- Video frame drops are observed on RTSP streaming
for low sampling rate clips
- For PCM offload audio track uses server proxy position
value to caluclate render position.
Due to high minimum fragment size value of PCM offload,
low sampling rate clips are using bigger audio buffer
than desired 20ms PCM offload buffer. This results
improper audio render position return to Nuplayer,
which leads to video frame drops
- Reduce minimum fragement size for PCM offload from 4k
to 512
Change-Id: I661f1ffe7cfd41edead82d06063802a3ad7b9b66
Voice call setup latency value is exceeding 500 msec. It is due to
primary output selecting handset instead of voice handset causing
the extra delay. Fix this by selecting voice handset when the mode
is set to MODE_IN_CALL.
CRs-Fixed: 855783
Change-Id: Ibfaee75d658bffbcf2e9cdeaa589e0487c214b41
Sometimes no RTP packets transferred in VoLTE call on headset.
Voice call path setup is failed due to no backends enabled.
This happens when routing is triggered on combo device for
primary output and mode is MODE_IN_CALL at that time.
Fix this issue by selecting proper device based on in call
information.
CRs-Fixed: 859796
Change-Id: I9a5a6b9296d168e9292b94fda147491ee35c0edd
- all changes for native audio backward
compatibility and related
- add set parameters from UI
- enable dynamic device switching
- featurize the code
- various backend concurrency changes
Change-Id: Id0f824c4b4c033f42008a4e8868652c8f6fe5c42
- currently buffer size is Aligned with 96 which is not multiple
of some channels(5,7).
- Buffer size must be multiple of (number of channels * bytes per sample).
For writes to succeed, the buffer must be written at address which is
multiple of 32.
- Alignments of (number of channels * bytes per sample)*32 satisfies both
of the above requirements.
Change-Id: I20de875615141a4a331383a5348abd28b97306f7
- Changes for device/Usecase selection
- Add new and combo devices for headphones
- Combo device handling
- Add new backend and logic to handle multiple
backends
- Remove hardcoded mixers and select them
dynamically based on the device
Change-Id: I7207291f2c27da13ba0cc77c66de1144f4b9888a
SLIMBUS_0 is the backend on 8952 external codec PLs.
Whereas TERT_MI2S is the backend on 8952 internal codec PLs.
Since both these PLs are supported on same code base,
provision is required to select backend based on codec type.
Change-Id: I9c0dd7d952ab9d6b9bf738074d97597622e09590
Newer msm chips may support multiple codecs, leading
to existence of multiple mixer paths files once for each.
Below change implements a generic algorithm to accommodate
this change.
Get the codec internal name i.e. tomtom, tasha etc.
from the sound card name and form the mixer paths
file name dynamically. This is generic way of picking
any codec name based mixer files in future with no
code change. This code assumes mixer files are formed with
format as mixer_paths_internalcodecname.xml.
e.g. mixer_paths_tasha.xml etc.
If this dynamically read mixer files fails to open then it
falls back to default mixer file i.e. mixer_paths.xml. This is
done to preserve backward compatibility but not mandatory as
long as the mixer files are named as per above assumption.
Change-Id: Iefceaee525093e8b6a4fb06f6dd4649a262fbf12
With new Voice architecture, a voice session per subscription
will support different call types such as Voice/VoLTE/VoWLAN etc.
These new voice sessions will be tagged with new VSID.
Change-Id: I7e9c90fb3935ebce5ff281f2d1c25c66aa96e55e
Fluence module on ADSP is upgraded to support source
tracking, sound focus and audio zoom features. This
change adds support for the same in audio HAL and is
needed to support the features end-to-end.
Source Tracking is the capability to identify the
source of speech. Sound Focus is the capability to
configure which sectors in the 360 degrees plane to
listen to. Audio Zoom is the capability to combine
the sound focus with other technologies so that a user
can listen in on a particular source in a scene.
Change-Id: I019b22d6541e6d1a10552d808c3320a998b86e48
- 3rd parameter in strncat is not calculated properly. Replace with
strlcat() and fix the concatenation.
- Rename mixer control name to clarify where it applies.
Change-Id: I70af10190aaaf63b2a7d7e16a96ecda603b163dd
CRs-fixed: 811371
Currently, ALIGN macro can align properly only if the buffer size
is a power of 2.
To have same PCM samples for all channels, the buffer size requires to
be multiple of (number of channels * bytes per sample).
For writes to succeed, the buffer must be written at address which is
multiple of 32.
Alignment of 96 satsfies both of the above requirements.
Change-Id: I5ab33ef198b5b8f18ae9f09c4361e5cff84e700e
CRs-Fixed: 795936
* quic/LA.BR.1: (37 commits)
hal: Performance mode during recording usecase
audio: Support for compress offload recovery
audio: Fix for clip skip issue during SSR
hal: add checks for calibration buffer allocation failures
mm-audio: aenc-aac: fix integer overflow for encoded buffer timestamp calculation
hal: Add support for 8909 QRD skue
hal: Fix for FM mute issue on start of FM record
hal: Fix array length computation for backend id array.
hal : set Non SA+ app type for PCM Capture
hal : Fix for SA+ app type selection for PCM Capture usecase
hal : Fix for SA+ app type selection for PCM Capture usecase
hal: use 0 as default return value for out_get_render_position
audio: hal: Add support for pm8916 on msm8909
hal: Define a new combo device for AANC and Fluence
hal: Define a new combo device for AANC and Fluence
hal: use 0 as default return value for out_get_render_position
hal: Update DS2 implementation to support ACDB based license mechanism
hal : Overwrite the APP type for PCM RX and TX path
audio: HAL to support for peripheral manager
hal : Set the input device for VOIP calls using audio path
...
Conflicts:
hal/Android.mk
hal/audio_extn/audio_extn.c
hal/audio_extn/audio_extn.h
hal/audio_extn/dolby.c
hal/audio_extn/utils.c
hal/audio_hw.c
hal/msm8916/hw_info.c
hal/msm8916/platform.c
hal/msm8974/platform.c
hal/platform_api.h
Change-Id: Ibfa171e8f3af713dbb2cffbaf2ca2b0df3e8ae73
Aligning PCM buffer size to 1024 bytes is not required in case
of PCM offloading. It actually creates problem when playing
6 ch, 96KHz, 16-bit stream PCM in offload mode, due to aligning
PCM data becomes unequal for the six channels.
Align the buffer to LCM of 2, 4, 6, 8 i.e. 24.
Change-Id: I0823ca2ccdd5fceaf23a182b763a1d69f91f602f
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Merge tag 'AU_LINUX_ANDROID_LA.BF64.1.05.00.00.021.095' into merge_branch
AU_LINUX_ANDROID_LA.BF64.1.05.00.00.021.095 based on quic/aosp/LA.BF64.1
* tag 'AU_LINUX_ANDROID_LA.BF64.1.05.00.00.021.095': (30 commits)
audio: Reject incall record request when mode is not IN_CALL
mm-audio: aenc-aac: fix integer overflow for encoded buffer timestamp calculation
hal: Add support for parsing hdmi edid information for msm8992
hal: Fix glitch in audio playback during device switch
hal: add support to query hpx state using getparameter
hal: add all compressed formats for app type selection
hal: Performance mode during recording usecase
hal: fix null pointer de-reference
hal : add 8992 for msm8992 audio
post_proc: Support Headphone:X in offload and non tunnel modes
post_proc: Add support for non-tunnel DSP audio effects
audio: Avoid mute during AFE PROXY Usecase
hal: Fix DTS HPX Get parameters
hal: bug fixes for PCM offload
hal: Fix progress bar issue during compress offload playback
hal: Fix setting stream channel map properly
hardware/qcom/audio: Integrate Eagle framework for HeadphoneX
hal: Remove setting DAP bypass /DAP on from HAL
hal: Avoid resetting device to speaker for HDMI passthrough case
hal: Audio HAL / APM changes for HDMI passthrough
...
Conflicts:
Android.mk
hal/Android.mk
hal/audio_extn/audio_extn.h
hal/audio_extn/utils.c
hal/audio_hw.c
hal/voice.c
Change-Id: Ia92b2002486894e8711a48480324479fe114aeb6
Currently the deep buffer path uses 4 * 20msec buffer.
During device switch the AudioPolicyManager delays routing
command by 2 * output latency (80msec) to ensure that all
the data written to driver/DSP is played out on current
device itself.
The stream side buffering in the DSP in legacy PCM mode is
95msec and hence the depth of pipeline is 80 + 95 = 175msec
which exceeds the device switch delay 160msec (2 * 80). So
the tail (data written to driver before headset plugin) is
heard on headset and perceived as glitch.
Ensure that the buffering in the kernel is greater than or
equal to the buffering in the DSP to fix the issue.
Change-Id: I01a3862d63ce4c258056620693dee08761c7e83f
CRs-Fixed: 771446
Currently the deep buffer path uses 4 * 20msec buffer.
During device switch the AudioPolicyManager delays routing
command by 2 * output latency (80msec) to ensure that all
the data written to driver/DSP is played out on current
device itself.
The stream side buffering in the DSP in legacy PCM mode is
95msec and hence the depth of pipeline is 80 + 95 = 175msec
which exceeds the device switch delay 160msec (2 * 80). So
the tail (data written to driver before headset plugin) is
heard on headset and perceived as glitch.
Ensure that the buffering in the kernel is greater than or
equal to the buffering in the DSP to fix the issue.
Change-Id: I01a3862d63ce4c258056620693dee08761c7e83f
CRs-Fixed: 771446
When pcm offload is done, override the buffer size
that was calculated and use the value from the system property
Make write call blocking if small buffers are used in offload
Update latency value for pcm offload with small buffer hint based
on period size and period count.
Change-Id: Ic74caa6bd172c8e4554384e9fa98a5137117f07c
-Add support for HDMI passthrough for DD/DDP contents to HAL and
APM.
-Add mixer control to set volume on volume module on passthrough
COPP. The mixer control takes zero volume for mute and unity
volume for max volume.
-DDP content requires sample rate to be four times native sample
rate. HAL sample rate is changed to four times native rate from
start output stream. During rapid pause/resume offload thread
calls standby when there is no active track and no activity for
over a second. On resume start output steam is called and
sample rate is converted to four times its current value.
This results in session to be started with invalid sampling
rate and playback failures.
-Add HDMI passthrough support for JOC format.
Expose DDP/JOC as pass-through supported format if sink
supports either DD or DDP. This is to allow support for
pass-through convert based on edid data.
-Deep buffer music stream does not switch back to HDMI after
ringtone playback ends. Ringtone, alarm, notification etc are
played on speaker if HDMI pass-through is enabled. The
decision is taken based on stream or strategy. The decision
to change the device from HDMI to speaker is taken based on
the stream type. When a patricular stream ends use the
reference count instead of stream type.
-DAP is not turned on when switching from passthrough stream to
deep buffer stream.DAP bypass call from HAL to DAP HAL expects
integer pointer. Address of a bool variable is passed into the
function. Corrupt value is set on driver instead of turn on/off
causing random failures in DAP on/DAP bypass. Use same data type
and use enumeration to make the function call readable.
CRs-Fixed: 761339
Change-Id: Ided9439ec5e87233a1fec3ff9f50a1e9ba4cb788
HW echo cancellation is not working after device switch due
to state mismatch between tinyalsa and the kernel.
Ensure that EC reference is disabled always before enabling
EC reference for the new device.
Change-Id: Ia1152402990db836d3f94278a7d36726310dc60c
added base64 encode and decode utility functions
Add API to set & get audio calibration.
Use audiocal for set & get key.
Change-Id: I2ea3362a75bd99018f404d8cf0771efa0a2aabf7
Add quad mic recording support for standalone audio recording
and for VoIP calls using audio path.
Change-Id: I852ca864928b73892a88eccf67c2b9803df9cdad
Linux Build Service Account
Naresh Tanniru
Venkata Narendra Kumar Gutta
Alexy Joseph
Laxminath Kasam
Apoorv Raghuvanshi
Preetam Singh Ranawat
Karthik Reddy Katta
Pradnya Chaphekar
Vidyakumar Athota
Narsinga Rao Chella
Tanya Finkel
Fred Oh
Mingming Yin
Banajit Goswami
Amit Shekhar
Shruthi Krishna
Ravi Kumar Alamanda
Dhananjay Kumar
Ben Romberger