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The AudioFlinger kept pausing the audio when playing compressed AC3 or DTS.
This caused pause/resume loops that were hard to break out of.
The AudioFlinger was thinking that the compressed audio was PCM
because the HAL was in PCM mode playing SPDIF data bursts.
It also thought that EAC3 was at 192000 Hz instead of 48000
Hz because the data bursts are played at a higher rate.
This CL adds more calls to the shim that separates the AudioFlinger.
Now the AudioFlinger gets information about the HAL sample rate,
channel masks and format from the shim instead of calling the HAL directly.
The AudioFlinger now uses a different threshold for detecting
underruns when the audio is compressed.
Bug: 19938315
Bug: 20891646
Change-Id: Ib16f539346d1c7a273ea4feb3d3afcc3dc60237d
Signed-off-by: Phil Burk <philburk@google.com>
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The HAL does not reset the frame position on standby().
But applications expect the frame position to be reset.
So we subtract the position at standby from the current position.
Bug: 21724210
Bug: 21930805
Change-Id: I0c4520ba1c6c06a580f45f6bafc8cf1d56969f07
Signed-off-by: Phil Burk <philburk@google.com>
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Pass format for AC3 or DTS to SPDIF encoder.
Bug: 18292317
Change-Id: I6d80eb548cb431bac3390ebb3235146171344291
Signed-off-by: Phil Burk <philburk@google.com>
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Create an interface layer between the AudioFlinger and the HAL
that manages the wrapping and format conversion.
Removed unnecessary includes.
Handle rate conversion in getRenderPosition().
Try to open HAL with encoded format before wrapping with SPDIF.
Bug: 17566660
Change-Id: I00ad888ca15ff0f85b85efb8167c7f5ea761a244
Signed-off-by: Phil Burk <philburk@google.com>
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