| Commit message (Collapse) | Author | Age | Files | Lines |
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Change-Id: Iaf33b147367d95bd8c93e1e760b9ae668ea4939c
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-Fixes crash on mediaserver if the device_count got from
audioalsa library is greater than DEVICE_COUNT in HAL
Change-Id: I802af5f976620502dfe4c0761b81ad42d6fde81d
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Change-Id: Iebbfceecffb8d2afe3db77dc5a4378cc2791e91f
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* Build with FM_RADIO enabled fail as it conflict with
FM_RADIO stream type
Change-Id: Idb39bb2d883cce2f1eee332d0ddf7be1e94e90ce
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* Add explict case for HANDSET, and use SPEAKER as fallback
Change-Id: I3f4a7f0b0ab78123e214292588449eea8aac2d58
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Change-Id: Idb0277a023f9c8683d3cd0c734916fbb4671ec32
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- Sony Xperia S have camcorder_tx but it point to the primary mic
which is placed on the bottom of the phone, we can use secondary
mic which is placed behind camera
Change-Id: I7892495b5fc201d527617ad0a686d1e886b49b99
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-Issues:PCM music does not mute for LPA ringtone playback over headset.
-Causes:Media is not muted on correct output during LPA ringtone preview.
-Fixes:Mute STRATEGRY_MEDIA on both hardware and LPA output(if active)
for ringtone preview.
Change-Id: Iffd01eee4b7e5e2f6e35bf52a2a1f6ffbe74b5f9
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Change-Id: Ic6d7138ea52d3f8e6f15b129a3d4107459b2096b
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* No longer needed with the patch below since you
can set it to true/false value now.
http://review.cyanogenmod.com/#/c/18089/
Change-Id: Ie7d84c450240d1b77935a9947713bda54fbb4cd1
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* Improves call quality
* Gets rid of call echo
Change-Id: I15ce70e9a0ca948d7917fd429c428161577ff4a0
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* There doesn't seem to be a good reason for using HANDSET as the
fallback route for playback. This causes problems when existing
VR/PCM_REC mode with some apps.
* Change the default route to SPEAKER and add explict case for HANDSET.
Change-Id: I09f784167154cb27d09d9a3848b9650af3cee07c
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* Some devices have a mic dedicated to the camcorder. Enable it.
Change-Id: I75fb7b2e36c37505a603f2aaed7362199535055a
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* Samsung has specially-tuned channels for VR, let's use them.
Change-Id: I225e02167744147cdea7b9bd46c823cf602a5d49
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fix profile selection on gtalk
Change-Id: Iafbd87b4e5a6e68a0945990876b0f85b36cd5983
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also handle aic3254 power
Change-Id: Ia8cf4ffb85a8ba5123701ae0336130dbae5dbbc2
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removed AudioStreamInMSM72xx::read spam
removed aic3254_set_volume, it's never used
revised aic3254 debug messages
always unmute mic when call start
Change-Id: Idf5631fc67c945490a57c0966fc8c1452b798d67
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* Add missing SND_DEVICE_FM_TX_AND_SPEAKER const
* Change msm_audio_7x30.h to msm_audio.h
NB: this require that your local include file is renamed.
For those using kernel 2.6.35, a few patches from 3.0 kernel is needed
for proper functioning LPA. See http://bit.ly/KJfMYd
Further more, you need the following in BoardConfig.mk:
COMMON_GLOBAL_CFLAGS += -DWITH_QCOM_LPA
TARGET_USES_QCOM_LPA := true
and in system.prop:
lpa.decode=true
lpa.use-stagefright=true
Change-Id: Iba1b70434fb795946388a3587c084754025db0b6
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* Adds support for the call and VOIP channels found on this hardware.
* Also raise the max number of devices since we are over the limit.
* VOIP channel is currently disabled until MVS is working.
Change-Id: I52617b4578196b4db5a00d859cfa5bbc92757a3d
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camcorder
- During Camcorder Recording + Voice call with headset connected,
audio is routing back to headset if the headset is removed and
plugged in back to the target. But cam corder is always expected
to record from handset tx.
- When headset is removed and plugged in back, setDeviceConnectionState()
is updating only output devices. Input devices not being updated as the
condition checked for updating the input devices is always false.
Also PhoneApp sends two intents headset connected and tty mode change
when headset is connected. So even if input device is updated in
SetDevice ConnectionState(), because of tty mode change intent audio
is again routing to headset.
- Fix this issue by changing the condition to update input devices and
ignore the tty mode change event if current device is not headset.
Change-Id: I9de6da353bc23e279835403c163d6f400e024d0b
CRs-fixed: 336124
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- Volume does not change when a voice call is the
first use case with all the tones disabled.
- When voice call is the first use case, acdb_loader_send_voice_cal()
is called before ACDB is initialized causing ACDB to be
initialized incorrectly.
- Fixed the issue by calling initACDB() before acdb_loader_send_voice_cal()
CRs-Fixed: 356557
Change-Id: Id696bc2548cc68b6e1ac023bb12f852af08c6d7e
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-When Voip call is triggered, observe voice mute
issue sometimes.
-When Voip call is started, there is a ringer
that comes first and speaker is enabled. After
accepting the call, PolicyManager initiate route
device to Handset for Voip Usecase. Meanwhile
VoipStreamoutDirect thread opens MVS driver and also
routes the voice stream to current active rx device
and enables it. So enable Device is happening in
above two contexts at same time causing issue.
Need to acquire Lock in Voip context to avoid this.
Change-Id: I95c09204ae615e7255ed18beae9bd5c0a154eb8a
CRs-Fixed: 349440
Conflicts:
audio/msm8660/AudioHardware.cpp
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* This should be a float, not an int. It was breaking LPA and probably
other things.
Change-Id: Icd021717995025fb9580d7b73693ee7381eb8622
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Change-Id: I7cc45c08340cf7a78e0ebb03c7af02dbec6ffab1
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- Store float voice volume got from audioflinger even if
and re-use on next call.
- Check mic mute state before mute/unmute it.
Change-Id: I0e50157986be4a7bc1bf7f2394d1af110540b59c
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- Store float voice volume got from audioflinger even if
msm_set_voice_rx_vol return an error and re-use it on next call.
- Check mic mute state before mute/unmute it.
- Remove redundant volume set.
Change-Id: Iaa65503e3dbe0464ac15c7e1cb00064b391c1123
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- No voice heard, during the first voice call after SSR. The SSR
is done during a concurrency of voice call and video streaming.
- The on going call before the SSR is not terminated, and the voice
driver doesn't gets notified about this and ignores a new request
for voice. Hence there is no voice heard in the first call after SSR.
- Fix is to terminate voice in audio HAL constructor there by
notifying the voice driver that the call has ended. If there is
no voice active, the driver would ignore the request.
CRs-fixed: 339562
Change-Id: I063b42c55e557bc2b0ddbc07bf040533cb61e944
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Change-Id: I7d4eae80e7603445813c4bf87d07a5e73223b3b5
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* Also fix build with QCOM_VOIP enabled.
Change-Id: Idfc752850a4f46b02b6c5ab74e3a28162de8e075
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Change-Id: Ic30702c5e85917ce18ab0d7b6be08c8dc3abb6c1
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Change-Id: Ic98a6336f6487589e434c228bfcbd1bb4a37634e
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- Merge changes related to STRATEGY_ENFORCED_AUDIBLE from
AudioPolicyManagerBase to 7630 specific AudioPolicyManager
Change-Id: I06012c008bb3128622f265bf1095f275d9587e18
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- Merge changes related to STRATEGY_ENFORCED_AUDIBLE from
AudioPolicyManagerBase to 7627a specific AudioPolicyManager
Change-Id: I9b74d586c41c2a1eb5fb3cde5489ab7acbfd1779
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Change-Id: Ia35e9d975735f2e3fb616681bc14b3ebbfa47610
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Change-Id: I8c0c5a64de4888597681e7bdef7302b541147965
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Change-Id: I7ddd676ca015a214b9778e9914aae032e6f55536
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Change-Id: Ib7729c4265ed79fb680ee4cae8c6bad9c605bca9
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- Make a voip call, mute the mic and end the call.
Now make another voip call, observe mic mute still
present.
- the mute flag related to voip session is not reset
when the call ended. It will result in carry forwarding
the mute to next new voip session and remains muted
until the mic Mute button is pressed again.
- Reset the mute flag of voip and voice session
at the end of respective call sessions.
CRs-Fixed: 343241
(cherry picked from commit dff334e51d45705c5f39caa76eb20431b0791c85)
Change-Id: Id09380338421fcb6c1d8098ac258505fc3d0f772
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Change-Id: Ib39dd8867353743afc8ca0ea301ce6b4dafb2dde
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CM doesn't support them, and this prevents the modules from building
Change-Id: Id04f9e6f04e4ae9ed390b829c88d358e5867585a
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Change-Id: Ifcd3219ca1f67805641230158e89df8ddf1e04b5
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Change-Id: I2047995dc91b3060b98a3ad20ffa0d71e5e2f6b3
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-When Gtalk voip call is used, observing distortion on
uplink voice.
-There is a sofware bug in the code while returning the
number of bytes read from HAL StreamInVoip read. The
upper layers rely on this value for calculating the delay
to read the next voice uplink packet. As this value is
returned 1280 bytes instead of 640, it is resulting in
20ms more delay for each packet read and causing
packet drops and disortion in uplink voice.
-Fix the issue by returning the actual number of bytes
that are read from MVS and filled into buffer.
CRs-Fixed: 331850
(cherry picked from commit b2a90147720040d6bdf8df0737370052507be7b0)
Change-Id: I662e3d47b919a20198b0203539ddb5d5bb2073ea
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