Commit Graph

3008 Commits

Author SHA1 Message Date
854e84c7fb Use webrtc/base/logging.h for video coding/processing.
Replaces system_wrappers' logging.h in video_coding and
video_processing.

BUG=webrtc:5118
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1435873003

Cr-Commit-Position: refs/heads/master@{#10664}
2015-11-17 00:39:10 +00:00
c91d173870 Revert of Several Tick counter improvements. (patchset #8 id:140001 of https://codereview.webrtc.org/1415923010/ )
Reason for revert:
Potentially breaks a threading test under DrMemory.  Rolling back while I investigate.

Original issue's description:
> Several Tick counter improvements.
>
> Move logic into cc file
> Simplify interval calculation
> Remove unused QUERY_PERFORMANCE_COUNTER windows implementation
> Remove double divide on each ::Now() invocation on mac
>
> Move TickTime and TickInterval funcitons to cc file in prep for refactoring.
>
> BUG=
> R=mflodman@webrtc.org, pbos@webrtc.org
>
> Committed: https://crrev.com/4c27e4b62da2047063d88eedfeec3e939fea7843
> Cr-Commit-Position: refs/heads/master@{#10661}

TBR=pbos@webrtc.org,mflodman@webrtc.org,noahric@chromium.org,thaloun@google.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review URL: https://codereview.webrtc.org/1450203002

Cr-Commit-Position: refs/heads/master@{#10663}
2015-11-17 00:28:56 +00:00
fa6228e221 Introduced the render sample queue for the aec and aecm.
BUG=webrtc:5099

Review URL: https://codereview.webrtc.org/1410833002

Cr-Commit-Position: refs/heads/master@{#10662}
2015-11-17 00:27:50 +00:00
4c27e4b62d Several Tick counter improvements.
Move logic into cc file
Simplify interval calculation
Remove unused QUERY_PERFORMANCE_COUNTER windows implementation
Remove double divide on each ::Now() invocation on mac

Move TickTime and TickInterval funcitons to cc file in prep for refactoring.

BUG=
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1415923010 .

Cr-Commit-Position: refs/heads/master@{#10661}
2015-11-16 22:37:59 +00:00
6f8ce060a2 common_video: rename interface -> include
To avoid breaking downstream, the "interface" directories were copied
into a new "common_video/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
The header guards are also identical to avoid mixing them up in the transition.

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

Review URL: https://codereview.webrtc.org/1418913006

Cr-Commit-Position: refs/heads/master@{#10659}
2015-11-16 21:52:31 +00:00
b27f590ece Create rtc::AtomicInt POD struct.
Prevents accidental non-atomic reads, increments and stores since
"volatile int" doesn't enforce atomic usage.

BUG=
R=kwiberg@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1420043008

Cr-Commit-Position: refs/heads/master@{#10657}
2015-11-16 19:03:06 +00:00
9a7c838ec4 Adding stddef.h to opus_inst.h.
This is to prevent size_t from undefined. This does not happen in current WebRTC since the sources that opus_inst.h gets used have proper definitions. But it would be good to add the definition in itself.

Review URL: https://codereview.webrtc.org/1446093003

Cr-Commit-Position: refs/heads/master@{#10653}
2015-11-16 16:07:04 +00:00
e155ae671c Move CNG and RED management into the Rent-A-Codec
This leaves CodecOwner without a job, so we eliminate it.

BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1443653004

Cr-Commit-Position: refs/heads/master@{#10650}
2015-11-16 12:50:02 +00:00
0b9e29c87d Remove include dirs from modules/{media_file,pacing}
Also move files out of media_file/source.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_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
R=asapersson@webrtc.org, perkj@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1435093002 .

Cr-Commit-Position: refs/heads/master@{#10647}
2015-11-16 10:12:32 +00:00
f8506cbdd8 rtcp::Ij renamed to rtcp::ExtendedJitterReport
to match name given in the RFC5450
  private member renamed to inter_arrival_jitters_ for the same reason.
rtcp::ExtendedJitterReport moved into own file
accessors and Parse function added
  to make class usable for parsing packet

Review URL: https://codereview.webrtc.org/1434213004

Cr-Commit-Position: refs/heads/master@{#10636}
2015-11-13 15:33:26 +00:00
0fa9b22789 Remove scoped_ptrs for VCM sender_ and receiver_.
Put VideoSender/VideoReceiver flat within the object, not as
scoped_ptrs, giving fewer allocations and looking a bit nicer.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1443613002

Cr-Commit-Position: refs/heads/master@{#10634}
2015-11-13 13:59:59 +00:00
df948f03b3 rtcp::ReportBlock refactored to contain parsing
Review URL: https://codereview.webrtc.org/1420283022

Cr-Commit-Position: refs/heads/master@{#10633}
2015-11-13 11:03:18 +00:00
0e7e259ebd Move BitrateAllocator from BitrateController logic to Call.
This is a step on the way to have variable bitrate for audio and is
intended to be as much of a no-op as possible.

BUG=webrtc:5079

Review URL: https://codereview.webrtc.org/1441673002

Cr-Commit-Position: refs/heads/master@{#10630}
2015-11-13 05:02:46 +00:00
68876f990e Introduces Android API level linting, fixes all current API lint errors.
This CL attempts to annotate accesses on >16 API levels using as
small scopes as possible. The TargetApi notations mean "yes, I know
I'm accessing a higher API and I take responsibility for gating the
call on Android API level". The Encoder/Decoder classes are annotated
on the whole class, but they're only accessed through JNI; we should
annotate on method level otherwise and preferably on private methods.

This patch also fixes some compiler-level deprecation warnings (i.e.
-Xlint:deprecation), but probably not all of them.

BUG=webrtc:5063
R=henrika@webrtc.org, kjellander@webrtc.org, magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1412673008 .

Cr-Commit-Position: refs/heads/master@{#10624}
2015-11-12 16:37:01 +00:00
5dda80abea Remove webrtc/modules/video_{capture,render}/include
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_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
R=pbos@webrtc.org, perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1439823002 .

Cr-Commit-Position: refs/heads/master@{#10619}
2015-11-12 11:47:02 +00:00
e71b24e421 OpenSL ES stability improvements.
This CL does two things:

1) Improves stability in the existing OpenSL ES implementation for devices that
supports OpenSL ES. The cost is a slight increase in latency since the focus here
has been on avoiding audio glitches.

2) Adds a new Java API to exclude usage of OpenSL ES to enable comparisons between
OpenSL ES and Java based audio backends.

BUG=b/22452539

Review URL: https://codereview.webrtc.org/1440623002

Cr-Commit-Position: refs/heads/master@{#10618}
2015-11-12 09:48:36 +00:00
96839648a0 Trivial initialization fix in AudioDeviceIOS
NOTRY=TRUE
TBR=tkchin
BUG=webrtc:5058

Review URL: https://codereview.webrtc.org/1435003002

Cr-Commit-Position: refs/heads/master@{#10616}
2015-11-12 09:01:24 +00:00
6b14f9377d Adjust parameter for VP9 resize unittest.
Needed for upcoming libvpx roll.

TBR=stefan@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1432773005 .

Cr-Commit-Position: refs/heads/master@{#10611}
2015-11-11 21:42:13 +00:00
9b72af94cd Remove webrtc/modules/audio_processing/{aec,aecm,ns}/include
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_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
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1440523002 .

Cr-Commit-Position: refs/heads/master@{#10608}
2015-11-11 19:16:28 +00:00
ee2bac26dd 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}
2015-11-11 18:34:07 +00:00
4dc941128f CodecManager::RegisterEncoder: Call SetFec on new encoder, not old
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1416633011

Cr-Commit-Position: refs/heads/master@{#10604}
2015-11-11 16:34:28 +00:00
00ac85e2e3 Update temporal up switch field for non-flexible mode according to updates in the RTP payload profile.
The U bit is no longer obtained from the SS data.

https://tools.ietf.org/id/draft-ietf-payload-vp9-01.txt

BUG=chromium:500602

Review URL: https://codereview.webrtc.org/1433273002

Cr-Commit-Position: refs/heads/master@{#10601}
2015-11-11 13:30:55 +00:00
5237aaf243 Convert usage of ARRAY_SIZE to arraysize.
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.

BUG=None
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1405023016

Cr-Commit-Position: refs/heads/master@{#10594}
2015-11-11 07:44:39 +00:00
be57983f4b Rename Maybe to Optional
And add examples of good and bad usage to the documentation.

R=aluebs@webrtc.org, henrik.lundin@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1432553007 .

Cr-Commit-Position: refs/heads/master@{#10588}
2015-11-10 21:34:32 +00:00
ed8275a44f CodecManager: Eliminate the stereo_send_ member
It can be computed from other members, notably the current encoder's
number of channels.

BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1423803007

Cr-Commit-Position: refs/heads/master@{#10585}
2015-11-10 17:47:41 +00:00
c94bd9bf86 If a desktop captured window switches on/off it full screen mode, the capture may be unexpectedly terminated. During the transition of full screen mode on/off, the window can be temporarily invisible.
BUG=498484

Review URL: https://codereview.webrtc.org/1426103005

Cr-Commit-Position: refs/heads/master@{#10583}
2015-11-10 15:33:58 +00:00
d153a37801 Remove contention between RTCP packets and encoding.
Receiving RTCP often caused the worker thread to stall for >20 ms
(>100ms observed) due to contention on VideoSender's send_crit_ (used to
protect encoding).

This change removes an unnecessary acquire of send_crit_ and caches
encoder settings in ViEEncoder instead of acquiring them through
::SendCodec() in VCM (which is blocking).

BUG=webrtc:5106
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1433703002 .

Cr-Commit-Position: refs/heads/master@{#10582}
2015-11-10 15:27:21 +00:00
cfc319be1d Reland of Work on flexible mode and screen sharing. (patchset #1 id:1 of https://codereview.webrtc.org/1438543002/ )
Reason for revert:
Failed test not related to this CL (test fails on
master at an earlier date), re-landing original CL..

(This time from my @webrtc account.)

Original issue's description:
> Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ )
>
> Reason for revert:
> Seems to break VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Linux Memcheck buildbot.
>
> Original issue's description:
> > Work on flexible mode and screen sharing.
> >
> > Implement VP8 style screensharing but with spatial layers.
> > Implement flexible mode.
> >
> > Files from other patches:
> > generic_encoder.cc
> > layer_filtering_transport.cc
> >
> > BUG=webrtc:4914
> >
> > Committed: https://crrev.com/77ccfb4d16c148e61a316746bb5d9705e8b39f4a
> > Cr-Commit-Position: refs/heads/master@{#10572}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,philipel@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4914
>
> Committed: https://crrev.com/0be8f1d347bdb171462df89c2a4c69b3f3eb7519
> Cr-Commit-Position: refs/heads/master@{#10578}

TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,terelius@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4914

Review URL: https://codereview.webrtc.org/1431283002

Cr-Commit-Position: refs/heads/master@{#10581}
2015-11-10 15:17:26 +00:00
c95c366f5a Move the Rent-A-Codec™ from CodecOwner to CodecManager
Future CLs will move it even further down the stack.

BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1431103002

Cr-Commit-Position: refs/heads/master@{#10580}
2015-11-10 14:35:28 +00:00
0be8f1d347 Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ )
Reason for revert:
Seems to break VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Linux Memcheck buildbot.

Original issue's description:
> Work on flexible mode and screen sharing.
>
> Implement VP8 style screensharing but with spatial layers.
> Implement flexible mode.
>
> Files from other patches:
> generic_encoder.cc
> layer_filtering_transport.cc
>
> BUG=webrtc:4914
>
> Committed: https://crrev.com/77ccfb4d16c148e61a316746bb5d9705e8b39f4a
> Cr-Commit-Position: refs/heads/master@{#10572}

TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,philipel@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4914

Review URL: https://codereview.webrtc.org/1438543002

Cr-Commit-Position: refs/heads/master@{#10578}
2015-11-10 13:31:22 +00:00
Per
327d8babc8 Add DecodedImageCallback::Decoded() function with custom decode time value.
On Android, we would like to use MediaCodec output buffers to hold decoded frames until they can be rendered to a texture. There can only be one texture buffer used at the same time and therefore the calculated decode time in VCMTiming will be wrong since that calculation will also include the time where the decoder waited for the upper layers (that depend on network jitter and actual render time) to release the frame.

This new method will be used in
https://codereview.webrtc.org/1422963003/

BUG=webrtc:4993
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1414693006 .

Cr-Commit-Position: refs/heads/master@{#10576}
2015-11-10 13:00:45 +00:00
805fc710f7 Let Rent-A-Codec™ create and own speech encoders
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1410333015

Cr-Commit-Position: refs/heads/master@{#10575}
2015-11-10 12:05:23 +00:00
3cea256806 Reland "Prevent Opus DTX from generating intermittent noise during silence"
The original CL is reviewed at
https://codereview.webrtc.org/1415173005/

A silly mistake was made at the last patch set, and the CL was reverted. This CL is to fix and reland it.

BUG=

Review URL: https://codereview.webrtc.org/1422213003

Cr-Commit-Position: refs/heads/master@{#10574}
2015-11-10 11:49:32 +00:00
77ccfb4d16 Work on flexible mode and screen sharing.
Implement VP8 style screensharing but with spatial layers.
Implement flexible mode.

Files from other patches:
generic_encoder.cc
layer_filtering_transport.cc

BUG=webrtc:4914

Review URL: https://codereview.webrtc.org/1328113004

Cr-Commit-Position: refs/heads/master@{#10572}
2015-11-10 10:19:20 +00:00
c12be3984f -Removed the indirect error message reporting in aec and aecm.
-Made the component error messages generic to be an unspecified error message.

BUG=webrtc:5099

Review URL: https://codereview.webrtc.org/1404743003

Cr-Commit-Position: refs/heads/master@{#10570}
2015-11-10 07:53:53 +00:00
b4a753fdb5 Revert of Prevent Opus DTX from generating intermittent noise during silence (patchset #10 id:250001 of https://codereview.webrtc.org/1415173005/ )
Reason for revert:
Breaks voe_auto_test on all three "large tests bots".
https://build.chromium.org/p/client.webrtc/builders/Win32%20Release%20%5Blarge%20tests%5D/builds/5630/steps/voe_auto_test/logs/stdio
https://build.chromium.org/p/client.webrtc/builders/Mac32%20Release%20%5Blarge%20tests%5D/builds/5599/steps/voe_auto_test/logs/stdio
https://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/5645/steps/voe_auto_test/logs/stdio

Notice these bots are no longer a part of the default trybot set, so they have to be run manually when working with code that affects their tests (they were removed as they queued up all the time in the CQ, and usually don't catch breakages).

Original issue's description:
> Prevent Opus DTX from generating intermittent noise during silence.
>
> Opus may have an internal error that causes this. Here we make a workaround by adding some small disturbance to the input signals to break a long sequence of zeros.
>
> BUG=webrtc:5127
>
> Committed: https://crrev.com/f475add57eada116bc960fe2935876ec8c077977
> Cr-Commit-Position: refs/heads/master@{#10565}

TBR=tina.legrand@webrtc.org,kwiberg@webrtc.org,solenberg@webrtc.org,minyue@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5127

Review URL: https://codereview.webrtc.org/1428613004

Cr-Commit-Position: refs/heads/master@{#10567}
2015-11-09 21:27:11 +00:00
c1cd2bbd79 Turned off progress report for finished processing when the progress report is explicitly deactivated
BUG=webrtc:5099

Review URL: https://codereview.webrtc.org/1407723002

Cr-Commit-Position: refs/heads/master@{#10566}
2015-11-09 18:38:12 +00:00
f475add57e Prevent Opus DTX from generating intermittent noise during silence.
Opus may have an internal error that causes this. Here we make a workaround by adding some small disturbance to the input signals to break a long sequence of zeros.

BUG=webrtc:5127

Review URL: https://codereview.webrtc.org/1415173005

Cr-Commit-Position: refs/heads/master@{#10565}
2015-11-09 18:08:20 +00:00
ab48ef3534 Remove legacy audio device glue files.
I cannot find any references to these old locations.

TESTED=
git cl try -c --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=henrikg@webrtc.org

Review URL: https://codereview.webrtc.org/1421723009 .

Cr-Commit-Position: refs/heads/master@{#10564}
2015-11-09 16:44:05 +00:00
1f1912d1f0 Added unittest of the locking functionality in the audio processing module
The test is currently disabled as it takes too long to run in a coffe-cup manner

BUG=webrtc:5099

Review URL: https://codereview.webrtc.org/1394803002

Cr-Commit-Position: refs/heads/master@{#10560}
2015-11-09 11:13:25 +00:00
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
19299fb28b Remove interface directories kept to avoid breaking downstream.
This is a follow-up CL for https://codereview.webrtc.org/1417683006
now that downstream code has been updated to use the 'include' directories
for header files instead.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel -m tryserver.webrtc --bot=ios_rel

Review URL: https://codereview.webrtc.org/1414793020

Cr-Commit-Position: refs/heads/master@{#10547}
2015-11-06 23:24:52 +00:00
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
56b1128c8f Change to use local Random object instead of global rand() in the RtcEventLog unit test.
Removed Rand(int low, int high) since that function outputs results that are non-random and/or outside the interval if low is negative.

Added new Uniform(uint32_t, uint32_t) function to replace Rand(int low, int high).

Changed various unit tests to use the new functions.
BUG=

Review URL: https://codereview.webrtc.org/1413053002

Cr-Commit-Position: refs/heads/master@{#10541}
2015-11-06 13:14:01 +00:00
c4a1c370aa Removed vie_defines.h
The defines still in use was only used in single files, so they were
moved to these specific cc-files.

Review URL: https://codereview.webrtc.org/1411573007

Cr-Commit-Position: refs/heads/master@{#10539}
2015-11-06 12:33:56 +00:00
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
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
c253a1c00e Reland of "Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile."
BUG=webrtc:5144
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1409753007

Cr-Commit-Position: refs/heads/master@{#10533}
2015-11-06 08:12:09 +00:00
b7a5c16d2c Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ )
This is the second revert. The first attempt in https://codereview.webrtc.org/1423693008/
was missing a subtle curly brace caused by a merge conflict.
I'm going to let this one go through the CQ.

Reason for revert:
This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.

See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.

Original issue's description:
> Add aecdump support to audioproc_f.
>
> Add a new interface to abstract away file operations. This CL temporarily
> removes support for dumping the output of reverse streams. It will be easy to
> restore in the new framework, although we may decide to only allow it with
> the aecdump format.
>
> We also now require the user to specify the output format, rather than
> defaulting to the input format.
>
> TEST=Bit-exact output to the previous audioproc_f version using an input wav
> file, and to the legacy audioproc using an aecdump file.
>
> Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> Cr-Commit-Position: refs/heads/master@{#10460}

TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1412963007

Cr-Commit-Position: refs/heads/master@{#10532}
2015-11-05 20:33:25 +00:00
006d93d3c6 Added protobuf message for loss-based BWE events, and wired it up to the send side bandwidth estimator.
BUG=

Review URL: https://codereview.webrtc.org/1411673003

Cr-Commit-Position: refs/heads/master@{#10531}
2015-11-05 20:02:19 +00:00