| Commit message (Collapse) | Author | Age | Files | Lines |
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Conflicts:
audio/audio_hw.c
init.tuna.rc
overlay/frameworks/base/core/res/res/xml/storage_list.xml
recovery/Android.mk
Change-Id: I0b13c6b5e61ea709b0e9aaa1c7b798684b33873a
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the ringtone offset has to be setted to analog side.
Buganizer : 6920555
According to Samsung's spec, the earphone ringtone volume level should
be 14dB lowere than the media playback volume.
On ICS, this behavior was working properly, but on JB this behavior is
not working properly.
Below is the analog and digital volume change from ICS to JB:
ICS : Digital Volume = Normal / Analog volume = lowered 14dB
JB : Digital Volume = increased 14dB (in comparison to ICS) / Analog
volume = lowered 14dB (same as ICS)
Hence the volume in JB has increased by 14dB when compared to ICS.
Bug 6920555.
Change-Id: Ibc248612db378b5b991221468d8f801257ba4103
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Bug: 6878923
Change-Id: Id49d6489e5a99dee088246d146ee38151ba9499c
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out_get_parameters() was calling strdup() on the string
returned by str_parms_to_str() before returning it
to the caller. This creates a new string which is never freed
as str_parms_to_str() already allocates a new string.
Change-Id: I4bcc4aa17ab55e830d7a0569151f717422f6459b
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Modifications for new audio device enums:
- Separated input and output device fields as output and input device
values are now on 32 bits.
- Changed audio device API version to 2.0
Also removed get_supported_devices() function not needed if audio_policy.conf
file is present.
Change-Id: I41b782e7450b4664048cc484a681b9327d8395da
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When an auxiliary mic channel is used, the echo reference
should use only the main channels to be consistent with the
way the reverse effect processing is configured.
Change-Id: I28ee1e2a9852fdd0e904fb01bedf90f3372683c9
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Change-Id: Icf113e2e863a79cb3d870fac5781539702cdbfa8
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Bug: 6881638
Change-Id: I76255c2cd5845671c2342e22932c692342257208
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Change-Id: I90a50b58dd23fe522724df53f08b4f9687150da6
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Acquire the audio device mutex before calling into ril
library in adev_set_voice_volume() to avoid concurrency with
other calls to ril from select_mode() or set_incall_device().
Bug 6626532.
Change-Id: I2347477b39ce46137a654047266b70dd691c021c
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Since that was incorrectly detected, it was failing to use
toro audio settings and was using maguro ones
Change-Id: I0a5524011365e3ef39ab110b0e608dbce9edbae6
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The volume for media was not in sync with speaker phone and was
too quiet.
Change-Id: I2e18a75e85335f4f2b82dc8532551a3690f99890
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seems this is the max we can go without getting any negative effect.
Change-Id: Ic769b1ac200d2b60588cb3ccaf65c7bce7251c1a
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Switching from BT SCO to earpiece does not seem to
work when in call and an output stream is active.
This change modifies out_set_parameters() to force the
output stream into standby when a new audio path is
selected while in call.
Bug 6676684.
Change-Id: I2817f80ea3fa3a0e00e9705fdb6d9a7e3183549b
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Bug 6615379.
Change-Id: I5ef2cc168bbe26b40c49e602d6345c1b64c2b1b0
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Workaround for HDMI multi channel channel swap on first
playback after opening the output stream: force re-opening
the HDMI pcm driver after writing a few periods.
Bug 4282214.
Change-Id: Ibe1452a8905a27bc3f95564a45cfb9bb490b65ae
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Added a dedicated audio output stream for multichannel HDMI.
This output stream is used when an HDMI sink supporting 6 or 8
PCM channels is connected and 5.1 or 7.1 multichannel content it played.
Change-Id: I7ad1cd6be4c2b3a9e24a4811aa87e7223badedc4
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Add back the capability to change the deep buffer size according to screen state.
This solves various issues related to audio focus, volume and pause control
that arise with large audio buffers.
Those issues should be ultimately addressed by changes in the audio framework.
Change-Id: I6889ecf0e5d8740745152261f27343e1ff533e7b
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Fixed 2 issues with media volume:
1 - since we use mm port for music and tones port for other use cases
the digital volume should be applied to both "DL2 Tones Playback Volume"
and "DL2 Media Playback Volume".
2 - the total gain applied to audio originating from the AP is the
combination of digital gain in ABE and analog gain in codec. Some use cases
like telephony have a higher priority than media and apply a different (higher)
analog gain. As this analog gain is common to all sources, digital media gain
should be adjusted accordingly to avoid volume bursts while in call and playing
music. This is particularly important in speaker phone mode.
Change-Id: I90200282edca7098603edca2d56821290988cb20
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Fixed memory leak introduced by commit 4e7a573f in case
of error in adev_open_output_stream().
Change-Id: I4acc070d748cea228da846f95c7826160e0196a5
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Implement one output stream with short buffers and
one output stream with deep buffers.
The stream with short buffers is selected for most use cases and
provides short latency. It uses TONES_DL port and IOCTL write mode.
The stream with deep buffers is used for music playback.
It uses MM_DL port and MMAP NOIRQ write mode.
The deep buffer stream is not used when the device selection is
BT SCO, HDMI or SPDIF.
The echo reference is only taken from the short buffer stream.
Change-Id: I60ef720e52e96970b8b6618f9f43f24baadce60b
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Use 4 buffers of 96 frames each = 4 ms at 48 kHz.
Keep the 44.1 kHz -> 48 kHz up-sampler in HAL.
Disable mmap mode and non-IRQ mode; this gives better variance for cycle times.
Reduce number of buffers from 4 to 2, works OK in non-mmap mode but not mmap mode.
Update comments based on code review.
Tested with audio input.
Not yet tested with echo cancellation.
Change-Id: I69db00ab408cd2aad5788d602eb01fc0c7e4e78b
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Change-Id: Ia6b6caf67f3c2e53431d7b65c3a30c57975faa2a
Signed-off-by: Mike Lockwood <lockwood@google.com>
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Change-Id: Ia2d0f55fc065e7071d9f5207e0dc91b63f554759
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Change-Id: I1169d279b4a59355cf4362a7128b053bf940c158
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Added support for audio pre processing libraries
implementing dual mic solutions.
When a pre processor is enabled, its multi channel capabilities are
queried and compared to capture channel combinations supported by the
device and other enabled pre processings.
The most favorable configuration is chosen and pcm capture driver is
restarted with the appropriate channel config.
Also made various capture and process buffers naming and allocation more
consistent.
Change-Id: I90be4798951d0a34dc77d6bdc93ef15cad3ff5af
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Added audio policy manager configuration file.
Change-Id: I62163e203a42596ac69b2971c5c0fa99817b33b3
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Commit 78a7609d fixed audio drop at the start of ringtone.
This commit fixes another similar issue with camera shutter sound being
dropped while in call over headset.
There was a workaround for this second issue in audio policy manager but this was
not satisfactory as it was impacting all devices for a problem that is
Prime specific.
Change-Id: I42b37c7da4a232323b520a8a55ac5b3086b5a230
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Fix error in get_capture_delay() that was not taking into account
the fact that frames in in->buffer are at driver sampling rate while
frames in in->proc_buf are at requested sampling rate.
Change-Id: I09e627bd316daedab5ffea3dd638254eaa270a5b
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selected
* commit 'd28a1a802b1965ab4b9014c658240faafe219994':
audio: fix audio drop when speaker is selected
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When changing audio path to speaker while playback is active,
several hundred ms of audio are dropped. This is mostly noticeable
when a ringtone starts playing.
This change is a workaround forcing the output in standby when speaker
is selected.
The root cause must still be indentified and fixed.
Change-Id: Idef8dc1cdbf2da499a414d0b60244f91ef66e73b
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Change-Id: Ib1d5af6687479c8d189a3407c229a6ac0ed5c03b
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Change-Id: Ifb68db236cb6b9e039eadf573e177add1de62d8c
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Change-Id: If9c95a4808785e58ee4595e5c762d01d87f1936d
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Change-Id: Id432e901f8107a00a7f371e5882b1290a1154961
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Change-Id: I5179699b22224473bd158e90f864e4e73895b5dc
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Change-Id: I2e2a5f625956dc5d09dbdc3f6f2d9a010ecc7bad
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Change-Id: I711920dde1560ca202ef878ee93a2af61545524b
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Change-Id: I7a30fe3f66933aed8b5a6185553112575b4de1a7
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See https://android-git.corp.google.com/g/#/c/157220
Bug: 5449033
Change-Id: I2e1c43800c19b718cc7ee94ec299c62bc14873b4
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See https://android-git.corp.google.com/g/157065
Bug: 5449033
Change-Id: I02cfaca251935e4a50ad4302a72c4273be41db22
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* commit '31688e73c947845cea86079aefa2dfab68b56c93':
audio HAL: release audio pre processing buffers.
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Buffers allocated for audio pre processing are not released
when an input stream is closed.
Issue 5753047.
Change-Id: Ie8fd5f49d97e9bebc70fc38de0844a79074ac526
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The clock sync func is unused for both HSPA and LTE device.
Change-Id: Ia9f369a0151cb3bb15242544e5f5442b893253bc
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ics-mr1
* commit 'ec429c1320e97145b42a8f334ed5506d316bb412':
audio: force speaker route for call when docked
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As we did for the HDMI audio case, force the speaker route for
calls when in a digital dock because we cannot directly route
the modem audio output through the S/PDIF output because it is
a McASP device.
Fixes bug 5434090
Change-Id: I52ff7877a8be778b9e74eebb3ad2c9f13b634bca
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* commit '56e8b292cb2aa15804eb436d48c71e1a98b36550':
audio: decrease headset gain by 14dB for ringtone mode
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This is to prevent audio shock in AUDIO_MODE_RINGTONE.
Change-Id: Ic21c347a64ee0e2668dbff49dc6addcb93e4d82f
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See https://android-git.corp.google.com/g/#/c/143865
Bug: 5449033
Change-Id: Iab0aa050fba57491f5cb7ed928f44a0fda7d1ea4
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Mute and unmute VX_UL gain to avoid pop noises in the tx path
during call switch to the modem during the switch it means when
audio path changes(Example: Analog path switches from EAR<->HS<->HF).
Change-Id: I567d4156a5b9aa7b51d068fe279f942376a5a40c
Signed-off-by: venkappa mala <venkappa.m@samsung.com>
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