This test verifies bit exactness for the send-side of ACM. The test
setup is a chain of three different test classes:
test::AcmSendTest -> AcmSenderBitExactness -> test::AcmReceiveTest
The receiver side is driving the test by requesting new packets from
AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
packet from test::AcmSendTest::NextPacket, which inserts audio from the
input file until one packet is produced. (The input file loops
indefinitely.) Before passing the packet to the receiver, the
AcmSenderBitExactness class verifies the packet header and updates a
payload checksum with the new payload. The decoded output from the
receiver is also verified with a (separate) checksum.
The current CL only adds tests for 30 ms and 60 ms iSAC. More codecs
will be added in coming changes.
BUG=3521
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6949 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL turns the background noise generation in NetEq off by default. The noise generation used to kick in during long-duration packet losses, when there was no point in extrapolating the latest audio any longer. However, this sometimes produces annoying noise in situations where silence would have been preferable.
With this change, a long packet-loss concealment will be faded out to zeros instead of a low noise.
Reference files are updated where needed.
BUG=3519
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6882 4adac7df-926f-26a2-2b94-8c16560cd09d
This change prepares for switching default background noise (bgn) mode
from on to off. The actual switch will be done later.
In this change, the bgn mode is included as a setting in NetEq's config
struct. We're also removing the connection between playout modes and
bgn modes in ACM. In practice this means that bgn mode will change from
off to on for streaming mode, but since the playout modes are not used
it does not matter.
BUG=3519
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6843 4adac7df-926f-26a2-2b94-8c16560cd09d
---
Fixes for re-enabling more MSVC level 4 warnings: webrtc/ edition
This contains fixes for the following sorts of issues:
* Possibly-uninitialized local variable
* Signedness mismatch
* Assignment inside conditional
This also contains a small number of other cleanups to nearby code. In
particular several warning-disables for MSVC are removed because they don't seem
to be necessary (either that warning is not enabled or the code does not trigger
it).
BUG=crbug.com/81439
TEST=none
R=henrika@webrtc.org, pkasting@chromium.org
Review URL: https://webrtc-codereview.appspot.com/18769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6667 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL introduces a bit-exactness test for the receive-side of the
AudioCoding Module. The main part of the test is done in the helper
class AcmReceiveTest. The test is executed from the test fixture
AcmReceiverBitExactness.
The test inserts packets from a pre-encoded RTP file. The output is
summed up into a checksum, which is verified versus a reference at the
end of the test. Alternatively, if the flag --generate_output is given,
the output is written to a file for subjective verification.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6549 4adac7df-926f-26a2-2b94-8c16560cd09d
* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.
Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.
Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.
BUG=3111
R=henrik.lundin@webrtc.org, turaj@webrtc.org, xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc
Review URL: https://webrtc-codereview.appspot.com/14559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
Phenomenon:
When packet loss rate was fed to a codec that does not implement packet loss adaptive encoding, VoE logs an error.
Reason:
The ACM function SetPacketLossRate(int rate) return -1 unnecessarily too often. It was intended for more severe errors like
1. codec is not ready
2. input rate is out of range
BUG=webrtc:3413
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6283 4adac7df-926f-26a2-2b94-8c16560cd09d
From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.
Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.
Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.
BUG=crbug/367114
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
This change includes max_packets_in_buffer and max_delay_ms in the
NetEq config struct. The packet buffer is also no longer limited in
terms of payload sizes (bytes), only number of packets.
The old constants governing the packet buffer limits are deleted.
BUG=3083
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5989 4adac7df-926f-26a2-2b94-8c16560cd09d
The value of AudioFrame::energy_ must be set to -1 in order to have the
energy calculated later on in the AudioConferenceMixer module. This was
not the case in ACM2, where the value was set to 0 instead. This
resulted in bad audio for multi-party calls (5 or more participants).
Implemented a unit test to verify ACM output frame.
BUG=3255
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5969 4adac7df-926f-26a2-2b94-8c16560cd09d
