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Change-Id: I8ffca0460195263d159aa13015c246122d8556a2
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up-sample coefficient were generated with a cut-off frequency of 24KHz
intead of ~20KHz, which caused more aliasing in the audible band.
also increased the attenuation to 1.3 dB on both up and down
sampling coefficient to avoid clipping.
Change-Id: Ie8aeecf1429190541b656810c6716b6aae5ece2e
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Change-Id: I550e5a59e51c11e1095ca338222b094f92b96878
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resolved.
* commit 'dd4e72b44ddb7a1591bd2f81c46d20bb0939e9eb':
Increase bitrate now that other issues have been resolved.
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* commit '2295ce2c6fcf07ffaa134440691a8cea5fe35b43':
Increase bitrate now that other issues have been resolved.
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Bug: 7426218
Change-Id: I67dfa1e4b85f326f355ad0ec5b6c699e87b45564
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dongles).
* commit 'dd974cc4f335b30731c452885bff3c396ed54724':
Disable silence suppression (causes issues with dongles).
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* commit 'e815552c9cb17be79cb3b7eb6fbdfa9ad81c861b':
Disable silence suppression (causes issues with dongles).
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Bug: 7426218
Change-Id: Ie1517a8017bae1f9a9b6c224cd3170dfcc5fb941
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Change-Id: I66852d90d384f1d9e77b51ad1a1ebdbaf61d0607
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cubic resampler was disabled because it hadn't been qualified,
however after I did some tests, it does improve significantly
the sound quality over the order-1 resampler, even if it is
still quite bad.
also HIGH_QUALITY resampler was partially disabled, it's now
fully enabled. It's a big improvement over the cubic resampler
in terms of aliasing noise (it's not as good in the pass-band).
Change-Id: I70e3658c255896588642697be9eb594ff4ec0f8b
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- we increase the interpolation precision from 4 to 7 bits
this doesn't increase CPU power required, it only increases the
size of the filter table but significantly reduces the noise
introduced by the quantization of the impulse response.
- the parameters of the filter are set such that aliasing is
rejected at 80 dB below 20 KHz. Because we don't use a lot of
coefficient (to save compute power), there are quite a bit of
attenuation in the pass-band: starting at 9KHz for the
down-sampler (48 to 44.1), and starting at 13 KHz for the
up-sampler (44.1 to 48) -- the transition band is about 15 KHz.
Change-Id: I855548d2aab8a0fb0d2a2da3a364b6842d7d3838
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Change-Id: I557350abb32b0480f5da7dcecadfbe9edbe53361
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- all parameters can be changed on the command-line
- added float output
- added debug option
- added an option to generate a polyphase filter coefficients
- added an attenuation option in dBFS
- added a lot of comments and references
- fixed kaiser window parameter
also the default should generate a filter with 80 dB rejection
(of the 24 KHz aliasing) above 20 KHz and a 15 KHz transition
band around ~20 KHz (for 48 KHz sampling rate).
It's not very good but corresponds to the current code.
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The effects libraries were exporting many more symbols than needed.
This reduces the exported symbols to just the needed ones (basically
just "AELI"), which happens to also save about 28KB.
Change-Id: I115077e52e8dc845282e6f62a522908d26dd72d6
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Change-Id: I6603aef5e3821a8f911e3f33ef8565d04bd1e2e5
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* commit '203836b7584ccb7787202578259d27d70a8a3ede':
Camera2: Don't hold locks while waiting to sync.
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* commit 'b8d6efba8b87e6eeb049651cb67f9827198e658e':
Camera2: Don't hold locks while waiting to sync.
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Bug: 7409877
Change-Id: Ia3a0bc4f0ab4e19fca868ba04a870cf8e8ee7adb
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Add reconnectAtOffset(), DrmInitialization() and getDrmInfo().
Also rearrange the code a bit so all the methods that just call through
to the wrapped DataSource are in the header.
Change-Id: If25b674df317b0f6da5d36241c694e32abb0a01c
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separately." into jb-mr1-dev
* commit 'e3e639b1714d35628f9b75276839ad55d8afc74c':
Don't attempt to seek audio and video separately.
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jb-mr1-dev
* commit 'ac1a450a76c2f42f66c876200cb724d1e93ef4c7':
Don't attempt to seek audio and video separately.
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The widevine extractor doesn't deal too well with that...
Change-Id: Iadfeede4fe0c086af788c5639782854e4fbb98ff
related-to-bug: 7262386
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into jb-mr1-dev
* commit '9e83f44c0b310a72a080c006ebc3c07bdccc7027':
Camera2: Increase mode update timeout.
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* commit '6ccc3c343e2751ebfe9fa77215bf84e1a97e38fc':
Camera2: Increase mode update timeout.
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During camera startup, it might take a few hundred milliseconds before
requests start to be dequeued by the HAL. Increase the timeout for
synchronizing mode changes and triggers so that triggers near startup
don't time out.
Bug: 6970465
Change-Id: I9dc35378e8018ec18ae31be874fcb094f8a9a0e9
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capture" into jb-mr1-dev
* commit '9cdf6db50c0dab30f32291bf00c702ca79132674':
stagefright: discard audio at start of capture
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jb-mr1-dev
* commit '5f6f5b7d6be5d7882102bf0fc3c691ef8fbe03a1':
stagefright: discard audio at start of capture
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The new camcorder start sound is longer than previous one and we
must discard more audio when capture starts.
Ideally, camcorder should use synchronous record start.
Bug 7394330.
Change-Id: I219b4e231aba706776dc7ccc4f1c996eaf22f61a
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* commit '3aaa4e30f7663673f3453d3f4671b1119f1802fd':
Increase allowed number of VHQ resamplers to 3
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* commit 'c4640c9eef850bb1c754bd6b477f1cc8350c6081':
Increase allowed number of VHQ resamplers to 3
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Bug: 7378660
Change-Id: I69e33ca2eb4bb9bd38e2c63df62cd1130d68baf6
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resampler to produce garbage" into jb-mr1-dev
* commit '7f4d5714f3f0ab4efaf2d0929a9f984467ef95c7':
Fix a typo that caused the high quality resampler to produce garbage
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produce garbage" into jb-mr1-dev
* commit 'e384d75b95b9db30d64a18ada93e070cf07de306':
Fix a typo that caused the high quality resampler to produce garbage
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into jb-mr1-dev
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the problem is that if libaudio_resampler is present, it is those
coefficients that will always be selected, but the correct
meta-data.
Bug: 7385994
Change-Id: Ieebeb37b4dfb62a1a051bc29fae2ce056dbc6621
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* commit 'dc5f12a217279d60444fc7243fe89ba4311eea88':
WFD use of HDCP is now optional.
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* commit '0328ec08dc1e90caa2a9e0c4e107d8ddaa74af20':
WFD use of HDCP is now optional.
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if the sink supports it we'll use HDCP (and fail if necessary), if it doesn't
we won't. If an HDCP session is established we'll tell our observer that
the connection is secure, otherwise we don't.
Change-Id: I7cbef384f2cf0a6ac65801c581eea227b9ef4c46
related-to-bug: 7368436
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