Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 16 matching lines @@
 // If we are on Android, iOS and/or ARM, use a lower complexity setting as
 // default, to save encoder complexity.
 const int kDefaultComplexity = 5;
 #else
 const int kDefaultComplexity = 9;
 #endif
 
 // We always encode at 48 kHz.
 const int kSampleRateHz = 48000;
 
-int16_t ClampInt16(size_t x) {
-  return static_cast<int16_t>(
-      std::min(x, static_cast<size_t>(std::numeric_limits<int16_t>::max())));
-}
-
-int16_t CastInt16(size_t x) {
-  DCHECK_LE(x, static_cast<size_t>(std::numeric_limits<int16_t>::max()));
-  return static_cast<int16_t>(x);
-}
-
 }  // namespace
 
 AudioEncoderOpus::Config::Config()
     : frame_size_ms(20),
       num_channels(1),
       payload_type(120),
       application(kVoip),
       bitrate_bps(64000),
       fec_enabled(false),
       max_playback_rate_hz(48000),
       complexity(kDefaultComplexity),
       dtx_enabled(false) {
 }
 
 bool AudioEncoderOpus::Config::IsOk() const {
   if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
     return false;
   if (num_channels != 1 && num_channels != 2)
     return false;
   if (bitrate_bps < kMinBitrateBps || bitrate_bps > kMaxBitrateBps)
     return false;
   if (complexity < 0 || complexity > 10)
     return false;
   return true;
 }
 
 AudioEncoderOpus::AudioEncoderOpus(const Config& config)
     : num_10ms_frames_per_packet_(
-          rtc::CheckedDivExact(config.frame_size_ms, 10)),
+          static_cast<size_t>(rtc::CheckedDivExact(config.frame_size_ms, 10))),
       num_channels_(config.num_channels),
       payload_type_(config.payload_type),
       application_(config.application),
       dtx_enabled_(config.dtx_enabled),
-      samples_per_10ms_frame_(rtc::CheckedDivExact(kSampleRateHz, 100) *
-                              num_channels_),
+      samples_per_10ms_frame_(static_cast<size_t>(
+          rtc::CheckedDivExact(kSampleRateHz, 100) * num_channels_)),
       packet_loss_rate_(0.0) {
   CHECK(config.IsOk());
   input_buffer_.reserve(num_10ms_frames_per_packet_ * samples_per_10ms_frame_);
   CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, num_channels_, application_));
   SetTargetBitrate(config.bitrate_bps);
   if (config.fec_enabled) {
     CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
   } else {
     CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
   }
@@ skipping 22 matching lines @@
 size_t AudioEncoderOpus::MaxEncodedBytes() const {
   // Calculate the number of bytes we expect the encoder to produce,
   // then multiply by two to give a wide margin for error.
   size_t bytes_per_millisecond =
       static_cast<size_t>(bitrate_bps_ / (1000 * 8) + 1);
   size_t approx_encoded_bytes =
       num_10ms_frames_per_packet_ * 10 * bytes_per_millisecond;
   return 2 * approx_encoded_bytes;
 }
 
-int AudioEncoderOpus::Num10MsFramesInNextPacket() const {
+size_t AudioEncoderOpus::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
-int AudioEncoderOpus::Max10MsFramesInAPacket() const {
+size_t AudioEncoderOpus::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 int AudioEncoderOpus::GetTargetBitrate() const {
   return bitrate_bps_;
 }
 
 void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
   bitrate_bps_ = std::max(std::min(bits_per_second, kMaxBitrateBps),
                           kMinBitrateBps);
@@ skipping 49 matching lines @@
 
 AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
     uint32_t rtp_timestamp,
     const int16_t* audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   if (input_buffer_.empty())
     first_timestamp_in_buffer_ = rtp_timestamp;
   input_buffer_.insert(input_buffer_.end(), audio,
                        audio + samples_per_10ms_frame_);
-  if (input_buffer_.size() < (static_cast<size_t>(num_10ms_frames_per_packet_) *
-                              samples_per_10ms_frame_)) {
+  if (input_buffer_.size() <
+      (num_10ms_frames_per_packet_ * samples_per_10ms_frame_)) {
     return EncodedInfo();
   }
   CHECK_EQ(input_buffer_.size(),
-           static_cast<size_t>(num_10ms_frames_per_packet_) *
-           samples_per_10ms_frame_);
+           num_10ms_frames_per_packet_ * samples_per_10ms_frame_);
   int status = WebRtcOpus_Encode(
       inst_, &input_buffer_[0],
-      rtc::CheckedDivExact(CastInt16(input_buffer_.size()),
-                           static_cast<int16_t>(num_channels_)),
-      ClampInt16(max_encoded_bytes), encoded);
+      rtc::CheckedDivExact(input_buffer_.size(),
+                           static_cast<size_t>(num_channels_)),
+      max_encoded_bytes, encoded);
   CHECK_GE(status, 0);  // Fails only if fed invalid data.
   input_buffer_.clear();
   EncodedInfo info;
   info.encoded_bytes = static_cast<size_t>(status);
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
   info.send_even_if_empty = true;  // Allows Opus to send empty packets.
   info.speech = (status > 0);
   return info;
 }
@@ skipping 40 matching lines @@
   return Reconstruct(conf);
 }
 
 bool AudioEncoderMutableOpus::SetMaxPlaybackRate(int frequency_hz) {
   auto conf = config();
   conf.max_playback_rate_hz = frequency_hz;
   return Reconstruct(conf);
 }
 
 }  // namespace webrtc