henrik.lundin@webrtc.org 8d350d4bc4 Add new AcmGenericCodecTest and verify output from Encode function ...

The test specifically verifies that the output is as expected when
DTX/CNG is used.

COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38219004

Cr-Commit-Position: refs/heads/master@{#8497}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8497 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-02-25 10:06:20 +00:00

..

acm_codec_database.cc

Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :

2015-02-23 21:29:45 +00:00

acm_codec_database.h

Delete all codec-specific subclasses of ACMGenericCodec

2015-02-23 09:26:51 +00:00

acm_common_defs.h

Compile ACM2 and ACM1.

2013-09-19 00:12:23 +00:00

acm_generic_codec_opus_test.cc

Merge ACMGenericCodec and ACMGenericCodecWrapper

2015-02-23 16:33:49 +00:00

acm_generic_codec_test.cc

Add new AcmGenericCodecTest and verify output from Encode function

2015-02-25 10:06:20 +00:00

acm_generic_codec.cc

Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call"

2015-02-25 10:03:19 +00:00

acm_generic_codec.h

Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call"

2015-02-25 10:03:19 +00:00

acm_neteq_unittest.cc

Wrap ACM2 code inside acm2 namespace. This gurantees that one ACM would not use components of others by accident.

2013-10-06 04:47:28 +00:00

acm_receive_test_oldapi.cc

New ACM test to trigger audio glitch when switching output sample rate

2014-10-14 10:49:58 +00:00

acm_receive_test_oldapi.h

New ACM test to trigger audio glitch when switching output sample rate

2014-10-14 10:49:58 +00:00

acm_receive_test.cc

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

acm_receive_test.h

Reland "Converting five tests to use new AudioCoding interface" (r7258)

2014-09-23 12:05:34 +00:00

acm_receiver_unittest_oldapi.cc

Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call"

2015-02-25 10:03:19 +00:00

acm_receiver_unittest.cc

Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call"

2015-02-25 10:03:19 +00:00

acm_receiver.cc

Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call"

2015-02-25 10:03:19 +00:00

acm_receiver.h

Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call"

2015-02-25 10:03:19 +00:00

acm_resampler.cc

Added include of assert.h for files calling assert but missing the include.

2014-04-29 17:54:17 +00:00

acm_resampler.h

Add an output capacity parameter to ACMResampler::Resample10Msec()

2014-04-24 19:05:33 +00:00

acm_send_test_oldapi.cc

Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting.

2015-02-02 23:54:40 +00:00

acm_send_test_oldapi.h

Setting Opus target application.

2015-01-20 16:01:50 +00:00

acm_send_test.cc

Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting.

2015-02-02 23:54:40 +00:00

acm_send_test.h

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

audio_coding_module_impl.cc

Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call"

2015-02-25 10:03:19 +00:00

audio_coding_module_impl.h

Remove more unused code in ACM

2015-02-24 12:02:18 +00:00

audio_coding_module_unittest_oldapi.cc

Re-land "Switch to using AudioEncoderIsac instead of ACMISAC"

2015-02-19 15:10:49 +00:00

audio_coding_module_unittest.cc

Re-land "Switch to using AudioEncoderIsac instead of ACMISAC"

2015-02-19 15:10:49 +00:00

audio_coding_module.cc

Starting to implement the new ACM API

2014-09-08 13:13:19 +00:00

audio_coding_module.gypi

Delete all codec-specific subclasses of ACMGenericCodec

2015-02-23 09:26:51 +00:00

call_statistics_unittest.cc

Change gflags and gmock includes to be full paths.

2014-09-04 09:55:40 +00:00

call_statistics.cc

Remove std:: prefixes from C functions in webrtc/.

2014-03-07 15:23:34 +00:00

call_statistics.h

Update talk to 58174641 together with http://review.webrtc.org/4319005/.

2013-12-13 19:17:43 +00:00

initial_delay_manager_unittest.cc

Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :

2015-02-23 21:29:45 +00:00

initial_delay_manager.cc

Fix the chain that propagates the audio frame's rtp and ntp timestamp including:

2014-06-05 20:34:08 +00:00

initial_delay_manager.h

Fix the chain that propagates the audio frame's rtp and ntp timestamp including:

2014-06-05 20:34:08 +00:00

nack_unittest.cc

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

nack.cc

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

nack.h

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

OWNERS

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00