
return -1;
}
if (stt->agcMode < kAgcModeFixedDigital &&
(stt->lowLevelSignal == 0 || stt->agcMode != kAgcModeAdaptiveDigital))
{
if (WebRtcAgc_ProcessAnalog(agcInst,
inMicLevel,
outMicLevel,
stt->vadMic.logRatio,
echo,
saturationWarning) == -1)
{
return -1;
}
if ((stt->agcMode < kAgcModeFixedDigital) && ((stt->lowLevelSignal == 0)
|| (stt->agcMode != kAgcModeAdaptiveDigital)))
{
if (WebRtcAgc_ProcessAnalog(agcInst, inMicLevelTmp, outMicLevel,
stt->vadMic.logRatio, echo, saturationWarning) == -1)
{
return -1;
}
}
}
#ifdef WEBRTC_AGC_DEBUG_DUMP
fprintf(stt->agcLog,
"%5d\t%d\t%d\t%d\t%d\n",
stt->fcount,
inMicLevelTmp,
*outMicLevel,
stt->maxLevel,
stt->micVol);
fprintf(stt->agcLog,
"%5d\t%d\t%d\t%d\t%d\n",
stt->fcount,
inMicLevel,
*outMicLevel,
stt->maxLevel,
stt->micVol);
#endif
/* update queue */
if (stt->inQueue > 1)
{
memcpy(stt->env[0], stt->env[1], 10 * sizeof(int32_t));
memcpy(stt->Rxx16w32_array[0], stt->Rxx16w32_array[1], 5 * sizeof(int32_t));
}
if (stt->inQueue > 0)
{
stt->inQueue--;
}
/* If 20ms frames are used the input mic level must be updated so that
* the analog AGC does not think that there has been a manual volume
* change. */
inMicLevelTmp = *outMicLevel;
/* Store a positive saturation warning. */
if (*saturationWarning == 1)
{
satWarningTmp = 1;
}
/* update queue */
if (stt->inQueue > 1)
{
memcpy(stt->env[0], stt->env[1], 10 * sizeof(int32_t));
memcpy(stt->Rxx16w32_array[0],
stt->Rxx16w32_array[1],
5 * sizeof(int32_t));
}
/* Trigger the saturation warning if displayed by any of the frames. */
*saturationWarning = satWarningTmp;
if (stt->inQueue > 0)
{
stt->inQueue--;
}
return 0;
}

7
webrtc/modules/audio_processing/agc/digital_agc.c
View File

@ -319,11 +319,7 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const int16_t *in_near,
{
L = 8;
L2 = 3;
} else if (FS == 16000)
{
L = 16;
L2 = 4;
} else if (FS == 32000)
} else if (FS == 16000 || FS == 32000)
{
L = 16;
L2 = 4;
@ -332,7 +328,6 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const int16_t *in_near,
return -1;
}
// TODO(andrew): again, we don't need input and output pointers...
if (in_near != out)
{
// Only needed if they don't already point to the same place.

57
webrtc/modules/audio_processing/agc/include/gain_control.h
View File

@ -50,14 +50,13 @@ extern "C"
#endif
/*
* This function processes a 10/20ms frame of far-end speech to determine
* if there is active speech. Far-end speech length can be either 10ms or
* 20ms. The length of the input speech vector must be given in samples
* (80/160 when FS=8000, and 160/320 when FS=16000 or FS=32000).
* This function processes a 10 ms frame of far-end speech to determine
* if there is active speech. The length of the input speech vector must be
* given in samples (80 when FS=8000, and 160 when FS=16000 or FS=32000).
*
* Input:
* - agcInst : AGC instance.
* - inFar : Far-end input speech vector (10 or 20ms)
* - inFar : Far-end input speech vector
* - samples : Number of samples in input vector
*
* Return value:
@ -69,22 +68,19 @@ int WebRtcAgc_AddFarend(void* agcInst,
int16_t samples);
/*
* This function processes a 10/20ms frame of microphone speech to determine
* if there is active speech. Microphone speech length can be either 10ms or
* 20ms. The length of the input speech vector must be given in samples
* (80/160 when FS=8000, and 160/320 when FS=16000 or FS=32000). For very low
* input levels, the input signal is increased in level by multiplying and
* overwriting the samples in inMic[].
* This function processes a 10 ms frame of microphone speech to determine
* if there is active speech. The length of the input speech vector must be
* given in samples (80 when FS=8000, and 160 when FS=16000 or FS=32000). For
* very low input levels, the input signal is increased in level by multiplying
* and overwriting the samples in inMic[].
*
* This function should be called before any further processing of the
* near-end microphone signal.
*
* Input:
* - agcInst : AGC instance.
* - inMic : Microphone input speech vector (10 or 20 ms) for
* L band
* - inMic_H : Microphone input speech vector (10 or 20 ms) for
* H band
* - inMic : Microphone input speech vector for L band
* - inMic_H : Microphone input speech vector for H band
* - samples : Number of samples in input vector
*
* Return value:
@ -99,17 +95,14 @@ int WebRtcAgc_AddMic(void* agcInst,
/*
* This function replaces the analog microphone with a virtual one.
* It is a digital gain applied to the input signal and is used in the
* agcAdaptiveDigital mode where no microphone level is adjustable.
* Microphone speech length can be either 10ms or 20ms. The length of the
* input speech vector must be given in samples (80/160 when FS=8000, and
* 160/320 when FS=16000 or FS=32000).
* agcAdaptiveDigital mode where no microphone level is adjustable. The length
* of the input speech vector must be given in samples (80 when FS=8000, and 160
* when FS=16000 or FS=32000).
*
* Input:
* - agcInst : AGC instance.
* - inMic : Microphone input speech vector for (10 or 20 ms)
* L band
* - inMic_H : Microphone input speech vector for (10 or 20 ms)
* H band
* - inMic : Microphone input speech vector for L band
* - inMic_H : Microphone input speech vector for H band
* - samples : Number of samples in input vector
* - micLevelIn : Input level of microphone (static)
*
@ -130,23 +123,19 @@ int WebRtcAgc_VirtualMic(void* agcInst,
int32_t* micLevelOut);
/*
* This function processes a 10/20ms frame and adjusts (normalizes) the gain
* both analog and digitally. The gain adjustments are done only during
* active periods of speech. The input speech length can be either 10ms or
* 20ms and the output is of the same length. The length of the speech
* vectors must be given in samples (80/160 when FS=8000, and 160/320 when
* FS=16000 or FS=32000). The echo parameter can be used to ensure the AGC will
* not adjust upward in the presence of echo.
* 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 or FS=32000). 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.
*
* Input:
* - agcInst : AGC instance
* - inNear : Near-end input speech vector (10 or 20 ms) for
* L band
* - inNear_H : Near-end input speech vector (10 or 20 ms) for
* H band
* - inNear : Near-end input speech vector for L band
* - inNear_H : Near-end input speech vector for H band
* - samples : Number of samples in input/output vector
* - inMicLevel : Current microphone volume level
* - echo : Set to 0 if the signal passed to add_mic is