| Commit message (Collapse) | Author | Age | Files | Lines |
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Fix compilation issues that appear when enabling LOG_NDEBUG.
Change-Id: I87e9e5ac66157759dd6f521fab0dd346089a011a
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RTSP stack would parse "BYE" message from RTCP packets and
notify NuPlayer which will send the notification to the client.
Change-Id: I461960f28610f998b71a6e1322fe79f9856c7a92
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Access units of seekable RTSP will be mapped to normal playtime in
RTSPSource, so the negative media time can be adjusted and played
normally.
Change-Id: I12793dbbf367650e66532195324adb5b5ad8fe85
CRs-Fixed: 866580
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- When network is poor, the response of RTSP teardown won't be received
in time, so ANR will happen.
- With this patch, a teardown message will be sent when timeout expires,
in order to avoid ANR.
Change-Id: I3f9efd9fefa66104ad452559ced5ff5218d73a66
CRs-Fixed: 650866
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Fix crash issues while repeated pause/resume. The root cause is:
While doing repeated pause-resume operation in RTSP streaming,
some times, the accessunit timeout check is being posted when
the streaming paused, and even after receving the EOS.
Due to this check, after the EOS, the number of packets received
will be zero and the session gets Teardown while the
streaming is paused. Upon resume, it leads to check failure and crash.
Cancel the timeout check if pause is not issued to MyHandler when
nearing EOS.
Change-Id: I897d8e18b0dbc29fc1099e7b3d1b5ccc16426a4d
CRs-Fixed: 636312, 676054
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- Add property "rtsp.transport.TCP"
Value "true" - TCP
Value "false" - UDP
default value - false
- This property can be used to test rtsp
streaming over TCP directly for network
compatibility testing
Change-Id: Ic27b440067af49e7e3bcc45d86eb7a17ae9db54f
CRs-Fixed: 435538
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Add extensions for customization support in rtsp stack.
Provide default implementations in AVMediaServiceUtils.
Change-Id: I67adeb54b35d1f01911625bb9bad27e94ad0caf0
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Change access modifiers and add overridables in rtsp stack.
Make ARTSPConnection/ARTPConnection extensible for IPV6 support.
Provide default implementations in AVMediaServiceExensions and
AVMediaServiceFactory.
Change-Id: Iaa67070d1832d56e0569dabfd8327c1998f04493
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The original RTSP seek implementation involves pausing and restarting
a session. This change clears data/eos status after an rtsp session
is paused for a seek, and delays the seek to return after data/eos
status are cleared.
Bug: 22207372
Change-Id: I1bdf65653f90436f7ee5d7fe85eeadc1598a0d56
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Bug: 17474566
Change-Id: I0dbd7a6a54edaf5b4fe5bd324d38f791a346b2fd
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bug: 17310253
Change-Id: I6ce8c4740a3509d82323ccc05f82cb842368caee
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Change-Id: I1b2f6b65c5abbc366068a60b8909104f31b94228
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Bug: 19607784
Change-Id: I94cddcb81f671422ad4982a23dc4acfe57a9f1aa
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We should come back and replace AString with std::string and switch to the
"real" StringPrintf family, but this fixes the ODR violation that was
preventing us from booting.
Bug: 19265750
Change-Id: I798eb9ca5dd634e44625af5e583439ef9f0cdc35
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Bug: 17556472
Change-Id: I0387c78727d9a18abddcfdb4b480f4b1412bbc9f
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* commit 'f1ac623fcc6bbda2faff9752cd611182a897afe1':
Implemented support for RTSP 301 Redirect
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RTSP 301 (Permament Redirect) support has been implemented.
Change-Id: If82ffc87f4e7dcbdf98e0a662a35cc086750fc1b
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* commit 'f4431278a9613f55ecd944ab2e3eb615b372f269':
Remove streaming URI from default logs
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Streaming URI should not be visible in default logcat logs
Change-Id: I104cc56b5335f8c5621013e4c5be8028f0379833
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Change-Id: Ie3bae3f037730e316d7fca12e7a3527973f752ef
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access unit" into klp-dev
* commit '58dd07863571951408b67fa0a7f17cb23606fb1c':
Send kWhatConnected in onTimeUpdate() before first access unit
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Bug: 10642588
Change-Id: If2b4fbbf250d5307e304f31c7aa4ac480e279484
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set."
* commit 'af66fae15f8c386ad884e5fa83db4eaef4c4f2ee':
Fix crash in MyHandler when sockets are not set.
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-When going quickly in and out of the video view during an rtsp
streaming session, a race condition occurs and MyHandler tries to
connect to a socket that has been reset. To avoid this,
checks are added.
- If there are errors during setupTrack 1, it is no use
setting up track 2. It will cause new errors.
- No assert for socket connect since there is a normal
status check already.
Change-Id: Ie06221d6c0d78ce0449f76c782ed5120fa646bfd
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Change-Id: I29171368f1b69333ef7eae53ada2fab94e3e28b9
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Mediaserver sockets are now routed as if the connection was in the
requesting app in per user routing.
Change-Id: I60f4649c3c4145a65264b54c1aa2c6c7741efaba
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Change-Id: I3d77b86f7e616af62a826fc37126706ad8ff6158
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If RTP accessunit comes earlier than play response,
the normal play time mapping posted in func onAccessUnitComplete is wrong.
This leads wrong timestamp of the first few frames.
This issue is found in the 3 CtsVerifier RTSP streaming cases.
Change-Id: I640eea375b1f3f4730238f9d561c3b40ec682395
Signed-off-by: Yajun Zeng <beanz@marvell.com>
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If DESCRIBE response is received with status 200 but no content,
MyHandler will still set content data for session description
parsing. This will cause NULL Pointer crash.
This fix checks whether DESCRIBE response has content before
parsing session description.
Change-Id: I114ae6fd54ce804e61718f62618ca9008425a433
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This fix handles problems with several asynchronous calls
within streaming. This case is when the phone has sent a
request to the server and while the response is being sent
back by the server the request is aborted by the user.
The fix is an if case that checks if we have aborted while
waiting for a response from the server. If we have aborted
we should ignore the late response instead of continuing.
Change-Id: I1264bb992f6abcaee1f10a89479e08b54a95e3c2
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and put the logic to create that string in one location instead of many...
Change-Id: I1f729f2e7376cd3b45eea0e48f7bd10084b41b39
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response"
* commit '59ac7b3056db57e5a8e851b7946a181c5fc34852':
Fix for crash if no content in DESCRIBE response
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If DESCRIBE response is received with status 200 but no content,
MyHandler will still set content data for session description
parsing. This will cause NULL Pointer crash.
This fix checks whether DESCRIBE response has content before
parsing session description.
Change-Id: I114ae6fd54ce804e61718f62618ca9008425a433
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If pausing an RTSP stream, an RTSP Pause request is sent and then
if the stream is immediately resumed again, an RTSP Play request
will be sent to the server.
But the new data after the pause will not be buffered until
Sender Reports have arrived again on both channels.
Meanwhile the player will resume playback and continue consuming
the already existing buffer.
This means that there is a risk that the buffer is emptied while
waiting for sender reports.
This commit simply adds a delay before the RTSP pause request is
sent, allowing some additional RTSP buffering that might be needed
when the stream is resumed again.
Also, if the stream is resumed again before the RTSP pause request
is sent, there is no need for any RTSP pause request, hence it is
omitted.
Change-Id: I928c8bfb5e99a6a146dcda4e51e528973ecbe065
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The fix takes care of several near end of stream use cases:
seek, pause and fake timestamps.
Change-Id: I5f5fa881b1f619dfd5e1afd2af957082345c59eb
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If a=control: is present at session-level in the SDP response,
RFC2326:C.1.1 defines the URL to be used for aggregate commands.
This includes PLAY and PAUSE but not TEARDOWN.
Change-Id: Iaa1dc2271d00df39dc83477a99fda6fbeb73c5b4
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When a stream is paused, RTSP Pause is also sent to the server.
Otherwise the buffering might continue until the memory runs out.
When the stream is resumed, RTSP Play will be sent in order to
resume the buffering.
Change-Id: I5dc1761140827c532451638c3fd3f34271e5b9ab
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Added buffering start and end notifications for RTSP.
MEDIA_INFO_BUFFERING_START is sent when buffering is started
and MEDIA_INFO_BUFFERING_END is sent when the buffer has
filled up.
This patch also adds RTSP end of stream handling.
EOS is signalled when BYE is received OR when
detecting end of stream even if no actual EOS is received.
Change-Id: I5cccb6845060ae6afd66d9f735b89da81476cd13
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has successfully completed and only then signals that preparation is
complete.
Change-Id: I1a60f718e673fe1462c69369c40eafbed6a14326
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The information is used to decide on visibility of pause button and
to handle the duration clock correctly.
Change-Id: I286ac992fd171c7fc313e429326d38b6fc80e3fb
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Added support for playing SDP files from http links. Previously,
SDP files only worked when started from rtsp links
(rtsp://a.b.c/abc.sdp), but they are just as common in http links.
patch provided by "Oscar Rydhé <oscar.rydhe@sonyericsson.com>"
Change-Id: Ic73af3a9a002009dbe8b04c267a4621bf7fe2f46
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This fix handles problems with several asynchronous calls
within streaming. This case is when the phone has sent a
request to the server and while the response is being sent
back by the server the request is aborted by the user.
The fix is an if case that checks if we have aborted while
waiting for a response from the server. If we have aborted
we should ignore the late response instead of continuing.
Change-Id: I1264bb992f6abcaee1f10a89479e08b54a95e3c2
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ABuffer objects through messages.
Change-Id: I9f8b4e4c4767d0d70a0105e0c0813b754379b49d
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i.e. the "SR" RTCP packet is sent for only one of the two tracks.
fake timestamps if that's the case, previously we'd only fake timestamps
if we didn't receive _any_ "SR" packets.
Change-Id: Id63d4940d453ba6c04c62e02ab9a0ad843936bc1
related-to-bug: 5669027
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See https://android-git.corp.google.com/g/#/c/157220
Bug: 5449033
Change-Id: Ic9c19d30693bd56755f55906127cd6bd7126096c
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See https://android-git.corp.google.com/g/157065
Bug: 5449033
Change-Id: I00a4b904f9449e6f93b7fd35eac28640d7929e69
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See https://android-git.corp.google.com/g/156801
Bug: 5449033
Change-Id: Ib08fe86d23db91ee153e9f91a99a35c42b9208ea
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make sure we emulate timestamps" into ics-mr1
* commit 'aa82c39bdb4ad9c1fdcb09f3bea11be5197d3ce6':
Fix Bitreader "putBits" implementation, make sure we emulate timestamps
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