This also decouples EventMessageEncoder's serialization schema from the emesg spec (it happens to still match the emsg-v0 spec, but this is no longer required).
PiperOrigin-RevId: 261877918
If we keep streams in chunk sources after selecting new tracks, we also keep
a reference to a stale disabled TrackSelection object. Fix this by updating
the TrackSelection object when keeping the stream. The static part of the
selection (i.e. the subset of selected tracks) stays the same in all cases.
Issue:#6256
PiperOrigin-RevId: 261696082
- When in STATE_SEEK with targetGranule==0, seeking would exit
without checking that the input was positioned at the correct
place.
- Seeking could fail due to trying to read beyond the end of the
stream.
- Seeking was not robust against IO errors during the skip phase
that occurs after the binary search has sufficiently converged.
PiperOrigin-RevId: 261317035
A previous change switched to calculation of the bitrate based on the
first MPEG audio header in the stream. This had the effect of fixing
seeking to be consistent with playing from the start for streams where
every frame has the same padding value, but broke streams where the
encoder (correctly) modifies the padding value to match the declared
bitrate in the header.
Issue: #6238
PiperOrigin-RevId: 261163904
Android considers ALAC initialization data to consider of the magic
cookie only, where-as FFmpeg requires a full atom. Standardize around
the Android definition, since it makes more sense (the magic cookie
being contained within an atom is container specific, where-as the
decoder shouldn't care what container the media stream is carried in)
Issue: #5938
PiperOrigin-RevId: 261124155
Checking inputPosition == 0 isn't sufficient because the synchronization
at the top of read() may advance the input (i.e. in the case where there's
some garbage prior to the seek frame).
PiperOrigin-RevId: 261086901
When using speed adjustment it was possible for playback to get stuck at a
period transition when the channel count changed: SonicAudioProcessor would be
drained at the point of the period transition in preparation for creating a new
AudioTrack with the new channel count, but during draining the incorrect (new)
channel count was used to calculate the output buffer size for pending data from
Sonic. This meant that, for example, if the channel count changed from stereo to
mono we could have an output buffer size that stored an non-integer number of
audio frames, and in turn this would cause writing to the AudioTrack to get
stuck as the AudioTrack would prevent writing a partial audio frame.
Use Sonic's current channel count when draining output to fix the issue.
PiperOrigin-RevId: 260156541
1. Using the Locale on API<21 doesn't make any sense because it's a no-op
anyway. Slightly restructured the code to avoid that.
2. API<21 often reports languages with non-standard underscores instead of
dashes. Normalize that too.
3. Some invalid language tags on API>21 get normalized to "und". Use original
tag in such a case.
Issue:#6153
PiperOrigin-RevId: 258773463
2-letter codes (ISO 639-1) are the standard Android normalization and thus we
should prefer them to 3-letter codes (although both are technically allowed
according the BCP47).
This helps in two ways:
1. It simplifies app interaction with our normalized language codes as the
Locale class makes it easy to convert a 2-letter to a 3-letter code but
not the other way round.
2. It better normalizes codes on API<21 where we previously had issues with
language+country codes (see tests).
3. It allows us to normalize both ISO 639-2/T and ISO 639-2/B codes to the same
language.
PiperOrigin-RevId: 258729728
If we use the default start position, we currently resolve it immediately
even if we need to play an ad first, and later try to project forward again
if we believe that the default start position should be used.
This causes problems if a specific start position is set and the later
projection after the preroll ad shouldn't take place.
The problem is solved by keeping the content position as TIME_UNSET (= default
position) if an ad needs to be played first. The content after the ad can
then be resolved to its current default position if needed.
PiperOrigin-RevId: 258583948
Sending MESSAGE_PREPARE_SOURCE should happen last in the constructor.
It was previously happening before initialization finished (and in
particular before pendingMediaPeriods was instantiated).
Issue: #6146
PiperOrigin-RevId: 257158275
Currently, we sometimes apply new playback parameters directly and sometimes
through the list of playbackParameterCheckpoints. Only when using the checkpoints,
we also reset the offset and corresponding position for speedup position
calculation. However, these offsets need to be changed in all cases to prevent
calculation errors during speedup calculation[1].
This change channels all playback parameters changes through the checkpoints to
ensure the offsets get updated accordingly. This fixes an issue introduced in
31911ca54a.
[1] - The speed up is calculated using the ratio of input and output bytes in
SonicAudioProcessor.scaleDurationForSpeedUp. Whenever we set new playback
parameters to the audio processor these two counts are reset. If we don't reset
the offsets too, the scaled timestamp can be a large value compared to the input
and output bytes causing massive inaccuracies (like the +20 seconds in the
linked issue).
Issue:#6117
PiperOrigin-RevId: 256533780
Before this change we'd release the audio track and create a new one as soon
as audio processors had drained when reconfiguring.
Fix this behavior by stop()ing the AudioTrack to play out all written data.
Issue: #2446
PiperOrigin-RevId: 244812402
We currently handle most the control code logic after handling special
characters. This includes filtering out repeated control codes and checking
for the correct channel. As the special character sets are control codes as well,
these checks should happen before parsing the characters.
Issue:#6133
PiperOrigin-RevId: 256993672
1. Only output video starting from a keyframe
2. When calculating the timestamp offset to adjust live streams to start
at t=0, use the timestamp of the first tag from which a sample is actually
output, rather than just the first audio/video tag. The test streams in
the referenced GitHub issue start with a video tag whose packet type is
AVC_PACKET_TYPE_SEQUENCE_HEADER (i.e. does not contain a sample) and whose
timestamp is set to 0 (i.e. isn't set). The timestamp is set correctly on
tags that from which a sample is actually output.
Issue: #6111
PiperOrigin-RevId: 256147747
We currently don't display the last frame because the seek time is behind the
last frame's timestamps and it's thus marked as decodeOnly.
This case can be detected by checking whether all data sent to the codec is
marked as decodeOnly at the time we read the end of stream signal. If so, we
can re-enable the last frame. This should work for almost all cases because the
end-of-stream signal is read in the same feedInputBuffer loop as the last
frame and we therefore haven't released the last frame buffer yet.
Issue:#2568
PiperOrigin-RevId: 251425870
