The last selection criteria is the audio bitrate to prefer higher-quality
streams. We shouldn't apply this criterium though if the languages of the
tracks are different.
Issue:#6335
PiperOrigin-RevId: 265064756
Currently the value of FLAG_ALLOW_CACHE_FRAGMENTATION is defined as "1 << 4" but commented as "8". Either the value of FLAG_ALLOW_CACHE_FRAGMENTATION should be "1 << 3", or the comment should be 16. Here I am modifying the comment since it does not affect any current behavior.
PiperOrigin-RevId: 265011839
The app is able to pass a more specialized array type, so the Arrays.copyOf call
produces an array into which it's not valid to store arbitrary AudioProcessors.
Create a new array and copy into it to avoid this problem.
PiperOrigin-RevId: 264779164
This matches the documentation on MetadataDecoder.decode:
"@return The decoded metadata object, or null if the metadata could not be decoded."
PiperOrigin-RevId: 263767144
entries are used in .equals(), so it's good to have them printed in toString() too (for test failures) and it makes logging easier too.
PiperOrigin-RevId: 263335503
The current max video buffer is 13MB which is too small for high quality
streams and doesn't allow the DefaultLoadControl to buffer up to its default
max buffer time of 50 seconds.
Also move util method and constants only used by DefaultLoadControl into this
class.
PiperOrigin-RevId: 263328088
I think we need to start clearing the holder as part of the
DRM rework. When we do this, it'll only be valid to read
from the holder immediately after it's been populated.
PiperOrigin-RevId: 262362725
We're no longer tied to the emsg spec, so we can skip unused fields
and assume ms for duration.
Also remove @Nullable annotation from EventMessageEncoder#encode, it
seems the current implementation never returns null
PiperOrigin-RevId: 262135009
We already allow mixed mime type and mixed sample rate adaptation on request,
so for completeness, we can also allow mixed channel count adaptation.
Issue:#6257
PiperOrigin-RevId: 261930046
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
