Herring board exhibits a strong echo even in non speakerphone modes.
To compensate the lack of AEC or AES when not in speakerphone, the mic gain
had been reduced in the ADC. But this has an adverse effect on other VoIP applications
that have their own AEC and are penalized by the weak mic gain.
This workaround enables an acceptable mic gain for other VoIP apps while offering a
SIP call experience which is not worse than it was with the residual echo that was
present even with mic gain reduction.
Change-Id: I33fd37858758e94e42ef5b545d3f0dc233220bf1
Since String.replaceFirst uses regex and since SIP user names are
allowed to include regex charaters such as '+', the code must
fist convert the string to a literal pattern String before using
replaceFirst method.
Change-Id: I25eac852bd620724ca1c5b2befc023af9dae3c1a
This change unhides RTP related classes including AudioCodec,
AudioGroup, AudioStream, and RtpStream. This allows developers
to control audio streams directly and also makes conference
calls possible with the combination of the public SIP APIs.
Change-Id: Idfd4edf65a1cbf3245ec2786fbc03b06438b0fb3
Cherripick from master CL 79833, 79417, 78864, 80332, 87500
Add new audio mode and recording source for audio communications
other than telelphony.
The audio mode MODE_IN_CALL signals the system the device a phone
call is currently underway. There was no way for audio video
chat or VoIP applications to signal a call is underway, but not
using the telephony resources. This change introduces a new mode
to address this. Changes in other parts of the system (java
and native) are required to take this new mode into account.
The generic AudioPolicyManager is updated to not use its phone
state variable directly, but to use two new convenience methods,
isInCall() and isStateInCall(int) instead.
Add a recording source used to designate a recording stream for
voice communications such as VoIP.
Update the platform-independent audio policy manager to pass the
nature of the audio recording source to the audio policy client
interface through the AudioPolicyClientInterface::setParameters()
method.
SIP calls should set the audio mode to MODE_IN_COMMUNICATION,
Audio mode MODE_IN_CALL is reserved for telephony.
SIP: Enable built-in echo canceler if available.
1. Always initialize AudioRecord with VOICE_COMMUNICATION.
2. If echo canceler is available, disable our echo suppressor.
Note that this CL is intentionally not correcting the
getAudioSourceMax() return value in MediaRecorder.java as the
new source is hidden here.
Change-Id: Ie68cd03c50553101aa2ad838fe9459b2cf151bc8
The previous implementation registers receivers when SipService starts up.
If the user doesn't use SIP at all, SipService will still process connecivity
and wifi state change events, which involves holding wake lock and thus
consumes power unnecessarily.
With this CL, SipService is completely idle if the user doesn't use SIP at all.
It registers receivers only when at least one account is opened.
Bug: 3326998
Change-Id: Idea43747f8204b0ccad3fc05a1b1c0b29c9b2557
The previous implementation registers receivers when SipService starts up.
If the user doesn't use SIP at all, SipService will still process connecivity
and wifi state change events, which involves holding wake lock and thus
consumes power unnecessarily.
With this CL, SipService is completely idle if the user doesn't use SIP at all.
It registers receivers only when at least one account is opened.
Bug: 3326998
Change-Id: Ib70e0cf2c808e0ebab4c3c43dcab5532d24e5eeb
Originally a stream does not send packets when it is receive-only or there is
nothing to mix. However, this causes some problems with certain firewalls and
proxies. A firewall might remove a port mapping when there is no outgoing
packet for a preiod of time, and a proxy might wait for incoming packets from
both sides before start forwarding. To solve these problems, we send out a
silence packet on the stream for every second. It should be good enough to
keep the stream alive with relatively low resources.
Bug: 3119690
Change-Id: Ib9c55e5dddfba28928bd9b376832b68bda24c0e4
This is to make SipManager.isVoipSupported() effective.
Also add NPE check now that we may return null SipAudioCall when VOIP is not
supported.
Bug: 3251016
Change-Id: Icd551123499f55eef190743b90980922893c4a13
SipURI returns port -1 when port is not present in the URI.
Don't call SipProfile.Builder.setPort() when that happens.
Bug: 3291248
Change-Id: I8e608cbc56ea82862df55fdba885f6a864db83ab
SipURI returns port -1 when port is not present in the URI.
Don't call SipProfile.Builder.setPort() when that happens.
Change-Id: Ic5fe7301195705a77010038cae20d6629b33135e
+ Also fix race between ending and changing (holding/unholding) a SIP call.
+ Remove an unused method.
Bug : 3128233
Change-Id: Ie18d8333a88f0d9906d54988243d909b58e07e4b
