| e66950506c473e660f2e5762d7a71e13808be387 |
|
30-Mar-2012 |
Chia-chi Yeh <chiachi@android.com> |
RTP: refactor a little bit and fix few minor bugs.
Change-Id: I063644507f26996ded462972afcb550a4528dac8
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| f743e1f6abdb018fc58c467cdf35cbb8b81cf8c4 |
|
14-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Whitespace
Fix indentation, and add blank lines in key places for clarity
Change-Id: I57a0a8142394f83203161aa9b8aa9276abf3ed7c
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 597f8282ee1b86ba8f7384eb3060bac3b3f7cf92 |
|
12-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: I543e730aff2d03c18c26b116c9fe9419259808af
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 3762c311729fe9f3af085c14c5c1fb471d994c03 |
|
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/157220
Bug: 5449033
Change-Id: Ic9c19d30693bd56755f55906127cd6bd7126096c
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 8564c8da817a845353d213acd8636b76f567b234 |
|
06-Jan-2012 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/157065
Bug: 5449033
Change-Id: I00a4b904f9449e6f93b7fd35eac28640d7929e69
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 5baa3a62a97544669fba6d65a11c07f252e654dd |
|
20-Dec-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/156016
Bug: 5449033
Change-Id: I4c4e33bb9df3e39e11cd985e193e6fbab4635298
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 71f2cf116aab893e224056c38ab146bd1538dd3e |
|
20-Oct-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/143865
Bug: 5449033
Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 6d8b9b84ac83acfc193fd633ba961168867124fa |
|
14-Sep-2011 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: Update parameters for larger packet intervals."
|
| be57bfe853d07369f429b600039ea474b9ea5e31 |
|
07-Sep-2011 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Update parameters for larger packet intervals.
Also remove some duplicated code.
Change-Id: I64576e5442a962eb4b0dfa83b52a8127567ba597
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 81a5ec5b94d889656cc2f212102c441b91b2e3c0 |
|
08-Sep-2011 |
Chia-chi Yeh <chiachi@android.com> |
Merge "RTP: support payloads with larger packetization interval."
|
| 35d05dcba1e829782813b6ec21afceb5cffc22e6 |
|
06-Sep-2011 |
Chia-chi Yeh <chiachi@android.com> |
RTP: support payloads with larger packetization interval.
RFC 3551 section 4.2 said that a receiver should accept packets
representing between 0 and 200ms of audio data. Now we add the
ability to decode multiple frames in a payload as long as the
jitter buffer is not full. This change covers G711, GSM, and
GSM-EFR. AMR will be added later.
Bug: 3029736
Change-Id: Ifd194596766d14f02177925c58432cd620e44dd7
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 74e0a990ae3196b8195db2a399c22516c7dd0823 |
|
29-Aug-2011 |
Eric Laurent <elaurent@google.com> |
VoIP JNI: Force AEC on for tuna board
Force AEC on for tuna board because of the strong feedback
of Rx audio path, even when playing over earpiece or headset.
Change-Id: I9c14257d56103ba82d6cdb0b7d5a3f315638136e
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 5fb3ba60afe68060ac1ed291f4a108fef8c622c3 |
|
25-Jul-2011 |
Eric Laurent <elaurent@google.com> |
Issue 3370834: No Echo canceler for SIP
Added detection of platfrom AEC in AudioGroup. If an AEC
is present, the SIP stack will use it, otherwise the echo suppressor
of the stack will be used.
Change-Id: I4aa45a8868466120f5f9fae71b491fe4ae1162c2
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 34bb419e5946ab28112e9e27a4d1b3928d31e0e2 |
|
11-May-2011 |
Dima Zavin <dima@android.com> |
update for new audio.h header location
Change-Id: Ic4c62c4037800802427eb7d3c7f5eb8b25d18876
Signed-off-by: Dima Zavin <dima@android.com>
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 24fc2fb1c541e954b83fd31ea9f786a5e9b45501 |
|
20-Apr-2011 |
Dima Zavin <dima@android.com> |
audio/media: convert to using the audio HAL and new audio defs
Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 7a492a9ad42947a3a7b777b0eb6eec56f5bb942b |
|
05-Apr-2011 |
Eric Laurent <elaurent@google.com> |
am b7a76e84: am a482d83c: Merge "Issue 4157048: mic gain for VoIP/SIP calls." into gingerbread
* commit 'b7a76e84fde7fe534d46aaaa71e3224798354009':
Issue 4157048: mic gain for VoIP/SIP calls.
|
| d7a724e6d89420408200c20937baa3b2bd902742 |
|
30-Mar-2011 |
Eric Laurent <elaurent@google.com> |
Issue 4157048: mic gain for VoIP/SIP calls.
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
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 2ba92c71b5684dce700cf848bf157153c156df1d |
|
15-Nov-2010 |
Jean-Michel Trivi <jmtrivi@google.com> |
do not merge bug 3370834 Cherrypick from master
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
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| d87be273aaea32995c87a6cbc6250cbfeeddd84d |
|
06-Jan-2011 |
Chia-chi Yeh <chiachi@android.com> |
Enable built-in echo canceler if available.
1. Always initialize AudioRecord with VOICE_COMMUNICATION.
2. If echo canceler is available, disable our echo suppressor.
Change-Id: Idf18d3833189a8478c1b252ebe6ce55e923280b3
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| d0da38079617e867db5d2bbdaaaa4cd49027d4eb |
|
05-Jan-2011 |
Chia-chi Yeh <chiachi@android.com> |
am dc78e3fe: am 3cf71376: RTP: Send silence packets on idle streams for every second.
* commit 'dc78e3fe7f2ffbc810cd54e86e3a83e279d74984':
RTP: Send silence packets on idle streams for every second.
|
| 3cf71376421f942d06b30101fbf0df7f3b23fbdd |
|
04-Jan-2011 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Send silence packets on idle streams for every second.
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
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 53aa6ef70d8692277f9403f94d43918ad9712dd0 |
|
30-Nov-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Prepare to unhide the APIs.
Polish things a little bit.
Change-Id: I2c3cea8b34b9c858879bc722ea1f38082ba22b8d
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 8a68b52b9873f1f3d7114576c9f39a2b7b402152 |
|
21-Oct-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Fix non-zero DC in EchoSuppressor caused while aggregating samples.
Rewrite using integer arithmetic to get full 32-bit precision instead
of 23-bit in single precision floating-points.
Bug: 3029745
Change-Id: If67dcc403923755f403d08bbafb41ebce26e4e8b
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| a8a10096a1501e901676632d78f699cdebe9f4f6 |
|
04-Oct-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Add a baseline echo suppressor.
Change-Id: I832f1f572f141fd928afe671b12d0b59f2a8e0b1
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 67ecb5b90c7d944a485ed35f3e968ab0ae49f5b4 |
|
01-Oct-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Start AudioRecord before AudioTrack to avoid being disabled.
Change-Id: I96be89fda41d77e2cf5bfc1c2f14e2b109001b57
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 21ae1ad6a695d6f1f253797fcf2a77b975b82cd3 |
|
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Minor fixes with polishing.
Change-Id: I50641373989e512fb489b5017edbcfd7848fe8b9
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 3520bd43139f4571cf96af126dba13681633bcb0 |
|
30-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Adjust the jitter buffer to 512ms.
Change-Id: Ia91c1aa1a03b65dbd329ea98383f370844e2b0c0
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| fe5298992a52f93bb8365d345cdd82d88a4b49f2 |
|
29-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Revise the workaround of private addresses and fix bugs.
Change-Id: Ie654b569f47049aa452eca8d3e6d4a98ac18469c
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 9083c84af1742cfc9228add21ec72310e67e6086 |
|
28-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Delay the initialization of AudioTrack and AudioRecord.
Related to http://b/3043844.
Change-Id: I2c4fd9f64e6eba597d68b2ea1ceeff83103697db
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| bd2294204e3edaede3fe81eb9b11c05c4fafe627 |
|
23-Sep-2010 |
Chung-yih Wang <cywang@google.com> |
Fix the unhold issue especially if one is behind NAT.
+call startAudio() when call is established.
Change-Id: Ib6a1e34017fb83007ce275da1991058e8b803833
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 7a69aeffda29bd1a7ebc5993eeb4e9ee224f096a |
|
22-Sep-2010 |
repo sync <chiachi@android.com> |
RTP: Add log throttle for "no data".
Change-Id: I14d9886a40fa780514cbc6c5bac6fb2a670f55f4
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 4033a67d0e99d422336574fc5c982d349632b117 |
|
16-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: Update native part to reflect the API change.
Change-Id: Ic2858920ad77d7312f2429f89ca509a481363431
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 557b04de23238fb496b5ca58e21331c842e95660 |
|
08-Sep-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: prevent buffer overflow in AudioRecord.
This change simply reduces the receive timeout of DeviceSocket. It works
because AudioRecord will block us till there is enough data, which makes
AudioSocket overlap AudioRecord.
Change-Id: I4700224fb407e148ef359a9d99279e10240128d0
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 7fa7ee11f6c274903241897c284337ba8b158988 |
|
26-Aug-2010 |
Chia-chi Yeh <chiachi@android.com> |
Revert "RTP: integrate the echo canceller from speex."
This reverts commit 4ae6ec428f3570b9020b35ada6a62f94af66d888.
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 4ae6ec428f3570b9020b35ada6a62f94af66d888 |
|
24-Aug-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: integrate the echo canceller from speex.
Currently the filter_length is set to one second.
Will change that when we have a better idea.
Change-Id: Ia942a8fff00b096de8ff0049a448816ea9a68068
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 2880ef86e5210832ef44f2d45c46ada1891372e5 |
|
24-Aug-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: reduce the latency by overlapping AudioRecord and AudioTrack.
Change-Id: I00d750ee514ef68d5b2a28bd1893417ed70ef1fc
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| b8790323473bef75a27d2da6fde2497b3bfe19eb |
|
19-Aug-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: fix few leaks when fail to add streams into a group.
Change-Id: Iefb3fe219ad48641da37a83c8d14e9ebf1d3086c
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 3459d3037cc0c482a27422f1cc000b5e9d289ae8 |
|
18-Aug-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: remove froyo-compatible code.
Change-Id: I6822a4e4749a5909959658c29253242b4018aeb0
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|
| 4c5d28cee0537c83ff0e5bc0daaae78f68dfc7c8 |
|
06-Aug-2010 |
Chia-chi Yeh <chiachi@android.com> |
RTP: move into frameworks.
Change-Id: Ic9c17b460448c746b21526ac10b647f281ae48e9
/frameworks/base/voip/jni/rtp/AudioGroup.cpp
|