Index: webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h |
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h |
index 2e15fb562000ba40793fb36c086a43658c196e58..e840f3b1b186cbe839acbfa8c9cdc6ce70c4a9d3 100644 |
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h |
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h |
@@ -23,8 +23,6 @@ struct CodecInst; |
template <typename T> |
class AudioEncoderIsacT final : public AudioEncoder { |
public: |
- using AudioEncoder::EncodeInternal; |
- |
// Allowed combinations of sample rate, frame size, and bit rate are |
// - 16000 Hz, 30 ms, 10000-32000 bps |
// - 16000 Hz, 60 ms, 10000-32000 bps |
@@ -62,9 +60,9 @@ class AudioEncoderIsacT final : public AudioEncoder { |
size_t Num10MsFramesInNextPacket() const override; |
size_t Max10MsFramesInAPacket() const override; |
int GetTargetBitrate() const override; |
- EncodedInfo EncodeInternal(uint32_t rtp_timestamp, |
- rtc::ArrayView<const int16_t> audio, |
- rtc::Buffer* encoded) override; |
+ EncodedInfo EncodeImpl(uint32_t rtp_timestamp, |
+ rtc::ArrayView<const int16_t> audio, |
+ rtc::Buffer* encoded) override; |
void Reset() override; |
private: |