Update a ton of audio code to use size_t more correctly and in general reduce

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

Index: webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index 4c11197f49f97ebca6cabd7809baaba0fe923cc3..1cefeb662661091a782be499dc8e02699616d7ab 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -344,7 +344,7 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
 
 int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
   enum NetEqOutputType type;
-  int samples_per_channel;
+  size_t samples_per_channel;
   int num_channels;
   bool return_silence = false;
 
@@ -394,7 +394,7 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
   }
 
   // NetEq always returns 10 ms of audio.
-  current_sample_rate_hz_ = samples_per_channel * 100;
+  current_sample_rate_hz_ = static_cast<int>(samples_per_channel * 100);
 
   // Update if resampling is required.
   bool need_resampling = (desired_freq_hz != -1) &&
@@ -403,18 +403,19 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
   if (need_resampling && !resampled_last_output_frame_) {
     // Prime the resampler with the last frame.
     int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
-    samples_per_channel =
+    int samples_per_channel_int =
         resampler_.Resample10Msec(last_audio_buffer_.get(),
                                   current_sample_rate_hz_,
                                   desired_freq_hz,
                                   num_channels,
                                   AudioFrame::kMaxDataSizeSamples,
                                   temp_output);
-    if (samples_per_channel < 0) {
+    if (samples_per_channel_int < 0) {
       LOG(LERROR) << "AcmReceiver::GetAudio - "
                      "Resampling last_audio_buffer_ failed.";
       return -1;
     }
+    samples_per_channel = static_cast<size_t>(samples_per_channel_int);
   }
 
   // The audio in |audio_buffer_| is tansferred to |audio_frame_| below, either
@@ -422,17 +423,18 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
   // TODO(henrik.lundin) Glitches in the output may appear if the output rate
   // from NetEq changes. See WebRTC issue 3923.
   if (need_resampling) {
-    samples_per_channel =
+    int samples_per_channel_int =
         resampler_.Resample10Msec(audio_buffer_.get(),
                                   current_sample_rate_hz_,
                                   desired_freq_hz,
                                   num_channels,
                                   AudioFrame::kMaxDataSizeSamples,
                                   audio_frame->data_);
-    if (samples_per_channel < 0) {
+    if (samples_per_channel_int < 0) {
       LOG(LERROR) << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
       return -1;
     }
+    samples_per_channel = static_cast<size_t>(samples_per_channel_int);
     resampled_last_output_frame_ = true;
   } else {
     resampled_last_output_frame_ = false;
@@ -448,7 +450,7 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
 
   audio_frame->num_channels_ = num_channels;
   audio_frame->samples_per_channel_ = samples_per_channel;
-  audio_frame->sample_rate_hz_ = samples_per_channel * 100;
+  audio_frame->sample_rate_hz_ = static_cast<int>(samples_per_channel * 100);
 
   // Should set |vad_activity| before calling SetAudioFrameActivityAndType().
   audio_frame->vad_activity_ = previous_audio_activity_;
@@ -787,10 +789,11 @@ bool AcmReceiver::GetSilence(int desired_sample_rate_hz, AudioFrame* frame) {
     frame->sample_rate_hz_ = current_sample_rate_hz_;
   }
 
-  frame->samples_per_channel_ = frame->sample_rate_hz_ / 100;  // Always 10 ms.
+  frame->samples_per_channel_ =
+      static_cast<size_t>(frame->sample_rate_hz_ / 100);  // Always 10 ms.
   frame->speech_type_ = AudioFrame::kCNG;
   frame->vad_activity_ = AudioFrame::kVadPassive;
-  int samples = frame->samples_per_channel_ * frame->num_channels_;
+  size_t samples = frame->samples_per_channel_ * frame->num_channels_;
   memset(frame->data_, 0, samples * sizeof(int16_t));
   return true;
 }