304 Commits

Author SHA1 Message Date
kwiberg
39d8bee397 Make ACMCodecDB private to RentACodec
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1414203010

Cr-Commit-Position: refs/heads/master@{#10549}
2015-11-07 00:22:50 +00:00
kwiberg
d6c0f8cac1 Remove ACMCodecDB::Codec, and make the rest of ACMCodecDB private
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1423043005

Cr-Commit-Position: refs/heads/master@{#10546}
2015-11-06 22:28:08 +00:00
kwiberg
fb3d8b3df2 Remove ACMCodecDB::CodecFreq
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1408773005

Cr-Commit-Position: refs/heads/master@{#10536}
2015-11-06 09:24:16 +00:00
kwiberg
288886b2ec Pass audio to AudioEncoder::Encode() in an ArrayView
Instead of in separate pointer and size arguments.

Review URL: https://codereview.webrtc.org/1418423010

Cr-Commit-Position: refs/heads/master@{#10535}
2015-11-06 09:21:39 +00:00
kwiberg
93a2febe73 Move ACMCodecDB::ValidPayloadType to RentACodec
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1430043003

Cr-Commit-Position: refs/heads/master@{#10525}
2015-11-05 15:39:42 +00:00
kwiberg
98cc88c873 Correctly handle the error case where the CodecId has a negative value
Negative values should be treated the same as too-large positive values: by returning an empty Maybe.

TBR=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1407383010

Cr-Commit-Position: refs/heads/master@{#10509}
2015-11-04 17:56:22 +00:00
Henrik Kjellander
ff761fba82 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}
2015-11-04 07:32:04 +00:00
kwiberg
4b938e5c69 Hide ACMCodecDB::database_ behind accessors
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1425233003

Cr-Commit-Position: refs/heads/master@{#10498}
2015-11-03 20:38:31 +00:00
kwiberg
1fd4a4ab35 Let AudioCodingModule::SendCodec return Maybe<CodecInst>
And deal with the consequences thereof...

Review URL: https://codereview.webrtc.org/1406123011

Cr-Commit-Position: refs/heads/master@{#10497}
2015-11-03 19:20:57 +00:00
kwiberg
de94d08d57 Hide ACMCodecDB::codec_settings_ behind an accessor
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1419013005

Cr-Commit-Position: refs/heads/master@{#10494}
2015-11-03 13:46:14 +00:00
henrik.lundin
678c903eb2 Delete AcmReceiver::SetInitialDelay
This method is no longer called. With that gone, a number of other
methods and member variables are obsoleted, and removed.

Methods deleted:
AcmReceiver::InsertStreamOfSyncPackets
AcmReceiver::GetNumSyncPacketToInsert()
AcmReceiver::GetSilence, never called

Member variables deleted:
missing_packets_sync_stream_
late_packets_sync_stream_
av_sync_
initial_delay_manager_

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1419573013

Cr-Commit-Position: refs/heads/master@{#10484}
2015-11-02 16:31:30 +00:00
henrik.lundin
1d5c9bd800 Remove unused method AcmReceiver:RedPayloadType
BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1415313007

Cr-Commit-Position: refs/heads/master@{#10481}
2015-11-02 12:46:35 +00:00
henrik.lundin
9bc2667fa6 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}
2015-11-02 11:26:03 +00:00
henrik.lundin
74f0f3551e Delete a chain of methods in ViE, VoE and ACM
The end goal is to remove AcmReceiver::SetInitialDelay. This change is
in preparation for that goal. It turns out that
AcmReceiver::SetInitialDelay was only invoked through the following
call chain, where each method in the chain is never referenced from
anywhere else (except from tests in some cases):

ViEChannel::SetReceiverBufferingMode
-> ViESyncModule::SetTargetBufferingDelay
-> VoEVideoSync::SetInitialPlayoutDelay
-> Channel::SetInitialPlayoutDelay
-> AudioCodingModule::SetInitialPlayoutDelay
-> AcmReceiver::SetInitialDelay

The start of the chain, ViEChannel::SetReceiverBufferingMode was never
referenced.

This change deletes all the methods above except
AcmReceiver::SetInitialDelay itself, which will be handled in a
follow-up change.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1421013006

Cr-Commit-Position: refs/heads/master@{#10471}
2015-11-01 19:43:38 +00:00
kwiberg
102c6a61bc Replace rtc:🦗:Settable with rtc::Maybe
The former is very similar to the latter, but less general (mostly in
naming).

This CL, which is the first to use Maybe at scale, also removes the implicit conversion from T to Maybe<T>, since it was agreed that the increased verbosity increased legibility.

Review URL: https://codereview.webrtc.org/1430433004

Cr-Commit-Position: refs/heads/master@{#10461}
2015-10-30 09:47:44 +00:00
kwiberg
ee1879ca40 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}
2015-10-29 13:20:33 +00:00
henrik.lundin
48ed930975 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}
2015-10-29 12:36:32 +00:00
Henrik Kjellander
74640895fa audio_coding: rename interface -> include
BUG=webrtc:5095
R=henrik.lundin@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417173004 .

Cr-Commit-Position: refs/heads/master@{#10444}
2015-10-29 10:31:11 +00:00
Henrik Kjellander
98f53510b2 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}
2015-10-28 17:17:50 +00:00
kwiberg
fce4a945b8 RentACodec: New class that takes over part of ACMCodecDB's job
Following CLs will finish the takeover completely. After that,
RentACodec will also start creating and owning codecs, at which point
its name will start making sense.

BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1412683006

Cr-Commit-Position: refs/heads/master@{#10432}
2015-10-27 18:40:29 +00:00
henrik.lundin
e9eca8f5ae Removing AudioCoding class, a.k.a the new ACM API
We have decided not to do a switch from old (AudioCodingModule) to new
(AudioCoding) API. Instead, we will gradually evolve the old API to
meet the new design goals.

As a consequence of this decision, the AudioCoding and AudioCodingImpl
classes are deleted. Also removing associated unit test sources. No
test coverage is lost with this operation, since the tests for the
"old" API are testing more than the deleted tests did.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1415163002

Cr-Commit-Position: refs/heads/master@{#10406}
2015-10-26 12:26:45 +00:00
pbos
22993e1a0c Unify FrameType and VideoFrameType.
Prevents some heap allocation and frame-type conversion since interfaces
mismatch. Also it's less confusing to have one type for this.

BUG=webrtc:5042
R=magjed@webrtc.org, mflodman@webrtc.org, henrik.lundin@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1371043003

Cr-Commit-Position: refs/heads/master@{#10320}
2015-10-19 09:39:15 +00:00
henrik.lundin
9ea2147e5c Delete iSAC-fb from AudioCodingModule
This is no longer used. Related code in NetEq and the iSAC codec itself
will be deleted in follow-up CLs.

BUG=4210

Review URL: https://codereview.webrtc.org/1404623002

Cr-Commit-Position: refs/heads/master@{#10264}
2015-10-13 13:28:04 +00:00
kwiberg
98ab3a46d6 Don't link with audio codecs that we don't use
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.

This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means,
likely just the linker omitting compilation units with no incoming
references.

(This was previously landed as revisions 10046 and 10060, and got
reverted because it broke several of the Chromium FYI bots.)

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1368843003

Cr-Commit-Position: refs/heads/master@{#10127}
2015-10-01 04:54:29 +00:00
Henrik Lundin
6ba8e4a4f2 ACM: Remove a few local enums that were no longer used
BUG=webrtc:3520
R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1375863002 .

Cr-Commit-Position: refs/heads/master@{#10114}
2015-09-30 08:59:36 +00:00
henrik.lundin
1bd0e03ce5 ACM: Removing runtime APIs related to playout mode
The playout mode in NetEq can still be set through the constructor
configuration.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1362943004

Cr-Commit-Position: refs/heads/master@{#10089}
2015-09-28 13:12:21 +00:00
solenberg
3fd7be4cb1 Revert of Don't link with audio codecs that we don't use (patchset #4 id:60001 of https://codereview.webrtc.org/1349393003/ )
Reason for revert:
Breaking Chromium FYI bots.

Original issue's description:
> Don't link with audio codecs that we don't use
>
> We used to link with all audio codecs unconditionally (except Opus);
> this patch makes gyp and gn only link to the ones that are used.
>
> (This unfortunately fails to have a measurable impact on Chromium
> binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
> fix were already being excluded from Chromium by some other means
> (likely just the linker omitting compilation units with no incoming
> references).)
>
> BUG=webrtc:4557
>
> Committed: https://crrev.com/f66a9251424351ea6d631c54dd1feb64cc13d809
> Cr-Commit-Position: refs/heads/master@{#10046}

TBR=henrik.lundin@webrtc.org,tina.legrand@webrtc.org,kjellander@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1368933002

Cr-Commit-Position: refs/heads/master@{#10069}
2015-09-25 08:36:11 +00:00
solenberg
a53e383d7d Revert of CodecOwner: Don't look at definitions for classes we don't link with (patchset #1 id:1 of https://codereview.webrtc.org/1364233002/ )
Reason for revert:
Breaking Chromium FYI bots.

Original issue's description:
> CodecOwner: Don't look at definitions for classes we don't link with
>
> It's good hygiene and just generally the right thing to do. And
> apparently at least sometimes required by Microsoft's compiler.
>
> Committed: https://crrev.com/f4d38ea57aa739b525066b095468cb4af1d2799b
> Cr-Commit-Position: refs/heads/master@{#10060}

TBR=henrik.lundin@webrtc.org,tommi@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1368083002

Cr-Commit-Position: refs/heads/master@{#10068}
2015-09-25 08:31:01 +00:00
kwiberg
574d5daa6d CodecOwner::SetEncoders: Return error code when given bad arguments
Instead of FATAL on a bad codec specification, log and return an error
code. This is a band-aid until callers are taught to only give it good
specifications.

BUG=webrtc:5033, chromium:526478

Review URL: https://codereview.webrtc.org/1364193002

Cr-Commit-Position: refs/heads/master@{#10066}
2015-09-25 05:54:00 +00:00
kwiberg
f4d38ea57a CodecOwner: Don't look at definitions for classes we don't link with
It's good hygiene and just generally the right thing to do. And
apparently at least sometimes required by Microsoft's compiler.

Review URL: https://codereview.webrtc.org/1364233002

Cr-Commit-Position: refs/heads/master@{#10060}
2015-09-24 17:21:02 +00:00
kwiberg
ec249d4eae ACMCodecDB: Remove unused stuff, and move private stuff to anonymous namespace
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1360123002

Cr-Commit-Position: refs/heads/master@{#10048}
2015-09-24 11:32:11 +00:00
kwiberg
f66a925142 Don't link with audio codecs that we don't use
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.

(This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means
(likely just the linker omitting compilation units with no incoming
references).)

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1349393003

Cr-Commit-Position: refs/heads/master@{#10046}
2015-09-24 10:18:48 +00:00
kwiberg
8967183bf7 Simple cleanups of AudioDecoder and AudioEncoder classes
* Make sure they're all final and don't allow copying or assignment.

  * Get rid of the single-channel PCM decoder classes.

  * Move some includes from .h to .cc files where possible.

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1353803002

Cr-Commit-Position: refs/heads/master@{#10021}
2015-09-22 21:06:34 +00:00
Karl Wiberg
7404368998 Move AudioDecoderIsac* to its own files
Currently, it's sitting in AudioEncoderIsac*'s files, which is less
than obvious. This CL puts the encoder and decoder in separate files
together with the C implementation; CLs are afoot to make it so for
the other built-in codecs as well.

BUG=webrtc:4557
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1339253003 .

Cr-Commit-Position: refs/heads/master@{#10018}
2015-09-22 17:31:52 +00:00
kwiberg
6faf5bebba Move AudioDecoderPcm* next to AudioEncoderPcm*
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1348613003

Cr-Commit-Position: refs/heads/master@{#10015}
2015-09-22 13:16:56 +00:00
henrik.lundin
061b79af60 ACM: Remove functions related to DTMF
The functions were essentially no-op. Also removing forward declaration
of ACMDTMFDetection, which was not used.

BUG=3520

Review URL: https://codereview.webrtc.org/1356543003

Cr-Commit-Position: refs/heads/master@{#9982}
2015-09-18 08:29:17 +00:00
Ivo Creusen
ae856f2c9f Added support for logging the SSRC corresponding to AudioPlayout events.
To do this, the logging of this event was moved from the ACM to
VoiceEngine Channel. A new LogAudioPlayoutEvent function was added on
the RtcEventLog interface, and the LogDebugEvent function was removed
since it is no longer being used.

BUG=webrtc:4741
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, kwiberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/1340283002 .

Cr-Commit-Position: refs/heads/master@{#9972}
2015-09-17 14:34:15 +00:00
henrikg
91d6edef35 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}
2015-09-17 07:24:51 +00:00
kwiberg
1f9baab753 Remove the preprocessor symbol WEBRTC_CODEC_AVT (it was always defined)
BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1338283002

Cr-Commit-Position: refs/heads/master@{#9960}
2015-09-17 02:29:51 +00:00
kwiberg
844a91081e Remove the preprocessor symbol WEBRTC_CODEC_PCM16 (it was always defined)
BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1336923002

Cr-Commit-Position: refs/heads/master@{#9955}
2015-09-16 16:42:26 +00:00
henrikg
3c089d751e Add RTC_ prefix to contructormagic 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.

* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS

Related CL: https://codereview.webrtc.org/1335923002/

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1345433002

Cr-Commit-Position: refs/heads/master@{#9953}
2015-09-16 12:37:52 +00:00
kwiberg
39720f2669 ACM CodecOwner: Test that we reset speech encoder when enabling CNG or RED
If we don't, we'll end up crashing if they're enabled when the speech
encoder is in the middle of encoding a packet, since CNG and RED
assume that the speech encoder starts out with an empty buffer
(because they need to be in sync with it).

BUG=chromium:490368

Review URL: https://codereview.webrtc.org/1331853002

Cr-Commit-Position: refs/heads/master@{#9917}
2015-09-10 12:44:52 +00:00
kwiberg
9b66d3ba60 MockAudioEncoder: Use a dedicated marker method for test expectations
This makes the sequence of expected calls easier to read. Also, we can
save one line and get rid of a gmock warning by expecting the
MockAudioEncoder object to be destroyed at the end of the test instead
of making a final marker call.

Review URL: https://codereview.webrtc.org/1331793003

Cr-Commit-Position: refs/heads/master@{#9916}
2015-09-10 12:09:49 +00:00
kwiberg
c99ebc1490 Remove AudioEncoder methods SetMaxBitrate and SetMaxPayloadSize
And the corresponding ACM methods SetISACMaxRate and
SetISACMaxPayloadSize. They were only used in tests.

Review URL: https://codereview.webrtc.org/1311533010

Cr-Commit-Position: refs/heads/master@{#9903}
2015-09-09 07:54:10 +00:00
ivoc
b04965ccf8 Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call.
An option was added to voe_cmd_test to make a RtcEventLog dump.

BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1267683002

Cr-Commit-Position: refs/heads/master@{#9901}
2015-09-09 07:09:49 +00:00
kwiberg
3f5f1c2ad3 Change return type of AudioEncoder::SetMaxPlaybackRate to void
There's no point in returning a status code, since the max playback rate
is only a suggestion that the encoder is free to disregard.

Review URL: https://codereview.webrtc.org/1332573003

Cr-Commit-Position: refs/heads/master@{#9900}
2015-09-09 06:15:41 +00:00
kwiberg
12cfc9b4da Fold AudioEncoderMutable into AudioEncoder
It makes more sense to combine the two interfaces, since there wasn't
a clear line separating them. The result is a combined interface with
just over a dozen methods, half of which need to be implemented by
every subclass, while the other half have sensible (and trivial)
default implementations and are implemented only by the few subclasses
that need non-default behavior.

Review URL: https://codereview.webrtc.org/1322973004

Cr-Commit-Position: refs/heads/master@{#9894}
2015-09-08 12:57:59 +00:00
kwiberg
6aae75728a On FATAL, log which unsupported encoder the caller wanted us to create
Hopefully, this will make it easier to figure out what's wrong the
next time this happens.

BUG=526478

Review URL: https://codereview.webrtc.org/1313073008

Cr-Commit-Position: refs/heads/master@{#9844}
2015-09-02 12:05:06 +00:00
Henrik Lundin
05f71fcb61 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}
2015-09-01 09:52:06 +00:00
Karl Wiberg
4376648df0 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}
2015-08-27 13:22:21 +00:00