| 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;
|
|
|