Fix a mistake which came in as part of a merge conflict resolution
during code review of the recent unicast mode refactor of LibAAH_RTP.
Nop packet which were supposed to carry TS transformations for the
pause state accidentally got flagged as Flush operations. The flush
packet successfully carried the TS transformation, but also had the
undesired side effect of constantly flushing the stream.
Change-Id: I4c6aa0043fc274a1d7e880ed1d19cf277f22194b
Signed-off-by: John Grossman <johngro@google.com>
Gapless playback for appropriately tagged mp3 and m4a files.
Currently this is implemented in OMXCodec, which most players
use, but should be easy to support in other players as well by
using the SkipCutBuffer utility class.
Change-Id: I748c669adc1cfbe5ee9a7dea2fad945d48882551
It was only used to decide whether to issue a warning.
The warning was issued the first time track was created but
not at re-creation. Now it is a verbose message every time,
not a warning since it happens all the time with key clicks on A2DP.
Change-Id: I9d39f53c0a7eb84b666e55b1b76ff830cf8f37ba
EOS was being treated as a flush operation which was causing problems.
In particular, the transmitter was delcaring that playback was
complete early (by the clock lead time of the system, which was 1
second in this case). Also, the receiver was treating the EOS message
just like the flush message, immediately destroying the substreams
associated with the program without letting them play out first.
Change the transmitter to send the EOS message like it always does,
but have it wait until the media time of the last sample has arrived
before reporting playback complete to the app level of things.
On the receiver side of things, don't treat the EOS message like the
flush message. Instead, have the EOS message simply put the substream
into EOS mode, allowing it to signal EOS to its decoder and shut off
the isAboutToUnderflow hack.
Change-Id: Ibe3ac01044373f83edb7a5f4b70478bd78c16d11
Bionic/Android support eventfd, so there is really no reason to have
PipeEvent around any more. This change gets rid of it in LibAAH_RTP
and replaces it with eventfds.
Change-Id: I841fcb71bf5015d521d7517c69f44eac0ea92278
Signed-off-by: John Grossman <johngro@google.com>
Add support for unicast mode to the AAH RXPlayer. At the API level,
things should be pretty simple. To use unicast mode, instead of
passing the multicast address and port in the data source URL, just
pass the unicast address and port of the transmitters command and
control port. For example, instead of
aahRX://224.128.60.5:8867
one might instead pass
aahRX://192.168.63.5:55476
Change-Id: I7b40716983d7a91def86dcf40f093dda4255aae3
Signed-off-by: John Grossman <johngro@google.com>
Fix a bug discovered while working on adding unicast mode to the TX/RX
players. Also some general cleanup/consolidation regarding timeout
code.
The bug went like this. When a TX player had hit EOS, it would send
an EOS command payload to its receivers. Later, when application
level code shutdown and cleaned up the player, it would send another.
In situations where there is massive packet loss, there is a chance
that not only did both of the EOS packets get dropped, but that they
never got filled in by the retry algorithm because the receiver gave
up on the RTP gap due to an aboutToUnderflow situation in at least one
of its active substreams.
When this happens, there are two major problems. First, all of the
substreams associated with the TX player which has now gone away have
become effectively leaked. They will only get cleaned up if the
entire RTP stream (the TX Group) goes away for 10 seconds or more, or
when the RX Player itself is reset by application level code or a
fatal error. These substreams are holding decoder and renderer
resources which are probably in very short supply, which is a Bad
Thing.
Second, there is now at least one substream in the RX player which is
never going to receive another payload (its TX player source is gone),
but is still considered to be active by the rx player. Assuming that
this substream's program was in the play state when the track ended,
there is now at least one substream which is always
"aboutToUnderflow". From here on out, when the retry algorithm is
attempting to decide whether or not it has the time to attempt to fill
in a gap in the muxed RTP sequence, it always decides that it does not
have the time because of the orphaned substream which is stuck in its
about to underflow state. This effectively means that the retry
algorithm is completely shut off until the rx player gets reset
somehow (something which does not happen during normal operation).
Since the environment had to be extremely lossy to trigger this chain
of events in the first place, and its probably no better now, your
playback is just going to be chock full of gaps which produces
horrible stuttering in the presentation stage of the system.
Two new failsafes have been introduced to keep the double EOS drop
from causing this. First, a timeout has been introduced on the
substream level, in addition to the already existing RTP level
timeout. If a substream fails to receive an activity for 10 seconds
(same timeout as the master RTP timeout), it will be automatically
flushed and purged.
Second, the nature of the master RTP timeout on the transmitter side
has been changed. Instead of just sending an empty NOP command packet
to indicate that the main RTP stream is still alive, the transmitter
now sends a new time of command packet; the Active Program Update
packet. This packet contains a list of all the active program ID
attached to this TX group. Upon receiving one of these APU packets,
RX players reset the inactivity timers for all substreams which are
members of the programs listed in the packet, but they also
immediately purge any substreams associated with programs not present
in the APU.
Between the two of these, no matter how nasty and selective the packet
smashing gremlins in your system happen to be, substreams will always
eventually clean up and avoid getting stuck in a perma-stutter
situation.
Also in this CL:
+ Extract some common utility code into a utils.cpp file so that it
can be shared across the library.
+ Stop using custom timeout logic in the RXPlayer. Instead, use the
common Timeout helper class in utils.cpp.
Signed-off-by: John Grossman <johngro@google.com>
Change-Id: I350869942074f2cae020f719c2911d9092ba8055
Significantly refactor the TXGroup code to allow transmit groups to
operate in a unicast fanout mode in addition to the traditional pure
multicast mode. Important changes include...
+ Each transmit group active in the system now has its own socket to
send and receive traffic on. In the past, this socket was used to
listen for retry requests from clients. Now it is also used to
listen for group membership reports (IGMPv3 style) from unicast
clients. Having an individual socket per transmit group allows
unicast clients to join the group needing only the IP address and
port of the transmitters socket, and not needing any additional
"group id" to be sent to the client beforehand.
+ Setup for the transmitter is now slightly different. As before, to
setup for multicast mode, a user can call setRetransmitEndpoint
passing an IPv4 multicast address and specific port to transmit to.
It used to also be the case that a user could pass a specific
unicast address and port to transmit to as well. This is no longer
allowed. Instead, to operate in unicast mode, a user passes 0.0.0.0
(IPADDR_ANY) as the IP address. In addition, they need to pass
either 0 for a port to create a new unicast mode TX group, or they
need to pass a specific port to cause the player to attempt to use
an existing unicast mode TX group. The specific port should be the
command and control port of the TX group which was bound to when the
group was originally created.
+ A magic invoke was added to allow clients to fetch the command and
control port on which a TX Player's TX Group is listening.
The API described above is most likely temporary and should eventually
be replaced with one where TX groups are formal top level objects with
their own independent interface and life-cycle management.
Signed-off-by: John Grossman <johngro@google.com>
Change-Id: Ib4e9737c10660d36c50f1825c9824fff5390b1c7
Rename AAH_TXSender to AAH_TXGroup in preparation for refactoring to
support unicast retransmission.
Signed-off-by: John Grossman <johngro@google.com>
Change-Id: I3984db27d1c61c6155d5d7cb9c38eead421b9249
createTrack and openRecord don't need the "old" flags parameter,
which was either audio_policy_output_t or audio_in_acoustics_t
shifted left by 16 bits. But they do need "new" flags, which
are defined by the application use case. Initially, the only
application use case flag is timed output, but others are planned.
For output, the audio_policy_output_t flags are passed to
AudioSystem::getOutput, which returns an audio_io_handle_t, and that
handle is then passed to createTrack. So createTrack doesn't need the
old flags parameter.
For input, the audio_in_acoustics_t flags are passed to
AudioSystem::getInput, which returns an audio_io_handle_t, and that
handle is then passed to openRecord. So openRecord doesn't need the
old flags parameter.
Change-Id: I18a9870911846cca69d420c19fe6a9face2fe8c4
MediaRecorder.setPreviewDisplay() is not required if applications
use MediaRecorder.setCamera(). Besides, this causes a problem when
apps use Camera.setPreviewTexture. Camera service thinks the
surface texture from Camera.setPreviewTexture and the surface from
MediaRecorder.setPreviewDisplay are different.
bug:5988937
Change-Id: Ia345705b6679ef349db6e354feaa3cc0fe8bcd8c
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
Conflicts:
media/libmediaplayerservice/MediaPlayerService.cpp
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
Conflicts:
media/libmedia/AudioTrack.cpp
