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freeswitch/src/mod/codecs/mod_isac/isac.h
Anthony Minessale 224943f0b6 remove case from mod_iSAC sheesh....
2012-01-24 16:58:28 -06:00

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/*
* 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_ */
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