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honeycomb-mr1
* commit 'a7a9c4cbbc2315a59ad27b43c83c66e272dcc2f2':
docs: add package description for RTP
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Change-Id: I02c181a48101be288fb4aabf497f573f00038f90
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calls." into gingerbread
* commit 'b7a76e84fde7fe534d46aaaa71e3224798354009':
Issue 4157048: mic gain for VoIP/SIP calls.
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gingerbread
* commit 'a482d83ccf35ccd6cc29a9e1ace3d77b5f28d013':
Issue 4157048: mic gain for VoIP/SIP calls.
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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
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special allowed chars."
* commit '6f67e7bf831147257e078dd72a22f2e43e009122':
Making it possible to call SIP calls with special allowed chars.
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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
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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
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* changes:
SipService: registers broadcast receivers on demand.
SipService: release wake lock for cancelled tasks.
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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
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Bug: 3327004
Change-Id: I0691cd70edf61f815ecb0613aca85babd89f6cc4
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bug:3326867
Change-Id: I2a62c75fb3f5e9c6ec2e00b29396e93b0c183d9b
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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
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Change-Id: If600df0eb1e6135aed9f3b2eacfb6bc9ed5d78ff
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bug:3487791
Change-Id: I7d8d146f8542cd7df387547c7ce3d5ded27f8e97
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bug:3461707
Change-Id: I69a4f84dde3929c754c838fd12e624b774f44826
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bug:3326867
Change-Id: I766e6e28f6ad3e84de2c9e24850d472ad00271cc
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honeycomb
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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
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Bug: 3327004
Change-Id: Ice47f973b5f2969f26eaa83a3e4795b2e153ba8b
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bug:3326867
Change-Id: Ic67dd7d4858f28224e4f01ad8b65bcd3a3c15f10
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1. Always initialize AudioRecord with VOICE_COMMUNICATION.
2. If echo canceler is available, disable our echo suppressor.
Change-Id: Idf18d3833189a8478c1b252ebe6ce55e923280b3
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Change-Id: I8f68e01e5f8c73bb8afd44312cbfadb55aab4330
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every second.
* commit 'dc78e3fe7f2ffbc810cd54e86e3a83e279d74984':
RTP: Send silence packets on idle streams for every second.
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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
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SipManager." into gingerbread
* commit 'aec9a33f1cfc7c32690bc8e24aefaeb137ab9859':
Check if VoIP API is supported in SipManager.
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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
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* commit 'd90bc225b9d6e4f8f69d984aa63062a7b20ac65c':
Remove SIP realm/domain check
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as the realm may be different from the domain.
Bug: 3283834
Change-Id: I64c9f0d6d626afdb397c5d378d30afa9d6a64ca9
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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
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SipURI returns port -1 when port is not present in the URI.
Don't call SipProfile.Builder.setPort() when that happens.
Change-Id: Ic5fe7301195705a77010038cae20d6629b33135e
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used for requests." into gingerbread
* commit 'c9cc9ab590ef879877e466c0b5f5823e11bb4c47':
Fix SIP bug of different transport/port used for requests.
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bug: http://b/3156148
Change-Id: I4fa5b274d2e90ebde12d9e99822dc193a65bad32
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settings" into gingerbread
* commit '7da1ffc9d2a51ef6120389a06351fd770ab45f46':
Set AudioGroup mode according to audio settings
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Set AudioGroup mode according to holding, mute and speaker phone settings.
Bug: 3119690
Change-Id: I02803ae105409b7f8482e6c2ef3e67623bd54e03
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* commit '6034f9b2664799cb4f983657a78023b49efff825':
Fix race between ending and answering a SIP call.
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+ Also fix race between ending and changing (holding/unholding) a SIP call.
+ Remove an unused method.
Bug : 3128233
Change-Id: Ie18d8333a88f0d9906d54988243d909b58e07e4b
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call." into gingerbread
* commit 'ed34b244f1665b604d2a291db504415b10a514d7':
Do not suppress error feedback during a SIP call.
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Bug: 3124788
Change-Id: Ia0a06f72336d1795515428eba0c9f875c32d13d1
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Polish things a little bit.
Change-Id: I2c3cea8b34b9c858879bc722ea1f38082ba22b8d
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implementation." into gingerbread
* commit 'c41b27e2748ee19620636a14721a1dc14c3b418c':
Correct SipService.isOpened() implementation.
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Make it return true for all existing accounts.
Rename mOpened to mOpenedToReceiveCalls to make it less confusing.
Bug: 3155849
Change-Id: I327f411bf76afd73434ad1fa2ffef3db1e35d778
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stack." into gingerbread
* commit '5c85338dcf85462534d85440ded100a8012ff9dd':
Notify SipSessions before closing SIP stack.
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Bug: 3116480
Change-Id: I748d63382ade250aed27ccb09ea68c76a433fd27
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gingerbread
* commit '0e58a9529895e270dae90e69486a59e41de714b8':
Throw proper exceptions in SipManager
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