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Remove unused constants, so chrome can enable a warning for that. Patch from thakis@

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R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2296006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4844 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
niklas.enbom@webrtc.org 2013-09-25 22:05:05 +00:00
parent cecaae2e4c
commit 3e7703640f
9 changed files with 2 additions and 25 deletions
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5
webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
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@ -27,11 +27,6 @@ static const int kSegments = 5;
// A division factor of 1/5 in Q15.
static const int16_t kDivFactor = 6553;
// Filter coefficicients in Q15.
static const int16_t kDampFilter[PITCH_DAMPORDER] = {
-2294, 8192, 20972, 8192, -2294
};
// Interpolation coefficients; generated by design_pitch_filter.m.
// Coefficients are stored in Q14.
static const int16_t kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {

1
webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
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@ -32,7 +32,6 @@ namespace webrtc {
namespace {
const int kRtpHeaderSize = 12;
const int kNeteqInitSampleRateHz = 16000;
const int kNackThresholdPackets = 2;

3
webrtc/modules/audio_processing/aec/aec_core.c
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@ -31,9 +31,6 @@
// Buffer size (samples)
static const size_t kBufSizePartitions = 250; // 1 second of audio in 16 kHz.
// Noise suppression
static const int converged = 250;
// Metrics
static const int subCountLen = 4;
static const int countLen = 50;

2
webrtc/modules/audio_processing/aecm/aecm_core.c
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@ -44,6 +44,7 @@ static const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END = {
};
#endif
#ifdef AECM_WITH_ABS_APPROX
//Q15 alpha = 0.99439986968132 const Factor for magnitude approximation
static const uint16_t kAlpha1 = 32584;
//Q15 beta = 0.12967166976970 const Factor for magnitude approximation
@ -56,6 +57,7 @@ static const uint16_t kBeta2 = 11072;
static const uint16_t kAlpha3 = 26951;
//Q15 beta = 0.57762063060713 const Factor for magnitude approximation
static const uint16_t kBeta3 = 18927;
#endif
// Initialization table for echo channel in 8 kHz
static const int16_t kChannelStored8kHz[PART_LEN1] = {

2
webrtc/modules/remote_bitrate_estimator/rtp_to_ntp.cc
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@ -18,8 +18,6 @@ namespace webrtc {
namespace synchronization {
const double kNtpFracPerMs = 4.294967296E6;
RtcpMeasurement::RtcpMeasurement()
: ntp_secs(0), ntp_frac(0), rtp_timestamp(0) {}

5
webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
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@ -19,11 +19,6 @@
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
namespace
{
const float FRAC = 4.294967296E9;
}
namespace webrtc {
using namespace RTCPUtility;
using namespace RTCPHelp;

7
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -22,13 +22,6 @@
extern MatlabEngine eng; // Global variable defined elsewhere.
#endif
// Local for this file.
namespace {
const float kFracMs = 4.294967296E6f;
} // namespace
#ifdef _WIN32
// Disable warning C4355: 'this' : used in base member initializer list.
#pragma warning(disable : 4355)

1
webrtc/video_engine/vie_capturer.cc
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@ -31,7 +31,6 @@
namespace webrtc {
const int kThreadWaitTimeMs = 100;
const int kMaxDeliverWaitTime = 500;
ViECapturer::ViECapturer(int capture_id,
int engine_id,

1
webrtc/video_engine/vie_remb.cc
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@ -22,7 +22,6 @@
namespace webrtc {
const int kRembTimeOutThresholdMs = 2000;
const int kRembSendIntervallMs = 1000;
const unsigned int kRembMinimumBitrateKbps = 50;