Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.

---

Fixes for re-enabling more MSVC level 4 warnings: webrtc/ edition

This contains fixes for the following sorts of issues:
* Possibly-uninitialized local variable
* Signedness mismatch
* Assignment inside conditional

This also contains a small number of other cleanups to nearby code. In
particular several warning-disables for MSVC are removed because they don't seem
to be necessary (either that warning is not enabled or the code does not trigger
it).

BUG=crbug.com/81439
TEST=none
R=henrika@webrtc.org, pkasting@chromium.org

Review URL: https://webrtc-codereview.appspot.com/18769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6667 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/base/nssstreamadapter.cc

@@ -952,17 +952,11 @@ NSSContext *NSSContext::global_nss_context;
 // Static initialization and shutdown
 NSSContext *NSSContext::Instance() {
   if (!global_nss_context) {
-    NSSContext *new_ctx = new NSSContext();
+    scoped_ptr<NSSContext> new_ctx(new NSSContext());
-
+    new_ctx->slot_ = PK11_GetInternalSlot();
-    if (!(new_ctx->slot_ = PK11_GetInternalSlot())) {
+    if (new_ctx->slot_)
-      delete new_ctx;
+      global_nss_context = new_ctx.release();
-      goto fail;
   }
-
-    global_nss_context = new_ctx;
-  }
-
- fail:
   return global_nss_context;
 }
 

webrtc/modules/audio_coding/codecs/isac/main/source/isac.c

@@ -2243,8 +2243,6 @@ int16_t WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
   } else {
     ISACUBStruct* instUB = &(instISAC->instUB);
     ISACLBStruct* instLB = &(instISAC->instLB);
-    double bottleneckLB;
-    double bottleneckUB;
     int32_t bottleneck = instISAC->bottleneck;
     int16_t codingMode = instISAC->codingMode;
     int16_t frameSizeMs = instLB->ISACencLB_obj.new_framelength /
@@ -2263,6 +2261,8 @@ int16_t WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
       instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX_30;
     } else if ((encoder_operational_rate == kIsacSuperWideband) &&
                (instISAC->encoderSamplingRateKHz == kIsacWideband)) {
+      double bottleneckLB = 0;
+      double bottleneckUB = 0;
       if (codingMode == 1) {
         WebRtcIsac_RateAllocation(bottleneck, &bottleneckLB, &bottleneckUB,
                                   &(instISAC->bandwidthKHz));

webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc

@@ -838,7 +838,7 @@ int16_t ACMGenericCodec::ProcessFrameVADDTX(uint8_t* bitstream,
   // Calculate number of samples in 10 ms blocks, and number ms in one frame.
   int16_t samples_in_10ms = static_cast<int16_t>(freq_hz / 100);
   int32_t frame_len_ms = static_cast<int32_t>(frame_len_smpl_) * 1000 / freq_hz;
-  int16_t status;
+  int16_t status = -1;
 
   // Vector for storing maximum 30 ms of mono audio at 48 kHz.
   int16_t audio[1440];

webrtc/modules/audio_coding/main/acm2/acm_opus.cc

@@ -80,7 +80,7 @@ ACMOpus::ACMOpus(int16_t codec_id)
   if (codec_id_ != ACMCodecDB::kOpus) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
                  "Wrong codec id for Opus.");
-    sample_freq_ = -1;
+    sample_freq_ = 0xFFFF;
     bitrate_ = -1;
   }
   return;

webrtc/modules/audio_coding/main/acm2/acm_speex.cc

@@ -30,7 +30,7 @@ ACMSPEEX::ACMSPEEX(int16_t /* codec_id */)
       vbr_enabled_(false),
       encoding_rate_(-1),
       sampling_frequency_(-1),
-      samples_in_20ms_audio_(-1) {
+      samples_in_20ms_audio_(0xFFFF) {
   return;
 }
 

webrtc/modules/audio_processing/agc/digital_agc.c

@@ -310,7 +310,7 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const int16_t *in_near,
     int32_t gain32, delta;
     int16_t logratio;
     int16_t lower_thr, upper_thr;
-    int16_t zeros, zeros_fast, frac;
+    int16_t zeros = 0, zeros_fast, frac = 0;
     int16_t decay;
     int16_t gate, gain_adj;
     int16_t k, n;

webrtc/modules/interface/module_common_types.h

@@ -20,11 +20,6 @@
 #include "webrtc/common_types.h"
 #include "webrtc/typedefs.h"
 
-#ifdef _WIN32
-// Remove warning "new behavior: elements of array will be default initialized".
-#pragma warning(disable : 4351)
-#endif
-
 namespace webrtc {
 
 struct RTPAudioHeader {
@@ -34,21 +29,10 @@ struct RTPAudioHeader {
   uint8_t channel;                    // number of channels 2 = stereo
 };
 
-enum {
+const int16_t kNoPictureId = -1;
-  kNoPictureId = -1
+const int16_t kNoTl0PicIdx = -1;
-};
+const uint8_t kNoTemporalIdx = 0xFF;
-enum {
+const int kNoKeyIdx = -1;
-  kNoTl0PicIdx = -1
-};
-enum {
-  kNoTemporalIdx = -1
-};
-enum {
-  kNoKeyIdx = -1
-};
-enum {
-  kNoSimulcastIdx = 0
-};
 
 struct RTPVideoHeaderVP8 {
   void InitRTPVideoHeaderVP8() {
@@ -67,7 +51,7 @@ struct RTPVideoHeaderVP8 {
                               // kNoPictureId if PictureID does not exist.
   int16_t tl0PicIdx;          // TL0PIC_IDX, 8 bits;
                               // kNoTl0PicIdx means no value provided.
-  int8_t temporalIdx;         // Temporal layer index, or kNoTemporalIdx.
+  uint8_t temporalIdx;        // Temporal layer index, or kNoTemporalIdx.
   bool layerSync;             // This frame is a layer sync frame.
                               // Disabled if temporalIdx == kNoTemporalIdx.
   int keyIdx;                 // 5 bits; kNoKeyIdx means not used.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -85,7 +85,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
           CriticalSectionWrapper::CreateCriticalSection()),
       default_module_(
           static_cast<ModuleRtpRtcpImpl*>(configuration.default_module)),
-      padding_index_(-1),  // Start padding at the first child module.
+      padding_index_(static_cast<size_t>(-1)),  // Start padding at first child.
       nack_method_(kNackOff),
       nack_last_time_sent_full_(0),
       nack_last_seq_number_sent_(0),

webrtc/modules/rtp_rtcp/source/tmmbr_help.cc

@@ -266,12 +266,9 @@ TMMBRHelp::FindTMMBRBoundingSet(int32_t numCandidates, TMMBRSet& candidateSet)
                 numBoundingSet++;
             }
         }
-        if (numBoundingSet != 1)
+        return (numBoundingSet == 1) ? 1 : -1;
-        {
-            numBoundingSet = -1;
     }
-    } else
+
-    {
     // 1. Sort by increasing packetOH
     for (int i = candidateSet.sizeOfSet() - 1; i >= 0; i--)
     {
@@ -439,7 +436,7 @@ TMMBRHelp::FindTMMBRBoundingSet(int32_t numCandidates, TMMBRSet& candidateSet)
 
         // 9. Go back to step 5 if any tuple remains in candidate list
     } while (numCandidates > 0);
-    }
+
     return numBoundingSet;
 }
 

webrtc/system_wrappers/source/thread_win.cc

@@ -159,7 +159,7 @@ void ThreadWindows::Run() {
   if (set_thread_name_) {
     WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, id_,
                  "Thread with name:%s started ", name_);
-    SetThreadName(-1, name_); // -1, set thread name for the calling thread.
+    SetThreadName(static_cast<DWORD>(-1), name_); // -1 == caller thread.
   } else {
     WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, id_,
                  "Thread without name started");

webrtc/video_engine/vie_codec_impl.cc

@@ -421,7 +421,7 @@ unsigned int ViECodecImpl::GetDiscardedPackets(const int video_channel) const {
   ViEChannel* vie_channel = cs.Channel(video_channel);
   if (!vie_channel) {
     shared_data_->SetLastError(kViECodecInvalidChannelId);
-    return -1;
+    return static_cast<unsigned int>(-1);
   }
   return vie_channel->DiscardedPackets();
 }

webrtc/video_engine/vie_defines.h

@@ -120,14 +120,6 @@ inline int ChannelId(const int moduleId) {
 #if defined(_WIN32)
   #define RENDER_MODULE_TYPE kRenderWindows
 
-  // Warning pragmas.
-  // new behavior: elements of array 'XXX' will be default initialized.
-  #pragma warning(disable: 4351)
-  // 'this' : used in base member initializer list.
-  #pragma warning(disable: 4355)
-  // Frame pointer register 'ebp' modified by inline assembly code.
-  #pragma warning(disable: 4731)
-
   // Include libraries.
   #pragma comment(lib, "winmm.lib")
 

webrtc/voice_engine/channel.cc

@@ -3765,7 +3765,7 @@ Channel::PrepareEncodeAndSend(int mixingFrequency)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
                      "Channel::PrepareEncodeAndSend() invalid audio frame");
-        return -1;
+        return 0xFFFFFFFF;
     }
 
     if (channel_state_.Get().input_file_playing)
@@ -3819,7 +3819,7 @@ Channel::EncodeAndSend()
     {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
                      "Channel::EncodeAndSend() invalid audio frame");
-        return -1;
+        return 0xFFFFFFFF;
     }
 
     _audioFrame.id_ = _channelId;
@@ -3832,7 +3832,7 @@ Channel::EncodeAndSend()
     {
         WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
                      "Channel::EncodeAndSend() ACM encoding failed");
-        return -1;
+        return 0xFFFFFFFF;
     }
 
     _timeStamp += _audioFrame.samples_per_channel_;
@@ -4182,7 +4182,7 @@ Channel::MixOrReplaceAudioWithFile(int mixingFrequency)
         // Currently file stream is always mono.
         // TODO(xians): Change the code when FilePlayer supports real stereo.
         _audioFrame.UpdateFrame(_channelId,
-                                -1,
+                                0xFFFFFFFF,
                                 fileBuffer.get(),
                                 fileSamples,
                                 mixingFrequency,

webrtc/voice_engine/transmit_mixer.cc

@@ -1236,7 +1236,7 @@ int32_t TransmitMixer::MixOrReplaceAudioWithFile(
         // Currently file stream is always mono.
         // TODO(xians): Change the code when FilePlayer supports real stereo.
         _audioFrame.UpdateFrame(-1,
-                                -1,
+                                0xFFFFFFFF,
                                 fileBuffer.get(),
                                 fileSamples,
                                 mixingFrequency,