28 Commits

Author SHA1 Message Date
kwiberg
3258db26ed Split iSAC encoder/decoder: Test more cases (and make sure they work)
This patch tests separate iSAC encoder and decoder in more cases (32
kHz in addition to 16 kHz, and 30 ms adaptive and 60 ms nonadaptive).

In order to handle 32 kHz adaptive, the decoder needs to be told of
the encoder's sample rate (16 kHz worked already because that's the
default). And since we can't set the encoder's frame size without also
setting its bit rate, we need a way to set the decoder's bit rate as
well.

It turned out to be way too messy to continue verifying that the
bandwidth estimator does something reasonable in all these cases,
because it seems it doesn't. So the GetSetBandwidthInfo is now just
responsible for ensuring that split encoder/decoder behaves the same
as conjoined encoder/decoder; the job of verifying that the bandwidth
estimator does its job properly falls on some other test (that doesn't
exist yet).

Review URL: https://codereview.webrtc.org/1225093005

Cr-Commit-Position: refs/heads/master@{#9583}
2015-07-15 01:54:43 +00:00
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
Peter Kasting
aba07ef6d9 Reland "Upconvert various types to int.", isac portion.
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which
reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24.  Specifically, the
files in webrtc/modules/audio_coding/codecs/isac/ 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/1179093002

Cr-Commit-Position: refs/heads/master@{#9422}
2015-06-12 01:19:37 +00:00
Peter Kasting
728d9037c0 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 21:31:48 +00:00
Peter Kasting
cb180976dd Revert "Upconvert various types to int."
This reverts commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24.

BUG=499241
TBR=hlundin

Review URL: https://codereview.webrtc.org/1179953003

Cr-Commit-Position: refs/heads/master@{#9418}
2015-06-11 19:42:42 +00:00
Peter Kasting
83ad33a8ae 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
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54629004

Cr-Commit-Position: refs/heads/master@{#9405}
2015-06-10 00:20:09 +00:00
Karl Wiberg
dcccab3ebb New interface: AudioEncoderMutable
With implementations for all codecs. It has no users yet. This new
interface is the same as AudioEncoder (in fact it is a subclass) but
it allows changing some parameters after construction.

COAUTHOR=henrik.lundin@webrtc.org
BUG=4228
R=jmarusic@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51679004

Cr-Commit-Position: refs/heads/master@{#9149}
2015-05-07 10:35:18 +00:00
minyue@webrtc.org
7f7d7e3427 Prevent crash in NetEQ when decoder overflow.
NetEQ can crash when decoder gives too many output samples than it can handle. A practical case this happens is when multiple opus packets are combined.

The best solution is to pass the max size to the ACM decode function and let it return a failure if the max size if too small.

BUG=4361
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45619004

Cr-Commit-Position: refs/heads/master@{#8730}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8730 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-16 12:31:19 +00:00
henrik.lundin@webrtc.org
0c5b137e7e Remove support for iSAC RCU
The current way that iSAC RCU is packetized and sent as a RED packet,
with the same payload type for primary and redundant payloads, does
not follow the specification for RED. As it is now, it is impossible
for a receiver to know if an incoming RED packet with iSAC payloads
inside consists of two "primary" (but time-shifted) payloads, or one
primary and one RCU payload. The RED standard stipulates that the
former option is the correct interpretation, while our implementation
currently applies the latter.

This CL removes support for iSAC RCU from Audio Coding Module, but
leaves it in the iSAC codec itself (i.e., in the C implementation).

BUG=4402
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45569004

Cr-Commit-Position: refs/heads/master@{#8713}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8713 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-13 08:28:54 +00:00
henrik.lundin@webrtc.org
1d25c87199 Reland r8577 "Collapse AudioEncoderDecoderIsacRed into ..."
This effectively reverts r8578.

TBR=jmarusic@webrtc.org

Original commit message:
Collapse AudioEncoderDecoderIsacRed into AudioEncoderDecoderIsac

With this change, support for iSAC-RED is incorporated into the
regular AudioEncoderDecoderIsac class.

COAUTHOR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44539004

Cr-Commit-Position: refs/heads/master@{#8589}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8589 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 08:55:42 +00:00
henrik.lundin@webrtc.org
bcef431902 Revert r8577 "Collapse AudioEncoderDecoderIsacRed into ..."
Some of the build bots seems to have reacted to this change.

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42169004

Cr-Commit-Position: refs/heads/master@{#8578}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8578 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 20:13:48 +00:00
henrik.lundin@webrtc.org
1fc28f2305 Collapse AudioEncoderDecoderIsacRed into AudioEncoderDecoderIsac
With this change, support for iSAC-RED is incorporated into the regular
AudioEncoderDecoderIsac class.

COAUTHOR=kwiberg@webrtc.org
R=jmarusic@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43549004

Cr-Commit-Position: refs/heads/master@{#8577}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8577 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 19:31:17 +00:00
henrik.lundin@webrtc.org
f45c8ca88b Reland r8248 "Introduce ACMGenericCodecWrapper"
This effectively reverts r8249.

This new class inherits from ACMGenericCodec. The purpose is to wrap
AudioEncoder objects into an ACMGenericCodec interface. This is a
temporary construction that will be used during the ACM redesign work.

BUG=4228
COAUTHOR=kwiberg@webrtc.org
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38919004

Cr-Commit-Position: refs/heads/master@{#8255}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8255 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-05 18:30:16 +00:00
henrik.lundin@webrtc.org
bdebccf384 Fix a number of things in AudioEncoderDecoderIsac*
- Add max_bit_rate and max_payload_size_bytes to config structs.
- Fix support for 48 kHz sample rate.
- Fix iSAC-RED.
- Add method UpdateDecoderSampleRate().
- Update locking structure with a separate lock for local member
variables used by the encoder methods.

BUG=3926
COAUTHOR:kwiberg@webrtc.org

R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41659004

Cr-Commit-Position: refs/heads/master@{#8204}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8204 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 14:11:09 +00:00
henrik.lundin@webrtc.org
eed7a22bbf Make an AudioEncoder subclass for iSAC redundant encoding
Adding unit test, too.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36559005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7946 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 09:52:36 +00:00
kwiberg@webrtc.org
88bdec8c3a AudioEncoder subclass for iSACfix
This patch refactors AudioEncoderDecoderIsac so that it can share
almost all code with the very similar AudioEncoderDecoderIsacFix.

BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7912 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 12:49:37 +00:00
kwiberg@webrtc.org
396a5e0001 WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[]
This patch changes WebRtcIsac_Decode, WebRtcIsac_DecodeRcu, and
WebRtcIsacfix_Decode so that they read the encoded data from a uint8
array instead of a uint16 array.

BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7431 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 11:23:24 +00:00
kwiberg@webrtc.org
3f7f899a15 WebRtcIsac_UpdateBwEstimate et al.: Type byte streams as uint8, not uint16
This patch changes the signature of WebRtcIsac_UpdateBwEstimate,
WebRtcIsacfix_UpdateBwEstimate, and WebRtcIsacfix_UpdateBwEstimate1 so
that they expect the encoded data to be uint8 arrays, not uint16,
which is more natural. The implementations of the functions are left
unchanged for now.

BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7430 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 11:07:06 +00:00
kwiberg@webrtc.org
1172988c79 Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[]
The affected functions are

  WebRtcIsacfix_ReadFrameLen
  WebRtcIsacfix_GetNewBitStream
  WebRtcIsacfix_ReadBwIndex

and

  WebRtcIsac_ReadFrameLen
  WebRtcIsac_GetNewBitStream
  WebRtcIsac_ReadBwIndex
  WebRtcIsac_GetRedPayload

BUG=909
R=aluebs@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7429 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 10:53:42 +00:00
turaj@webrtc.org
532ed43e85 Prevent reading outside iSAC bitstream, if the stream is corrupted.
BUG=chrome_373312(#24)
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7379 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 00:21:02 +00:00
kwiberg@webrtc.org
7ee24a7906 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
We have to fix both at once, since there's a macro that calls one of
them or the other.

BUG=909
R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7266

Review URL: https://webrtc-codereview.appspot.com/19229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7285 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 10:31:02 +00:00
andrew@webrtc.org
a3c4d4dd2c Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..."
This was causing apparently legitimate failures on the following bots:
http://chromegw/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/2599
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29%28dbg%29/builds/2023
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29%28dbg%29/builds/1825
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29/builds/2013
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29/builds/1795

> WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
> 
> We have to fix both at once, since there's a macro that calls one of
> them or the other.
> 
> BUG=909
> R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19229004

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7267 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 01:32:57 +00:00
kwiberg@webrtc.org
8c5740b485 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
We have to fix both at once, since there's a macro that calls one of
them or the other.

BUG=909
R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7266 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 23:04:14 +00:00
andresp@webrtc.org
262e676a08 Reland rev 7041 with BUILD.gn files.
Original description:
  Audio codecs to include webrtc/typedefs.h

  Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h

  CL Generated with:
  $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g"

BUG=3777
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 13:28:48 +00:00
henrike@webrtc.org
1b8b4c4959 Revert 7041 " Audio codecs to include webrtc/typedefs.h"
Breaks gn build, see e.g. http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Linux%20GN/builds/1248/steps/compile/logs/stdio

R=turaj@webrtc.org
TBR=andresp@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/19219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7046 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 19:42:16 +00:00
andresp@webrtc.org
9730d3aae9 Audio codecs to include webrtc/typedefs.h
Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h

CL Generated with:
$ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g"

BUG=3777
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7041 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 14:37:18 +00:00
pbos@webrtc.org
0946a56023 WebRtc_Word32 => int32_t etc. in audio_coding/
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1271006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-09 00:28:06 +00:00
andrew@webrtc.org
14b43beb7c Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00