Index: webrtc/modules/audio_coding/codecs/audio_encoder.cc |
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc |
index 0520918f9f1971bb1983f90d8267d39fc86ad87e..6f793e253142e0e233c7389ef5810ed18d57aa37 100644 |
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc |
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc |
@@ -32,7 +32,7 @@ AudioEncoder::EncodedInfo AudioEncoder::Encode( |
static_cast<size_t>(NumChannels() * SampleRateHz() / 100)); |
const size_t old_size = encoded->size(); |
- EncodedInfo info = EncodeInternal(rtp_timestamp, audio, encoded); |
+ EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded); |
RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes); |
return info; |
} |
@@ -59,7 +59,7 @@ AudioEncoder::EncodedInfo AudioEncoder::DEPRECATED_Encode( |
return info; |
} |
-AudioEncoder::EncodedInfo AudioEncoder::EncodeInternal( |
+AudioEncoder::EncodedInfo AudioEncoder::EncodeImpl( |
uint32_t rtp_timestamp, |
rtc::ArrayView<const int16_t> audio, |
rtc::Buffer* encoded) |
@@ -80,7 +80,7 @@ AudioEncoder::EncodedInfo AudioEncoder::EncodeInternal( |
uint8_t* encoded) |
{ |
rtc::Buffer temp_buffer; |
- EncodedInfo info = EncodeInternal(rtp_timestamp, audio, &temp_buffer); |
+ EncodedInfo info = EncodeImpl(rtp_timestamp, audio, &temp_buffer); |
RTC_DCHECK_LE(temp_buffer.size(), max_encoded_bytes); |
std::memcpy(encoded, temp_buffer.data(), info.encoded_bytes); |
return info; |