| 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:
|
|
|