| 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 50d369ba35f397253201027ce39b374ee35c3a8c..6b361b4146eb98bcf90dd47c3936e822c467c3c3 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
|
| @@ -13,7 +13,6 @@
|
|
|
| #include <vector>
|
|
|
| -#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
|
| #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
|
| #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
|
|
|
| @@ -94,36 +93,5 @@ class AudioEncoderIsacT final : public AudioEncoder {
|
| RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT);
|
| };
|
|
|
| -template <typename T>
|
| -class AudioDecoderIsacT final : public AudioDecoder {
|
| - public:
|
| - AudioDecoderIsacT();
|
| - explicit AudioDecoderIsacT(LockedIsacBandwidthInfo* bwinfo);
|
| - ~AudioDecoderIsacT() override;
|
| -
|
| - bool HasDecodePlc() const override;
|
| - size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
|
| - void Reset() override;
|
| - int IncomingPacket(const uint8_t* payload,
|
| - size_t payload_len,
|
| - uint16_t rtp_sequence_number,
|
| - uint32_t rtp_timestamp,
|
| - uint32_t arrival_timestamp) override;
|
| - int ErrorCode() override;
|
| - size_t Channels() const override;
|
| - int DecodeInternal(const uint8_t* encoded,
|
| - size_t encoded_len,
|
| - int sample_rate_hz,
|
| - int16_t* decoded,
|
| - SpeechType* speech_type) override;
|
| -
|
| - private:
|
| - typename T::instance_type* isac_state_;
|
| - LockedIsacBandwidthInfo* bwinfo_;
|
| - int decoder_sample_rate_hz_;
|
| -
|
| - RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
|
| -};
|
| -
|
| } // namespace webrtc
|
| #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
|
|
|