Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.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.
  */
 
 #include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 
 #include <limits>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
 
 namespace webrtc {
 
 namespace {
 
-int16_t NumSamplesPerFrame(int num_channels,
-                           int frame_size_ms,
-                           int sample_rate_hz) {
-  int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000;
-  RTC_CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max())
-      << "Frame size too large.";
-  return static_cast<int16_t>(samples_per_frame);
-}
-
 template <typename T>
 typename T::Config CreateConfig(const CodecInst& codec_inst) {
   typename T::Config config;
   config.frame_size_ms = codec_inst.pacsize / 8;
   config.num_channels = codec_inst.channels;
   config.payload_type = codec_inst.pltype;
   return config;
 }
 
 }  // namespace
 
 bool AudioEncoderPcm::Config::IsOk() const {
   return (frame_size_ms % 10 == 0) && (num_channels >= 1);
 }
 
 AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
     : sample_rate_hz_(sample_rate_hz),
       num_channels_(config.num_channels),
       payload_type_(config.payload_type),
       num_10ms_frames_per_packet_(
           static_cast<size_t>(config.frame_size_ms / 10)),
-      full_frame_samples_(NumSamplesPerFrame(config.num_channels,
-                                             config.frame_size_ms,
-                                             sample_rate_hz_)),
+      full_frame_samples_(
+          config.num_channels * config.frame_size_ms * sample_rate_hz / 1000),
       first_timestamp_in_buffer_(0) {
   RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
   RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
       << "Frame size must be an integer multiple of 10 ms.";
   speech_buffer_.reserve(full_frame_samples_);
 }
 
 AudioEncoderPcm::~AudioEncoderPcm() = default;
 
 size_t AudioEncoderPcm::MaxEncodedBytes() const {
   return full_frame_samples_ * BytesPerSample();
 }
 
 int AudioEncoderPcm::SampleRateHz() const {
   return sample_rate_hz_;
 }
 
-int AudioEncoderPcm::NumChannels() const {
+size_t AudioEncoderPcm::NumChannels() const {
   return num_channels_;
 }
 
 size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
@@ skipping 50 matching lines @@
                                     size_t input_len,
                                     uint8_t* encoded) {
   return WebRtcG711_EncodeU(audio, input_len, encoded);
 }
 
 size_t AudioEncoderPcmU::BytesPerSample() const {
   return 1;
 }
 
 }  // namespace webrtc