Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
       3             :  *
       4             :  *  Use of this source code is governed by a BSD-style license
       5             :  *  that can be found in the LICENSE file in the root of the source
       6             :  *  tree. An additional intellectual property rights grant can be found
       7             :  *  in the file PATENTS.  All contributing project authors may
       8             :  *  be found in the AUTHORS file in the root of the source tree.
       9             :  */
      10             : 
      11             : #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
      12             : #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
      13             : 
      14             : #include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
      15             : 
      16             : namespace webrtc {
      17             : 
      18             : struct IsacFloat {
      19             :   using instance_type = ISACStruct;
      20             :   static const bool has_swb = true;
      21           0 :   static inline int16_t Control(instance_type* inst,
      22             :                                 int32_t rate,
      23             :                                 int framesize) {
      24           0 :     return WebRtcIsac_Control(inst, rate, framesize);
      25             :   }
      26           0 :   static inline int16_t ControlBwe(instance_type* inst,
      27             :                                    int32_t rate_bps,
      28             :                                    int frame_size_ms,
      29             :                                    int16_t enforce_frame_size) {
      30           0 :     return WebRtcIsac_ControlBwe(inst, rate_bps, frame_size_ms,
      31           0 :                                  enforce_frame_size);
      32             :   }
      33           0 :   static inline int16_t Create(instance_type** inst) {
      34           0 :     return WebRtcIsac_Create(inst);
      35             :   }
      36           0 :   static inline int DecodeInternal(instance_type* inst,
      37             :                                    const uint8_t* encoded,
      38             :                                    size_t len,
      39             :                                    int16_t* decoded,
      40             :                                    int16_t* speech_type) {
      41           0 :     return WebRtcIsac_Decode(inst, encoded, len, decoded, speech_type);
      42             :   }
      43           0 :   static inline size_t DecodePlc(instance_type* inst,
      44             :                                  int16_t* decoded,
      45             :                                  size_t num_lost_frames) {
      46           0 :     return WebRtcIsac_DecodePlc(inst, decoded, num_lost_frames);
      47             :   }
      48             : 
      49           0 :   static inline void DecoderInit(instance_type* inst) {
      50           0 :     WebRtcIsac_DecoderInit(inst);
      51           0 :   }
      52           0 :   static inline int Encode(instance_type* inst,
      53             :                            const int16_t* speech_in,
      54             :                            uint8_t* encoded) {
      55           0 :     return WebRtcIsac_Encode(inst, speech_in, encoded);
      56             :   }
      57           0 :   static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) {
      58           0 :     return WebRtcIsac_EncoderInit(inst, coding_mode);
      59             :   }
      60           0 :   static inline uint16_t EncSampRate(instance_type* inst) {
      61           0 :     return WebRtcIsac_EncSampRate(inst);
      62             :   }
      63             : 
      64           0 :   static inline int16_t Free(instance_type* inst) {
      65           0 :     return WebRtcIsac_Free(inst);
      66             :   }
      67           0 :   static inline void GetBandwidthInfo(instance_type* inst,
      68             :                                       IsacBandwidthInfo* bwinfo) {
      69           0 :     WebRtcIsac_GetBandwidthInfo(inst, bwinfo);
      70           0 :   }
      71           0 :   static inline int16_t GetErrorCode(instance_type* inst) {
      72           0 :     return WebRtcIsac_GetErrorCode(inst);
      73             :   }
      74             : 
      75           0 :   static inline int16_t GetNewFrameLen(instance_type* inst) {
      76           0 :     return WebRtcIsac_GetNewFrameLen(inst);
      77             :   }
      78           0 :   static inline void SetBandwidthInfo(instance_type* inst,
      79             :                                       const IsacBandwidthInfo* bwinfo) {
      80           0 :     WebRtcIsac_SetBandwidthInfo(inst, bwinfo);
      81           0 :   }
      82           0 :   static inline int16_t SetDecSampRate(instance_type* inst,
      83             :                                        uint16_t sample_rate_hz) {
      84           0 :     return WebRtcIsac_SetDecSampRate(inst, sample_rate_hz);
      85             :   }
      86           0 :   static inline int16_t SetEncSampRate(instance_type* inst,
      87             :                                        uint16_t sample_rate_hz) {
      88           0 :     return WebRtcIsac_SetEncSampRate(inst, sample_rate_hz);
      89             :   }
      90             :   static inline void SetEncSampRateInDecoder(instance_type* inst,
      91             :                                              uint16_t sample_rate_hz) {
      92             :     WebRtcIsac_SetEncSampRateInDecoder(inst, sample_rate_hz);
      93             :   }
      94             :   static inline void SetInitialBweBottleneck(instance_type* inst,
      95             :                                              int bottleneck_bits_per_second) {
      96             :     WebRtcIsac_SetInitialBweBottleneck(inst, bottleneck_bits_per_second);
      97             :   }
      98           0 :   static inline int16_t UpdateBwEstimate(instance_type* inst,
      99             :                                          const uint8_t* encoded,
     100             :                                          size_t packet_size,
     101             :                                          uint16_t rtp_seq_number,
     102             :                                          uint32_t send_ts,
     103             :                                          uint32_t arr_ts) {
     104           0 :     return WebRtcIsac_UpdateBwEstimate(inst, encoded, packet_size,
     105           0 :                                        rtp_seq_number, send_ts, arr_ts);
     106             :   }
     107           0 :   static inline int16_t SetMaxPayloadSize(instance_type* inst,
     108             :                                           int16_t max_payload_size_bytes) {
     109           0 :     return WebRtcIsac_SetMaxPayloadSize(inst, max_payload_size_bytes);
     110             :   }
     111           0 :   static inline int16_t SetMaxRate(instance_type* inst, int32_t max_bit_rate) {
     112           0 :     return WebRtcIsac_SetMaxRate(inst, max_bit_rate);
     113             :   }
     114             : };
     115             : 
     116             : }  // namespace webrtc
     117             : #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_