Prepare to convert various types to size_t.

Prepare to convert various types to size_t.

This makes some behaviorally-invariant changes to make certain code that
currently only works correctly with signed types work safely regardless of the
signedness of the types in question.  This is preparation for a future change
that will convert a variety of types to size_t.

There are also some formatting changes (e.g. converting "enum hack" usage to real consts) to make it simpler to just change "int" to "size_t" in the future to change the types of those constants.

BUG=none
R=andrew@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=ajm

Committed: https://crrev.com/f045e4da43e671ae511aa1d9b6ef2968256a745d
Cr-Commit-Position: refs/heads/master@{#9413}

[Peter Kasting, 2015-06-10 00:57:34]

pkasting@chromium.org changed reviewers:
+ andrew@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org

[Peter Kasting, 2015-06-10 00:58:54]

This is a copy of the latest patch on
https://webrtc-codereview.appspot.com/53649004/ , moved here so I can continue
to run tryjobs on it.

TBR to ajm for webrtc/common_audio/, webrtc/modules/audio_processing/,
webrtc/modules/interface/ OWNERS as he reviewed those already on the other CL.

juberti: talk/app/webrtc/ OWNERS
kwiberg: webrtc/modules/audio_coding/ OWNERS

[Andrew MacDonald, 2015-06-10 01:27:37]

lgtm

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/interface/...
File webrtc/modules/interface/module_common_types.h (right):

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/interface/...
webrtc/modules/interface/module_common_types.h:446: const int length =
samples_per_channel * num_channels;
Not sure I looked at this file before. I assume samples_per_channel will become
a size_t later?

[Peter Kasting, 2015-06-10 01:58:37]

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/interface/...
File webrtc/modules/interface/module_common_types.h (right):

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/interface/...
webrtc/modules/interface/module_common_types.h:446: const int length =
samples_per_channel * num_channels;
On 2015/06/10 01:27:37, andrew wrote:
> Not sure I looked at this file before. I assume samples_per_channel will
become
> a size_t later? 

Yes, as will |length|.

[juberti1, 2015-06-10 05:13:45]

lgtm

[kwiberg-webrtc, 2015-06-10 07:49:10]

lgtm for webrtc/modules/audio_coding/

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/audio_codi...
File webrtc/modules/audio_coding/codecs/ilbc/encode.c (right):

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/codecs/ilbc/encode.c:431: start_count = (Nfor > 1) ?
0 : (2 - NFor);
Nit: it'd be slightly easier to read if you said Nfor >= 2

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/codecs/ilbc/encode.c:431: start_count = (Nfor > 1) ?
0 : (2 - NFor);
See previous comment.

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/audio_codi...
File webrtc/modules/audio_coding/neteq/expand.cc (right):

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/neteq/expand.cc:553: // |distortion_lag| and
|correlation_lag| are equal.
We probably don't need this comment anymore. :-)

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/audio_codi...
File webrtc/modules/audio_coding/neteq/merge.cc (right):

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/neteq/merge.cc:345: start_index = (input_length >
start_index) ? 0 : (start_index - input_length);
Not for now, perhaps, but it strikes me that you could have used a saturating
unsigned subtract in quite a few places.

[Peter Kasting, 2015-06-11 04:14:53]

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/audio_codi...
File webrtc/modules/audio_coding/codecs/ilbc/encode.c (right):

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/codecs/ilbc/encode.c:431: start_count = (Nfor > 1) ?
0 : (2 - NFor);
On 2015/06/10 07:49:10, kwiberg1 wrote:
> Nit: it'd be slightly easier to read if you said Nfor >= 2

Done (for both).

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/audio_codi...
File webrtc/modules/audio_coding/neteq/expand.cc (right):

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/neteq/expand.cc:553: // |distortion_lag| and
|correlation_lag| are equal.
On 2015/06/10 07:49:10, kwiberg1 wrote:
> We probably don't need this comment anymore. :-)

Heh.  Removed.

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/audio_codi...
File webrtc/modules/audio_coding/neteq/merge.cc (right):

https://codereview.webrtc.org/1174813003/diff/20001/webrtc/modules/audio_codi...
webrtc/modules/audio_coding/neteq/merge.cc:345: start_index = (input_length >
start_index) ? 0 : (start_index - input_length);
On 2015/06/10 07:49:10, kwiberg1 wrote:
> Not for now, perhaps, but it strikes me that you could have used a saturating
> unsigned subtract in quite a few places.

Yeah.  There isn't one available right now but it'd be easy to write:

  size_t SaturatedSubtractSizeT(size_t a, size_t b) {
    return (b > a) ? 0 : a - b;
  }

If you want me to do this, and tell me the appropriate place it should go, I can
add it and make use of it when I switch to size_t.

I'm going to go ahead and land this for now but feel free to follow up and I
will implement whatever you suggest.

[Peter Kasting, 2015-06-11 04:15:59]

Committed patchset #3 (id:40001) manually as
f045e4da43e671ae511aa1d9b6ef2968256a745d (presubmit successful).

[commit-bot: I haz the power, 2015-06-11 04:16:03]

Patchset 3 (id:??) landed as
https://crrev.com/f045e4da43e671ae511aa1d9b6ef2968256a745d
Cr-Commit-Position: refs/heads/master@{#9413}