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Unified Diff: webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h

Issue 1417173004: audio_coding: rename interface -> include (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Restored incorrectly renamed header guards and fixed an old error Created 5 years, 2 months ago
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Index: webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h b/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
deleted file mode 100644
index 1f5aeb36f41de5694fbb256993779219aea96490..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
+++ /dev/null
@@ -1,724 +0,0 @@
-/*
- * 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 <stddef.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.
- */
-
- void 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,
- size_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,
- size_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 : Number of samples in decoded PLC vector
- */
-
- size_t WebRtcIsac_DecodePlc(
- ISACStruct* ISAC_main_inst,
- int16_t* decoded,
- size_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);
-
- void WebRtcIsac_SetInitialBweBottleneck(ISACStruct* ISAC_main_inst,
- int bottleneck_bits_per_second);
-
- /******************************************************************************
- * 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,
- 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_INTERFACE_ISAC_H_ */
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