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Duplicated sampling frequency multiplier to aecpc_t struct to avoid a getter.

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TEST=audioproc_unittest, trybots
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1099011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3517 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
bjornv@webrtc.org 2013-02-14 21:17:12 +00:00
parent 4013ac478e
commit 6f6acd9f80
2 changed files with 18 additions and 14 deletions
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29
webrtc/modules/audio_processing/aec/echo_cancellation.c
View File

@ -11,7 +11,7 @@
/*
* Contains the API functions for the AEC.
*/
#include "echo_cancellation.h"
#include "webrtc/modules/audio_processing/aec/include/echo_cancellation.h"
#include <math.h>
#ifdef WEBRTC_AEC_DEBUG_DUMP
@ -20,12 +20,12 @@
#include <stdlib.h>
#include <string.h>
#include "aec_core.h"
#include "aec_resampler.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/audio_processing/aec/echo_cancellation_internal.h"
#include "ring_buffer.h"
#include "typedefs.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_processing/aec/aec_core.h"
#include "webrtc/modules/audio_processing/aec/aec_resampler.h"
#include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
#include "webrtc/typedefs.h"
// Maximum length of resampled signal. Must be an integer multiple of frames
// (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
@ -204,6 +204,9 @@ WebRtc_Word32 WebRtcAec_Init(void *aecInst, WebRtc_Word32 sampFreq, WebRtc_Word3
aecpc->highSkewCtr = 0;
aecpc->sampFactor = (aecpc->scSampFreq * 1.0f) / aecpc->splitSampFreq;
// Sampling frequency multiplier (SWB is processed as 160 frame size).
aecpc->rate_factor = aecpc->splitSampFreq / 8000;
// Default settings.
aecConfig.nlpMode = kAecNlpModerate;
aecConfig.skewMode = kAecFalse;
@ -395,7 +398,7 @@ WebRtc_Word32 WebRtcAec_Process(void *aecInst, const WebRtc_Word16 *nearend,
}
nFrames = nrOfSamples / FRAME_LEN;
nBlocks10ms = nFrames / aecpc->aec->mult;
nBlocks10ms = nFrames / aecpc->rate_factor;
if (aecpc->ECstartup) {
if (nearend != out) {
@ -433,7 +436,7 @@ WebRtc_Word32 WebRtcAec_Process(void *aecInst, const WebRtc_Word16 *nearend,
// PART_LEN samples. Use 75% of the average value of the system
// delay as buffer size to start with.
aecpc->bufSizeStart = WEBRTC_SPL_MIN((3 * aecpc->sum *
aecpc->aec->mult * 8) / (4 * aecpc->counter * PART_LEN),
aecpc->rate_factor * 8) / (4 * aecpc->counter * PART_LEN),
kMaxBufSizeStart);
// Buffer size has now been determined.
aecpc->checkBuffSize = 0;
@ -443,7 +446,7 @@ WebRtc_Word32 WebRtcAec_Process(void *aecInst, const WebRtc_Word16 *nearend,
// For really bad systems, don't disable the echo canceller for
// more than 0.5 sec.
aecpc->bufSizeStart = WEBRTC_SPL_MIN((aecpc->msInSndCardBuf *
aecpc->aec->mult * 3) / 40, kMaxBufSizeStart);
aecpc->rate_factor * 3) / 40, kMaxBufSizeStart);
aecpc->checkBuffSize = 0;
}
}
@ -521,7 +524,7 @@ WebRtc_Word32 WebRtcAec_Process(void *aecInst, const WebRtc_Word16 *nearend,
#ifdef WEBRTC_AEC_DEBUG_DUMP
{
int16_t far_buf_size_ms = (int16_t)(aecpc->aec->system_delay /
(sampMsNb * aecpc->aec->mult));
(sampMsNb * aecpc->rate_factor));
(void)fwrite(&far_buf_size_ms, 2, 1, aecpc->bufFile);
(void)fwrite(&aecpc->knownDelay, sizeof(aecpc->knownDelay), 1,
aecpc->delayFile);
@ -808,7 +811,7 @@ WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst)
}
static int EstBufDelay(aecpc_t* aecpc) {
int nSampSndCard = aecpc->msInSndCardBuf * sampMsNb * aecpc->aec->mult;
int nSampSndCard = aecpc->msInSndCardBuf * sampMsNb * aecpc->rate_factor;
int current_delay = nSampSndCard - aecpc->aec->system_delay;
int delay_difference = 0;
@ -819,7 +822,7 @@ static int EstBufDelay(aecpc_t* aecpc) {
// be negative.
// 1) Compensating for the frame(s) that will be read/processed.
current_delay += FRAME_LEN * aecpc->aec->mult;
current_delay += FRAME_LEN * aecpc->rate_factor;
// 2) Account for resampling frame delay.
if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {

3
webrtc/modules/audio_processing/aec/echo_cancellation_internal.h
View File

@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_
#include "modules/audio_processing/aec/aec_core.h"
#include "webrtc/modules/audio_processing/aec/aec_core.h"
typedef struct {
int delayCtr;
@ -25,6 +25,7 @@ typedef struct {
short skewMode;
int bufSizeStart;
int knownDelay;
int rate_factor;
short initFlag; // indicates if AEC has been initialized