generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Fix compilation errors under clang 3.5.

Browse Source 
Enables building tip-of-tree clang which introduces new warnings that
cause compilation errors in our code base (-Werror).

BUG=
R=andrew@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5630 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2014-03-03 16:47:03 +00:00
parent 31413dc635
commit 0117d1c48c
8 changed files with 12 additions and 17 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/audio_processing/aec/aec_core.c
View File

@ -741,7 +741,7 @@ int WebRtcAec_GetDelayMetricsCore(AecCore* self, int* median, int* std) {
// Calculate the L1 norm, with median value as central moment.
for (i = 0; i < kHistorySizeBlocks; i++) {
l1_norm += (float)(fabs(i - my_median) * self->delay_histogram[i]);
l1_norm += (float)abs(i - my_median) * self->delay_histogram[i];
}
*std = (int)(l1_norm / (float)num_delay_values + 0.5f) * kMsPerBlock;

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

@ -288,12 +288,7 @@ int32_t WebRtcAgc_InitDigital(DigitalAgc_t *stt, int16_t agcMode)
int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc_t *stt, const int16_t *in_far,
int16_t nrSamples)
{
// Check for valid pointer
if (&stt->vadFarend == NULL)
{
return -1;
}
assert(stt != NULL);
// VAD for far end
WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples);

6
webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
View File

@ -249,10 +249,10 @@ void OveruseDetector::UpdateKalman(int64_t t_delta,
const double residual = t_ts_delta - slope_*h[0] - offset_;
const bool stable_state =
(BWE_MIN(num_of_deltas_, 60) * fabsf(offset_) < threshold_);
(BWE_MIN(num_of_deltas_, 60) * fabs(offset_) < threshold_);
// We try to filter out very late frames. For instance periodic key
// frames doesn't fit the Gaussian model well.
if (fabsf(residual) < 3 * sqrt(var_noise_)) {
if (fabs(residual) < 3 * sqrt(var_noise_)) {
UpdateNoiseEstimate(residual, min_frame_period, stable_state);
} else {
UpdateNoiseEstimate(3 * sqrt(var_noise_), min_frame_period, stable_state);
@ -358,7 +358,7 @@ BandwidthUsage OveruseDetector::Detect(double ts_delta) {
return kBwNormal;
}
const double T = BWE_MIN(num_of_deltas_, 60) * offset_;
if (fabsf(T) > threshold_) {
if (fabs(T) > threshold_) {
if (offset_ > 0) {
if (time_over_using_ == -1) {
// Initialize the timer. Assume that we've been

4
webrtc/modules/video_coding/main/source/jitter_estimator.cc
View File

@ -162,7 +162,7 @@ VCMJitterEstimator::UpdateEstimate(int64_t frameDelayMS, uint32_t frameSizeBytes
// deviation is probably due to an incorrect line slope.
double deviation = DeviationFromExpectedDelay(frameDelayMS, deltaFS);
if (abs(deviation) < _numStdDevDelayOutlier * sqrt(_varNoise) ||
if (fabs(deviation) < _numStdDevDelayOutlier * sqrt(_varNoise) ||
frameSizeBytes > _avgFrameSize + _numStdDevFrameSizeOutlier * sqrt(_varFrameSize))
{
// Update the variance of the deviation from the
@ -257,7 +257,7 @@ VCMJitterEstimator::KalmanEstimateChannel(int64_t frameDelayMS,
{
return;
}
double sigma = (300.0 * exp(-abs(static_cast<double>(deltaFSBytes)) /
double sigma = (300.0 * exp(-fabs(static_cast<double>(deltaFSBytes)) /
(1e0 * _maxFrameSize)) + 1) * sqrt(_varNoise);
if (sigma < 1.0)
{

4
webrtc/modules/video_coding/main/source/receiver.cc
View File

@ -156,12 +156,12 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(
// Assume that render timing errors are due to changes in the video stream.
if (next_render_time_ms < 0) {
timing_error = true;
} else if (abs(next_render_time_ms - now_ms) > max_video_delay_ms_) {
} else if (labs(next_render_time_ms - now_ms) > max_video_delay_ms_) {
WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
VCMId(vcm_id_, receiver_id_),
"This frame is out of our delay bounds, resetting jitter "
"buffer: %d > %d",
static_cast<int>(abs(next_render_time_ms - now_ms)),
static_cast<int>(labs(next_render_time_ms - now_ms)),
max_video_delay_ms_);
timing_error = true;
} else if (static_cast<int>(timing_->TargetVideoDelay()) >

2
webrtc/modules/video_coding/main/source/rtt_filter.cc
View File

@ -114,7 +114,7 @@ bool
VCMRttFilter::JumpDetection(uint32_t rttMs)
{
double diffFromAvg = _avgRtt - rttMs;
if (abs(diffFromAvg) > _jumpStdDevs * sqrt(_varRtt))
if (fabs(diffFromAvg) > _jumpStdDevs * sqrt(_varRtt))
{
int diffSign = (diffFromAvg >= 0) ? 1 : -1;
int jumpCountSign = (_jumpCount >= 0) ? 1 : -1;

2
webrtc/modules/video_coding/main/source/video_sender_unittest.cc
View File

@ -52,7 +52,7 @@ struct Vp8StreamInfo {
MATCHER_P(MatchesVp8StreamInfo, expected, "") {
bool res = true;
for (int tl = 0; tl < kMaxNumberOfTemporalLayers; ++tl) {
if (abs(expected.framerate_fps[tl] - arg.framerate_fps[tl]) > 0.5) {
if (fabs(expected.framerate_fps[tl] - arg.framerate_fps[tl]) > 0.5) {
*result_listener << " framerate_fps[" << tl
<< "] = " << arg.framerate_fps[tl] << " (expected "
<< expected.framerate_fps[tl] << ") ";

2
webrtc/video/call_perf_tests.cc
View File

@ -184,7 +184,7 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
// estimated as being synchronized. We don't want to trigger on those.
if (time_since_creation < kStartupTimeMs)
return;
if (abs(latest_audio_ntp - latest_video_ntp) < kInSyncThresholdMs) {
if (labs(latest_audio_ntp - latest_video_ntp) < kInSyncThresholdMs) {
if (first_time_in_sync_ == -1) {
first_time_in_sync_ = now_ms;
webrtc::test::PrintResult("sync_convergence_time",