10 Commits

Author SHA1 Message Date
yujo
36b1a5fcec Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:
const int16_t* data() const;
int16_t* mutable_data();

- data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames.
- mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_.

These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation.

This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later.

BUG=webrtc:7343
TBR=henrika

Review-Url: https://codereview.webrtc.org/2750783004
Cr-Commit-Position: refs/heads/master@{#18543}
2017-06-12 19:45:32 +00:00
kwiberg
65cb70d939 Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq
In short, what I did was to

  * Remove acm_common_defs.h (the stuff in it was used only by
    acm_codec_database.cc).

  * Move audio_coding_module_typedefs.h to a new build target.

  * Move the NetEqDecoder enum (and the associated
    NetEqDecoderToSdpAudioFormat function) to a new file in a new
    build target.

BUG=webrtc:7243, webrtc:7244

Review-Url: https://codereview.webrtc.org/2723253005
Cr-Commit-Position: refs/heads/master@{#17005}
2017-03-03 14:16:28 +00:00
kwiberg
da2bf4e150 Stop using old AudioCodingModule::RegisterReceiveCodec overloads
BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2388153004
Cr-Commit-Position: refs/heads/master@{#14753}
2016-10-24 20:47:16 +00:00
kwiberg
ac9f876bc0 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
gmock.h and gtest.h were moved (or rather, got wrappers so that we
could put some icky compatibility hacks in one place instead of 500)
in this CL: https://codereview.webrtc.org/2358993004/

NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2381013002
Cr-Commit-Position: refs/heads/master@{#14464}
2016-10-01 05:29:53 +00:00
kwiberg
77eab70470 Enable the -Wundef warning for clang
NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2358993004
Cr-Commit-Position: refs/heads/master@{#14425}
2016-09-29 00:42:08 +00:00
henrik.lundin
d4ccb00b9e Propagate muted parameter to VoE::Channel
Deleted the temporary ACM method without the muted parameter, and had
to modify several tests for this. The muted parameter is not yet propagated to the AudioConferenceMixer; this is the next step.

BUG=webrtc:5609
TBR=perkj@webrtc.org

Review-Url: https://codereview.webrtc.org/1985743002
Cr-Commit-Position: refs/heads/master@{#12779}
2016-05-17 19:22:03 +00:00
kwiberg
3747838afb Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/test/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1695763004

Cr-Commit-Position: refs/heads/master@{#11618}
2016-02-15 04:41:04 +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
Henrik Lundin
4d68208a20 Reduce the runtime of some ACM tests in modules_tests
By reducing the length of the audio input, the total runtime of
$ out/Debug/modules_tests --gtest_filter=AudioCodingModuleTest.*
is reduced by more than 10x, when run single-threaded.

The PCMFile helper class is extended with a FastForward method (to
skip initial silence in the test files) and a limiter on how much to
read.

BUG=webrtc:2463
R=ivoc@webrtc.org

Review URL: https://codereview.webrtc.org/1513223002 .

Cr-Commit-Position: refs/heads/master@{#10973}
2015-12-10 15:24:50 +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