Currently able to play Ogg Vorbis, PCM WAV and other lossless files seamlessly
by reusing the initial AudioTrack for subsequent players.
Change-Id: Ie7cf6b9076bdf4f9211574456d192c02c04fecc7
Fix indentation to be multiple of 4.
Make it easier to search:
sp< not sp < to
"switch (...)" instead of "switch(...)" (also "if" and "while")
Remove redundant blank line at start or EOF.
Remove whitespace at end of line.
Remove extra blank lines where they don't add value.
Use git diff -b or -w to verify.
Change-Id: I966b7ba852faa5474be6907fb212f5e267c2874e
This affects:
- IAudioFlinger::openOutput
- AudioTrack::AudioTrack
- AudioTrack::set
- apps that call these
Change-Id: I26fb281bac6cb87593d17697bc9cb37a835af205
The 3rd parameter (param2) to AudioFlingerClient::ioConfigChanged
is used as an input. So changed it from void * to const void *.
It is then cast to const OutputDescriptor *
or const audio_stream_type_t * depending on the event.
Change-Id: Ieec0d284f139b74b3389b5ef69c7935a8e5650ee
Current AudioTrack implementation enforces that the requested audio
buffer size is at least corresponding the audio latency.
This requirement is too strong and leads to problems with current
stagefright and AudioSink implementations when playing over output
streams with long latency.
Ultimately, the AudioSink design should be changed to specify a minimum
buffer size in time or frames units but not in buffer count units.
Change-Id: I8ba603956f92ac49143a8249572665aa548f2f0f
This is a cherry-pick of I6ab07d89b2eeb0650e634b8c3b7a0b36aba4e7dd
with merge conflicts addressed by hand and additional changes made in
response to code review feedback.
Move in the direction of a more publishable API for configuring a
media player for retransmission. It used to be that we used a custom
invoke and a modified URL (prefixed with aahTX://). There are many
issues with this technique and it was never meant to stand the test of
time.
This CL gets rid of all that. A new (but currently hidden) method was
introduced to the java level MediaPlayer API, called
setRetransmitTarget(InetSocketAddress), which allows an app writer to
set the retransmit target. For now, this method needs to be called
before a call to setDataSource (which is pretty unusual for the
MediaPlayer API) because this mid level code uses this as a cue to
instantiate an aahTX player instead of relying on the data source to
select a player. When retranmit functionality becomes part of the
existing android player implemenation, this
set-retrans-before-set-data-source behavior can go away, along with
the aahTX player itself.
Change-Id: I3b46c5227bbf69acb2f3cc4f93cfccad9777be98
Signed-off-by: John Grossman <johngro@google.com>
prctl(PR_SET_NAME) limits to 15 characters. Before we had names like
"Binder Thread #" and the counter was cut off :-( Also remove redundant
"thread" at end of name; it's always a thread.
Change-Id: I1f99c2730ba0787ed9b59c15914356cddf698e2f
warning: pointer of type 'void *' used in arithmetic
warning: enumeral and non-enumeral type in conditional expression
Change-Id: I7b8d626a636145ef648e3b5d0e77068216dd012e
Bring the Visualizer class into line with the SDK documentation by
returning ERROR_DEAD_OBJECT instead of ERROR_INVALID_OPERATION when
the Visualizer loses its binder connection to the mediaserver because
of a mediaserver restart.
Also add a new callback interface to allow clients to be
asynchronously notified in the case of server death. Right now, the
interface definition and the registration method are flagged as hidden
pending API council review/approval.
See http://b/issue?id=5717519 for details.
Change-Id: Ic15856f27ed5a950a583ac11ca81f79bd7e9b1a0
Signed-off-by: John Grossman <johngro@google.com>
Bring in changes to audio flinger made to support timed audio tracks
and HW master volume control.
Change-Id: Ide52d48809bdbed13acf35fd59b24637e35064ae
Signed-off-by: John Grossman <johngro@google.com>
Unconditional delete for raw pointers.
Use "if (sp != 0)" not "if (sp.get() != 0)" or "if (sp != NULL)".
Use "if (raw != NULL)" not "if (raw)".
Change-Id: I531a8da7c37149261ed2f34b862ec4896a4b785b
Other:
- add a comment to nextUniqueId
- made ThreadBase::mId const, since it is only assigned in constructor.
Change-Id: I4e8b7bec4e45badcde6274d574b8a9aabd046837
This is just documentation, as C++ method const-ness doesn't mean anything
for a binder API. Instead, here const means "no side effects".
Change-Id: Iaa9cd2fe477db10ae9a40cac4f79f0faa9b4e5e6
The client callback threads had mutexes called AudioTrackThread::mLock
and ClientRecordThread::mLock. These mutexes were only used by start()
and stop(), and were unused by the thread itself. But start() and
stop() already have their own protection provided by AudioTrack::mLock
and AudioRecord::mLock. So the thread mutexes can be removed.
Change-Id: I098406d381645d77fba06a15511e179a327848ef
Also remove defaults in startToneCommand(), they're not needed and the
default for tone type was nonsense.
Change-Id: I70fa8cee4f3dbb8c66ceb3719c8d3d2f447f05b9
Was a mix of audio_source_t, uint8_t, and int.
Related fixes:
- fix comments in MediaRecorder.java
- AudioPolicyService server side was not checking source parameter at
all, so if the client wrapper was bypassed, invalid values could be
passed into audio HAL
- JNI android_media_AudioRecord_setup was checking source for positive
values, but not negative values. This test is redundant, since already
checked at Java and now checked by AudioPolicyService also, but might
as well make it correct.
Change-Id: Ie5e25d646dcd59a86d7985aa46cfcb4a1ba64a4a
Was int or uint32_t.
When AudioFlinger::format can't determine the correct format,
return INVALID rather than DEFAULT.
Init mFormat to INVALID rather than DEFAULT in the constructor.
Subclass constructors will set mFormat to the correct value.
Change-Id: I9b62640aa107d24d2d27925f5563d0d7407d1b73
