henrik.lundin@webrtc.org fa58745445 Delete all codec-specific subclasses of ACMGenericCodec ...

They have all been replaced by AudioEncoder subclasses, accessed throgh
ACMGenericCodecWrapper objects. After this change, the only subclass of
ACMGenericCodec is ACMGenericCodecWrapper. (The two will be consolidated
in a future cl.)

This CL also deletes acm_opus_unittest.cc. This test file was already
replaced audio_encoder_opus_unittest.cc	in r8244.

BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40729004

Cr-Commit-Position: refs/heads/master@{#8457}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8457 4adac7df-926f-26a2-2b94-8c16560cd09d

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

..

acm_codec_database.cc

Delete all codec-specific subclasses of ACMGenericCodec

2015-02-23 09:26:51 +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

Re-implementing AcmOpusTest as AcmGenericCodecOpusTest

2015-02-18 13:15:45 +00:00

acm_generic_codec.cc

Fix a problem with reading uninitialized memory in ACM

2015-02-19 13:56:15 +00:00

acm_generic_codec.h

Re-implementing AcmOpusTest as AcmGenericCodecOpusTest

2015-02-18 13:15:45 +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

Delete all codec-specific subclasses of ACMGenericCodec

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

acm_receiver_unittest.cc

Delete all codec-specific subclasses of ACMGenericCodec

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

acm_receiver.cc

Adding two new stats to VoiceReceiverInfo

2015-02-18 15:24:39 +00:00

acm_receiver.h

Adding two new stats to VoiceReceiverInfo

2015-02-18 15:24:39 +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

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

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

audio_coding_module_impl.h

Adding two new stats to VoiceReceiverInfo

2015-02-18 15:24:39 +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

Change gflags and gmock includes to be full paths.

2014-09-04 09:55:40 +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