Fix compilation errors under clang 3.5.

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

webrtc/modules/audio_processing/aec/aec_core.c

@@ -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;
 

webrtc/modules/audio_processing/agc/digital_agc.c

@@ -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
+    assert(stt != NULL);
-    if (&stt->vadFarend == NULL)
-    {
-        return -1;
-    }
-
     // VAD for far end
     WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples);
 

webrtc/modules/remote_bitrate_estimator/overuse_detector.cc

@@ -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

webrtc/modules/video_coding/main/source/jitter_estimator.cc

@@ -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)
     {

webrtc/modules/video_coding/main/source/receiver.cc

@@ -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()) >

webrtc/modules/video_coding/main/source/rtt_filter.cc

@@ -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;

webrtc/modules/video_coding/main/source/video_sender_unittest.cc

@@ -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] << ") ";

webrtc/video/call_perf_tests.cc

@@ -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",