| 6955870806624479723addfae6dcf5d13968796c |
|
13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| e1ca167217cdce7c6cb7f675fd466cae1e3e6e69 |
|
08-Jan-2016 |
henrik.lundin <henrik.lundin@webrtc.org> |
Add tracing to NetEqImpl::GetAudio
BUG=webrtc:5167
R=pbos@webrtc.org
NOTRY=true
Review URL: https://codereview.webrtc.org/1571693002
Cr-Commit-Position: refs/heads/master@{#11183}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| a689b44c17dd22635188da325352414f4a5ab6b7 |
|
17-Dec-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
Add tracing to NetEqImpl::InsertPacket
BUG=webrtc:5167
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1525423004
Cr-Commit-Position: refs/heads/master@{#11065}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 4cf61dd116288e9f119209c59e07f1d9439d8d05 |
|
09-Dec-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
NetEq: Add codec name and RTP timestamp rate to DecoderInfo
The new fields are default-populated for built-in decoders, but for
external decoders, the name can now be given when registering the
decoder.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1484343003
Cr-Commit-Position: refs/heads/master@{#10952}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4 |
|
23-Nov-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
NetEq: Add new method last_output_sample_rate_hz
This change moves the logics for keeping track of the last ouput
sample rate from AcmReceiver to NetEq, where it fits better. The
getter function AcmReceiver::current_sample_rate_hz() is renamed to
last_output_sample_rate_hz().
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1467163002
Cr-Commit-Position: refs/heads/master@{#10754}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 672304a654f6e774e90139f5a20add64a462f7c3 |
|
20-Nov-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
NetEq: Remove overly verbose logging
This change removes all LS_VERBOSE logs that will print once every
packet or more often.
TBR=pbos@webrtc.org
BUG=webrtc:5227
Review URL: https://codereview.webrtc.org/1461903004
Cr-Commit-Position: refs/heads/master@{#10733}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| ee2bac26dd3eb4463126098f87701ff66098b288 |
|
11-Nov-2015 |
kwiberg <kwiberg@webrtc.org> |
AcmReceiver::InsertPacket and NetEq::InsertPacket: Take ArrayView arguments
Instead of separate pointer and size arguments.
Review URL: https://codereview.webrtc.org/1429943004
Cr-Commit-Position: refs/heads/master@{#10606}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
|
04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 9bc2667fa6deee5d4162b13a878481640a58cce5 |
|
02-Nov-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
ACM/NetEq: Restructure how post-decode VAD is enabled
This change avoids calling neteq_->EnableVad() and DisableVad from the
AcmReceiver constructor. Instead, the new member
enable_post_decode_vad is added to NetEq's config struct. It is
disabled by defualt, but ACM sets it to enabled. This preserves the
behavior both of NetEq stand-alone (i.e., in tests) and of ACM.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1425133002
Cr-Commit-Position: refs/heads/master@{#10476}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| ee1879ca40ffe4af9bb9613e03eacc5c2c4881fc |
|
29-Oct-2015 |
kwiberg <kwiberg@webrtc.org> |
Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
This operation was relatively simple, since no one was doing anything
fishy with this enum. A large number of lines had to be changed
because the enum values now live in their own namespace, but this is
arguably worth it since it is now much clearer what sort of constant
they are.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1424083002
Cr-Commit-Position: refs/heads/master@{#10449}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 48ed930975ef7e84023044ed584c4eff914e6c9a |
|
29-Oct-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
ACM: Move NACK functionality inside NetEq
Negative acknowledgement (NACK) has up to now been implemented in
ACM. But, since NetEq is in charge of the actual packet buffer, it
makes more sense to have the NACK functionlaity in there.
This CL does the following:
- Move nack.{h,cc} and the unit tests from main/acm2 to neteq.
- Move the NACK related code in ACM into NetEq.
- NACK related functions in AcmReceiver are changed to simple
forwarding APIs.
- Remove unused members in AcmReceiver.
- Remove unused API functions in NetEq.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1410073006
Cr-Commit-Position: refs/heads/master@{#10448}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 6d92bf59f3f8c0ce8ad445c11aaaf955eae752cc |
|
23-Sep-2015 |
minyuel <minyue@webrtc.org> |
Returning correct duration estimate on Opus DTX packets.
Bug 4985 revealed two flaws
1. Opus duration estimate did not return correct length for DTX packets,
2. NetEq DoCodecInternalCng did not assign enough buffer.
P.S.
Generalizing problem 1, current NetEq decode function checks memory size by calling the duration estimate function. This is not ideal. A better way is to let codec's decode function to receive buffer size and return failure if it is not enough. This can be made in a separate CL.
BUG=webrtc:4985
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1334303005 .
Cr-Commit-Position: refs/heads/master@{#10031}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 05f71fcb61df680ab0e0dab06ed6578ce062fa21 |
|
01-Sep-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
NetEq: Fixing a corner case with depleted sync buffer
In some cases, the number of samples (per channel) in NetEq's sync
buffer could fall below the allowed minimum (5 samples for narrowband,
scaling for other rates). If the number of samples extracted from the
buffer was smaller than the desired number, an error is
returned. However, if the decoder returns fewer samples than expected,
it could happen that the sync buffer level falls under the minimum,
but enough samples are extracted. This triggered an assert. With this
change, the minimum level of the sync buffer is always enforced.
A test is implemented to trigger the problem. It made the assert fire
without this fix, but it now passes.
BUG=webrtc:4840
R=minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1324453002 .
Cr-Commit-Position: refs/heads/master@{#9828}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 98f3cc54daf0385e8b76875aa4ea0cb16987bfc5 |
|
28-Aug-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
NetEq: Removing two asserts
These asserts cover error cases that are also handled by the code
after the assert. Should not have both assert and error handling.
BUG=webrtc:4840
Review URL: https://codereview.webrtc.org/1321023002
Cr-Commit-Position: refs/heads/master@{#9804}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 116c84e1b0b7d584a0f04f11de9b2bdb71b25040 |
|
27-Aug-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
NetEq: Fixing a bug that caused rtc::checked_cast to trigger
This is a bug that was introduced in
https://codereview.webrtc.org/1230503003, where the variable "int
temp_bufsize" was changed to a size_t. If the packet buffer was
flushed while inserting a packet, temp_bufsize became negative, which
was tested later in an if-statement. Now, with size_t instead, it
would just become very large, and the if-statement would never see a
negative value. The effect was that the packet size in samples could
be updated with a very large positive number, causing an overflow
which triggered rtc::checked_cast in
StatisticsCalculator::GetNetworkStatistics, line 220.
Also adding a test to reproduce the crash. Without the fix, the test
results in the above mentioned checked_cast to trigger. With the fix,
everything works fine.
BUG=chromium:525260
Review URL: https://codereview.webrtc.org/1307893004
Cr-Commit-Position: refs/heads/master@{#9802}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 9c3efd00523a81d0f2b582799fbe67afe44139b2 |
|
27-Aug-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
Reland: Implement NetEq's CurrentDelay function
This was not implemented before. It returns the current total delay
(packet buffer and sync buffer) of NetEq. This is the same information
that was already available in
NetEqNetworkStatistics::current_buffer_size_ms, that can be obtained
through NetEq::NetworkStatistics(). But, since the current delay is a
key metric of NetEq, it is convenient to have it available in a
simpler way.
This is a re-landing of r9359,
https://webrtc-codereview.appspot.com/51149004, which was reverted in
r9360. The refactoring made in r9669 facilitated the relanding.
TBR=minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1313873003
Cr-Commit-Position: refs/heads/master@{#9801}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 4376648df021fd82f25a38694e33678f802d06ea |
|
27-Aug-2015 |
Karl Wiberg <kwiberg@google.com> |
AudioDecoder: Replace Init() with Reset()
The Init() method was previously used to initialize and reset
decoders, and returned an error code. The new Reset() method is used
for reset only; the constructor is now responsible for fully
initializing the AudioDecoder.
Reset() doesn't return an error code; it turned out that none of the
functions it ended up calling could actually fail, so this CL removes
their error return codes as well.
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1319683002 .
Cr-Commit-Position: refs/heads/master@{#9798}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 1bb8cf846d9a0bfe74fceae34ebef60f56d12fa4 |
|
25-Aug-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
NetEq/ACM: Refactor how packet waiting times are calculated
With this change, the aggregates for packet waiting times are
calculated in NetEq's StatisticsCalculator insead of in
AcmReceiver. This simplifies things somewhat, and avoids having to
copy the raw data on polling.
R=ivoc@webrtc.org, minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1296633002 .
Cr-Commit-Position: refs/heads/master@{#9778}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| dce40cf804019a9898b6ab8d8262466b697c56e0 |
|
24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| bef77e234fa53a52b830b5833948711f75ab8bbb |
|
18-Aug-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
NetEq: Implement logging of Delayed Packet Outage Events
Measures the duration of each packet loss concealment (a.k.a. expand)
event that is not followed by a merge operation.
Having decoded and played packet m−1, the next expected packet is
m. If packet m arrives after some time of packet loss concealment, we
have a delayed packet outage event. However, if instead packet n>m
arrives, we have a lost packet outage event. In NetEq, the two outage
types results in different operations. Both types start with expand
operations to generate audio to play while the buffer is empty. When a
lost packet outage happens, the expand operation(s) are followed by
one merge operation. For delayed packet outages, merge is not done,
and the expand operations are immediately followed by normal
operations.
This change also includes unit tests for the new statistics.
BUG=webrtc:4915, chromium:488124
R=minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1290113002 .
Cr-Commit-Position: refs/heads/master@{#9725}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| d67a219bec0c4cde149014984d5dfe168fe0a346 |
|
03-Aug-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Switch to base/logging.h in neteq_impl.cc
This change includes base/logging.h instead of the old and deprecated
system_wrappers/interface/logging.h. This requires some changes of the
actual logging invocations.
For reference the following regexps where used (in Eclipse) for a few
of the replacements:
find: LOG_FERR1\(\s*([^,]*),\s*([^,]*),\s*(.*)\);
replace: LOG($1) << "$2 " << $3;
find: LOG_FERR2\(\s*([^,]*),\s*([^,]*),\s*([^,]*),\s*(.*)\);
replace: LOG($1) << "$2 " << $3 << " " << $4;
BUG=4735
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50229004 .
Cr-Commit-Position: refs/heads/master@{#9669}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 36b7cc32643bae0379d8102ce05dae82ecc466a1 |
|
12-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reland "Upconvert various types to int.", neteq portion.
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which
reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. Specifically, the
files in webrtc/modules/audio_coding/neteq/ 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/1181073002
Cr-Commit-Position: refs/heads/master@{#9427}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 728d9037c016c01295177fa700fc7927f0bb80bb |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| b7e5054414ff524f9db81dab7917729b8c4c8bcb |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Match existing type usage better.
This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones. For example:
* Change a few type declarations to better match how the majority of code uses those objects.
* Eliminate "< 0" check for unsigned values.
* Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar.
* Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects.
* Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t.
* Similarly, add casts when passing a larger type to a function taking a smaller one.
* Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar.
* Use "false" instead of "0" for setting a bool.
* Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps. For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t.
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=andrew, asapersson, henrika
Review URL: https://codereview.webrtc.org/1168753002
Cr-Commit-Position: refs/heads/master@{#9419}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| cb180976dd0e9672cde4523d87b5f4857478b5e9 |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| f045e4da43e671ae511aa1d9b6ef2968256a745d |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
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
Review URL: https://codereview.webrtc.org/1174813003
Cr-Commit-Position: refs/heads/master@{#9413}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 83ad33a8aed1fb00e422b6abd33c3e8942821c24 |
|
10-Jun-2015 |
Peter Kasting <pkasting@google.com> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 5abd3e1f986c627a852bf823d15feaa5f619a559 |
|
03-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Revert r9359 "Implement NetEq's CurrentDelay function"
This reverts commit d8a03facf6986a011c8f889c63d87f9216a1e912, since it
broke the Chrome build. Will have to swap to using base/logging.h in
neteq_impl.cc before re-landing this change.
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50219004
Cr-Commit-Position: refs/heads/master@{#9360}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| d8a03facf6986a011c8f889c63d87f9216a1e912 |
|
03-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Implement NetEq's CurrentDelay function
This was not implemented before. It returns the current total delay (packet buffer and sync buffer) of NetEq. This is the same information that was already available in NetEqNetworkStatistics::current_buffer_size_ms, that can be obtained through NetEq::NetworkStatistics(). But, since the current delay is a key metric of NetEq, it is convenient to have it available in a simpler way.
R=kwiberg@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51149004
Cr-Commit-Position: refs/heads/master@{#9359}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| cf808d2366e58b33540931d182f36800d9a15b0d |
|
27-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Add new fast mode for NetEq's Accelerate operation
This change instroduces a mode where the Accelerate operation will be
more aggressive. When enabled, it will allow acceleration at lower
correlation levels, and possibly remove multiple pitch periods at
once.
The feature is enabled through NetEq::Config, and is off by
default. This means that bit-exactness tests are currently not
affected.
A unit test was added for the Accelerate class, with and without fast
mode enabled.
BUG=4691
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50039004
Cr-Commit-Position: refs/heads/master@{#9295}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 905495cfaaed396c816f0a62e31e0311ce86e1b0 |
|
25-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Introduce NetEq::Config::ToString and use it in NetEq's constructor
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54559004
Cr-Commit-Position: refs/heads/master@{#9279}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| d8399e630f3f4886d455e2c4d2307794b60261c0 |
|
25-May-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
Also provide sample rate when registering decoders
This replaces the old practice of looking up the sample rate in a
table, which won't work when we add support for external decoders.
COAUTHOR=henrik.lundin@webrtc.org
BUG=4474
R=jmarusic@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54469004
Cr-Commit-Position: refs/heads/master@{#9276}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 7f6c4d42a2605d1da39af3f957a46cf57b043b84 |
|
09-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
Fix clang style warnings in webrtc/modules/audio_coding/neteq
Mostly this consists of marking functions with override when
applicable, and moving function bodies from .h to .cc files.
BUG=163
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44109004
Cr-Commit-Position: refs/heads/master@{#8960}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 6dba1ebd14d8cd96e6e56adad868b33fdedecc53 |
|
18-Mar-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Make AudioDecoder stateless
The channels_ member varable is removed from the base class, and the
associated accessor function is changed to Channels() which is a pure
virtual function.
R=jmarusic@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43779004
Cr-Commit-Position: refs/heads/master@{#8775}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8775 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 7f7d7e3427cc70e1b8b050283ef031e28c83699a |
|
16-Mar-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 1eda4e3db60f484d179cee359e150c4f0c9c7c67 |
|
25-Feb-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call"
This should be safe to land now that issue 4143 was resolved (in r8492).
This change effectively reverts 8488.
TBR=kwiberg@webrtc.org
Original commit message:
This CL changes the way the decoder sample rate is set and updated. In
practice, it only concerns the iSAC (float) codec.
One single iSAC decoder instance is used for both wideband and
super-wideband decoding, and the instance must be told to switch
output frequency if the payload type changes. This used to be done
through a call to UpdateDecoderSampleRate, but is now instead done in
the Decode call as an extra parameter.
Review URL: https://webrtc-codereview.appspot.com/39289004
Cr-Commit-Position: refs/heads/master@{#8496}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8496 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 903182bd8e782b162900b99bc7e25c35edebdb67 |
|
24-Feb-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Revert r8476 "Set decoder output frequency in AudioDecoder::Decode call"
This change uncovered issue 4143, evading the Memcheck suppression
since the signature is changed in the Decode function.
A fix for this is in the making; see
https://review.webrtc.org/36309004. This CL will be re-landed once the
fix is in place.
BUG=4143
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42089004
Cr-Commit-Position: refs/heads/master@{#8488}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8488 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| b9c18d56438eefb71ff68d47880d2b49fd380bc7 |
|
24-Feb-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Set decoder output frequency in AudioDecoder::Decode call
This CL changes the way the decoder sample rate is set and updated. In
practice, it only concerns the iSAC (float) codec.
One single iSAC decoder instance is used for both wideband and
super-wideband decoding, and the instance must be told to switch
output frequency if the payload type changes. This used to be done
through a call to UpdateDecoderSampleRate, but is now instead done in
the Decode call as an extra parameter.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34349004
Cr-Commit-Position: refs/heads/master@{#8476}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8476 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| d324546ced76d4e792338af4f7d02a5cd8819f92 |
|
23-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes
BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36179004
Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 2c1bcf2cb4a9e19a337e52fd576242e04168d5e9 |
|
17-Feb-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Adding decoded_fec_rate to NetEQ Network Statistics.
A statistic is introduced to reflect the actual benefits of Opus FEC. It shows what percentage of samples in the rendered audio come from FEC data.
BUG=3867
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34969004
Cr-Commit-Position: refs/heads/master@{#8384}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8384 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| c11348b5d7163a040d777d1cfedafde758d461e0 |
|
10-Feb-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Fixing a bug in expand_rate calculation for stereo signal.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41849004
Cr-Commit-Position: refs/heads/master@{#8307}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8307 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 0e81fdf5d2c2665bc3d23e07cfd9ea7f7d36aed9 |
|
03-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting.
BUG=chromium:82439
TEST=none
R=henrik.lundin@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40569004
Cr-Commit-Position: refs/heads/master@{#8229}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8229 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 026b892e724c3f47bde92d773d84099768e57ec8 |
|
30-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Using << on an int8_t or uint8_t will output a character rather than a number.
Places that do this need to cast to int to get the desired behavior.
BUG=none
TEST=none
R=henrik.lundin@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40579004
Cr-Commit-Position: refs/heads/master@{#8223}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8223 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 7d2b6a9346a4ecc944857d6b37e40979594650cf |
|
28-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Enable Clang warning implicit-fallthrough and annotate the code.
BUG=4242
R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34899004
Cr-Commit-Position: refs/heads/master@{#8187}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8187 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| e04a93bcf5e1b608c798a6a3148224b8035f0119 |
|
09-Dec-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Move the AudioDecoder interface out of NetEq
It belongs with the codecs, next to the AudioEncoder interface.
R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org
Previously committed here: https://code.google.com/p/webrtc/source/detail?r=7798
and reverted here: https://code.google.com/p/webrtc/source/detail?r=7799
Review URL: https://webrtc-codereview.appspot.com/27309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7839 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 3800e13a3a7031220e2d21990858d4d08581e393 |
|
03-Dec-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Revert r7798 ("Move the AudioDecoder interface out of NetEq")
Apparently, it caused all sorts of problems I don't have time to
straighten out right now.
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 00ba1a7dfd66e096ee5fb5e4e084c5565738426f |
|
03-Dec-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Move the AudioDecoder interface out of NetEq
It belongs with the codecs, next to the AudioEncoder interface.
R=henrik.lundin@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7798 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 4591fbd09f9cb6e83433c49a12dd8524c2806502 |
|
20-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 6ff3ac1db8b357e50cd6631e3f7cb0caf2aff986 |
|
20-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Fix problems if first packet into NetEq is rejected
This CL fixes the problem described in issue 4021. In summary, of the
very first packet coming in to NetEq gets rejected because the RTP
payload type is unknown, subsequent GetAudio calls would trigger asserts
(in debug builds). The problem was that the first_packet_ variable was
reset and new_codec_ was set, even though the packet was discarded
further down the line. Now, these variables are modified after the
packet has been verified.
BUG=4021
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7724 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 40af3a56e409f8b9fc231907a20514cba7e32f27 |
|
19-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Revert "Add DCHECK to ensure that NetEq's packet buffer is not empty"
This reverts r7719. It failed to compile in Chromium.
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7720 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 6f6ef72950b9bda79392e83d7b1495d4ff07b4a2 |
|
19-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Add DCHECK to ensure that NetEq's packet buffer is not empty
This DCHECK ensures that one packet was inserted after the buffer was
flushed.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7719 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 52b42cb069a035f10e951195c28cf6d05d1fd91c |
|
04-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Fix problem with late packets in NetEq
Since r7255, it could happen that an old packet would block the decoding
process until enough packet was received for the buffer to flush. This
CL fixes that by:
- Partially reverting r7255;
- Remove recent old packets before taking a decision for GetAudio;
- Remove all old packets after a packet has been extracted for decoding;
- Adding tests for reordered packets.
BUG=chrome:423985
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7612 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| b0f4b3da055cb09813d52f417f64ce2275887fea |
|
04-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Improving error message from neteq_rtpplay
If a packet with unknown RTP payload type is inserted, this CL
will make sure that the error message is a little more detailed
and gives a better understadning of what to do.
BUG=2692
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7603 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 7cbc4f969aa1f145b1538c0f0144ad3cc81b69e3 |
|
07-Oct-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Set NetEq playout mode through the Config struct
This change opens up the possibility to set the playout mode when
creating the NetEq object. The old methods SetPlayoutMode and
PlayoutMode are still available, but are deprecated.
BUG=3520
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7381 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| 612171527e87e903816b5e169a97a0858a39b50e |
|
22-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Ensure that NetEq recovers after a large timestamp jump
Before this change it could happen that a large jump in timestamp (a
jump not correlated to wall-clock change) caused the audio to go silent
without recovering. The reason was that all incoming packets after the
jump were considered too old compared to the last decoded packet, and
were deleted. With CL changes two things:
1. If the only available packet in the buffer is an old packet, NetEq
will do Expand instead of immediate reset. This is to avoid that one
late packet triggers a reset.
2. Old packets are discarded only when the decision to decode a packet
has been taken. This is to allow the buffer to grow and eventually
flush if no decodable packet has been found for some time.
This CL also includes a new unit test for this situation.
BUG=3785
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7255 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|
| ea25784107b9202300a57d838d2c56e158220eef |
|
07-Aug-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Change how background noise mode in NetEq is set
This change prepares for switching default background noise (bgn) mode
from on to off. The actual switch will be done later.
In this change, the bgn mode is included as a setting in NetEq's config
struct. We're also removing the connection between playout modes and
bgn modes in ACM. In practice this means that bgn mode will change from
off to on for streaming mode, but since the playout modes are not used
it does not matter.
BUG=3519
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6843 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
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| 9c55f0f957534144d2b8a64154f0a479249b34be |
|
09-Jun-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename neteq4 folder to neteq
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.
This CL effectively reverts r6257 "Rename neteq4 folder to neteq".
BUG=2996
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
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| 1b9df05c8521d1d807b08d7c00eb2f7e5b097fdf |
|
28-May-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6257 "Rename neteq4 folder to neteq"
> Rename neteq4 folder to neteq
>
> BUG=2996
> R=turaj@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/12569005
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6259 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
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| a90f6d67f72359cf63b59480fa87a13aae808c03 |
|
28-May-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename neteq4 folder to neteq
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12569005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6257 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.cc
|