2011-07-07 08:21:25 +00:00
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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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2017-09-15 06:47:31 +02:00
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#ifndef MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
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#define MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
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2011-07-07 08:21:25 +00:00
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2018-07-25 16:05:48 +02:00
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#include <stdint.h>
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2011-07-07 08:21:25 +00:00
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/*
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* Solution to support multiple instances
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*/
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typedef struct WebRtcG722EncInst G722EncInst;
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typedef struct WebRtcG722DecInst G722DecInst;
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/*
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* Comfort noise constants
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*/
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#define G722_WEBRTC_SPEECH 1
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#define G722_WEBRTC_CNG 2
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#ifdef __cplusplus
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extern "C" {
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#endif
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/****************************************************************************
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* WebRtcG722_CreateEncoder(...)
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*
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* Create memory used for G722 encoder
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*
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* Input:
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* - G722enc_inst : G722 instance for encoder
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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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2013-04-09 00:28:06 +00:00
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int16_t WebRtcG722_CreateEncoder(G722EncInst** G722enc_inst);
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2011-07-07 08:21:25 +00:00
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/****************************************************************************
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* WebRtcG722_EncoderInit(...)
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*
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* This function initializes a G722 instance
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*
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* Input:
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* - G722enc_inst : G722 instance, i.e. the user that should receive
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* be initialized
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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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2013-04-09 00:28:06 +00:00
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int16_t WebRtcG722_EncoderInit(G722EncInst* G722enc_inst);
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2011-07-07 08:21:25 +00:00
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/****************************************************************************
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* WebRtcG722_FreeEncoder(...)
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*
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* Free the memory used for G722 encoder
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*
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* Input:
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* - G722enc_inst : G722 instance for encoder
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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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Reland "Upconvert various types to int.", misc. codecs portion.
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which
reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. Specifically, the
files in webrtc/modules/audio_coding/codecs/ that are not in ilbc/ or isac/, as
well as webrtc/modules/audio_coding/main/test/opus_test.cc, are relanded.
The original commit message is below:
Upconvert various types to int.
Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t.
Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C."
This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change.
BUG=none
TBR=kwiberg
Review URL: https://codereview.webrtc.org/1179093003
Cr-Commit-Position: refs/heads/master@{#9424}
2015-06-11 19:02:46 -07:00
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int WebRtcG722_FreeEncoder(G722EncInst* G722enc_inst);
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2011-07-07 08:21:25 +00:00
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/****************************************************************************
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* WebRtcG722_Encode(...)
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*
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* This function encodes G722 encoded data.
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*
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* Input:
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* - G722enc_inst : G722 instance, i.e. the user that should encode
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* a packet
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* - speechIn : Input speech vector
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* - len : Samples in speechIn
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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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Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
2015-06-11 14:31:38 -07:00
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* Return value : Length (in bytes) of coded data
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2011-07-07 08:21:25 +00:00
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*/
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Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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size_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
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const int16_t* speechIn,
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size_t len,
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uint8_t* encoded);
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2011-07-07 08:21:25 +00:00
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/****************************************************************************
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* WebRtcG722_CreateDecoder(...)
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*
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* Create memory used for G722 encoder
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*
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* Input:
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* - G722dec_inst : G722 instance for decoder
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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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2013-04-09 00:28:06 +00:00
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int16_t WebRtcG722_CreateDecoder(G722DecInst** G722dec_inst);
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2011-07-07 08:21:25 +00:00
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/****************************************************************************
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* WebRtcG722_DecoderInit(...)
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*
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2015-08-27 15:22:11 +02:00
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* This function initializes a G722 instance
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2011-07-07 08:21:25 +00:00
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*
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* Input:
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2015-08-27 15:22:11 +02:00
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* - inst : G722 instance
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2011-07-07 08:21:25 +00:00
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*/
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2015-08-27 15:22:11 +02:00
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void WebRtcG722_DecoderInit(G722DecInst* inst);
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2011-07-07 08:21:25 +00:00
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/****************************************************************************
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* WebRtcG722_FreeDecoder(...)
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*
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* Free the memory used for G722 decoder
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*
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* Input:
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* - G722dec_inst : G722 instance for decoder
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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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Reland "Upconvert various types to int.", misc. codecs portion.
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which
reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. Specifically, the
files in webrtc/modules/audio_coding/codecs/ that are not in ilbc/ or isac/, as
well as webrtc/modules/audio_coding/main/test/opus_test.cc, are relanded.
The original commit message is below:
Upconvert various types to int.
Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t.
Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C."
This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change.
BUG=none
TBR=kwiberg
Review URL: https://codereview.webrtc.org/1179093003
Cr-Commit-Position: refs/heads/master@{#9424}
2015-06-11 19:02:46 -07:00
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int WebRtcG722_FreeDecoder(G722DecInst* G722dec_inst);
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2011-07-07 08:21:25 +00:00
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/****************************************************************************
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* WebRtcG722_Decode(...)
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*
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* This function decodes a packet with G729 frame(s). Output speech length
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* will be a multiple of 80 samples (80*frames/packet).
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*
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* Input:
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* - G722dec_inst : G722 instance, i.e. the user that should decode
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* a packet
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* - encoded : Encoded G722 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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* - speechType : 1 normal, 2 CNG (Since G722 does not have its own
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* DTX/CNG scheme it should always return 1)
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*
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Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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* Return value : Samples in decoded vector
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2011-07-07 08:21:25 +00:00
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*/
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|
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|
|
|
Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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size_t WebRtcG722_Decode(G722DecInst* G722dec_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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2011-07-07 08:21:25 +00:00
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/****************************************************************************
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* WebRtcG722_Version(...)
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*
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* Get a string with the current version of the codec
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*/
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2013-04-09 00:28:06 +00:00
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int16_t WebRtcG722_Version(char* versionStr, short len);
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2011-07-07 08:21:25 +00:00
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#ifdef __cplusplus
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}
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#endif
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2017-09-15 06:47:31 +02:00
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#endif /* MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_ */
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