Commits · 1a4c79eada496dc43918440608f15410eaecd28c · e / devices / smdk4412 / android_frameworks_av · GitLab

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

Sep 03, 2010
- 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
- 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
- Merge "Fix issue 2952766." into gingerbread · 44eb0968
  Eric Laurent authored Aug 30, 2010
  
  44eb0968
- Remove camera metering mode API. · 541d7655
  Wu-cheng Li authored Aug 27, 2010
  541d7655
- Merge "Reverse the default setting of media.stagefright.enable-{rtsp,record}... · 1c842b26
  Andreas Huber authored Aug 30, 2010
  1c842b26
- Reverse the default setting of media.stagefright.enable-{rtsp,record} in... · a1ffe49c
  Andreas Huber authored Aug 30, 2010
  a1ffe49c
- Merge "LVM release 1.07 delivery." into gingerbread · 38496993
  Eric Laurent authored Aug 30, 2010
  
  38496993
- Finetune some rtsp timeout constants. · e56121bc
  Andreas Huber authored Aug 30, 2010
  e56121bc
- Merge "ALoopers can now be named (useful to distinguish threads)." into gingerbread · c01ec022
  Andreas Huber authored Aug 30, 2010
  
  c01ec022
Aug 28, 2010
- Fix issue 2952766. · c1c88e27
  Eric Laurent authored Aug 27, 2010
  c1c88e27
Aug 27, 2010
- ALoopers can now be named (useful to distinguish threads). · a814c1fd
  Andreas Huber authored Aug 27, 2010
  a814c1fd
- Workaround for a QCOM issue where the output buffer size advertised by the AVC encoder · 824c9ff4
  James Dong authored Aug 26, 2010
  824c9ff4
- Merge "Suppress the video recording start signal - bug 2950297" into gingerbread · 352c4683
  James Dong authored Aug 27, 2010
  
  352c4683
- Merge "Better support for rtsp (normal play-)time display. Better seek... · f8860bf3
  Andreas Huber authored Aug 27, 2010
  f8860bf3
- Merge "We accidentally always aborted after 10 secs, even if the connection... · 165dc4c4
  Andreas Huber authored Aug 27, 2010
  165dc4c4
- Better support for rtsp (normal play-)time display. Better seek support,... · 8d342970
  Andreas Huber authored Aug 27, 2010
  8d342970