| Commit message (Collapse) | Author | Age | Files | Lines |
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Adapted from a patch by @MWisBest
Change-Id: I1b0cb2db0e5473088eb42b623bfd902332b1ec47
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Just a hunch, see the comment...
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This basically halves the long period size. I wish there was a different way
to fix this but I don't think there is.
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Move the detection in start_output_stream_deep_buffer down to allow a call to
pcm_set_avail_min now. Did a few micro-optimizations as well.
Conflicts:
audio/audio_hw.c
Change-Id: I90d5663040986ffd597f37ae66334467adacea3b
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(buffer_frames is only used for OUT_RESAMPLER)
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Previously this was corrected on the first write, however that was causing
a buffer underrun sometimes when a stream was started. This avoids that.
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Conflicts:
audio/policy/audio_policy.default.conf
Change-Id: I4d13946e88cacbf5e4ca383d5d0756262442efd2
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Fixes MM-UL not working.
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By moving this block down, the sampling rates are properly defined for this check.
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This audio HAL has always required some sort of resampling.
THIS. ENDS. NOW.
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This needs further testing before enabling by default, but so far it's been OK.
Conflicts:
audio/audio_hw.c
Change-Id: Ic4f86440ff4d01ab4d0d9f977bdec22f10f60555
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This fixes a bunch of annoying junk in the logcat about fast path being denied to UI sounds due to them having a 48000Hz sample rate.
Also allows playback of 96kHz audio, as Android refuses to resample to anything lower than a divide by two.
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Change-Id: I26e7436513bacf951a0cf4fc252521a8f1cdd3d8
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Pass device address (and audio source for inputs) to
open_output_stream() and open_input_stream() audio HAL functions.
Bug: 14815883.
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On direct output streams the audio HAL must implement
the volume function. In the case of HDMI the only function
required is to mute audio when volume is 0 as volume
is defined as fixed on digital output streams.
Bug 8541062
Change-Id: I4b4e28a910e7b321b3a68567e9ad03fede065ca8
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Added support for simultaneous capture from front and back
mics.
Change-Id: Ica1b75fe432f419272ae92e8ab04b1d34524c189
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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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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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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: 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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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 '2cb034ebbf5eb4f9ead26150d288bf6d90dc2fee':
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: 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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