Commits · acb5621440968ddf62651a2e6c1ddb69cf675972 · e / devices / smdk4412 / android_frameworks_av · GitLab

Donate to e Foundation | Murena handsets with /e/OS | Own a part of Murena! Learn more

This project is mirrored from https://github.com/LineageOS/android_frameworks_av. Pull mirroring updated Aug 22, 2025.

Sep 09, 2010
- TimedEventQueue now explicitly sets its scheduling policy to foreground as it should. · acb56214
  Andreas Huber authored Sep 09, 2010
  acb56214
- Merge "Instead of asserting return a runtime error if the maximum sample size... · 31d2a4b0
  Andreas Huber authored Sep 09, 2010
  31d2a4b0
- Merge "When 32-bit offset is used, if the requested max file size is greater... · 4c73f1f9
  James Dong authored Sep 09, 2010
  4c73f1f9
- Instead of asserting return a runtime error if the maximum sample size cannot be determined. · 49110ce0
  Andreas Huber authored Sep 09, 2010
  49110ce0
- Instead of asserting, publish no tracks if an MP3Extractor is used on non-mp3 content. · 772bcc27
  Andreas Huber authored Sep 09, 2010
  772bcc27
- When 32-bit offset is used, · d2518e0c
  James Dong authored Sep 08, 2010
  d2518e0c
- Merge "HW audio encoder expects timestamp via kKeyTime from each input buffer" into gingerbread · fbf71624
  James Dong authored Sep 08, 2010
  
  fbf71624
- HW audio encoder expects timestamp via kKeyTime from each input buffer · 3c3763d2
  James Dong authored Sep 08, 2010
  3c3763d2
Sep 08, 2010
- Modify type of some environmental reverb parameters · 54c38fdc
  Eric Laurent authored Sep 08, 2010
  54c38fdc
Sep 03, 2010
- Merge "LVM release 1.08 delivery." into gingerbread · f9c0ae84
  Eric Laurent authored Sep 03, 2010
  
  f9c0ae84
- Merge "Not all audio source has the drift time information" into gingerbread · ddba3f04
  James Dong authored Sep 03, 2010
  
  ddba3f04
- LVM release 1.08 delivery. · 2d3bf535
  Eric Laurent authored Sep 02, 2010
  2d3bf535
- Ogg files can be tagged to be automatically looping, this setting always... · 8ae49d87
  Andreas Huber authored Sep 03, 2010
  8ae49d87
- Merge "Properly buffer a certain amount of data on streaming sources before... · 1a4c79ea
  Andreas Huber authored Sep 03, 2010
  1a4c79ea
- Properly buffer a certain amount of data on streaming sources before finishing prepare(). · 8650e196
  Andreas Huber authored Sep 03, 2010
  8650e196
- Not all audio source has the drift time information · caa68a57
  James Dong authored Sep 03, 2010
  caa68a57
- Remove unused/debugging code from MP4 file writer · b4d5320b
  James Dong authored Sep 02, 2010
  b4d5320b
- Better file size estimate · 1f90c4b3
  James Dong authored Sep 02, 2010
  1f90c4b3
- Merge "Calculate audio media drift time from AudioSource" into gingerbread · bd057752
  James Dong authored Sep 02, 2010
  
  bd057752
Sep 02, 2010
- Merge "Fix problem in AudioEffect::command() status." into gingerbread · 34c8d617
  Eric Laurent authored Sep 02, 2010
  
  34c8d617
- Fix problem in AudioEffect::command() status. · aeae3de9
  Eric Laurent authored Sep 02, 2010
  aeae3de9
- Calculate audio media drift time from AudioSource · d707fcb3
  James Dong authored Sep 01, 2010
  d707fcb3
Sep 01, 2010
- Merge "Better support for buffered streaming of rtsp content, if buffer drops... · 9b93478f
  Andreas Huber authored Sep 01, 2010
  9b93478f
- Merge "Make sure that if initialization fails, AudioSource still behaves well." into gingerbread · e91b4624
  James Dong authored Sep 01, 2010
  
  e91b4624
- Better support for buffered streaming of rtsp content, if buffer drops below a... · c9e89487
  Andreas Huber authored Sep 01, 2010
  c9e89487
- Make sure that if initialization fails, AudioSource still behaves well. · 6e20bdf7
  James Dong authored Sep 01, 2010
  6e20bdf7
- Merge "Keep gtalk video chat specific code consistent with rtsp changes." into gingerbread · bcbe5af6
  Andreas Huber authored Sep 01, 2010
  
  bcbe5af6
- Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer. · 82f7321b
  Andreas Huber authored Sep 01, 2010
  82f7321b
- Keep gtalk video chat specific code consistent with rtsp changes. · 389636ce
  Andreas Huber authored Sep 01, 2010
  389636ce
Aug 31, 2010
- Audio Effects: fix problems in volume control. · 8f45bd72
  Eric Laurent authored Aug 31, 2010
  8f45bd72
- Merge "Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr." into gingerbread · 06124758
  Andreas Huber authored Aug 31, 2010
  
  06124758
- Merge "Better detection of connection problems - timeout if no rtcp packets... · 69a4f8ba
  Andreas Huber authored Aug 31, 2010
  69a4f8ba
- Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr. · 4dba3e90
  Andreas Huber authored Aug 31, 2010
  4dba3e90
- Add some encoding parameters for the "record" utility · f74c8f9e
  James Dong authored Aug 31, 2010
  f74c8f9e
- Better detection of connection problems - timeout if no rtcp packets arrive... · e7d3e90d
  Andreas Huber authored Aug 31, 2010
  e7d3e90d
- fixedfft: Only includes cpu-features.h when __arm__ is defined. · 5edae619
  Chia-chi Yeh authored Aug 31, 2010
  5edae619
Aug 30, 2010
- Merge "Recent changes to the rtsp code require every buffer fed to the packet... · 5d5f5dfc
  Andreas Huber authored Aug 30, 2010
  5d5f5dfc
- Recent changes to the rtsp code require every buffer fed to the packet source... · b1860547
  Andreas Huber authored Aug 30, 2010
  b1860547
- Merge "Instead of closing the connection altogether if no UDP packets arrive... · e26cd862
  Andreas Huber authored Aug 30, 2010
  e26cd862
- Instead of closing the connection altogether if no UDP packets arrive after a... · 7aef0337
  Andreas Huber authored Aug 30, 2010
  7aef0337