The headset state indicated by HeadsetObserver in the broadcast intent ACTION_HEADSET_PLUG was not 0 or 1 as specified in the java doc but contained a bit field indicating the type of headset connected.
Modified HeadsetObserver to broacast a state conforming to java doc.
Added an extra to intent ACTION_HEADSET_PLUG to indicate if headset has a microphone or not.
Removed handling of non standard headset indications from HeadsetObserver.
Removed platform specific devices from output devices defined in AudioSystem.
Modified AudioService to use new ACTION_HEADSET_PLUG intent extra instead of bitfield in state.
commit 08259dd3dc9026887f9bbfedaf45866eb56ea9bc
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 5 12:02:31 2009 -0800
DO NOT MERGE: Use PV for metadata extraction even if stagefright is used for playback.
commit 991832fe4dc012e51d3d9ed8d647c7f09991858f
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 5 11:24:11 2009 -0800
DO NOT MERGE: Do not assert if we encounter OMX_StateInvalid. All bets are off though.
commit cec45cf302d9218fe79956cbe8a462d7ca3a10bb
Author: Andreas Huber <andih@google.com>
Date: Mon Oct 26 16:11:54 2009 -0700
DO NOT MERGE: When freeing an OMX node, attempt to transition it from its current state all the way to "Loaded" in order to properly free any allocated buffers.
commit 34a1e885ef9113d68acbc26d36fcc47fdebbed84
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 5 11:10:49 2009 -0800
DO NOT MERGE: Fix heap corruptin in OMXNodeInstance.
commit 5a47f7439a1298b330541a7e4e647a8b44487388
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 5 11:08:19 2009 -0800
DO NOT MERGE: Fix seek-on-initial-read behaviour of OMXCodec.
commit 45bed64722501b9f411a2940aff5aff4cc4d2e98
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 5 11:02:23 2009 -0800
DO NOT MERGE: Renaming string.h to stagefright_string.h to avoid conflicts.
commit 6738e306a50196f31a73d4fc7b7c45faff639903
Author: Andreas Huber <andih@google.com>
Date: Thu Oct 15 13:46:54 2009 -0700
DO NOT MERGE: Reimplement the OMX backend for stagefright.
Besides a major cleanup and refactoring, OMX is now a singleton living in the media server, it listens for death notifications of node observers/clients that allocated OMX nodes and performs/attempts cleanup.
Changed APIs to conform to the rest of the system.
Merge commit '67b692920c18f99b096dce285adc6f7439fa866c' into eclair-mr2
* commit '67b692920c18f99b096dce285adc6f7439fa866c':
Fix issue 2203561: Sholes: audio playing out of earpiece.
Create a new IAudioTrack interface to AudioFlinger when start() fails due to a broken pipe error.
Do the same if start fails due to the same error after time out in obtainBuffer().
Do not indicate that the AudioTrack is started to AudioPolicyManager if IAudioTrack start fails.
This avoids that an AudioTrack keeps a dead IAudioTrack after a media server crash.
Same modifications for AudioRecord.
Add a flag to ToneGenerator indicating that the callback thread can call Java. Without it, when the media server crashes and restarts, the AudioSystem error callback will crash in JNI if the IAudiotrack is created from AudioTrack callback thread.
Merge commit '7ed0ceeba54712f76e9a4f2dd4c9197d76813488' into eclair-mr2
* commit '7ed0ceeba54712f76e9a4f2dd4c9197d76813488':
Add new audio sources to support the A1026 recording configurations.
Merge commit 'bf96aaadd46fb5b0884070177faa16ec4f22e2ba' into eclair-mr2
* commit 'bf96aaadd46fb5b0884070177faa16ec4f22e2ba':
Fix issue 2192181: AudioFlinger must provide separated methods to set VOICE_CALL stream volume and down link audio volume.
Also DataSources now must provide a method initCheck()
and DataSource::reat_at has been renamed to readAt to conform to
standard API naming guidelines.
While our hardware decoders clearly outperform the software decoders in terms
of raw throughput, their startup latency makes them less suitable for thumbnail
extraction.
Added setVoiceVolume() method to AudioSystem, AudioFlinger, IAudioFlinger, AudioPolicyService.
Removed call to AudioHardwareInterface::setVoiceVolume() from AudioFlinger::setStreamVolume().
Add a quirk mode to OMXCodec that makes it aware of this fact for proper display. Also integrate back a change from eclair-mr2 that delays releasing an output buffer briefly after posting it to surface flinger, as we don't know how long it'll take it to actually display the buffer's content.
Besides a major cleanup and refactoring, OMX is now a singleton living in the media server, it listens for death notifications of node observers/clients that allocated OMX nodes and performs/attempts cleanup.
Changed APIs to conform to the rest of the system.
This change is a complement to the main fix in kernel driver for the same issue (partner change #1250).
It removes clicks sometimes heard after the end of the tones while audio flinger is sending 0s to the audio output stream.
The problem was that the sleep time between two writes was more than the duration of one audio output stream buffer which could cause some underrun.
Also fixed a recent regression in ToneGenerator that made that the end of previous tone was repeated at the beginning of current one under certain timing circumstances when the maximum tone duration was specified.
This currently assumes 44k stereo (won't crash on other formats, but won't give the correct results either), and links statically with libspeex to get FFT data, increasing the size of libmedia by about 45kb.
When the AudioTrack callback notification size is relatively high (Which is the case on Sholes and over A2DP), it is likely that the end of tone is reached during the first callback. In this case, the AudioTrack is stopped before exiting the callback which causes 2 problems:
- 1: If the AudioFlinger thread is scheduled before we exit the ToneGenerator callback, the track can be stopped and reset before the data is actually marked as present in the buffer by the AudioTrack callback => no audio will be processed by AudioFlinger.
- 2: In this case, the data write index in the AudioTrack buffer is incremented after the track was reset by the AudioFlinger which leaves unplayed data in the buffer. This data will be played the next time the AudioTrack is started if not flushed in between.
The fix consists in adding an intermediate state to ToneGenerator state machine so that we exit the callback function when the stop condition is reached and stop the AudioTrack the next time we execute the callback.
This api allows to instantiate a renderer by specifying the hosting java Surface object. This hides the implementation details of (java-)Surface, (native-)Surface and friends.
Add a parameter to ToneGenerator.startTone() allowing the caller to specify the tone duration. This is used by the phone application to have a precise control on the DTMF tone duration which was not possible with the use of delayed messaged.
Also modified AudioFlinger output threads so that 0s are written to the audio output stream when no more tracks are ready to mix instead of just sleeping. This avoids an issue where the end of a previous DTMF tone could stay in audio hardware buffers and be played just before the beginning of the next DTMF tone.
