Files
webrtc/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
Karl Wiberg 2224294c52 iSAC: Functions for importing and exporting bandwidth est. info
They make it possible to send bandwidth estimation info from decoder
to encoder even if they are separate objects (which we want them to be
because multithreading).

R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1208923002.

Cr-Commit-Position: refs/heads/master@{#9535}
2015-07-03 02:04:46 +00:00

723 lines
25 KiB
C

/*
* 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_
#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
#include "webrtc/typedefs.h"
typedef struct WebRtcISACStruct ISACStruct;
#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
*/
int16_t 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
*/
int16_t 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
*/
int16_t 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
*/
int16_t 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
*/
int16_t WebRtcIsac_EncoderInit(
ISACStruct* ISAC_main_inst,
int16_t 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
*/
int WebRtcIsac_Encode(
ISACStruct* ISAC_main_inst,
const int16_t* speechIn,
uint8_t* 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
*/
int16_t 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
*/
int16_t WebRtcIsac_UpdateBwEstimate(
ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int32_t packet_size,
uint16_t rtp_seq_number,
uint32_t send_ts,
uint32_t 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.
*/
int WebRtcIsac_Decode(
ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* 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
*/
int16_t WebRtcIsac_DecodePlc(
ISACStruct* ISAC_main_inst,
int16_t* decoded,
int16_t 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
*/
int16_t WebRtcIsac_Control(
ISACStruct* ISAC_main_inst,
int32_t rate,
int 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
*/
int16_t WebRtcIsac_ControlBwe(
ISACStruct* ISAC_main_inst,
int32_t rateBPS,
int frameSizeMs,
int16_t 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)
*
*/
int16_t WebRtcIsac_ReadFrameLen(
ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int16_t* 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
*/
int16_t 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.
*/
int16_t WebRtcIsac_GetUplinkBw(
ISACStruct* ISAC_main_inst,
int32_t* 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
*/
int16_t WebRtcIsac_SetMaxPayloadSize(
ISACStruct* ISAC_main_inst,
int16_t 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
*/
int16_t WebRtcIsac_SetMaxRate(
ISACStruct* ISAC_main_inst,
int32_t maxRate);
/******************************************************************************
* WebRtcIsac_DecSampRate()
* Return the sampling rate of the decoded audio.
*
* Input:
* - ISAC_main_inst : iSAC instance
*
* Return value : sampling frequency in Hertz.
*
*/
uint16_t WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst);
/******************************************************************************
* WebRtcIsac_EncSampRate()
*
* Input:
* - ISAC_main_inst : iSAC instance
*
* Return value : sampling rate in Hertz.
*
*/
uint16_t 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 : sampling rate in Hertz.
*
* Return value : 0 if successful
* -1 if failed.
*/
int16_t WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
uint16_t samp_rate_hz);
/******************************************************************************
* 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 : sampling rate in Hertz.
*
* Return value : 0 if successful
* -1 if failed.
*/
int16_t WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
uint16_t sample_rate_hz);
/******************************************************************************
* 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 payload, 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.
*/
int16_t WebRtcIsac_GetNewBitStream(
ISACStruct* ISAC_main_inst,
int16_t bweIndex,
int16_t jitterInfo,
int32_t rate,
uint8_t* encoded,
int16_t 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.
*
*/
int16_t WebRtcIsac_GetDownLinkBwIndex(
ISACStruct* ISAC_main_inst,
int16_t* bweIndex,
int16_t* 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.
*
*/
int16_t WebRtcIsac_UpdateUplinkBw(
ISACStruct* ISAC_main_inst,
int16_t 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
*
*/
int16_t WebRtcIsac_ReadBwIndex(
const uint8_t* encoded,
int16_t* 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
*
*/
int16_t 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
*
*
*/
int16_t WebRtcIsac_GetRedPayload(
ISACStruct* ISAC_main_inst,
uint8_t* 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
*/
int WebRtcIsac_DecodeRcu(
ISACStruct* ISAC_main_inst,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType);
/* Fills in an IsacBandwidthInfo struct. */
void WebRtcIsac_GetBandwidthInfo(ISACStruct* inst, IsacBandwidthInfo* bwinfo);
/* Uses the values from an IsacBandwidthInfo struct. */
void WebRtcIsac_SetBandwidthInfo(ISACStruct* inst,
const IsacBandwidthInfo* bwinfo);
#if defined(__cplusplus)
}
#endif
#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_ */