| 78bd91b15ee8ea5aa2ab5a8cad7e892cb2d01c1b |
|
26-Feb-2016 |
Lajos Molnar <lajos@google.com> |
stagefright: set pcm-encoding for raw audio tracks
Bug: 27172163
Change-Id: I4db7ace4e8cbf01939dedefbb1c54730e393530f
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 742a9baf0fbecd01a27b8741ab370552d1ab1771 |
|
09-Feb-2016 |
Lajos Molnar <lajos@google.com> |
stagefright: Remove unused hooks from AudioPlayer
These were used by AwesomePlayer.
Bug: 17108024
Change-Id: I9fbea10172628d80e93451d9b38d245fd3f3b20f
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| e4d40ad91f3516bde106658673b2cb076b5dfcb7 |
|
13-Jan-2016 |
Lajos Molnar <lajos@google.com> |
stagefright: Remove deprecated AwesomePlayer
Bug: 17108024
Change-Id: Ia721bdfa87b07612ad8e76cbdcda6de2b8ce2a74
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| b2487f03f12dcafdb801fc0007c8df8412397f44 |
|
01-Sep-2015 |
Marco Nelissen <marcone@google.com> |
Extractor service
Run extractors in a separate process. Currently all data is copied through a
binder transaction, and WVMExtractor is still run in the mediaserver process.
Change-Id: Ic5dbce87126dd321ad792f4dd540c2ff6b068d13
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 3a474aa67fc31505740526dd249d96204c08bf79 |
|
25-Apr-2015 |
Lajos Molnar <lajos@google.com> |
stagefright: support setting/getting playback/sync config in MediaSync
Bug: 18249558
Bug: 19666434
Bug: 20057497
Change-Id: I5868b17423d7c20cfaf4a399f3eb67bfba440605
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| e125bba94afe74df330c661ab0a64b1e9e2ac5b7 |
|
19-Jun-2014 |
aarti jadhav-gaikwad <agaikwad@nvidia.com> |
stagefright: Return correct play position after EOS
In case of offload getOutputPlayPositionUs_l returns
zero on reaching EOS. Return duration as position.
Bug: 15770459
Change-Id: I7aeafdc315a86ccd197d555d604e1069eead3153
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 2829edccd7d2bb8244246f316face82b650b8949 |
|
18-Jun-2014 |
aarti jadhav-gaikwad <agaikwad@nvidia.com> |
stagefright: Pass audio aac sub formats in offloadinfo
Pass audio aac sub formats in offloadinfo according to
aac profile. Audio HAL can take decision about offload
using DSP capabilities
Change-Id: If269a3654b5d2b09c183212b0646ef03e06f2d8f
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| a5750e0dad9e90f2195ce36f2c4457fa04b2b83e |
|
19-Jun-2014 |
Mark Salyzyn <salyzyn@google.com> |
libstagefright: 64-bit compile warnings
Change-Id: I3d1146714fa23be3d4e696599b6f70cac1f9d28b
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 2301acc6a9c7a3af4ad01f3d1d0f76f13eca7350 |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Update comments
Change-Id: I5776313b9b49072cd666d28880f0d07cc73f827b
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 84333e0475bc911adc16417f4ca327c975cf6c36 |
|
08-Feb-2014 |
Andreas Huber <andih@google.com> |
warnings be gone.
Change-Id: Ie3bae3f037730e316d7fca12e7a3527973f752ef
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 6f59db12a64f4496866952a251122ccb77a36c6b |
|
27-Jul-2013 |
Eric Laurent <elaurent@google.com> |
update offloaded audio track sampling rate
AudioPlayer must read the sampling rate from offloaded audio sinks
whenever a new time position is computed as the decoder can update
the sampling rate on the fly.
Change-Id: I997e5248cfd4017aeceb4e11689324ded2a5bc88
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 91b0ca1a5bea44dd9b5196910186dd2927821994 |
|
27-Sep-2013 |
Eric Laurent <elaurent@google.com> |
fix playback position after switching to offload
After switching from offloaded track to PCM track
while paused (e.g. when connecting A2DP), playback
restarts from the beginning of the song when resuming.
Save current position before recreating an AudioPlayer
in AwesomePlayer::play_l() and seek to the saved position before
starting playback.
Also fix a problem where the position is not reported properly
by AudioPlayer if a seek is pending and queried just after start
and before the first buffer is read from the MediaSource.
Bug: 8174034.
Change-Id: I254e65418ff903a9bf2e2111b89a00e2e54876c5
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 50d9a8f7de5f79fa8a36489a53846d6653997e38 |
|
05-Aug-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
AudioPlayer: timestamp fixes for compress offload
Use realtime queried from AudioTrack as the only time for compress
offload playback.
Change-Id: I314447637715c4864690c94173260165369146cb
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 94ea60f975c3eb7ce6d2a4430538a42a5fc3babd |
|
14-May-2013 |
Richard Fitzgerald <rf@opensource.wolfsonmicro.com> |
stagefright: offload playback support
Offloading of compressed audio decoding to audio DSP
is implemented for audio only, non streamed content.
when the datasource is
AudioPlayer:
- Create an offloaded sink when playing a compressed
source
- Send metadata to audio HAL
- Return sink start error to AwesomePlayer so that a
new player for PCM audio can be created in case of problem.
- Forward stream end and tear down callback events to AwesomePlayer
- Stop the sink and wait for stream end callback when EOS is reached.
- Pause and restart the sink if needed before flushing when seeking
(otherwise flush is a no op).
- For current media time, directly query the render position from the
sink and offset by the start position (seek to time)
AwesomePlayer:
- When initializing the audio decoder, check with audio policy manager
if offloading is supported. If yes, create the software decoder in
case a reconfiguration is needed but connect the audio track directly
to the AudioPlayer.
- In case of error when starting the AudioPlayer, reconnect the software
decoder (OMXSource) and recreate a PCM AudioPlayer.
- Handle AudioPlayer tear down event by detroying and recreating the
AudioPlayer to allow transitions between situations were offloading
is supported or not.
- Force tear down of offloaded AudioPlayer when paused for a certain time:
This will close the sink and allow the DSP to power down.
Utils:
- Added helper methods:
- send meta data to audio ia sink setParameters
- query audio policy manager if offloading is supported for a
given audio content
Change-Id: I115842ce424f947b966d45e253a74d3fd5df9aae
Signed-off-by: Eric Laurent <elaurent@google.com>
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| ad3af3305f024bcbbd55c894a4995e449498e1ba |
|
25-Mar-2013 |
Richard Fitzgerald <rf@opensource.wolfsonmicro.com> |
Public API changes for audio offload support.
NOTE: this does _not_ include all private member variables added
to classes as part of offload support. Only public/protected functions
and stubs functions/variables needed to make the changes buildable.
- isOffloadSupported() added to audio policy service
A stub implementation is required to build, this always returns false
- setParameters() added to IAudioTrack
A stub implementation is required to build, this always returns
INVALID_OPERATION
- CBlk flag for stream end
- Change AudioSystem::getRenderPosition() to take an audio_output_t
so caller can specify which output to query
- Add AudioSystem::isOffloadSupported()
This is fully implemented down to the AudioFlinger function
AudioPolicyServer::isOffloadSupported() which is just a stub
that always returns false.
- Add EVENT_STREAM_END to AudioTrack interface.
STREAM_END is used to signal when the hardware has actually finished
playing all the data it was sent.
- Add event type enumeration to media player interface AudioSink callbacks
so that the same callback can be used to handle multiple types of
event. For offloaded tracks we also have to handle STREAM_END and
TEAR_DOWN events
- Pass audio_offload_info_t to various functions used for opening outputs,
tracks and audio players. This passes additional information about the
compressed stream down to the HAL when using offload.
For publicly-available APIs this is an optional parameter (for some of
the internal and low-level APIs around the HAL interface it is mandatory)
- Add getParameters() and setParameters() API to AudioTrack
Currently dummy implementations.
- Change AudioPlayer contructor so that it takes a set of bitflags defining what
options are required. This replaces the original bool which only specified
whether to use deep buffering.
- Changes to StageFright class definition related to handling tearing-down of
an offloaded track when we need to switch back to software decode
- Define new StageFright utility functions used for offloaded tracks
Currently dummy implementations.
- AudioFlinger changes to use extended audio_config_t.
Fills in audio_offload_info_t member if this info is passed in when
opening an output.
- libvideoeditor changes required to add the new event type parameter
to AudioSink callback functions
- libmediaplayerservice changes required to add the new event type parameter
to AudioSink callback functions
Change-Id: I3ab41138aa1083d81fe83b886a9b1021ec7320f1
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Eric Laurent <elaurent@google.com>
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 2799d743ee2ae5a25fe869a7f9c052acc029559f |
|
30-May-2013 |
Glenn Kasten <gkasten@google.com> |
Use sp<AudioTrack> instead of raw AudioTrack *
This change prepares for the new implementation of AudioTrack client, which
will require clients to use only sp<AudioTrack>, not raw AudioTrack *.
A raw delete will cause a race condition during AudioTrack destruction.
AudioTrack was made a RefBase by commit b68a91a70bc8d0d18e7404e14443d4e4020b3635
on 2011/11/15, when it was needed by OpenSL ES (for the callback protector).
At that time, the only other client that was also converted from
AudioTrack * to sp<AudioTrack> was android.media.AudioTrack JNI in
project frameworks/base (file android_media_AudioTrack.cpp).
Details:
* Use .clear() instead of delete followed by = NULL.
* ALOG %p need .get().
* sp<> don't need to be listed in constructor initializer, if initially 0.
* Use == 0 for sp<> vs == NULL for raw pointers.
* Use if (sp != 0) instead of if (raw).
Change-Id: Ic7cad25795d6e862e112abdc227b6d33afdfce17
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 4575beb3dea80e271eaa6619234fdc02e914e6e6 |
|
22-May-2012 |
Andreas Huber <andih@google.com> |
Revert "Stop audio source if AudioPlayer could not be started"
This reverts commit 022048e95f3f1d65cb776cf21de2001423fcf7d9.
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 022048e95f3f1d65cb776cf21de2001423fcf7d9 |
|
17-May-2012 |
James Dong <jdong@google.com> |
Stop audio source if AudioPlayer could not be started
related-to-bug: 6500580
Change-Id: If47e442db2f1ea400e9b598502bb48513f28cd6d
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| a99a5bca365277271915cbaeea811ad87131270d |
|
16-May-2012 |
Andreas Huber <andih@google.com> |
Stop extrapolating audio time from the anchor if we're paused or reached eos.
Change-Id: I55903ba625d9c075a9a6e7bcd5ff95d450b915b2
related-to-bug: 6479613
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| e4451a91a61a341014f5eff61db356156c3ecb37 |
|
15-May-2012 |
Andreas Huber <andih@google.com> |
Smoothen audio "real" time by compensating with system_time()
for the delay since the last buffer submission to the audio sink.
Change-Id: Ib1a8672f96ba8bbef07d23372f1076fbb1f0bdcc
related-to-bug: 6479613
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| b48aea123775fb6ec7e6944826fc99da8d8996a5 |
|
04-May-2012 |
Eric Laurent <elaurent@google.com> |
Revert "Merge "Improve AudioPlayer position reporting""
This reverts commit 887c5d2ac34d81bde66dadbd7b2ff07501744778, reversing
changes made to 5aaeb0d64fc98f9b019a4378eb39f0ee49ee6ec4.
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 2c3297ab6c4daaaa7b27eed8418c64cf168fe2a1 |
|
01-May-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
No warning on using channel order in AudioPlayer
Don't log a warning when an AudioPlayer derives a channel mask
from the channel count, only info-log when channel count is
higher than 2.
Change-Id: Ia1cbced660903483654eede300ef72f7f17dcf78
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 1948eb3ea6eee336e8cdab9b0c693f93f5f19993 |
|
14-Apr-2012 |
Eric Laurent <elaurent@google.com> |
Add support for deep audio buffers
Allow AudioSink to use deep audio buffering when the
source is audio only and its duration is more than
a certain threshold.
This helps improve battery life but implies higher
audio latency.
Change-Id: Ie79915b61c370292f05aabda9779356570e03cbb
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 0ca3cf94c0dfc173ad7886ae162c4b67067539f6 |
|
18-Apr-2012 |
Eric Laurent <elaurent@google.com> |
rename audio policy output flags
Change-Id: I27c46bd1d1b2b5f96b87af7d05b951fef18a1312
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| ab334fd351ae5a0e18903da123d63e565b536874 |
|
14-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
audio_channel_in/out_mask_from_count
and avoid ambiguous term "channels" where it might be confusing
as to whether it is a channel mask or channel count
Change-Id: I744fa08ccb6001a98c97bd638d2c9d56836c4234
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 36a4d2cc7a44eb42d0d9ef5fe50b6747a7a34cab |
|
13-Mar-2012 |
Marco Nelissen <marcone@google.com> |
Merge "Gapless playback, step 1."
|
| 6b74d671a1321a6ecc4a40b6c87beedfecc1ec44 |
|
29-Feb-2012 |
Marco Nelissen <marcone@google.com> |
Gapless playback, step 1.
Currently able to play Ogg Vorbis, PCM WAV and other lossless files seamlessly
by reusing the initial AudioTrack for subsequent players.
Change-Id: Ie7cf6b9076bdf4f9211574456d192c02c04fecc7
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 18868c5db2f90309c6d11e5837822135e4a0c0fa |
|
07-Mar-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_policy_output_flags_t consistently
This affects:
- IAudioFlinger::openOutput
- AudioTrack::AudioTrack
- AudioTrack::set
- apps that call these
Change-Id: I26fb281bac6cb87593d17697bc9cb37a835af205
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| e7d0c712f1c9fa0b0e413b8eb729049995290aee |
|
09-Mar-2012 |
Marco Nelissen <marcone@google.com> |
Improve AudioPlayer position reporting
The latency was not taken into account when updating mPositionTimeRealUs
inside of the fillBuffer hook, contrary to what the getRealTimeUsLocked()
method does. This caused the realTimeOffset calculated in the
getMediaTimeUs to always be negative, causing the reported position to
always be equal to mPositionTimeMediaUs, which is updated infrequently.
With this change, the reported position is updated more frequently, allowing
apps to perform smoother UI updates.
Change-Id: I61e05c1a8b53d46b9091afb0d18a6289d13a7a5e
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 6556e5bb99bda64f3ab92e8b540fccdd59e9ed8d |
|
06-Mar-2012 |
Eric Laurent <elaurent@google.com> |
Merge "MediaPlayerService: fix AudioSink latency"
|
| db354e58e65592777aa17caa47933e14838b8b35 |
|
06-Mar-2012 |
Eric Laurent <elaurent@google.com> |
MediaPlayerService: fix AudioSink latency
The AudioSink latency is currently cached when the associated AudioTrack
is created. However, the AudioTrack latency can change if the AudioTrack is moved
from one output stream to another.
The AudioPlayer must also periodically update its view of the latency
as it is needed to compensate the real audio time used for A/V sync.
This fixes an A/V sync problem seen when switching A2DP on and off while
playing a video.
Change-Id: I28b24049ca114e1af3e24791dcc900f463536ba4
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 786618ffe881aceb64d65a6a2e2d76ede6e01ec0 |
|
02-Mar-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Add channel mask in AudioSink
Add support for specifying a channel mask when opening an AudioSink.
This parameter does not replace the channel count parameter in order
to not have to duplicate the logic to derive a mask from the
channel count everywhere an AudioSink is used without a known mask.
A mask of 0 (CHANNEL_MASK_USE_CHANNEL_ORDER) means a mask will
be automatically derived from the number of channels.
Update existing AudioSink implementations to use the channel mask,
and users of AudioSink to specify the mask if available, and
CHANNEL_MASK_USE_CHANNEL_ORDER otherwise.
Change-Id: Ifa9bd259874816dbc25ead2b03ea52e873cff474
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 7a8b0ed6419e57dd8b41d3806893d63d3df91aab |
|
02-Feb-2012 |
Jean-Michel Trivi <jmtrivi@google.com> |
Playback rate on MediaPlayer
Add support for modifying the playback rate of a MediaPlayer
by altering the sample rate of its AudioTrack.
The playback rate is expressed in permille, where 1000 is the
playback at normal speed.
Change-Id: I981d060ab32f7bae7a767e82c60c88ae635dceed
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 3856b090cd04ba5dd4a59a12430ed724d5995909 |
|
20-Oct-2011 |
Steve Block <steveblock@google.com> |
Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/143865
Bug: 5449033
Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 6c375490a5536695770a85b57ec9f828d3c0c9d4 |
|
28-Sep-2011 |
James Dong <jdong@google.com> |
Flush resets the count of played audio frames
Change-Id: I0cda94646a71757fa113a199d29235e13efd7ac3
related-to-bug: 5331470
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| f7eade99250520f2c9c8366a20a9256c4b34abc1 |
|
08-Sep-2011 |
Andreas Huber <andih@google.com> |
Audio player must not be used as a timesource before it's started.
Otherwise it'll trigger a division-by-zero exception since the audio sample rate
is as yet unknown.
Change-Id: I0793aa7c1c348ffa2611272bb646eff6ecf6ff53
related-to-bug: 5242451
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 0b293e76c8fe4e973ccd8a872bc5320ba28d49cc |
|
26-Aug-2011 |
Andreas Huber <andih@google.com> |
Properly reset the number of frames we think we've written to the audio sink
in order to compute the final delay before posting end-of-stream correctly.
Change-Id: I4ff6d7142fdf1ea2d19d12ec78c1e84322d22e5f
related-to-bug: 5216930
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| d217a8c4632b3e3065f8c2a26b9ce4dc4c97171f |
|
02-Jun-2011 |
Glenn Kasten <gkasten@google.com> |
Revert "Use AudioTrack::event_type not int in callback"
This reverts commit 2225e4b7049fa3fb9d39a068b8268b63c952d7c1
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 013b5a67d6f8e12a1b06e74863daec1bee4d99c1 |
|
05-May-2011 |
Glenn Kasten <gkasten@google.com> |
Use AudioTrack::event_type not int in callback
Change-Id: Ifefc708d46874e04fd0d01cb6e2d43b987ee796c
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| c0dfc5b02d4179769bbdd25c10d430576ec09568 |
|
10-May-2011 |
Andreas Huber <andih@google.com> |
Properly honour any pending seek request when reading the first buffer after
starting the audio player.
Change-Id: Ibedc1804a8c444e4d0f5be4eb87b22ed5d2c367d
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| fce7a473248381cc83a01855f92581077d3c9ee2 |
|
20-Apr-2011 |
Dima Zavin <dima@android.com> |
audio/media: convert to using the audio HAL and new audio defs
Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 7fb4b427d5fca79c09d69b212ddca41539c215a2 |
|
04-Apr-2011 |
Andreas Huber <andih@google.com> |
Merge "This reverts an earlier change 4aeec2da39fce4de9b07b694d59fd308ffb3d491"
|
| 51c1e0e86a0ad95bf3d890a9a2f51e54b8ef9444 |
|
04-Apr-2011 |
Andreas Huber <andih@google.com> |
This reverts an earlier change 4aeec2da39fce4de9b07b694d59fd308ffb3d491
Instead of returning 0-filled buffers after EOS from AudioTrack we do this
work in AudioOutput instead. That way the EOS signal (0 frames returned)
is preserved in AudioCache which otherwise would lead to a heap size overflow
filling everything with zeroes.
Change-Id: I7e07429ba887957a4340dd4b21eef4bba76248cd
related-to-bug: 3514073
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| bd7b7177f88ae6e83bd7bb8bfd9b7018be923931 |
|
04-Apr-2011 |
Andreas Huber <andih@google.com> |
Avoid a race condition while signaling audio EOS/seek complete by acquiring the lock.
Change-Id: Idfea452ff661b264e013fe515bcd2a2762bd959c
related-to-bug: 4193552
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| b1d3d37c7990d22d0659e4c8c9aac56d4c09b6ea |
|
28-Mar-2011 |
Andreas Huber <andih@google.com> |
am a4a81aef: am e7cedad2: am 2edbdcc6: Merge "After a seek there\'s a brief period where audio time isn\'t reestablished." into honeycomb-mr1
* commit 'a4a81aef39b15649bda146476d98f23ac554887f':
After a seek there's a brief period where audio time isn't reestablished.
|
| fe9b71919cdddf898a516169db840751878098ca |
|
26-Mar-2011 |
Andreas Huber <andih@google.com> |
After a seek there's a brief period where audio time isn't reestablished.
Don't use audio time to trigger a "we're very late, skipping ahead" action
until it's reestablished.
Change-Id: Ia8534e5bed60203d49a457d14b2e55eb972b5fa1
related-to-bug: 4173683
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 84b343f29063fbfa2ee61b2e3d37ba059ca507d4 |
|
22-Mar-2011 |
Andreas Huber <andih@google.com> |
Delay signaling the end of audio playback until all frames have actually played.
Change-Id: I1fa07358a885a818fd0a5d7da425740f86095e10
related-to-bug: 3404000
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 6e1cc6e6e5477693a279185c372b90863778a10b |
|
26-Jan-2011 |
Andreas Huber <andih@google.com> |
Instead of not feeding the AudioTrack any data after EOS, play silence instead.
Change-Id: I9aa797cf5bb58eefed51d732f4e5d09450551158
related-to-bug: 3347428
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| b2e3954c94717e43b3dc9b880564f166cfbbc0a2 |
|
05-Oct-2010 |
Andreas Huber <andih@google.com> |
Make sure to call AudioTrack::stop() instead of AudioTrack::pause() after submitting all samples to AudioTrack to make sure those remaining samples are actually played out.
Change-Id: Id574a0203efcb5e565f1b0fe77869fc33b9a9d56
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| c7d368d990303dc5369c7c61579f88c5059dc8d7 |
|
28-Sep-2010 |
Andreas Huber <andih@google.com> |
Properly flush the AudioTrack/AudioSink on a seek request and make sure that both the mp3 decoder and aac software decoders start fresh after a seek without any dependency on previously decoded content.
Change-Id: I4909fdf19518dbabb6c340e2a31b50dfe6c5b067
related-to-bug: 3029947
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| ed54ad0f8619ae416b0968ade6248894cbfc4dba |
|
28-Sep-2010 |
Andreas Huber <andih@google.com> |
Instead of constantly polling the AudioPlayer to see if it reached EOS or finished seeking, initiate the notification from the AudioPlayer when the event happens.
Change-Id: I43875b6adaf96d4e982ef3dfc3d6c8f7034ac51d
related-to-bug: 3036592
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 3cc219dfc67b866e10828f0c17641668d47c1cd8 |
|
20-Jul-2010 |
Andreas Huber <andih@google.com> |
Squashed commit of the following:
commit 4abf16bb04dc9695fedf4007a84f903074312ccd
Author: Andreas Huber <andih@google.com>
Date: Tue Jul 20 09:21:17 2010 -0700
Support a single format change at the beginning of audio playback. This way the AAC+ decoder may change its output format from what is originally encoded in the audio stream and we'll still play it back correctly.
Change-Id: Icc790122744745e9a88099788d4818ca1e265a82
related-to-bug: 2826841
commit 09c74da63e6ad5cb5dafb70f62696d75d2978967
Author: James Dong <jdong@google.com>
Date: Sun Jul 18 17:57:01 2010 -0700
Fix MPEG4Extractor to extract sampling frequency correctly when SBR is enabled.
Change-Id: I883c81dad3ea465e71cb5590e89d763671a90ff8
commit f672bf2a782dc7d5fb6325d611a7fe17045dfe9a
Author: James Dong <jdong@google.com>
Date: Thu Jul 8 20:56:13 2010 -0700
Enable the support for decoding audio with AAC+ and eAAC+ features
bug - 282684
Change-Id: I73c8377af3cc4edd3ee7cea86dc3b1c369fbd78b
Change-Id: I012f1179e933b6d1345d2368f357576c722485f7
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| a98420e863c374d1f15309467f2a1fc58d979d3b |
|
13-Apr-2010 |
Andreas Huber <andih@google.com> |
Better diagnostics in the stagefright commandline tool, support for playing sine-tones using a filename of "sine:[samplingrate]", i.e. sine:44100. Support for playing audio through to the speakers by using "-o" in addition to "-a".
related-to-bug: 2553359
Change-Id: Ic49eb89feb62474d9513bc21571e0e8fe3a15e0b
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| e7e3b785a0e7819db4c895a4f60e9a4dd755880c |
|
09-Mar-2010 |
Andreas Huber <andih@google.com> |
The audio track was accidentally not participating in the prefetch since it wasn't started at the time prepare() was called. Also, properly report the cached duration even near the end when the source has no more data to fetch.
Change-Id: I66a92fec24c9bfb25f1c186f1c877127bae2b4f9
related-to-bug: 2444425
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 25155ff8ccf7898d08ab62fae46297e046a571f0 |
|
01-Mar-2010 |
Andreas Huber <andih@google.com> |
Ensure that the audio codec instance is actually going away before we attempt to instantiate another one, otherwise we might end up with the software codec even though the hardware codec would have been available shortly.
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 5295c0c55d41a2906ea7f65a3f22e6278cb17d4b |
|
23-Feb-2010 |
Andreas Huber <andih@google.com> |
Propagate errors all the way through the MediaSources and send either MEDIA_PLAYBACK_COMPLETE or MEDIA_ERROR depending on the final reason for running out of buffers to play back.
related-to-bug: 2463749
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 32dcebf970356eb8599ea965d71535ed0c212c2d |
|
18-Feb-2010 |
Kenny Root <kroot@google.com> |
Store AudioTrack return value for error condition
Store AudioTrack->initCheck()'s return value so we can return it after
the AudioTrack is deleted.
Change-Id: Ie042a7d6a54d7f9afd8a5767cfd167dbeb2e5c04
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 7d5b8a70c28c0d5746a600467b2887822dbff88e |
|
10-Feb-2010 |
Andreas Huber <andih@google.com> |
Implementation of stagefright-enabled MediaPlayerService::decode functionality
related-to-bug: 2359268
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| cb9859bcf5f00cee57de06e9968b88a69b2d6d9c |
|
05-Feb-2010 |
Gloria Wang <gwang@google.com> |
Fix for 2224736. Sometimes mSeeking is reset without calling setSeekTo first.
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 80a68deec52c9a1b47215ed68526206ec88197f8 |
|
04-Feb-2010 |
Andreas Huber <andih@google.com> |
Previous change broke A/V sync, this is a better change to eliminate negative media positions.
related-to-bug: 2414900
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 1862a33b246249630b654182afb5914da3480d4c |
|
03-Feb-2010 |
Andreas Huber <andih@google.com> |
Fix a deadlock caused by the AudioPlayer notifying the observer that the stream had ended at the same time the observer was shutting down the AudioPlayer.
related-to-bug: 2414536
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| ad6516d9247bd0a76a393810b041c3e4094f0e36 |
|
03-Feb-2010 |
Andreas Huber <andih@google.com> |
AudioPlayer didn't properly account for the latency when calculating the current media time.
related-to-bug: 2414900
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 88c030e0e0152791ff74f90249f55fce01371198 |
|
29-Jan-2010 |
Andreas Huber <andih@google.com> |
Properly forward errors returned by AudioTrack up to the player.
related-to-bug: 2300197
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| fc9ba09e3bb368f823d473f5e2bb9aa32dba6289 |
|
12-Jan-2010 |
Andreas Huber <andih@google.com> |
Squashed commit of the following:
commit f81bb1dac5ef107bb0d7d5d756fb1ffa532ba2cc
Author: Andreas Huber <andih@google.com>
Date: Mon Jan 11 14:55:56 2010 -0800
Support for duration metadata, midi and ogg-vorbis files (in mediascanner)
commit 0b1385a0dc156ce27985a1ff757c4c142fd7ec39
Author: Andreas Huber <andih@google.com>
Date: Mon Jan 11 14:20:45 2010 -0800
Refactor meta data logic. Container specific metadata is now also returned by the MediaExtractor.
commit f9818dfac39c96e5fefe8c8295e60580692d5990
Author: Andreas Huber <andih@google.com>
Date: Fri Jan 8 14:26:09 2010 -0800
A first pass at supporting metadata through ID3 tags.
commit 476e9e253633336ab790f943e2d6c0cd8991d76a
Author: Andreas Huber <andih@google.com>
Date: Thu Jan 7 15:48:44 2010 -0800
Initial checkin of ID3 (V2.2 and V2.3) parser for use in stagefright.
related-to-bug: 2295456
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| bfa6b2d7a1be1832ac40ed90aece1834f720b5c6 |
|
20-Nov-2009 |
Andreas Huber <andih@google.com> |
Squashed commit of the following:
commit 1efc38dc3c33fef57b759002db3965ed07a28cb0
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 19 14:36:14 2009 -0800
Sending the SEEK-COMPLETE notification temporarily broke seeking backwards in time behaviour. This is now fixed.
Also, get rid of the semi-random delay after posting buffers to surface flinger in favour of delaying the buffer release until the next frame is displayed.
commit 51973062eb5ee63fd64b845d72bac517cc3369cf
Author: Andreas Huber <andih@google.com>
Date: Wed Nov 18 14:01:43 2009 -0800
Fix one more unit test, properly send seek-complete notification only after seek actually completed.
commit cb22250b34b1fcfe1bf459723a761fd003950229
Author: Andreas Huber <andih@google.com>
Date: Wed Nov 18 12:31:36 2009 -0800
Fix seek-while-paused in AwesomePlayer, revert to using FileSource if MmapSource fails.
commit 25eb9241138ddf7bb27ce90657116c5f8a94d880
Author: Andreas Huber <andih@google.com>
Date: Wed Nov 18 12:30:40 2009 -0800
Support seeking and duration in AMRExtractor, assuming all frames are the same size.
commit 44192f2ebb7ea3bbd3ba5910025692dbc6a08faa
Author: Andreas Huber <andih@google.com>
Date: Wed Nov 18 10:21:44 2009 -0800
MediaPlayerImpl is dead, long live AwesomePlayer.
commit c5b52d3c0674f5dc94db506afbce52401cceddac
Author: Andreas Huber <andih@google.com>
Date: Wed Nov 18 09:42:23 2009 -0800
New implementation of the stagefright mediaplayer.
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 84ec55b736ae2278d6abdc1c7a0312de46ef5f3d |
|
06-Nov-2009 |
Andreas Huber <andih@google.com> |
am fde273ba: Merge change Ic997acac into eclair
Merge commit 'fde273baf6592a5fde15ee4dd4d798d90368e440' into eclair-mr2
* commit 'fde273baf6592a5fde15ee4dd4d798d90368e440':
Make AudioPlayer a little less verbose, defer starting audio playback until after the first video frame has been decoded (if there's video at all).
|
| dae04ca7c4b5590786ffc336721ee8714cc79fef |
|
06-Nov-2009 |
Andreas Huber <andih@google.com> |
Make AudioPlayer a little less verbose, defer starting audio playback until after the first video frame has been decoded (if there's video at all).
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 48c948b1137e7bbdb161b51908657ab72ac5e2da |
|
08-Oct-2009 |
Andreas Huber <andih@google.com> |
Change to a int64_t usecs representation for timestamps and duration throughout stagefright.
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 18291bc20e55e8f3fd5feb786771a8ed32c19c59 |
|
10-Sep-2009 |
Andreas Huber <andih@google.com> |
Media MIME types now have corresponding constants in MediaDefs.h, new codec enumeration API.
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 0c89199745bc1bf05b997fc7c342017807676b6f |
|
26-Aug-2009 |
Andreas Huber <andih@google.com> |
assert => CHECK in stagefright.
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 693d271e62a3726689ff68f4505ba49228eb94b2 |
|
14-Aug-2009 |
Andreas Huber <andih@google.com> |
Squashed commit of the following:
commit 5bb012f0065f7ffaaeb4f569d71f0e3a8d6b19c3
Author: Andreas Huber <andih@google.com>
Date: Fri Aug 14 10:40:08 2009 -0700
An attempt at fixing export using the qcom encoders. More quirks.
commit 0690e76bfa48118a68287ccf1bbfa82febaa620c
Author: Andreas Huber <andih@google.com>
Date: Fri Aug 14 09:08:28 2009 -0700
Callbacks are now dispatched from a separate thread in OMX.
commit c6571a039526df29b6343f9a1971dbc019088c61
Author: Andreas Huber <andih@google.com>
Date: Thu Aug 13 15:42:25 2009 -0700
Massive API changes throughout stagefright, smart pointers everywhere.
commit 900612af6a0555664d9ba195112cd859491265f4
Author: Andreas Huber <andih@google.com>
Date: Thu Aug 13 13:33:12 2009 -0700
OMXCodecs now properly shutdown.
commit 96732f05e1b0603dcd1b11f16a23512592eeb4f5
Author: Andreas Huber <andih@google.com>
Date: Thu Aug 13 12:04:04 2009 -0700
More work on JPEG decoding using the hardware OMX component.
commit 63839a073ac393e3a130434ba467969053b694ad
Author: Andreas Huber <andih@google.com>
Date: Wed Aug 12 13:13:31 2009 -0700
An attempt to drive the JPEG decoder OMX node.
commit 3ac2fe5ab2926eda81b2123610b2434c645294ff
Author: Andreas Huber <andih@google.com>
Date: Tue Aug 11 16:38:21 2009 -0700
Renamed StateMachine to OMXCodec and put it in its proper place.
commit 247da75a96bf8881956413023dd49a84d5b4f5b2
Author: Andreas Huber <andih@google.com>
Date: Tue Aug 11 16:06:19 2009 -0700
Statemachine is now a full-fledged MediaSource.
commit 045244f6771fa0b9b329495c953afda900a84b71
Author: Andreas Huber <andih@google.com>
Date: Fri Aug 7 09:16:54 2009 -0700
Properly setup the input format when exporting to AMR audio.
commit 271b984cb32c5cd9e46e3f90ae121f334e4b8da9
Author: Andreas Huber <andih@google.com>
Date: Thu Aug 6 09:59:38 2009 -0700
Added some code to test audio encoding to the OMX harness.
commit 79af4748e4af33bd66d3fbac606e332a69741cf4
Author: Andreas Huber <andih@google.com>
Date: Wed Aug 5 14:36:22 2009 -0700
Merge the old OMXDecoder and the new, shiny, StateMachine code.
commit 91cf5dd77a8762bc10a0b2ffce35e3bbeb262231
Author: Andreas Huber <andih@google.com>
Date: Tue Aug 4 17:41:43 2009 -0700
A new harness to test OMX node compliance (and quirks).
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 0024245e134467d120b40099da16c467dc365e76 |
|
28-Jul-2009 |
Andreas Huber <andih@google.com> |
Squashed commit of the following:
commit 374ea382ee3a9e3ce17e4c6357fc40d02e362810
Author: Andreas Huber <andih@google.com>
Date: Tue Jul 28 09:54:13 2009 -0700
PV's OMX implementation now uses (spec-compliant) microseconds instead of milliseconds in buffer timestamps.
commit 8d02f8ab5d7b022ad4ad34db2a9bdeea6ce2acfe
Author: Andreas Huber <andih@google.com>
Date: Mon Jul 27 14:24:26 2009 -0700
Support for using an overlay for video playback on TI hardware.
Appears to be currently constrained to CbYCrY 16-bit colorspace.
commit d17f321cb4b15c1fea378f33a7ef5998f23dd0fc
Author: Andreas Huber <andih@google.com>
Date: Mon Jul 27 09:45:38 2009 -0700
Added '--audio-only' commandline option to stagefright tool.
commit d8beef6be5c668c46451446d87e622933371cd75
Author: Andreas Huber <andih@google.com>
Date: Fri Jul 24 13:35:00 2009 -0700
Generalize the various workarounds for OMX nodes with their own unique interpretation of the spec.
commit c7dfd53eeadf8ed5a39bf2b19b014dcd62f3324d
Author: Andreas Huber <andih@google.com>
Date: Thu Jul 23 16:06:36 2009 -0700
Fixed erroneous TI AAC decoder behaviour wrt shutdown.
The AAC decoder appears to not return out buffers on an Executing->Idle transition, implemented a workaround that does a flush on all ports followed by the Executing->Idle. Oh, and flush with OMX_ALL doesn't properly work either. Fun.
commit d6377282e75060881336578f166f9c7feacf3f8f
Author: Andreas Huber <andih@google.com>
Date: Thu Jul 23 14:06:50 2009 -0700
Apparently the "channels" parameter in AudioTrack's constructor no longer counts channels but is a bitmask of enabled destinations, update the code accordingly.
commit ff698c79e851a2e57d362e9c3a09828af4048087
Author: Andreas Huber <andih@google.com>
Date: Wed Jul 22 16:54:44 2009 -0700
Fix MPEG4 decoding using TI's hardware decoder that I broke earlier...
commit 2ef78bb87cd856eb7f0b3d7dd68782a8650c12bf
Author: Andreas Huber <andih@google.com>
Date: Wed Jul 22 15:43:18 2009 -0700
Now that the qcom decoder properly advertises its own custom colorspace, update dependent code to reflect this fact.
commit bbaec96910727080fd7c8a6907c04facb9f5220f
Author: Andreas Huber <andih@google.com>
Date: Wed Jul 22 14:32:03 2009 -0700
Finishing up previous, incomplete change.
commit 76f14a1ae816b6f434771f8d12bdad81196f351e
Author: Andreas Huber <andih@google.com>
Date: Wed Jul 22 14:25:17 2009 -0700
The TI video decoder now properly decoder AVC/H.264 content.
commit e106130d8c100d5c94603e43864a7a93cca10252
Author: Andreas Huber <andih@google.com>
Date: Wed Jul 22 08:56:04 2009 -0700
Experimental support for the TI H.264 decoder, various modifications to OMXDecoder, non-functional still.
commit 241c3062dec3447db1a1ee74558cb4b9098fc404
Author: Andreas Huber <andih@google.com>
Date: Tue Jul 21 12:13:09 2009 -0700
Enable TI hardware OMX decoders (except for AVC/H.264 which still has issues).
This particular set of OMX nodes does not appear to properly return our buffers when sending the "disable" command on a port. Rather it reqires manually flushing that port and _then_ disabling it instead.
commit 1c34506a46e32ce25f2a86f3b4250dcfc037356a
Author: Andreas Huber <andih@google.com>
Date: Tue Jul 21 08:51:35 2009 -0700
Make it simpler to switch between the stagefright player and PVPlayer.
commit 249c6de05671d403f8dd51f095d49bf190430c9c
Author: Andreas Huber <andih@google.com>
Date: Mon Jul 20 14:38:15 2009 -0700
Prepare to use soon-to-be-available hardware decoders in the OMX decoder.
/frameworks/av/media/libstagefright/AudioPlayer.cpp
|
| 20111aa043c5f404472bc63b90bc5aad906b1101 |
|
15-Jul-2009 |
Andreas Huber <andih@google.com> |
Squashed commit of the following:
commit 8d179995654c19d9094e51989dc84ed0deba6e04
Author: Andreas Huber <andih@google.com>
Date: Tue Jul 14 16:30:40 2009 -0700
Protect stagefright code with BUILD_WITH_STAGEFRIGHT build-define.
commit 99b41acb58c34e8f024c30c833a09fd7b31c4e1f
Author: Andreas Huber <andih@google.com>
Date: Tue Jul 14 14:06:00 2009 -0700
OMX::list_nodes API, stagefright --list now lists all available OMX components.
commit 94a74c5b24aae6d7e446fff83d62ae3a57dc351b
Author: Andreas Huber <andih@google.com>
Date: Mon Jul 13 16:04:29 2009 -0700
Reenabled "record" commandline application to test encoding, support for h263 codec specific data when writing to .mp4.
commit d363dd74f611577567c66d25900d6264c306468c
Author: Andreas Huber <andih@google.com>
Date: Mon Jul 13 10:59:58 2009 -0700
Make non-socket communication in IOMX the default.
commit e89fc82c4fd138c88339887baf72f1d0e67f8660
Author: Andreas Huber <andih@google.com>
Date: Mon Jul 13 10:32:33 2009 -0700
The OMX interface now optionally supports communication through the binder only (no sockets involved)
commit 28611de11223fba65299787cdd33f7fdbd6a60a3
Author: Andreas Huber <andih@google.com>
Date: Fri Jul 10 15:09:13 2009 -0700
Split off the logic to render to a Surface (not ISurface!) into SurfaceRender.{cpp,h}
commit c0cfa885faaeb17d3fc5436f6447b61708bd6fe1
Author: Andreas Huber <andih@google.com>
Date: Fri Jul 10 12:21:46 2009 -0700
Fix display dimensions vs. decoded dimensions display issues, unfortunately breaks software rendering a bit...
commit ae209a2b5d09caed5dd383c910f238926aaa8fc6
Author: Andreas Huber <andih@google.com>
Date: Fri Jul 10 09:33:29 2009 -0700
Simple video decoder performance test in "stagefright" commandline tool.
commit 59fe7840111d6f46dd3c48f4950146aed962dbd2
Author: Andreas Huber <andih@google.com>
Date: Thu Jul 9 16:05:12 2009 -0700
StagefrightPlayer: Properly publish video dimensions by notifying the listener.
commit 51b6598f0e47be5e12d77f1d14e111e71c593638
Author: Andreas Huber <andih@google.com>
Date: Thu Jul 9 11:38:15 2009 -0700
MediaPlayerInterface::AudioSink now supports the specification of a callback for pull-based audio.
commit 31ce53375d748da3e7d30e993ce89dae97fd65ea
Author: Andreas Huber <andih@google.com>
Date: Wed Jul 8 17:23:37 2009 -0700
Proper support for pushing audio to a specified AudioSink interface. Still not bulletproof: video falls behind frequently.
commit a23e312b10631a5d47f5c9831205beac6b84ab36
Author: Andreas Huber <andih@google.com>
Date: Wed Jul 8 15:10:13 2009 -0700
IGenesisPlayer and friends are no more, java interface now uses local playback instead
of playing inside the media server process.
commit 42ee5c06ab5ba1c16f50472b0b26a0526635c73c
Author: Andreas Huber <andih@google.com>
Date: Wed Jul 8 13:17:03 2009 -0700
MP3 extractor now supports seeking (only fixed bitrate files currently supported)
commit 96a790405763bfa738d1c6845c99a9fdf579b66f
Author: Andreas Huber <andih@google.com>
Date: Tue Jul 7 10:53:54 2009 -0700
Keeping up-to-date with new master-gl changes, removing CameraSource for now, removing unnecessary commandline utilities from the build.
commit 6c9dcb7c628c0347154dd7f37913a3a5f04deb99
Author: Andreas Huber <andih@google.com>
Date: Mon Jul 6 16:40:02 2009 -0700
Finally take advantage of the fact that the Qcom OMX decoders already place the output buffers into memory accessible by the hardware yuv converter and avoid extra allocations and memcpys in that case. Split of the render logic into subclasses of VideoRenderer.
commit e8d46e81cffc15ffb0bdf03f511e4a4497ca7621
Author: Andreas Huber <andih@google.com>
Date: Mon Jul 6 12:18:29 2009 -0700
Added a few more cases of valid MPEG4 files to the sniffer, fixed end of stream notifications in MPEG4 files.
commit 88bffc5ed4033a607080f8777a1c0ec9e3e079de
Author: Andreas Huber <andih@google.com>
Date: Thu Jul 2 12:25:03 2009 -0700
MediaPlayerImpl now properly pauses/resumes instead of tearing down the decoder chains and starting back up from the beginning.
commit 732aa0a87c7e8d8a10776b275712649954962414
Author: Andreas Huber <andih@google.com>
Date: Wed Jul 1 14:58:56 2009 -0700
Remove pthread_mutex_* and pthread_cond_* in favour of Mutex and Condition.
commit 669e5944b2fe0542a3b8dba0c441a94503e99fb0
Author: Andreas Huber <andih@google.com>
Date: Wed Jul 1 14:43:02 2009 -0700
Proper support for end-of-stream signalling of MediaSources.
commit afb038f2dc594c742e554b2d42b86ffc7434e3fc
Author: Andreas Huber <andih@google.com>
Date: Tue Jun 30 16:48:03 2009 -0700
Support for the "sniffing" of datasources to determine their probable mime type (along with confidence). Slightly refactored the MP3Extractor to share common code with the sniffer.
commit ed3644b8bd56807e1db4b32433a93a2467f1f3c5
Author: Andreas Huber <andih@google.com>
Date: Fri Jun 26 13:27:52 2009 -0700
I don't think prepare should automatically start the player.
commit 11eb9b49dc4a0819d366682817cba68613f4fe12
Author: Andreas Huber <andih@google.com>
Date: Fri Jun 26 11:09:22 2009 -0700
Somewhat better timing information implemented in the MP3Extractor.
commit cab023a1751735c0ed26c8493498a91d8644c6c9
Author: Andreas Huber <andih@google.com>
Date: Thu Jun 25 15:41:56 2009 -0700
Initial checkin for support of mp3 content, extraction and playback. Media extractors now abstracted in MediaExtractor baseclass.
commit 40f4146bb3c647e56690dd9b0fa3ffdea60030dc
Author: Andreas Huber <andih@google.com>
Date: Wed Jun 24 10:41:12 2009 -0700
Make sure we only allocate surface buffers if there is video to be played.
commit 7f85e7d92fe81740e102e09905b19354f4ac178e
Author: Andreas Huber <andih@google.com>
Date: Wed Jun 24 10:19:32 2009 -0700
Clean up a failed merge and implement stub for new API in MediaPlayerBase.
commit f0e9d494a4c03402ea7b54891848ca5a50c421b2
Author: Andreas Huber <andih@google.com>
Date: Fri Jun 12 15:38:03 2009 -0700
Support yuv420->rgb565 conversion if hardware color conversion is unavailable.
commit 87223248b3099cd5261b94169f84cfd1eba56a8f
Author: Andreas Huber <andih@google.com>
Date: Fri Jun 12 09:35:56 2009 -0700
Remove unimplemented and currently unnecessary blocking option from MediaSource::ReadOptions.
commit e07e651f0325892b797f3eb64dbeca0f317afdfd
Author: Andreas Huber <andih@google.com>
Date: Wed Jun 10 16:26:42 2009 -0700
Enable assertions in HTTPDataSource.
commit 0dc97adebea1721156880f4b3d5c70f8d50c53b2
Author: Andreas Huber <andih@google.com>
Date: Wed Jun 10 12:34:39 2009 -0700
delete foo.
commit 7f05845dda496e7defee284d32509d127e785784
Author: Andreas Huber <andih@google.com>
Date: Wed Jun 10 12:34:08 2009 -0700
added foo.
commit e432970c51eeec4e5d549bdb3e2d3e48ffb4f0e7
Author: Andreas Huber <andih@google.com>
Date: Wed Jun 10 11:44:31 2009 -0700
Make sure to initialize all fields in _all_ constructors... Use a Mutex instead of the raw pthread_mutex_t.
commit d6286a6bbca639952491448fa3e609b775a8a814
Author: Andreas Huber <andih@google.com>
Date: Tue Jun 9 17:11:49 2009 -0700
send_command is now implemented twice... ugly hack to get both shutdown working correctly and startup as well... playback now "works" on the emulator, too.
commit 8d657e004677a7c5dae4479e8519bcaa94c18aad
Author: Andreas Huber <andih@google.com>
Date: Tue Jun 9 11:58:55 2009 -0700
Instead of erasing codec specific data once it's been output to the omx node, retain the data and just rewind after a stop.
commit 279fede6ba35cbc37c35616c53406bf2c9720202
Author: Andreas Huber <andih@google.com>
Date: Tue Jun 9 10:58:11 2009 -0700
Various fixes related to play/pause. Decoder now properly frees OMXMediaBuffers, made MemoryDealer destructor protected instead of public to prevent accidental stack-allocation of MemoryDealer objects.
commit 0f717f15a6328efd3f34082929cd8ad2351095a9
Author: Andreas Huber <andih@google.com>
Date: Tue Jun 9 08:44:12 2009 -0700
Make sure to rewind codec specific data on OMXDecoder::stop and reset AudioPlayer's notion of time mapping.
commit 0cfb63e33270daf9b215214608ab383f93a31823
Author: Andreas Huber <andih@google.com>
Date: Mon Jun 8 14:25:48 2009 -0700
Massive rename.
commit 30ed8f476c4431a5275538958b3e281bd17d5eab
Author: Andreas Huber <andih@google.com>
Date: Mon Jun 8 13:52:41 2009 -0700
send_command and fill/empty buffer commands must be executed in sequence, send_command is now dispatched over the socket instead of through the binder interface to ensure this.
commit b832d04011b2dd8a9632eb7e575bfe1addd1270d
Author: Andreas Huber <andih@google.com>
Date: Mon Jun 8 10:21:17 2009 -0700
MediaBuffers can now be "clone()", they sit on top of the same data but have their own range and metadata. Once a clone's refcount drops to 0 it decrements the refcount on the original buffer.
commit 4e944672808bdf1ed40a69f3f602f93ebf7b4049
Author: Andreas Huber <andih@google.com>
Date: Mon Jun 8 10:01:01 2009 -0700
Split MediaBuffer.h into MediaBufferGroup.h and MediaBuffer.h, same with their respective source files.
commit ddb7529b4de4828db51432bd405fddd76cfa3b6a
Author: Andreas Huber <andih@google.com>
Date: Fri Jun 5 16:30:25 2009 -0700
More fudging with A/V sync...
commit 1e5a4d8df08237b58c06c6527a33152ae931addf
Author: Andreas Huber <andih@google.com>
Date: Fri Jun 5 11:54:20 2009 -0700
API cleanup of MediaSource. Got rid of MediaRing as it really doesn't serve a purpose if OMX decoders are used.
commit 0953c65232aa9753d17d2e25c5f5526eb810cf13
Author: Andreas Huber <andih@google.com>
Date: Thu Jun 4 16:59:19 2009 -0700
Experimental TimedEventQueue and an attempt at a cleaner player implementation using it, OMXDecoder is now a little less verbose.
commit 2085b8ea9484a21fa11f94194a27405d413aeecc
Author: Andreas Huber <andih@google.com>
Date: Thu Jun 4 10:17:01 2009 -0700
Properly bail out of the AV sync loop if the MediaPlayerImpl is stopped.
commit 471f143458a8a0d93cb1bf9568c63c3967279dc6
Author: Andreas Huber <andih@google.com>
Date: Wed Jun 3 17:32:27 2009 -0700
Now this is much better AV sync, defer seeking the audio source until we know where the video source synced to.
commit 4010d371dbe62ce4971e13437776cba32cb0bb3d
Author: Andreas Huber <andih@google.com>
Date: Wed Jun 3 16:01:07 2009 -0700
Tweaked some of the heuristics in AV sync, OMXDecoder is now a little less verbose, seek to a sync sample at or _after_ the requested position.
commit f3d4af1d5a4ac31fa02271b000f6c7ec593f0da9
Author: Andreas Huber <andih@google.com>
Date: Wed Jun 3 15:33:58 2009 -0700
Change the TimeSource interface's APIs, prefer microseconds over milliseconds now, changed the way MediaPlayerImpl does AV sync.
commit 08bb46fa61d12e852a424698ddb0c90700328f6e
Author: Andreas Huber <andih@google.com>
Date: Wed Jun 3 11:57:24 2009 -0700
Using framework's Mutex and Condition classes instead of pthread functions, much saner locking in OMXDecoder.
commit 4d77b4e09e59bbf8c1af009ae7dc62b10af52fd6
Author: Andreas Huber <andih@google.com>
Date: Tue Jun 2 17:12:03 2009 -0700
MediaSources now publish their format.
commit 7e429a26c57bf936e1a08035d2d11839f9f00310
Author: Andreas Huber <andih@google.com>
Date: Tue Jun 2 16:38:29 2009 -0700
MetaData is now implemented in terms of KeyedVector, MetaData is now reference counted.
commit 0fa957aa5c9bcb6ec122ed7b5b97430039be6797
Author: Andreas Huber <andih@google.com>
Date: Mon Jun 1 14:39:55 2009 -0700
Sync sample seek support.
commit 78a5f090d02e7e6e1a146b29bdb7771eecaa2f86
Author: Andreas Huber <andih@google.com>
Date: Mon Jun 1 13:51:11 2009 -0700
Distinguish shutdown from port-flush since one requires buffers to be freed and the other one doesn't. Fixes a problem with seek.
commit e57781d3b2c985a74cd8b60e7c8f2d576459d038
Author: Andreas Huber <andih@google.com>
Date: Mon Jun 1 10:52:20 2009 -0700
First shot at support for seeking a MediaSource. Sync frames are not properly handled yet.
commit 0551cd78fda78bf66877fb26c5a55e5ae23d8367
Author: Andreas Huber <andih@google.com>
Date: Thu May 28 16:23:38 2009 -0700
Preliminary support for camera input.
commit f8ebafa65d10d5a06944de6b2e2992a33121b857
Author: Andreas Huber <andih@google.com>
Date: Thu May 28 12:13:47 2009 -0700
Allow NativeMediaPlayer access to Surface's getISurface() method.
commit b1a9fb36eeccd9f49d28c1c64b31fccbd6506eeb
Author: Andreas Huber <andih@google.com>
Date: Thu May 28 11:38:25 2009 -0700
Header files are now in their own genesis2 hierarchy below frameworks/base/include.
commit f49b8b234b8f6ce83148787eef540dd920f04aa6
Author: Andreas Huber <andih@google.com>
Date: Thu May 28 11:22:03 2009 -0700
moved MPEG4Writer into libgenesis2.
commit 5224254919dac4728677335ea60429fa16f518b4
Author: Andreas Huber <andih@google.com>
Date: Thu May 28 11:13:38 2009 -0700
Everything now lives in the android namespace, integrated errors with android errors, renamed MediaPlayer to MediaPlayerImpl.
commit f9a9d84f7e0465ef5d1938a068655ba39527cf99
Author: Andreas Huber <andih@google.com>
Date: Wed May 27 16:43:11 2009 -0700
Support for getPosition() and getDuration().
commit 3206cc5140bcbc88be039f8d94b954af63e69752
Author: Andreas Huber <andih@google.com>
Date: Wed May 27 15:37:45 2009 -0700
Support for {MediaPlayer, GenesisPlayer, Genesis2Player}::isPlaying()
commit 9f41b248e0dfce71bbc50f883750cec52ebfd1fe
Author: Andreas Huber <andih@google.com>
Date: Wed May 27 13:49:50 2009 -0700
Proper disconnection from the OMX service.
commit cdcf47a2c31599109ee42277c7fa14af3c600091
Author: Andreas Huber <andih@google.com>
Date: Tue May 26 17:27:06 2009 -0700
New player type for the genesis player in libmediaplayerservice. Playing video through an ISurface now.
commit 0586af20ac9ae0403dd190bafd290b0f90549723
Author: Andreas Huber <andih@google.com>
Date: Fri May 22 15:53:19 2009 -0700
Fix a buffer overflow error by adding padding in case we need to de-frame AVC data.
commit eac62fff6ad2ffb7b75c0d4a1b9620d33e93fb9b
Author: Andreas Huber <andih@google.com>
Date: Fri May 22 13:35:30 2009 -0700
Support for http URIs in MediaPlayer, compensate for AudioTrack latency.
commit a7db456e3173d94d6b5e457547e2a6a5f7a8e1cc
Author: Andreas Huber <andih@google.com>
Date: Fri May 22 11:33:25 2009 -0700
Support for URIs in MediaPlayer class, support for shoutcast streams in MediaPlayer, force shutdown of OMX nodes if they don't shutdown cleanly within a second.
commit 1f0268f4f14ad5215668f34e06a420850174d665
Author: Andreas Huber <andih@google.com>
Date: Fri May 22 08:33:16 2009 -0700
and foo.txt is gone again.
commit 0feca4e0919f345704d78ec78894bac5027bd5ed
Author: Andreas Huber <andih@dhcp-172-19-27-148.mtv.corp.google.com>
Date: Fri May 22 08:28:49 2009 -0700
added foo.txt
commit 9c7d04b35cdc895a2de3602dc46522f608bd205f
Author: Andreas Huber <andih@google.com>
Date: Fri May 22 08:28:04 2009 -0700
delete foo again
commit fec3ca1720992cce51b9a92b86b945da88a6b16c
Author: Andreas Huber <andih@google.com>
Date: Fri May 22 08:27:45 2009 -0700
weird
commit 30a6571302333146e537e537a3a8986245cca87c
Author: Andreas Huber <andih@dhcp-172-19-27-148.mtv.corp.google.com>
Date: Fri May 22 08:18:55 2009 -0700
reverse test
commit dea3d99dd0fe9386e8a3c51954d9d0fe2be09d38
Author: Andreas Huber <andih@google.com>
Date: Fri May 22 08:18:22 2009 -0700
test
commit a9eeafec6e46072cac41e54ebdbd153cb7ae7518
Author: Andreas Huber <andih@google.com>
Date: Thu May 21 16:37:50 2009 -0700
Use the video ring again for now.
commit e05ff3d3c1c4f89ef499a2b172138a775fd694e3
Author: Andreas Huber <andih@google.com>
Date: Thu May 21 15:42:19 2009 -0700
Export to .mp4 now works for MPEG4 video content. Fixed a bug in the sample table wrt co64 chunks.
commit 5c44c540374a4e50c4e8aaab2b7a34aed6e1391e
Author: Andreas Huber <andih@google.com>
Date: Thu May 21 09:56:11 2009 -0700
Fix the build, all binder-related headers have moved and there's a new library libbinde
r to link against now.
commit 23e0a24404d16c898c2c6b3c7495ad303844bc71
Author: Andreas Huber <andih@google.com>
Date: Thu May 21 08:42:04 2009 -0700
Initial checkin of MPEG4Writer, a class to export media sources into an mpeg4 file. MPEG4/AVC metadata (efss & friends) not currently working.
commit 9cd77adb1ac194197364ffad00a0d7215f458e1a
Author: Andreas Huber <andih@google.com>
Date: Wed May 20 12:31:45 2009 -0700
New shell command to test recording/encoding. Various changes to OMXDecoder to support encoding properly. Now each OMXDecoder needs a dedicated thread on the client side since onEvent responses could block which would cause deadlocks if a node depends on the output of another node for example.
commit 5c510fb55ffed7ec25305999ed9b6c1eec48d503
Author: Andreas Huber <andih@google.com>
Date: Mon May 18 15:40:06 2009 -0700
Make media decoding the default in the genesis2 commandline app instead of shoutcast playback.
commit f6c7a2040b8eb43326621f12c3a936e09b3a70e3
Author: Andreas Huber <andih@google.com>
Date: Mon May 18 11:03:04 2009 -0700
The king is dead, long live the kind... Replaced old OMX binder interface with new OMX binder interface/socket hybrid.
commit 0d4955eb5df9bfafb576d22f2ed572644d6d69ac
Author: Andreas Huber <andih@google.com>
Date: Thu May 14 12:39:34 2009 -0700
dynamic negotiation of NAL framing with the MPEG4Source, removed stale code from genesis2 commandline tool. Enable support form PV H264 software decoder.
commit 380369f54a8027cbc9349305184d6bf527a24efe
Author: Andreas Huber <andih@google.com>
Date: Wed May 13 17:32:00 2009 -0700
Explore the option of running everything inside the mediaserver process by encapsulating the player in an IGenesisPlayer binder interface.
commit b5cada56b9089a23e0c461352a97dc71e08b424d
Author: Andreas Huber <andih@google.com>
Date: Wed May 13 12:11:06 2009 -0700
Introduction of TimeSource class, clean shutdown of OMX Nodes.
commit 89f6e582e605dcb2149eb67f73fe9345d4b9be2e
Author: Andreas Huber <andih@google.com>
Date: Tue May 12 17:11:56 2009 -0700
Helper classes MediaRing and MediaPlayer.
commit 9aaed8739b4bb941473da367fb88f63224d2f1b7
Author: Andreas Huber <andih@google.com>
Date: Tue May 12 14:19:18 2009 -0700
Using mmap to map a file instead of going through FILE* improves performance, who knew?
commit ad097a78bbd002c59193ffdc39b64935928426c7
Author: Andreas Huber <andih@google.com>
Date: Mon May 11 16:42:42 2009 -0700
Added a performance test, refactored AudioDecoder a little.
commit bdd1b80ab3f3ba43e04f1f996e7fa84ed2e2728a
Author: Andreas Huber <andih@google.com>
Date: Fri May 8 16:19:29 2009 -0700
No more copying the output buffers coming out of the decoder, NAL separation also done in the extractor to avoid copying the data.
commit 858955b42c29b8033841a8caaa56c2a449810ef3
Author: Andreas Huber <andih@google.com>
Date: Fri May 8 14:02:38 2009 -0700
Change from a model in which the caller specifies the destination buffer to one where the MediaSource returns a buffer.
commit 95098b6d4f8bdc37dfb6a109ca58db4d9e414e2d
Author: Andreas Huber <andih@google.com>
Date: Fri May 8 13:21:45 2009 -0700
Changes to support multiple pieces of codec specific data to the decoder, de-frame nal units from the h.264 stream and insert start codes
commit 3941fcf87a48dda81683df727dfb1381f62d9184
Author: Andreas Huber <andih@google.com>
Date: Thu May 7 16:37:45 2009 -0700
some refactoring. Most streams play now.
commit d3152862840d4dd39068e5197bec8674f3d55dd5
Author: Andreas Huber <andih@google.com>
Date: Thu May 7 15:37:32 2009 -0700
Misc. changes to support H264 decoding using the qcom hardware decoder
commit d4ecdfe7af36e7468270d1c2cbbdc540efe0351e
Author: Andreas Huber <andih@google.com>
Date: Thu May 7 09:28:18 2009 -0700
Timestamps are now stored as a fraction, i.e. units and scale, also, PV appears to violate OMX specs by assuming timestamps are in milliseconds instead of microseconds.
commit fe6a50943f885e065d4061aa727a54417e9f36db
Author: Andreas Huber <andih@google.com>
Date: Wed May 6 14:18:27 2009 -0700
Refactored commandline app instantiation of the decoder.
commit da1ff2724d0e78e8b360d9a2c34c55dd0e2c0492
Author: Andreas Huber <andih@google.com>
Date: Wed May 6 10:41:14 2009 -0700
Proper audio/video synchronization.
commit 569ee3a23f600f4b6b87c6178c0223b06e97f201
Author: Andreas Huber <andih@google.com>
Date: Wed May 6 09:17:20 2009 -0700
Preliminary support for timestamps.
commit e4e302a89207550d8a7170d8c0afbd9c15047c84
Author: Andreas Huber <andih@google.com>
Date: Tue May 5 15:13:12 2009 -0700
Another quick hack job to make CachingDataSource thread-safe as well.
commit dfb376a58bd7c09aa04e89558186fcfe13fe696c
Author: Andreas Huber <andih@google.com>
Date: Tue May 5 15:08:14 2009 -0700
Quick rushjob to make FileSource and SampleTable thread safe.
commit 02b29e7ebdab1f8b959b6c177ceb3b89d998f375
Author: Andreas Huber <andih@google.com>
Date: Mon May 4 17:22:04 2009 -0700
Made video decoding work for mpeg4/h263 and mostly(?) with AVC content.
commit 3a57b8ff15e48f5a0af03bf0a99c32e240dbb7ef
Author: Andreas Huber <andih@google.com>
Date: Mon May 4 09:26:42 2009 -0700
Getting rid of old sample code to drive the OMXNode binder interface.
commit 5b046aed0cf524a57ea711bc1e43ea80e57b2cb3
Author: Andreas Huber <andih@google.com>
Date: Fri May 1 16:21:11 2009 -0700
Moved files from the commandline utility into the framework, renamed MP3Decoder to AudioDecoder
commit 7f3fbba21512a078b732cc52c7bd5ba5acb9a317
Author: Andreas Huber <andih@google.com>
Date: Fri May 1 15:56:19 2009 -0700
Fixed buffer ordering and an off-by-1 error in the SampleTable code. Plays all AACs now :)
commit 024d06baab8f2c540dde16f7e7a4b6dd57f638c7
Author: Andreas Huber <andih@google.com>
Date: Fri May 1 11:52:48 2009 -0700
Some AAC content in .3gp or .mp4 files plays correctly now, other does not. Implemented ESFS parser.
commit 1470427866b0d3e0ce6848b4edc01c13a09be289
Author: Andreas Huber <andih@google.com>
Date: Thu Apr 30 15:19:31 2009 -0700
Another checkpoint, amr decoding now functional.
commit e612d044439f56331b6a2bc4a88622e04d4d42e7
Author: Andreas Huber <andih@google.com>
Date: Wed Apr 29 12:11:29 2009 -0700
string wrapper around String8, various fixes, shoutcast now works again.
commit 2ea52dfb1a39bf3a223ffc87855e54919e2daff7
Author: Andreas Huber <andih@google.com>
Date: Wed Apr 29 11:08:18 2009 -0700
Initial check-in of OMXNode binder interface.
commit 7880d2957f74c0d7fba7679431ff5adb3780543b
Author: Andreas Huber <andih@google.com>
Date: Wed Apr 29 10:05:01 2009 -0700
Initial check-in of both libgenesis2 and the genesis2 commandline tool.
/frameworks/av/media/libstagefright/AudioPlayer.cpp
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