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_ |