Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc |
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc |
index 3e7d3ec738b59836a49109092430fd48840453f6..0806bb81d9839338ae931d47abfc29f71895f915 100644 |
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc |
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc |
@@ -137,15 +137,14 @@ AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal( |
uint8_t* encoded) { |
if (input_buffer_.empty()) |
first_timestamp_in_buffer_ = rtp_timestamp; |
- RTC_DCHECK_EQ(static_cast<size_t>(SamplesPer10msFrame()), audio.size()); |
+ RTC_DCHECK_EQ(SamplesPer10msFrame(), audio.size()); |
input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend()); |
if (input_buffer_.size() < |
- (static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame())) { |
+ (Num10msFramesPerPacket() * SamplesPer10msFrame())) { |
return EncodedInfo(); |
} |
- RTC_CHECK_EQ( |
- input_buffer_.size(), |
- static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame()); |
+ RTC_CHECK_EQ(input_buffer_.size(), |
+ Num10msFramesPerPacket() * SamplesPer10msFrame()); |
int status = WebRtcOpus_Encode( |
inst_, &input_buffer_[0], |
rtc::CheckedDivExact(input_buffer_.size(), |
@@ -214,11 +213,11 @@ void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) { |
RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.bitrate_bps)); |
} |
-int AudioEncoderOpus::Num10msFramesPerPacket() const { |
- return rtc::CheckedDivExact(config_.frame_size_ms, 10); |
+size_t AudioEncoderOpus::Num10msFramesPerPacket() const { |
+ return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10)); |
} |
-int AudioEncoderOpus::SamplesPer10msFrame() const { |
+size_t AudioEncoderOpus::SamplesPer10msFrame() const { |
return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels; |
} |