Update code to current Chromium master
This corresponds to: Chromium: 6555f9456074c0c0e5f7713564b978588ac04a5d webrtc: c8b569e0a7ad0b369e15f0197b3a558699ec8efa
This commit is contained in:
Arun Raghavan
724
webrtc/modules/audio_coding/codecs/isac/main/include/isac.h
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724
webrtc/modules/audio_coding/codecs/isac/main/include/isac.h
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@ -0,0 +1,724 @@
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_
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#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_
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#include <stddef.h>
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#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
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#include "webrtc/typedefs.h"
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typedef struct WebRtcISACStruct ISACStruct;
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#if defined(__cplusplus)
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extern "C" {
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#endif
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/******************************************************************************
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* WebRtcIsac_AssignSize(...)
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*
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* This function returns the size of the ISAC instance, so that the instance
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* can be created outside iSAC.
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*
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* Input:
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* - samplingRate : sampling rate of the input/output audio.
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*
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* Output:
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* - sizeinbytes : number of bytes needed to allocate for the
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* instance.
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*
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* Return value : 0 - Ok
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* -1 - Error
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*/
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int16_t WebRtcIsac_AssignSize(
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int* sizeinbytes);
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/******************************************************************************
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* WebRtcIsac_Assign(...)
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*
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* This function assignes the memory already created to the ISAC instance.
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*
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* Input:
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* - *ISAC_main_inst : a pointer to the coder instance.
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* - samplingRate : sampling rate of the input/output audio.
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* - ISAC_inst_Addr : the already allocated memory, where we put the
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* iSAC structure.
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*
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* Return value : 0 - Ok
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* -1 - Error
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*/
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int16_t WebRtcIsac_Assign(
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ISACStruct** ISAC_main_inst,
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void* ISAC_inst_Addr);
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/******************************************************************************
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* WebRtcIsac_Create(...)
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*
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* This function creates an ISAC instance, which will contain the state
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* information for one coding/decoding channel.
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*
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* Input:
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* - *ISAC_main_inst : a pointer to the coder instance.
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*
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* Return value : 0 - Ok
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* -1 - Error
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*/
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int16_t WebRtcIsac_Create(
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ISACStruct** ISAC_main_inst);
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/******************************************************************************
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* WebRtcIsac_Free(...)
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*
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* This function frees the ISAC instance created at the beginning.
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*
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* Input:
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* - ISAC_main_inst : an ISAC instance.
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*
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* Return value : 0 - Ok
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* -1 - Error
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*/
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int16_t WebRtcIsac_Free(
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ISACStruct* ISAC_main_inst);
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/******************************************************************************
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* WebRtcIsac_EncoderInit(...)
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*
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* This function initializes an ISAC instance prior to the encoder calls.
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*
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* Input:
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* - ISAC_main_inst : ISAC instance.
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* - CodingMode : 0 -> Bit rate and frame length are
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* automatically adjusted to available bandwidth
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* on transmission channel, just valid if codec
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* is created to work in wideband mode.
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* 1 -> User sets a frame length and a target bit
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* rate which is taken as the maximum
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* short-term average bit rate.
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*
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* Return value : 0 - Ok
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* -1 - Error
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*/
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int16_t WebRtcIsac_EncoderInit(
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ISACStruct* ISAC_main_inst,
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int16_t CodingMode);
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/******************************************************************************
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* WebRtcIsac_Encode(...)
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*
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* This function encodes 10ms audio blocks and inserts it into a package.
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* Input speech length has 160 samples if operating at 16 kHz sampling
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* rate, or 320 if operating at 32 kHz sampling rate. The encoder buffers the
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* input audio until the whole frame is buffered then proceeds with encoding.
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*
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*
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* Input:
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* - ISAC_main_inst : ISAC instance.
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* - speechIn : input speech vector.
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*
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* Output:
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* - encoded : the encoded data vector
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*
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* Return value:
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* : >0 - Length (in bytes) of coded data
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* : 0 - The buffer didn't reach the chosen
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* frame-size so it keeps buffering speech
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* samples.
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* : -1 - Error
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*/
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int WebRtcIsac_Encode(
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ISACStruct* ISAC_main_inst,
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const int16_t* speechIn,
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uint8_t* encoded);
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/******************************************************************************
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* WebRtcIsac_DecoderInit(...)
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*
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* This function initializes an ISAC instance prior to the decoder calls.
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*
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* Input:
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* - ISAC_main_inst : ISAC instance.
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*/
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void WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst);
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/******************************************************************************
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* WebRtcIsac_UpdateBwEstimate(...)
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*
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* This function updates the estimate of the bandwidth.
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*
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* Input:
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* - ISAC_main_inst : ISAC instance.
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* - encoded : encoded ISAC frame(s).
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* - packet_size : size of the packet.
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* - rtp_seq_number : the RTP number of the packet.
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* - send_ts : the RTP send timestamp, given in samples
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* - arr_ts : the arrival time of the packet (from NetEq)
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* in samples.
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*
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* Return value : 0 - Ok
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* -1 - Error
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*/
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int16_t WebRtcIsac_UpdateBwEstimate(
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ISACStruct* ISAC_main_inst,
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const uint8_t* encoded,
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size_t packet_size,
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uint16_t rtp_seq_number,
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uint32_t send_ts,
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uint32_t arr_ts);
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/******************************************************************************
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* WebRtcIsac_Decode(...)
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*
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* This function decodes an ISAC frame. At 16 kHz sampling rate, the length
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* of the output audio could be either 480 or 960 samples, equivalent to
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* 30 or 60 ms respectively. At 32 kHz sampling rate, the length of the
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* output audio is 960 samples, which is 30 ms.
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*
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* Input:
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* - ISAC_main_inst : ISAC instance.
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* - encoded : encoded ISAC frame(s).
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* - len : bytes in encoded vector.
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*
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* Output:
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* - decoded : The decoded vector.
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*
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* Return value : >0 - number of samples in decoded vector.
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* -1 - Error.
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*/
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int WebRtcIsac_Decode(
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ISACStruct* ISAC_main_inst,
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const uint8_t* encoded,
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size_t len,
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int16_t* decoded,
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int16_t* speechType);
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/******************************************************************************
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* WebRtcIsac_DecodePlc(...)
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*
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* This function conducts PLC for ISAC frame(s). Output speech length
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* will be a multiple of frames, i.e. multiples of 30 ms audio. Therefore,
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* the output is multiple of 480 samples if operating at 16 kHz and multiple
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* of 960 if operating at 32 kHz.
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*
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* Input:
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* - ISAC_main_inst : ISAC instance.
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* - noOfLostFrames : Number of PLC frames to produce.
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*
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* Output:
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* - decoded : The decoded vector.
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*
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* Return value : Number of samples in decoded PLC vector
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*/
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size_t WebRtcIsac_DecodePlc(
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ISACStruct* ISAC_main_inst,
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int16_t* decoded,
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size_t noOfLostFrames);
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/******************************************************************************
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* WebRtcIsac_Control(...)
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*
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* This function sets the limit on the short-term average bit-rate and the
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* frame length. Should be used only in Instantaneous mode. At 16 kHz sampling
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* rate, an average bit-rate between 10000 to 32000 bps is valid and a
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* frame-size of 30 or 60 ms is acceptable. At 32 kHz, an average bit-rate
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* between 10000 to 56000 is acceptable, and the valid frame-size is 30 ms.
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*
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* Input:
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* - ISAC_main_inst : ISAC instance.
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* - rate : limit on the short-term average bit rate,
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* in bits/second.
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* - framesize : frame-size in millisecond.
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*
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* Return value : 0 - ok
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* -1 - Error
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*/
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int16_t WebRtcIsac_Control(
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ISACStruct* ISAC_main_inst,
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int32_t rate,
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int framesize);
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void WebRtcIsac_SetInitialBweBottleneck(ISACStruct* ISAC_main_inst,
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int bottleneck_bits_per_second);
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/******************************************************************************
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* WebRtcIsac_ControlBwe(...)
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*
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* This function sets the initial values of bottleneck and frame-size if
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* iSAC is used in channel-adaptive mode. Therefore, this API is not
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* applicable if the codec is created to operate in super-wideband mode.
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*
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* Through this API, users can enforce a frame-size for all values of
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* bottleneck. Then iSAC will not automatically change the frame-size.
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*
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*
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* Input:
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* - ISAC_main_inst : ISAC instance.
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* - rateBPS : initial value of bottleneck in bits/second
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* 10000 <= rateBPS <= 56000 is accepted
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* For default bottleneck set rateBPS = 0
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* - frameSizeMs : number of milliseconds per frame (30 or 60)
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* - enforceFrameSize : 1 to enforce the given frame-size through
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* out the adaptation process, 0 to let iSAC
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* change the frame-size if required.
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*
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* Return value : 0 - ok
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* -1 - Error
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*/
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int16_t WebRtcIsac_ControlBwe(
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ISACStruct* ISAC_main_inst,
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int32_t rateBPS,
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int frameSizeMs,
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int16_t enforceFrameSize);
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/******************************************************************************
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* WebRtcIsac_ReadFrameLen(...)
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*
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* This function returns the length of the frame represented in the packet.
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*
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* Input:
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* - encoded : Encoded bit-stream
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*
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* Output:
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* - frameLength : Length of frame in packet (in samples)
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*
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*/
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int16_t WebRtcIsac_ReadFrameLen(
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ISACStruct* ISAC_main_inst,
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const uint8_t* encoded,
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int16_t* frameLength);
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/******************************************************************************
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* WebRtcIsac_version(...)
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*
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* This function returns the version number.
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*
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* Output:
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* - version : Pointer to character string
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*
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*/
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void WebRtcIsac_version(
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char *version);
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/******************************************************************************
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* WebRtcIsac_GetErrorCode(...)
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*
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* This function can be used to check the error code of an iSAC instance. When
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* a function returns -1 a error code will be set for that instance. The
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* function below extract the code of the last error that occurred in the
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* specified instance.
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*
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* Input:
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* - ISAC_main_inst : ISAC instance
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*
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* Return value : Error code
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*/
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int16_t WebRtcIsac_GetErrorCode(
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ISACStruct* ISAC_main_inst);
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/****************************************************************************
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* WebRtcIsac_GetUplinkBw(...)
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*
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* This function outputs the target bottleneck of the codec. In
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* channel-adaptive mode, the target bottleneck is specified through in-band
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* signalling retreived by bandwidth estimator.
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* In channel-independent, also called instantaneous mode, the target
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* bottleneck is provided to the encoder by calling xxx_control(...). If
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* xxx_control is never called the default values is returned. The default
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* value for bottleneck at 16 kHz encoder sampling rate is 32000 bits/sec,
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* and it is 56000 bits/sec for 32 kHz sampling rate.
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* Note that the output is the iSAC internal operating bottleneck which might
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* differ slightly from the one provided through xxx_control().
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*
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* Input:
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* - ISAC_main_inst : iSAC instance
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*
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* Output:
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* - *bottleneck : bottleneck in bits/sec
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*
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* Return value : -1 if error happens
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* 0 bit-rates computed correctly.
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*/
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int16_t WebRtcIsac_GetUplinkBw(
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ISACStruct* ISAC_main_inst,
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int32_t* bottleneck);
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/******************************************************************************
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* WebRtcIsac_SetMaxPayloadSize(...)
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*
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* This function sets a limit for the maximum payload size of iSAC. The same
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* value is used both for 30 and 60 ms packets. If the encoder sampling rate
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* is 16 kHz the maximum payload size is between 120 and 400 bytes. If the
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* encoder sampling rate is 32 kHz the maximum payload size is between 120
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* and 600 bytes.
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*
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* If an out of range limit is used, the function returns -1, but the closest
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* valid value will be applied.
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*
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* ---------------
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* IMPORTANT NOTES
|
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* ---------------
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||||
* The size of a packet is limited to the minimum of 'max-payload-size' and
|
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* 'max-rate.' For instance, let's assume the max-payload-size is set to
|
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* 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
|
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* translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
|
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* frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
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* i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
|
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* 170 bytes, i.e. min(170, 300).
|
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*
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* Input:
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* - ISAC_main_inst : iSAC instance
|
||||
* - maxPayloadBytes : maximum size of the payload in bytes
|
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* valid values are between 120 and 400 bytes
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* if encoder sampling rate is 16 kHz. For
|
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* 32 kHz encoder sampling rate valid values
|
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* are between 120 and 600 bytes.
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*
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* Return value : 0 if successful
|
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* -1 if error happens
|
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*/
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int16_t WebRtcIsac_SetMaxPayloadSize(
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ISACStruct* ISAC_main_inst,
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int16_t maxPayloadBytes);
|
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/******************************************************************************
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* 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,
|
||||
size_t len,
|
||||
int16_t* decoded,
|
||||
int16_t* speechType);
|
||||
|
||||
/* Fills in an IsacBandwidthInfo struct. |inst| should be a decoder. */
|
||||
void WebRtcIsac_GetBandwidthInfo(ISACStruct* inst, IsacBandwidthInfo* bwinfo);
|
||||
|
||||
/* Uses the values from an IsacBandwidthInfo struct. |inst| should be an
|
||||
encoder. */
|
||||
void WebRtcIsac_SetBandwidthInfo(ISACStruct* inst,
|
||||
const IsacBandwidthInfo* bwinfo);
|
||||
|
||||
/* If |inst| is a decoder but not an encoder: tell it what sample rate the
|
||||
encoder is using, for bandwidth estimation purposes. */
|
||||
void WebRtcIsac_SetEncSampRateInDecoder(ISACStruct* inst, int sample_rate_hz);
|
||||
|
||||
#if defined(__cplusplus)
|
||||
}
|
||||
#endif
|
||||
|
||||
|
||||
|
||||
#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_ */
|
||||
Reference in New Issue
Block a user