Commit Graph

528 Commits

Author SHA1 Message Date
0ca768b131 Adding DTX to WebRTC Opus wrapper (relanding).
This is relanding of r7846, which failed since the unit test depended on whether Opus is in fixed-point or float-point.

See the review of r7846 here:
https://webrtc-codereview.appspot.com/13219004/

Patch set 1 is the same as r7846. Further fixes are found in patch set 2 and later.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7878 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 16:09:35 +00:00
817e50dd7d Make an AudioEncoder subclass for PCM16B
The implementation depends on AudioEncoderPcm to reduce code
duplication.

BUG=3926
R=kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7872 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:47:19 +00:00
b3ad8cf6ca Make an AudioEncoder subclass for iSAC
BUG=3926

Previously committed: https://code.google.com/p/webrtc/source/detail?r=7675
and reverted: https://code.google.com/p/webrtc/source/detail?r=7676

R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7871 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:08:19 +00:00
abe3f1879c Checking whether ACM uses codec internal or WebRTC DTX.
It was not clear how one could know if ACM is using DTX from WebRTC or codec internal DTX.

This CL makes better use of IsInternalDTXReplacedWithWebRtc() which was designed for G.729 to export such information.

Before
IsInternalDTXReplacedWithWebRtc() gives true only if codec == G729 and G729's internal DTX is replaced with WebRTC DTX.

Now
IsInternalDTXReplacedWithWebRtc() gives true also when codec does not have internal DTX, i.e., must use WebRTC DTX, which is much more logical.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7870 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 08:53:21 +00:00
55d42c32a4 DCHECK: Reference condition parameter in release builds
So that caller's won't get warnings about unused variables for
variables that are only used in calls to DCHECK, such as

  int x = ...
  DCHECK_EQ(x, 17);

R=andrew@webrtc.org

Previously committed: https://code.google.com/p/webrtc/source/detail?r=7858
and reverted: https://code.google.com/p/webrtc/source/detail?r=7859

Review URL: https://webrtc-codereview.appspot.com/31169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7869 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 08:32:30 +00:00
d8ca723de7 Remove CELT support from audio_coding.
R=henrik.lundin@webrtc.org, juberti@webrtc.org
TBR=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/33579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7864 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 11:49:13 +00:00
3cd26b677a Revert r7858 ("DCHECK: Reference condition parameter in release builds")
Apparently Visual Studio is cleverer than I am at figuring out what
local variables are actually unused.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7859 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 08:57:14 +00:00
3148060e61 DCHECK: Reference condition parameter in release builds
So that caller's won't get warnings about unused variables for
variables that are only used in calls to DCHECK, such as

  int x = ...
  DCHECK_EQ(x, 17);

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7858 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 08:45:47 +00:00
ff1a3e36bd Make an AudioEncoder subclass for comfort noise
BUG=3926
R=bjornv@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7857 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 07:29:08 +00:00
19dd129c69 Revert 7846 "Adding DTX to WebRTC Opus wrapper"
> Adding DTX to WebRTC Opus wrapper
> 
> This is a step toward adding Opus DTX support in WebRTC.
> 
> Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See
> 
> https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html
> 
> We transmit the first 1-byte packet to let decoder be in-sync
> 
> BUG=webrtc:1014
> R=henrik.lundin@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/13219004

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7848 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 15:11:15 +00:00
4321f175f1 Adding DTX to WebRTC Opus wrapper
This is a step toward adding Opus DTX support in WebRTC.

Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See

https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html

We transmit the first 1-byte packet to let decoder be in-sync

BUG=webrtc:1014
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7846 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 13:27:39 +00:00
1784d7cfad Adding an codec interal CNG test in NetEq.
BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7843 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:46:39 +00:00
e04a93bcf5 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
2014-12-09 10:12:53 +00:00
8911bc52f1 Add AudioEncoder::Max10MsFramesInAPacket
BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7834 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 21:15:55 +00:00
130fef89dd Bugfix in AudioDecoderTest
When the encoded frame size (L ms) was larger than 10 ms, the test would
repeat the first 10 ms L/10 times for each encoded frame. This is now
fixed.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7833 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 21:07:59 +00:00
fcbe36a1d9 Add const qualifier to WebRtcPcm16b_Encode
BUG=909
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7831 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 18:26:49 +00:00
a1ef7bfa15 ATTRIBUTE_UNUSED expanded to empty on MSVS, so be sure to use the variable.
Ideally, this is a stopgap fix until ATTRIBUTE_UNUSED can be given a
proper definition.

TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7830 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 17:53:10 +00:00
cb858ba397 Make an AudioEncoder subclass for iLBC
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@google.com
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32649005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7828 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 17:11:44 +00:00
33ccdfa1f5 Relanding r7807.
r7807 was reverted to be excluded from the cause of a failure.

It has been verified and can reland now.

BUG=

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7810 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 12:14:12 +00:00
52bc4f4797 Revert 7807 "Removing unused opus wrapper APIs."
> Removing unused opus wrapper APIs.
> 
> WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().
> 
> WebRtcOpus_DecodePlcMaster/Slave() are also removed.
> 
> BUG=
> R=henrik.lundin@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28139004

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7809 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 11:00:50 +00:00
e54a6342dd Removing unused opus wrapper APIs.
WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().

WebRtcOpus_DecodePlcMaster/Slave() are also removed.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7807 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 08:47:25 +00:00
3a52458237 add WebRtcIsacfix_AutocorrNeon's intrinsics version
The modification only uses the unique part of the
 WebRtcIsacfix_AutocorrC function. Pass FiltersTest.AutocorrFixTest test
 on both ARMv7 and ARM64, and the single function performance is similar
 with original assembly version on different platforms. If not
 specified, the code is compiled by GCC 4.6. The result is the "X
 version / C version" ratio, and the less is better.

| run 100k times             | cortex-a7 | cortex-a15 |
| use C as the base on each  |  (1.2Ghz) |   (1.7Ghz) |
| CPU target                 |           |            |
|----------------------------+-----------+------------|
| Neon asm                   |       24% |        23% |
| Neon intrinsics (GCC 4.6)  |       33% |        32% |
| Neon intrinsics (GCC 4.8)  |       27% |        27% |

BUG=3850
R=andrew@webrtc.org, jridges@masque.com

Change-Id: Id6cd0671502fadbebd10b1f5493f5b16c988286f

Review URL: https://webrtc-codereview.appspot.com/27999004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7802 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 21:58:18 +00:00
8dc21dc238 Rename internal AudioEncoder::Encode method to EncodeInternal
BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7801 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 20:36:03 +00:00
3800e13a3a 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
2014-12-03 16:28:17 +00:00
00ba1a7dfd 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
2014-12-03 14:23:23 +00:00
fa914e283c Adding a duration printout to neteq_rtpplay
BUG=2692
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7796 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 13:28:53 +00:00
1751ee7d32 Remove -flax-vector-conversions flag for ARM NEON building.
Pass compilation on both ARMv7 and ARM64. The generated
binary (audioproc) is byte to byte (with symbol striped) same as
before. The output of audioproc -aecm is also byte to byte same between
C and NEON version on ARMv7 and ARM64.

Change-Id: Ibdf40fe085f6bad1311f59bf9318bbcf37dd7ce5

BUG=3850
R=andrew@webrtc.org, jridges@masque.com

Review URL: https://webrtc-codereview.appspot.com/30239004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7783 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02 19:36:14 +00:00
7f1dfa5b61 Adding a payload type to AudioEncoder objects
The type is set in the Config struct and is provided in the EncodedInfo
output struct from each Encode() call. The audio_decoder_unittest is
updated to verify correct propagation of the payload type.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7780 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02 12:08:39 +00:00
0cd5558f2b AudioEncoder subclass for G722
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7779 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02 11:45:51 +00:00
1db20a4180 Adding EncodedInfo struct to AudioEncoder::Encode
This struct will be expanded in future changes.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7771 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 14:44:50 +00:00
20446e7e56 Move and rename neteq/test/RTPcat to neteq/tools/rtpcat
BUG=2692
TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7770 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 14:23:01 +00:00
c93437ef96 Add test NetEqDecodingTest.CngFirst
This CL adds a test to verify that it is ok to start the stream with
a comfort noise packet.

BUG=4021
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7769 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 11:42:42 +00:00
83317146ba Adding a new test helper RtpFileWriter and use it in RTPcat
This new helper class writes RTP packets to file in rtpdump format.
A unit test is also included.

The new test class is used while re-writing the test tool RTPcat.

BUG=2692
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7768 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 11:25:04 +00:00
91d928e737 Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader
This is in preparation for creating a new class RtpFileWriter which
will use the same RtpPacket struct.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7749 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 15:50:30 +00:00
03499a0e95 Add wav output capability to neteq_rtpplay
This CL makes neteq_rtpplay capable of writing to wav files as well as
pcm files. This is done through the new class OutputWavFile, which
wraps a WavWriter object in an AudioSink interface.

BUG=2692
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7740 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-24 14:50:53 +00:00
1153322cf8 Build fix for MIPS Android Webview build.
Excluding optimizations to support MIPS32R6 platform for Android Webview build (see also https://code.google.com/p/webrtc/source/detail?r=7580).

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7729 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-21 16:28:32 +00:00
4591fbd09f 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
2014-11-20 22:28:14 +00:00
6ff3ac1db8 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
2014-11-20 14:14:49 +00:00
ed91068bf1 Create a NetEq test for when the first incoming payload type is unknown
This test is currently disabled. It triggers an issue where the NetEq
will trigger asserts on subsequent GetAudio calls if the first inserted
packet is rejected, for instance since the payload type is unknown to
NetEq.

BUG=4021
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7723 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 11:01:02 +00:00
40af3a56e4 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
2014-11-19 13:46:52 +00:00
6f6ef72950 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
2014-11-19 13:02:24 +00:00
966a708b93 Use RtpFileSource in NetEqDecodingTest
This CL removes the dependency on the old NETEQTEST_RTPpacket class
from the NetEqDecodingTest code, and also removes the dependency
from the modules_unittests target to neteq_test_tools.

BUG=2692
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7709 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 09:08:38 +00:00
52bb521b47 Update isolate files for Android APK tests.
This should speed up test execution on Android since only
the files needed by the test will be processed (instead
of the whole data + resources directories).

A few files for modules_unittests had to be explicitly added
for Android, since they were previously a part of the
add-whole-directories entries for the resources and data
directories.

BUG=webrtc:3741
TEST=Passing android+android_rel trybots.
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7694 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 08:35:05 +00:00
1431e4dd1c Revert 7675 "Make an AudioEncoder subclass for iSAC"
Above CL did not compile on Android. Followings are links to Android builds

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Builder%20%28dbg%29/builds/2648

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Clang%20%28dbg%29/builds/2369

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20ARM64%20%28dbg%29/builds/1320

> Make an AudioEncoder subclass for iSAC
> 
> BUG=3926
> R=henrik.lundin@webrtc.org, kjellander@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/25019004

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7676 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 01:44:13 +00:00
05feff013e Make an AudioEncoder subclass for iSAC
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7675 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-10 23:53:08 +00:00
8b2058e733 Remove the state_ member from AudioDecoder
The subclasses that need a state pointer should declare them---with
the right type, not void*, to get rid of all those casts.

Two small but not quite trivial cleanups are included because they
blocked the state_ removal:

  - AudioDecoderG722Stereo now inherits directly from AudioDecoder
    instead of being a subclass of AudioDecoderG722.

  - AudioDecoder now has a CngDecoderInstance member function, which
    is implemented only by AudioDecoderCng. This replaces the previous
    practice of calling AudioDecoder::state() and casting the result
    to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
    plainly visible in the AudioDecoder class declaration.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24169005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7644 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 07:54:31 +00:00
368215dacb Revert 7623 "Remove the state_ member from AudioDecoder"
Breaks Chrome compile:
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3861: 'LOG_FERR1': identifier not found
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3861: 'LOG_FERR1': identifier not found
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(169) : error C3867: 'webrtc::NetEqImpl::GetAudioInternal': function call missing argument list; use '&webrtc::NetEqImpl::GetAudioInternal' to create a pointer to member
...

> Remove the state_ member from AudioDecoder
> 
> The subclasses that need a state pointer should declare them---with
> the right type, not void*, to get rid of all those casts.
> 
> Two small but not quite trivial cleanups are included because they
> blocked the state_ removal:
> 
>   - AudioDecoderG722Stereo now inherits directly from AudioDecoder
>     instead of being a subclass of AudioDecoderG722.
> 
>   - AudioDecoder now has a CngDecoderInstance member function, which
>     is implemented only by AudioDecoderCng. This replaces the previous
>     practice of calling AudioDecoder::state() and casting the result
>     to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
>     plainly visible in the AudioDecoder class declaration.
> 
> R=henrik.lundin@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/24169005

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30879005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7629 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 00:45:58 +00:00
8a232f65dd Revert 7625 "Don't use DCHECK when you need the side effects..."
Reverting since 7623 might depend on this one

> Don't use DCHECK when you need the side effects...
> 
> R=pbos@webrtc.org
> TBR=henrik.lundin@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/32369004

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7628 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 00:43:59 +00:00
b8425bc4f3 Don't use DCHECK when you need the side effects...
R=pbos@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7625 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 22:10:18 +00:00
9e525585fd Remove the state_ member from AudioDecoder
The subclasses that need a state pointer should declare them---with
the right type, not void*, to get rid of all those casts.

Two small but not quite trivial cleanups are included because they
blocked the state_ removal:

  - AudioDecoderG722Stereo now inherits directly from AudioDecoder
    instead of being a subclass of AudioDecoderG722.

  - AudioDecoder now has a CngDecoderInstance member function, which
    is implemented only by AudioDecoderCng. This replaces the previous
    practice of calling AudioDecoder::state() and casting the result
    to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
    plainly visible in the AudioDecoder class declaration.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24169005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 21:18:47 +00:00