| Commit message (Collapse) | Author | Age | Files | Lines |
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Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
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o do not use edts/elst boxes since these optional boxes are ignored
o manipulate the first video/audio frame duration to make sure that the rest
of the audio/video is in sync (ideally, we should only manipulate
the vidoe frame duration, not the audio)
o reduce the initial audio mute/suppression period, which is used to
eliminate the "recording" sound.
bug - 3405882 and 3362703
Change-Id: Ib0acfb4f3843b365157288951dc122b006299c18
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bug - 3313754
Change-Id: I951dd0e21e34aa1412c391f003bc32103d0424b0
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bug - 3439313
Change-Id: Ie29d6e4945978ef27fc3e5849e467d895c7736d3
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bug - 3385198
Change-Id: I86ac8071eb28a538b333e102192193d1b9eda5eb
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o Skipping frames could lead to a lot of issues such as I frames is lost etc.
It is not being used anyway.
This reverts commit 53d4e0d58e2d5c18f6e026c705af833b9bdd7aba.
Conflicts:
media/libstagefright/AudioSource.cpp
media/libstagefright/CameraSource.cpp
Change-Id: I3abba1647de48db25bdc369066eb2a7ae4dedec2
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Change-Id: Ic0f1489f710929af50e7714867ae5153b3242dd8
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- This fixes media server crashes on droid
Change-Id: I7191cadc5275107425ec3ee3d437b2c5295858dc
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The problem was that the time to receive an output buffer
from an audio encoder is different because the encoder does not
need to read from the source for all output buffers. This leads
to large fluctuation in terms of wall clock duration between two
neighboring audio sample outputs from the audio encoder. As a
result, the media time for the video track after adjustment using
the drifting changes wildly sometimes.
This patch addresses this issue by only updating the media drift
time when an audio source input buffer is read. the wall clock
for the audio track is also calculated at the same time when
the input audio buffer is read at AudioSource.
bug - 2959800
Change-Id: I3174aa182f744784b540f0a7198524d4eee8bd7b
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Change-Id: I16dfc90bcb8a324d6ee9a38a5a1a31cc094c820a
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- bug 2950297
Change-Id: I0044d07178691feb904cf81e87c1b6d4b714dc1a
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Change-Id: I2687ad855aac758946954d0b3fe7aff9f7b5ae7c
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- Also collect stats on lost audio frames instead of time spent on reading
Change-Id: I6380b143e4fbdcd894491aaae523331e90d0f04f
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This is 1st part of the work to allow audio and video resync if
we found out that audio and video are out of sync during authoring
- also fixed a problem in AACEncoder::read() where the buffer acquired
from the buffer group does not release when error out at
reading from source.
Change-Id: I8a2740097fcfdf85e6178869afeb9f3687a99118
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- only start to track the max amplitude after the first call to getMaxAmplitude()
Change-Id: I64d3d9ca0542202a8535a211425e8bccceca50fc
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Change-Id: Iba0b35f57fdeac7ee1da16899406bf4b957a2c8c
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Change-Id: I5f362252c25e2251bbfa9818b711ee23b4975248
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Change-Id: Ibdcf7bea5fb66baa81878704ba4091dfcfe382ee
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Change-Id: I93a9d8bd260efc5e7fc135b726e3f1307c6df794
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- Extend the audio recording to AAC format
- Add support for setting some recording parameters
- Add stss box to the meta data in the recorded file
Change-Id: I41167bfd9d70ef9cd33906f8437b39c232b6d3b7
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pure-audio recording in stagefright.
related-to-bug: 2295449
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