Update audio code to use size_t more correctly,

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

Index: webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index 9c3183271b75bcbdaca46d20cd915d8ecee05323..57359d8499d3a6dbf3711f37470c1e8f0bf6dd58 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -76,22 +76,24 @@ bool IsCodecCN(int index) {
 }
 
 // Stereo-to-mono can be used as in-place.
-int DownMix(const AudioFrame& frame, int length_out_buff, int16_t* out_buff) {
+int DownMix(const AudioFrame& frame,
+            size_t length_out_buff,
+            int16_t* out_buff) {
   if (length_out_buff < frame.samples_per_channel_) {
     return -1;
   }
-  for (int n = 0; n < frame.samples_per_channel_; ++n)
+  for (size_t n = 0; n < frame.samples_per_channel_; ++n)
     out_buff[n] = (frame.data_[2 * n] + frame.data_[2 * n + 1]) >> 1;
   return 0;
 }
 
 // Mono-to-stereo can be used as in-place.
-int UpMix(const AudioFrame& frame, int length_out_buff, int16_t* out_buff) {
+int UpMix(const AudioFrame& frame, size_t length_out_buff, int16_t* out_buff) {
   if (length_out_buff < frame.samples_per_channel_) {
     return -1;
   }
-  for (int n = frame.samples_per_channel_; n > 0; --n) {
-    int i = n - 1;

[kwiberg-webrtc, 2015/07/12 17:39:26]

Hmm. This code already jumps through all the hoops needed for an unsigned loop
variable, even though it's signed. Curious...

[Peter Kasting, 2015/07/13 02:46:26]

That's probably a sign that I updated the code to jump through those hoops as a
precursor change, to minimize the deltas in this later enormous patch.

+  for (size_t n = frame.samples_per_channel_; n > 0; --n) {

[kwiberg-webrtc, 2015/07/12 17:39:26]

Maybe n != 0 now that it can't be negative.

[Peter Kasting, 2015/07/13 02:46:26]

Is one preferred over the other for some reason?  It seems like both should be
optimized the same way, but "> 0" is more restrictive from a reader's
perspective, which I think is a good thing, since it means you don't have to
mentally think, "oh, size_t is unsigned, so this can't be negative, so it really
means > 0".

[kwiberg-webrtc, 2015/07/15 01:05:00]

The compiler is surely smart enough to realize that they're the same operation
for unsigned numbers. My opinion is that != 0 is more readable for unsigned
numbers, since that's what the operation actually is; when you wrote > 0, I had
to double check that n was actually unsigned, since that operation would (in my
head) have made more sense for a signed number.

It seems our intuitions about what's most natural here differ. But the
difference isn't large, and the compiler can't possibly care, so choose based on
whether getting the last word or sucking up to reviewers is more important to
you. :-)

[Peter Kasting, 2015/07/15 06:30:09]

When I'm working with webrtc code (where I'm basically an interloper) and trying
to earn reviewer approval for montrous patchsets that change hundreds of files,
I'll choose sucking up to reviewers :)

[Peter Kasting, 2015/07/17 00:12:08]

Done.

+    size_t i = n - 1;
     int16_t sample = frame.data_[i];
     out_buff[2 * i + 1] = sample;
     out_buff[2 * i] = sample;
@@ -338,11 +340,10 @@ int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
 
 int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
                                                InputData* input_data) {
-  if (audio_frame.samples_per_channel_ <= 0) {
+  if (audio_frame.samples_per_channel_ == 0) {
     assert(false);
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                 "Cannot Add 10 ms audio, payload length is negative or "
-                 "zero");
+                 "Cannot Add 10 ms audio, payload length is zero");
     return -1;
   }
 
@@ -354,7 +355,7 @@ int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
   }
 
   // If the length and frequency matches. We currently just support raw PCM.
-  if ((audio_frame.sample_rate_hz_ / 100) !=
+  if (static_cast<size_t>(audio_frame.sample_rate_hz_ / 100) !=
       audio_frame.samples_per_channel_) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "Cannot Add 10 ms audio, input frequency and length doesn't"
@@ -477,17 +478,19 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
     // The result of the resampler is written to output frame.
     dest_ptr_audio = preprocess_frame_.data_;
 
-    preprocess_frame_.samples_per_channel_ = resampler_.Resample10Msec(
+    int samples_per_channel = resampler_.Resample10Msec(
         src_ptr_audio, in_frame.sample_rate_hz_,
         codec_manager_.CurrentEncoder()->SampleRateHz(),
         preprocess_frame_.num_channels_, AudioFrame::kMaxDataSizeSamples,
         dest_ptr_audio);
 
-    if (preprocess_frame_.samples_per_channel_ < 0) {
+    if (samples_per_channel < 0) {
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                    "Cannot add 10 ms audio, resampling failed");
       return -1;
     }
+    preprocess_frame_.samples_per_channel_ =
+        static_cast<size_t>(samples_per_channel);
     preprocess_frame_.sample_rate_hz_ =
         codec_manager_.CurrentEncoder()->SampleRateHz();
   }