| Index: webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
|
| diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
|
| index ba5959dbcd5c32571e7ab53db0af2c1f4034437f..127751c5ddb4e1b930c2d041e8d915e568f209f7 100644
|
| --- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
|
| +++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
|
| @@ -19,11 +19,13 @@
|
| namespace webrtc {
|
|
|
| namespace {
|
| -int16_t NumSamplesPerFrame(int num_channels,
|
| +int16_t NumSamplesPerFrame(size_t num_channels,
|
| int frame_size_ms,
|
| int sample_rate_hz) {
|
| - int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000;
|
| - CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max())
|
| + size_t samples_per_frame =
|
| + num_channels * frame_size_ms * sample_rate_hz / 1000;
|
| + CHECK_LE(samples_per_frame,
|
| + static_cast<size_t>(std::numeric_limits<int16_t>::max()))
|
| << "Frame size too large.";
|
| return static_cast<int16_t>(samples_per_frame);
|
| }
|
| @@ -56,7 +58,7 @@ int AudioEncoderPcm::SampleRateHz() const {
|
| return sample_rate_hz_;
|
| }
|
|
|
| -int AudioEncoderPcm::NumChannels() const {
|
| +size_t AudioEncoderPcm::NumChannels() const {
|
| return num_channels_;
|
| }
|
|
|
|
|