| /* |
| * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_ |
| |
| /* |
| * Define the fixed-point numeric formats |
| */ |
| #include "typedefs.h" |
| |
| typedef struct WebRtcISACStruct ISACStruct; |
| |
| enum IsacSamplingRate {kIsacWideband = 16, kIsacSuperWideband = 32}; |
| |
| |
| #if defined(__cplusplus) |
| extern "C" { |
| #endif |
| |
| /****************************************************************************** |
| * WebRtcIsac_AssignSize(...) |
| * |
| * This function returns the size of the ISAC instance, so that the instance |
| * can be created outside iSAC. |
| * |
| * Input: |
| * - samplingRate : sampling rate of the input/output audio. |
| * |
| * Output: |
| * - sizeinbytes : number of bytes needed to allocate for the |
| * instance. |
| * |
| * Return value : 0 - Ok |
| * -1 - Error |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_AssignSize( |
| int* sizeinbytes); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_Assign(...) |
| * |
| * This function assignes the memory already created to the ISAC instance. |
| * |
| * Input: |
| * - *ISAC_main_inst : a pointer to the coder instance. |
| * - samplingRate : sampling rate of the input/output audio. |
| * - ISAC_inst_Addr : the already allocated memory, where we put the |
| * iSAC structure. |
| * |
| * Return value : 0 - Ok |
| * -1 - Error |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_Assign( |
| ISACStruct** ISAC_main_inst, |
| void* ISAC_inst_Addr); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_Create(...) |
| * |
| * This function creates an ISAC instance, which will contain the state |
| * information for one coding/decoding channel. |
| * |
| * Input: |
| * - *ISAC_main_inst : a pointer to the coder instance. |
| * |
| * Return value : 0 - Ok |
| * -1 - Error |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_Create( |
| ISACStruct** ISAC_main_inst); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_Free(...) |
| * |
| * This function frees the ISAC instance created at the beginning. |
| * |
| * Input: |
| * - ISAC_main_inst : an ISAC instance. |
| * |
| * Return value : 0 - Ok |
| * -1 - Error |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_Free( |
| ISACStruct* ISAC_main_inst); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_EncoderInit(...) |
| * |
| * This function initializes an ISAC instance prior to the encoder calls. |
| * |
| * Input: |
| * - ISAC_main_inst : ISAC instance. |
| * - CodingMode : 0 -> Bit rate and frame length are |
| * automatically adjusted to available bandwidth |
| * on transmission channel, just valid if codec |
| * is created to work in wideband mode. |
| * 1 -> User sets a frame length and a target bit |
| * rate which is taken as the maximum |
| * short-term average bit rate. |
| * |
| * Return value : 0 - Ok |
| * -1 - Error |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_EncoderInit( |
| ISACStruct* ISAC_main_inst, |
| WebRtc_Word16 CodingMode); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_Encode(...) |
| * |
| * This function encodes 10ms audio blocks and inserts it into a package. |
| * Input speech length has 160 samples if operating at 16 kHz sampling |
| * rate, or 320 if operating at 32 kHz sampling rate. The encoder buffers the |
| * input audio until the whole frame is buffered then proceeds with encoding. |
| * |
| * |
| * Input: |
| * - ISAC_main_inst : ISAC instance. |
| * - speechIn : input speech vector. |
| * |
| * Output: |
| * - encoded : the encoded data vector |
| * |
| * Return value: |
| * : >0 - Length (in bytes) of coded data |
| * : 0 - The buffer didn't reach the chosen |
| * frame-size so it keeps buffering speech |
| * samples. |
| * : -1 - Error |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_Encode( |
| ISACStruct* ISAC_main_inst, |
| const WebRtc_Word16* speechIn, |
| WebRtc_Word16* encoded); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_DecoderInit(...) |
| * |
| * This function initializes an ISAC instance prior to the decoder calls. |
| * |
| * Input: |
| * - ISAC_main_inst : ISAC instance. |
| * |
| * Return value |
| * : 0 - Ok |
| * -1 - Error |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_DecoderInit( |
| ISACStruct* ISAC_main_inst); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_UpdateBwEstimate(...) |
| * |
| * This function updates the estimate of the bandwidth. |
| * |
| * Input: |
| * - ISAC_main_inst : ISAC instance. |
| * - encoded : encoded ISAC frame(s). |
| * - packet_size : size of the packet. |
| * - rtp_seq_number : the RTP number of the packet. |
| * - send_ts : the RTP send timestamp, given in samples |
| * - arr_ts : the arrival time of the packet (from NetEq) |
| * in samples. |
| * |
| * Return value : 0 - Ok |
| * -1 - Error |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_UpdateBwEstimate( |
| ISACStruct* ISAC_main_inst, |
| const WebRtc_UWord16* encoded, |
| WebRtc_Word32 packet_size, |
| WebRtc_UWord16 rtp_seq_number, |
| WebRtc_UWord32 send_ts, |
| WebRtc_UWord32 arr_ts); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_Decode(...) |
| * |
| * This function decodes an ISAC frame. At 16 kHz sampling rate, the length |
| * of the output audio could be either 480 or 960 samples, equivalent to |
| * 30 or 60 ms respectively. At 32 kHz sampling rate, the length of the |
| * output audio is 960 samples, which is 30 ms. |
| * |
| * Input: |
| * - ISAC_main_inst : ISAC instance. |
| * - encoded : encoded ISAC frame(s). |
| * - len : bytes in encoded vector. |
| * |
| * Output: |
| * - decoded : The decoded vector. |
| * |
| * Return value : >0 - number of samples in decoded vector. |
| * -1 - Error. |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_Decode( |
| ISACStruct* ISAC_main_inst, |
| const WebRtc_UWord16* encoded, |
| WebRtc_Word16 len, |
| WebRtc_Word16* decoded, |
| WebRtc_Word16* speechType); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_DecodePlc(...) |
| * |
| * This function conducts PLC for ISAC frame(s). Output speech length |
| * will be a multiple of frames, i.e. multiples of 30 ms audio. Therefore, |
| * the output is multiple of 480 samples if operating at 16 kHz and multiple |
| * of 960 if operating at 32 kHz. |
| * |
| * Input: |
| * - ISAC_main_inst : ISAC instance. |
| * - noOfLostFrames : Number of PLC frames to produce. |
| * |
| * Output: |
| * - decoded : The decoded vector. |
| * |
| * Return value : >0 - number of samples in decoded PLC vector |
| * -1 - Error |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_DecodePlc( |
| ISACStruct* ISAC_main_inst, |
| WebRtc_Word16* decoded, |
| WebRtc_Word16 noOfLostFrames); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_Control(...) |
| * |
| * This function sets the limit on the short-term average bit-rate and the |
| * frame length. Should be used only in Instantaneous mode. At 16 kHz sampling |
| * rate, an average bit-rate between 10000 to 32000 bps is valid and a |
| * frame-size of 30 or 60 ms is acceptable. At 32 kHz, an average bit-rate |
| * between 10000 to 56000 is acceptable, and the valid frame-size is 30 ms. |
| * |
| * Input: |
| * - ISAC_main_inst : ISAC instance. |
| * - rate : limit on the short-term average bit rate, |
| * in bits/second. |
| * - framesize : frame-size in millisecond. |
| * |
| * Return value : 0 - ok |
| * -1 - Error |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_Control( |
| ISACStruct* ISAC_main_inst, |
| WebRtc_Word32 rate, |
| WebRtc_Word16 framesize); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_ControlBwe(...) |
| * |
| * This function sets the initial values of bottleneck and frame-size if |
| * iSAC is used in channel-adaptive mode. Therefore, this API is not |
| * applicable if the codec is created to operate in super-wideband mode. |
| * |
| * Through this API, users can enforce a frame-size for all values of |
| * bottleneck. Then iSAC will not automatically change the frame-size. |
| * |
| * |
| * Input: |
| * - ISAC_main_inst : ISAC instance. |
| * - rateBPS : initial value of bottleneck in bits/second |
| * 10000 <= rateBPS <= 56000 is accepted |
| * For default bottleneck set rateBPS = 0 |
| * - frameSizeMs : number of milliseconds per frame (30 or 60) |
| * - enforceFrameSize : 1 to enforce the given frame-size through |
| * out the adaptation process, 0 to let iSAC |
| * change the frame-size if required. |
| * |
| * Return value : 0 - ok |
| * -1 - Error |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_ControlBwe( |
| ISACStruct* ISAC_main_inst, |
| WebRtc_Word32 rateBPS, |
| WebRtc_Word16 frameSizeMs, |
| WebRtc_Word16 enforceFrameSize); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_ReadFrameLen(...) |
| * |
| * This function returns the length of the frame represented in the packet. |
| * |
| * Input: |
| * - encoded : Encoded bit-stream |
| * |
| * Output: |
| * - frameLength : Length of frame in packet (in samples) |
| * |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_ReadFrameLen( |
| ISACStruct* ISAC_main_inst, |
| const WebRtc_Word16* encoded, |
| WebRtc_Word16* frameLength); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_version(...) |
| * |
| * This function returns the version number. |
| * |
| * Output: |
| * - version : Pointer to character string |
| * |
| */ |
| |
| void WebRtcIsac_version( |
| char *version); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_GetErrorCode(...) |
| * |
| * This function can be used to check the error code of an iSAC instance. When |
| * a function returns -1 a error code will be set for that instance. The |
| * function below extract the code of the last error that occurred in the |
| * specified instance. |
| * |
| * Input: |
| * - ISAC_main_inst : ISAC instance |
| * |
| * Return value : Error code |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_GetErrorCode( |
| ISACStruct* ISAC_main_inst); |
| |
| |
| /**************************************************************************** |
| * WebRtcIsac_GetUplinkBw(...) |
| * |
| * This function outputs the target bottleneck of the codec. In |
| * channel-adaptive mode, the target bottleneck is specified through in-band |
| * signalling retreived by bandwidth estimator. |
| * In channel-independent, also called instantaneous mode, the target |
| * bottleneck is provided to the encoder by calling xxx_control(...). If |
| * xxx_control is never called the default values is returned. The default |
| * value for bottleneck at 16 kHz encoder sampling rate is 32000 bits/sec, |
| * and it is 56000 bits/sec for 32 kHz sampling rate. |
| * Note that the output is the iSAC internal operating bottleneck which might |
| * differ slightly from the one provided through xxx_control(). |
| * |
| * Input: |
| * - ISAC_main_inst : iSAC instance |
| * |
| * Output: |
| * - *bottleneck : bottleneck in bits/sec |
| * |
| * Return value : -1 if error happens |
| * 0 bit-rates computed correctly. |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_GetUplinkBw( |
| ISACStruct* ISAC_main_inst, |
| WebRtc_Word32* bottleneck); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_SetMaxPayloadSize(...) |
| * |
| * This function sets a limit for the maximum payload size of iSAC. The same |
| * value is used both for 30 and 60 ms packets. If the encoder sampling rate |
| * is 16 kHz the maximum payload size is between 120 and 400 bytes. If the |
| * encoder sampling rate is 32 kHz the maximum payload size is between 120 |
| * and 600 bytes. |
| * |
| * If an out of range limit is used, the function returns -1, but the closest |
| * valid value will be applied. |
| * |
| * --------------- |
| * IMPORTANT NOTES |
| * --------------- |
| * The size of a packet is limited to the minimum of 'max-payload-size' and |
| * 'max-rate.' For instance, let's assume the max-payload-size is set to |
| * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps |
| * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms |
| * frame-size. Then a packet with a frame-size of 30 ms is limited to 150, |
| * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to |
| * 170 bytes, i.e. min(170, 300). |
| * |
| * Input: |
| * - ISAC_main_inst : iSAC instance |
| * - maxPayloadBytes : maximum size of the payload in bytes |
| * valid values are between 120 and 400 bytes |
| * if encoder sampling rate is 16 kHz. For |
| * 32 kHz encoder sampling rate valid values |
| * are between 120 and 600 bytes. |
| * |
| * Return value : 0 if successful |
| * -1 if error happens |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_SetMaxPayloadSize( |
| ISACStruct* ISAC_main_inst, |
| WebRtc_Word16 maxPayloadBytes); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_SetMaxRate(...) |
| * |
| * This function sets the maximum rate which the codec may not exceed for |
| * any signal packet. The maximum rate is defined and payload-size per |
| * frame-size in bits per second. |
| * |
| * The codec has a maximum rate of 53400 bits per second (200 bytes per 30 |
| * ms) if the encoder sampling rate is 16kHz, and 160 kbps (600 bytes/30 ms) |
| * if the encoder sampling rate is 32 kHz. |
| * |
| * It is possible to set a maximum rate between 32000 and 53400 bits/sec |
| * in wideband mode, and 32000 to 160000 bits/sec in super-wideband mode. |
| * |
| * If an out of range limit is used, the function returns -1, but the closest |
| * valid value will be applied. |
| * |
| * --------------- |
| * IMPORTANT NOTES |
| * --------------- |
| * The size of a packet is limited to the minimum of 'max-payload-size' and |
| * 'max-rate.' For instance, let's assume the max-payload-size is set to |
| * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps |
| * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms |
| * frame-size. Then a packet with a frame-size of 30 ms is limited to 150, |
| * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to |
| * 170 bytes, min(170, 300). |
| * |
| * Input: |
| * - ISAC_main_inst : iSAC instance |
| * - maxRate : maximum rate in bits per second, |
| * valid values are 32000 to 53400 bits/sec in |
| * wideband mode, and 32000 to 160000 bits/sec in |
| * super-wideband mode. |
| * |
| * Return value : 0 if successful |
| * -1 if error happens |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_SetMaxRate( |
| ISACStruct* ISAC_main_inst, |
| WebRtc_Word32 maxRate); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_DecSampRate() |
| * Return the sampling rate of the decoded audio. |
| * |
| * Input: |
| * - ISAC_main_inst : iSAC instance |
| * |
| * Return value : enumerator representing sampling frequency |
| * associated with the decoder, i.e. the |
| * sampling rate of the decoded audio. |
| * |
| */ |
| |
| enum IsacSamplingRate WebRtcIsac_DecSampRate( |
| ISACStruct* ISAC_main_inst); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_EncSampRate() |
| * |
| * Input: |
| * - ISAC_main_inst : iSAC instance |
| * |
| * Return value : enumerator representing sampling frequency |
| * associated with the encoder, the input audio |
| * is expected to be sampled at this rate. |
| * |
| */ |
| |
| enum IsacSamplingRate WebRtcIsac_EncSampRate( |
| ISACStruct* ISAC_main_inst); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_SetDecSampRate() |
| * Set the sampling rate of the decoder. Initialization of the decoder WILL |
| * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz |
| * which is set when the instance is created. |
| * |
| * Input: |
| * - ISAC_main_inst : iSAC instance |
| * - sampRate : enumerator specifying the sampling rate. |
| * |
| * Return value : 0 if successful |
| * -1 if failed. |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_SetDecSampRate( |
| ISACStruct* ISAC_main_inst, |
| enum IsacSamplingRate sampRate); |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_SetEncSampRate() |
| * Set the sampling rate of the encoder. Initialization of the encoder WILL |
| * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz |
| * which is set when the instance is created. The encoding-mode and the |
| * bottleneck remain unchanged by this call, however, the maximum rate and |
| * maximum payload-size will reset to their default value. |
| * |
| * Input: |
| * - ISAC_main_inst : iSAC instance |
| * - sampRate : enumerator specifying the sampling rate. |
| * |
| * Return value : 0 if successful |
| * -1 if failed. |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_SetEncSampRate( |
| ISACStruct* ISAC_main_inst, |
| enum IsacSamplingRate sampRate); |
| |
| |
| |
| /****************************************************************************** |
| * WebRtcIsac_GetNewBitStream(...) |
| * |
| * This function returns encoded data, with the recieved bwe-index in the |
| * stream. If the rate is set to a value less than bottleneck of codec |
| * the new bistream will be re-encoded with the given target rate. |
| * It should always return a complete packet, i.e. only called once |
| * even for 60 msec frames. |
| * |
| * NOTE 1! This function does not write in the ISACStruct, it is not allowed. |
| * NOTE 2! Currently not implemented for SWB mode. |
| * NOTE 3! Rates larger than the bottleneck of the codec will be limited |
| * to the current bottleneck. |
| * |
| * Input: |
| * - ISAC_main_inst : ISAC instance. |
| * - bweIndex : Index of bandwidth estimate to put in new |
| * bitstream |
| * - rate : target rate of the transcoder is bits/sec. |
| * Valid values are the accepted rate in iSAC, |
| * i.e. 10000 to 56000. |
| * - isRCU : if the new bit-stream is an RCU stream. |
| * Note that the rate parameter always indicates |
| * the target rate of the main paylaod, regardless |
| * of 'isRCU' value. |
| * |
| * Output: |
| * - encoded : The encoded data vector |
| * |
| * Return value : >0 - Length (in bytes) of coded data |
| * -1 - Error or called in SWB mode |
| * NOTE! No error code is written to |
| * the struct since it is only allowed to read |
| * the struct. |
| */ |
| WebRtc_Word16 WebRtcIsac_GetNewBitStream( |
| ISACStruct* ISAC_main_inst, |
| WebRtc_Word16 bweIndex, |
| WebRtc_Word16 jitterInfo, |
| WebRtc_Word32 rate, |
| WebRtc_Word16* encoded, |
| WebRtc_Word16 isRCU); |
| |
| |
| |
| /**************************************************************************** |
| * WebRtcIsac_GetDownLinkBwIndex(...) |
| * |
| * This function returns index representing the Bandwidth estimate from |
| * other side to this side. |
| * |
| * Input: |
| * - ISAC_main_inst : iSAC struct |
| * |
| * Output: |
| * - bweIndex : Bandwidth estimate to transmit to other side. |
| * |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_GetDownLinkBwIndex( |
| ISACStruct* ISAC_main_inst, |
| WebRtc_Word16* bweIndex, |
| WebRtc_Word16* jitterInfo); |
| |
| |
| /**************************************************************************** |
| * WebRtcIsac_UpdateUplinkBw(...) |
| * |
| * This function takes an index representing the Bandwidth estimate from |
| * this side to other side and updates BWE. |
| * |
| * Input: |
| * - ISAC_main_inst : iSAC struct |
| * - bweIndex : Bandwidth estimate from other side. |
| * |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_UpdateUplinkBw( |
| ISACStruct* ISAC_main_inst, |
| WebRtc_Word16 bweIndex); |
| |
| |
| /**************************************************************************** |
| * WebRtcIsac_ReadBwIndex(...) |
| * |
| * This function returns the index of the Bandwidth estimate from the bitstream. |
| * |
| * Input: |
| * - encoded : Encoded bitstream |
| * |
| * Output: |
| * - frameLength : Length of frame in packet (in samples) |
| * - bweIndex : Bandwidth estimate in bitstream |
| * |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_ReadBwIndex( |
| const WebRtc_Word16* encoded, |
| WebRtc_Word16* bweIndex); |
| |
| |
| |
| /******************************************************************************* |
| * WebRtcIsac_GetNewFrameLen(...) |
| * |
| * returns the frame lenght (in samples) of the next packet. In the case of channel-adaptive |
| * mode, iSAC decides on its frame lenght based on the estimated bottleneck |
| * this allows a user to prepare for the next packet (at the encoder) |
| * |
| * The primary usage is in CE to make the iSAC works in channel-adaptive mode |
| * |
| * Input: |
| * - ISAC_main_inst : iSAC struct |
| * |
| * Return Value : frame lenght in samples |
| * |
| */ |
| |
| WebRtc_Word16 WebRtcIsac_GetNewFrameLen( |
| ISACStruct* ISAC_main_inst); |
| |
| |
| /**************************************************************************** |
| * WebRtcIsac_GetRedPayload(...) |
| * |
| * Populates "encoded" with the redundant payload of the recently encoded |
| * frame. This function has to be called once that WebRtcIsac_Encode(...) |
| * returns a positive value. Regardless of the frame-size this function will |
| * be called only once after encoding is completed. |
| * |
| * Input: |
| * - ISAC_main_inst : iSAC struct |
| * |
| * Output: |
| * - encoded : the encoded data vector |
| * |
| * |
| * Return value: |
| * : >0 - Length (in bytes) of coded data |
| * : -1 - Error |
| * |
| * |
| */ |
| WebRtc_Word16 WebRtcIsac_GetRedPayload( |
| ISACStruct* ISAC_main_inst, |
| WebRtc_Word16* encoded); |
| |
| |
| /**************************************************************************** |
| * WebRtcIsac_DecodeRcu(...) |
| * |
| * This function decodes a redundant (RCU) iSAC frame. Function is called in |
| * NetEq with a stored RCU payload i case of packet loss. Output speech length |
| * will be a multiple of 480 samples: 480 or 960 samples, |
| * depending on the framesize (30 or 60 ms). |
| * |
| * Input: |
| * - ISAC_main_inst : ISAC instance. |
| * - encoded : encoded ISAC RCU frame(s) |
| * - len : bytes in encoded vector |
| * |
| * Output: |
| * - decoded : The decoded vector |
| * |
| * Return value : >0 - number of samples in decoded vector |
| * -1 - Error |
| */ |
| WebRtc_Word16 WebRtcIsac_DecodeRcu( |
| ISACStruct* ISAC_main_inst, |
| const WebRtc_UWord16* encoded, |
| WebRtc_Word16 len, |
| WebRtc_Word16* decoded, |
| WebRtc_Word16* speechType); |
| |
| |
| #if defined(__cplusplus) |
| } |
| #endif |
| |
| |
| |
| #endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_ */ |
