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* Merge "Not all audio source has the drift time information" into gingerbreadJames Dong2010-09-033-12/+15
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| * Not all audio source has the drift time informationJames Dong2010-09-033-12/+15
| | | | | | | | Change-Id: I74e502376348ca4a6ffaa7492bed35c1355e7e62
* | Ogg files can be tagged to be automatically looping, this setting always ↵Andreas Huber2010-09-033-4/+16
| | | | | | | | | | | | | | overrides the MediaPlayer's setLooping setting. Change-Id: Ifb564c6cdf6137eac14869f9ca7d471f05a5556a related-to-bug: 2974691
* | Merge "Properly buffer a certain amount of data on streaming sources before ↵Andreas Huber2010-09-032-88/+119
|\ \ | |/ |/| | | finishing prepare()." into gingerbread
| * Properly buffer a certain amount of data on streaming sources before ↵Andreas Huber2010-09-032-88/+119
| | | | | | | | | | | | | | finishing prepare(). Change-Id: I39bf3c6dafcbe003b51dea4795742dcd8548f207 related-to-bug: 2875110
* | Remove unused/debugging code from MP4 file writerJames Dong2010-09-031-204/+47
| | | | | | | | | | | | o also makes nal length in the recorded file modifiable at runtime Change-Id: I731b4dde7070d8d9628b36b523a5b2c011c7c2cf
* | Better file size estimateJames Dong2010-09-021-34/+103
|/ | | | | | | | | When the recorded file becomes large, the metadata size can no longer be ignored. This makes it possible to save the recorded file when the storage becomes almost full at the end of the recording session. Change-Id: Ief038080f825c9946ce550949c03e914aec1e31a
* Calculate audio media drift time from AudioSourceJames Dong2010-09-015-31/+60
| | | | | | | | | | | | | | | | | | | The problem was that the time to receive an output buffer from an audio encoder is different because the encoder does not need to read from the source for all output buffers. This leads to large fluctuation in terms of wall clock duration between two neighboring audio sample outputs from the audio encoder. As a result, the media time for the video track after adjustment using the drifting changes wildly sometimes. This patch addresses this issue by only updating the media drift time when an audio source input buffer is read. the wall clock for the audio track is also calculated at the same time when the input audio buffer is read at AudioSource. bug - 2959800 Change-Id: I3174aa182f744784b540f0a7198524d4eee8bd7b
* Merge "Better support for buffered streaming of rtsp content, if buffer ↵Andreas Huber2010-09-015-2/+77
|\ | | | | | | drops below a certain threshold we will temporarily pause playback until we have sufficient data." into gingerbread
| * Better support for buffered streaming of rtsp content, if buffer drops below ↵Andreas Huber2010-09-015-2/+77
| | | | | | | | | | | | | | a certain threshold we will temporarily pause playback until we have sufficient data. Change-Id: Ice8564e902e48c89c9c00f6651c5504b3c41fcad related-to-bug: 2556656
* | Merge "Make sure that if initialization fails, AudioSource still behaves ↵James Dong2010-09-011-4/+17
|\ \ | |/ |/| | | well." into gingerbread
| * Make sure that if initialization fails, AudioSource still behaves well.James Dong2010-09-011-4/+17
| | | | | | | | Change-Id: I16dfc90bcb8a324d6ee9a38a5a1a31cc094c820a
* | Merge "Keep gtalk video chat specific code consistent with rtsp changes." ↵Andreas Huber2010-09-011-0/+8
|\ \ | | | | | | | | | into gingerbread
| * | Keep gtalk video chat specific code consistent with rtsp changes.Andreas Huber2010-09-011-0/+8
| | | | | | | | | | | | Change-Id: I5f3f46c2150e16b26674432e427f79c04a69cd8e
* | | Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer.Andreas Huber2010-09-012-5/+74
|/ / | | | | | | | | Change-Id: I15e21eae50beb6057024ea42a7e9bf3b8d8a0603 related-to-bug: 2368598
* | Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and ↵Andreas Huber2010-08-314-6/+345
|\ \ | | | | | | | | | AAC-hbr." into gingerbread
| * | Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr.Andreas Huber2010-08-314-6/+345
| |/ | | | | | | | | Change-Id: Ied92ea8c2448a2cb1a732c72c21c69da1913dbc8 related-to-bug: 2556656
* | Better detection of connection problems - timeout if no rtcp packets arrive ↵Andreas Huber2010-08-312-43/+67
|/ | | | | | | within a certain time, not a final frame (which may take longer) Change-Id: I3c1ae79bb9342770e959ebdcdc6b748549b76330 related-to-bug: 2556656
* Merge "Recent changes to the rtsp code require every buffer fed to the ↵Andreas Huber2010-08-301-0/+1
|\ | | | | | | packet source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder." into gingerbread
| * Recent changes to the rtsp code require every buffer fed to the packet ↵Andreas Huber2010-08-301-0/+1
| | | | | | | | | | | | source to have a timestamp, we're ignoring timestamps for gtalk videochat but we still have to have a placeholder. Change-Id: Ib8615ce5a89a9a846ee2f9f96cdfb23462f72c7a
* | Instead of closing the connection altogether if no UDP packets arrive after ↵Andreas Huber2010-08-302-32/+83
|/ | | | | | | a certain time, try changing transports (to interleaved TCP). Also properly close the sockets on disconnection. Change-Id: Ie8d6a3865a0477e28d4b76bb9038e468451287b1 related-to-bug: 2556656
* Finetune some rtsp timeout constants.Andreas Huber2010-08-301-3/+11
| | | | Change-Id: Ice731c5097c2a2dee8a7f0cd45b547cd34f532c6
* Merge "ALoopers can now be named (useful to distinguish threads)." into ↵Andreas Huber2010-08-305-4/+13
|\ | | | | | | gingerbread
| * ALoopers can now be named (useful to distinguish threads).Andreas Huber2010-08-275-4/+13
| | | | | | | | Change-Id: Ieabaddb2e3a9e3a7a5bc36e55cd0721b60dbd50e
* | Workaround for a QCOM issue where the output buffer size advertised by the ↵James Dong2010-08-271-0/+13
| | | | | | | | | | | | | | | | | | | | AVC encoder is occasionally too small. bug - 2882917 Change-Id: Id59d8529084c5689a26f272e0cd3b1e955fd8a30
* | Merge "Suppress the video recording start signal - bug 2950297" into gingerbreadJames Dong2010-08-271-0/+46
|\ \ | |/ |/|
| * Suppress the video recording start signalJames Dong2010-08-261-0/+46
| | | | | | | | | | | | - bug 2950297 Change-Id: I0044d07178691feb904cf81e87c1b6d4b714dc1a
* | Better support for rtsp (normal play-)time display. Better seek support, ↵Andreas Huber2010-08-2714-65/+302
| | | | | | | | | | | | | | timeout if no packets arrive for too long. Change-Id: Id491541a6ae501604cda815f8e961a3bfe26db7d related-to-bug: 2556656
* | We accidentally always aborted after 10 secs, even if the connection was fine.Andreas Huber2010-08-271-2/+1
|/ | | | Change-Id: I3f2ae2f46ae62b84b1e253658d7182c04ee3dfae
* Merge "Support for RTP packets arriving interleaved with RTSP responses." ↵Andreas Huber2010-08-266-21/+216
|\ | | | | | | into gingerbread
| * Support for RTP packets arriving interleaved with RTSP responses.Andreas Huber2010-08-266-21/+216
| | | | | | | | Change-Id: Ib32fba257da32a199134cf8943117cf3eaa07a25
* | Merge "Make sure that timestamp does not go backward in MP4 file writer" ↵James Dong2010-08-261-2/+2
|\ \ | |/ |/| | | into gingerbread
| * Make sure that timestamp does not go backward in MP4 file writerJames Dong2010-08-241-2/+2
| | | | | | | | Change-Id: I90745b9df7f19d61f3ab826bf9d2419fe788554e
* | Merge "Fix support for per-frame unsynchronization in ID3V2.4 tags." into ↵Andreas Huber2010-08-252-4/+85
|\ \ | | | | | | | | | gingerbread
| * | Fix support for per-frame unsynchronization in ID3V2.4 tags.Andreas Huber2010-08-252-4/+85
| | | | | | | | | | | | | | | Change-Id: I6874b596f88817347756a375d9fb1c9bff418eca related-to-bug: 2949149
* | | Merge "Ensure that buffering updates eventually hit 100% after we download ↵Andreas Huber2010-08-251-12/+19
|\ \ \ | |/ / |/| | | | | everything." into gingerbread
| * | Ensure that buffering updates eventually hit 100% after we download everything.Andreas Huber2010-08-251-12/+19
| | | | | | | | | | | | | | | Change-Id: I43bb85b1128fa9c1bc8632970d7101006393bcc8 related-to-bug: 2844095
* | | Allow sniffers to return a packet of opaque data that the corresponding ↵Andreas Huber2010-08-2516-30/+80
|/ / | | | | | | | | | | | | extractor can take advantage of to not duplicate work already done sniffing. The mp3 extractor takes advantage of this now. Change-Id: Icb77ae3ee95a69c7da25b4d3b8696c0a2d33028a related-to-bug: 2948754
* | A first shot at proper support for seeking of rtsp streams.Andreas Huber2010-08-247-59/+144
|/ | | | | Change-Id: I9604f2d09feedc0074c0e715be58e719d4483760 related-to-bug: 2556656
* Merge "Better handling of rtsp connection and disconnection." into gingerbreadAndreas Huber2010-08-233-14/+99
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| * Better handling of rtsp connection and disconnection.Andreas Huber2010-08-233-14/+99
| | | | | | | | Change-Id: Ib126af6c14c5a212a51a5ee3c4a0a7d1860ad167
* | Runtime dump support for MediaWriterJames Dong2010-08-231-0/+32
|/ | | | Change-Id: I10b2c474de612ee4cef4b7c9eae2ee1dd8c2e895
* Only add 4 bytes offset for the output media buffer when SPS is not received ↵James Dong2010-08-191-5/+10
| | | | | | for SW AVC encoder Change-Id: Ia64c2751b6304e5d5891416bf23ff9b8ec54d5ef
* Don't drop a late frame which may lead to missing I frames in the MP4 fileJames Dong2010-08-191-1/+15
| | | | Change-Id: I8fef1454264230c1369561670236eb0a19ae4e76
* Return error from MPEG4Writer stop() if the check on codec specific data failedJames Dong2010-08-191-0/+26
| | | | Change-Id: Icbd08eec9b4201facbad56ff2040f0830cfb0115
* Merge "Make MediaWriter stop and pause return errors if necessary" into ↵James Dong2010-08-194-34/+79
|\ | | | | | | gingerbread
| * Make MediaWriter stop and pause return errors if necessaryJames Dong2010-08-194-34/+79
| | | | | | | | | | | | | | | | | | | | | | o Make the API consistent with SF framework, which the MediaSource provides a return status for stop o Also, helps to convey errors that occurred right when a premature stop() is called, leading to a potentially mal-formed output file. Change-Id: I52a932345f38570fdf8ea04d67d73dd94ccd30ef
* | Support for MP4V-ES packetization format according to RFC3016.Andreas Huber2010-08-196-0/+417
| | | | | | | | Change-Id: I5e182936c52f9eb80cdcf6132ead03705ee32d61
* | In the absence of width/height information in the sdp, extract the ↵Andreas Huber2010-08-198-96/+133
|/ | | | | | dimensions from the avc codec specific data. Change-Id: I98c4194593c7e6e24f6fc339c862245111800293
* Merge "Better support for rtsp streamed through VLC. Temporarily make the ↵Andreas Huber2010-08-195-11/+58
|\ | | | | | | socket blocking to read all of the session description." into gingerbread
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