11 Commits

Author SHA1 Message Date
ossu
e352578bc8 Moved injection of AudioDecoderFactory into voe::Channel.
Channel's API remains unchanged, but the creation of a BuiltinAudioDecoderFactory is now in Channel. The next step would be to amend Channel's API (through CreateChannel, I believe) to allow an AudioDecoderFactory to be sent along.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/1992763002
Cr-Commit-Position: refs/heads/master@{#12893}
2016-05-25 14:37:47 +00:00
henrik.lundin
834a6ea12b Add muted_output parameter to ACM
The new parameter indicates if the output in the AudioFrame is muted. If
so, the output samples are not written, but should be interpreted as all
zero.

A version of AudioCodingModule::PlayoutData10Ms() without the new
parameter is maintained while waiting for downstream dependencies to
conform.

BUG=webrtc:5609

Review-Url: https://codereview.webrtc.org/1976913002
Cr-Commit-Position: refs/heads/master@{#12719}
2016-05-13 10:45:31 +00:00
kwiberg
84be511ac0 Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
(This is a re-land of https://codereview.webrtc.org/1921233002, which
got reverted for breaking Chromium.)

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1923133002

Cr-Commit-Position: refs/heads/master@{#12522}
2016-04-27 08:20:08 +00:00
terelius
52d4e6bf5e Revert of Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/ (patchset #1 id:40001 of https://codereview.webrtc.org/1921233002/ )
Reason for revert:
Fails on Chromium FYI bots.

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/5392/

Original issue's description:
> Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
>
> BUG=webrtc:5520
>
> Committed: https://crrev.com/2c27a062ee46258abe9facc2cceee74f09bf6a99
> Cr-Commit-Position: refs/heads/master@{#12511}

TBR=tommi@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1924443002

Cr-Commit-Position: refs/heads/master@{#12513}
2016-04-26 16:32:09 +00:00
kwiberg
2c27a062ee Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1921233002

Cr-Commit-Position: refs/heads/master@{#12511}
2016-04-26 15:38:03 +00:00
henrik.lundin
9a410dd082 Change NetEq::GetPlayoutTimestamp to return an rtc::Optional<uint32_t>
This is in preparation for changes to when the playout timestamp is
valid.

BUG=webrtc:5669

Review URL: https://codereview.webrtc.org/1853183002

Cr-Commit-Position: refs/heads/master@{#12256}
2016-04-06 08:39:30 +00:00
kwiberg
4cdbd57fe3 AudioCodingModule: Add methods for injecting external encoder stacks
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1673213002

Cr-Commit-Position: refs/heads/master@{#12158}
2016-03-30 10:10:15 +00:00
Peter Kasting
6955870806 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}
2016-01-13 00:26:55 +00:00
pkasting
25702cb162 Misc. small cleanups.
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests).  Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks

All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.

BUG=none
TEST=none

Review URL: https://codereview.webrtc.org/1534193008

Cr-Commit-Position: refs/heads/master@{#11191}
2016-01-08 21:50:32 +00:00
henrik.lundin
4cf61dd116 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}
2015-12-09 14:21:02 +00:00
kjellander
3e6db2321c audio_coding: remove "main" directory
This is the last piece of the old directory layout of the modules.

Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.

BUG=webrtc: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 --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1481493004

Cr-Commit-Position: refs/heads/master@{#10803}
2015-11-26 12:45:01 +00:00