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* audio: fix error in capture path delay calculationEric Laurent2012-04-021-2/+5
| | | | | | | | 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
* am d28a1a80: am 467c02b6: am 78a7609d: audio: fix audio drop when speaker is ↵Eric Laurent2012-03-201-0/+12
|\ | | | | | | | | | | | | selected * commit '2cb034ebbf5eb4f9ead26150d288bf6d90dc2fee': audio: fix audio drop when speaker is selected
| * audio: fix audio drop when speaker is selectedEric Laurent2012-03-161-0/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | 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
* | audio_channel_in_mask_from_countGlenn Kasten2012-03-151-5/+1
| | | | | | | | Change-Id: Ib1d5af6687479c8d189a3407c229a6ac0ed5c03b
* | Prepare to move system/mediaGlenn Kasten2012-03-141-2/+2
| | | | | | | | Change-Id: Ifb68db236cb6b9e039eadf573e177add1de62d8c
* | Fix memory leaksGlenn Kasten2012-02-141-1/+2
| | | | | | | | Change-Id: If9c95a4808785e58ee4595e5c762d01d87f1936d
* | resolved conflicts for merge of 8c61349a to masterSimon Wilson2012-01-261-48/+117
|\ \ | |/ | | | | Change-Id: Id432e901f8107a00a7f371e5882b1290a1154961
| * audio: support multiple output PCMsSimon Wilson2012-01-251-48/+117
| | | | | | | | Change-Id: I5179699b22224473bd158e90f864e4e73895b5dc
* | Use audio_format_t consistentlyGlenn Kasten2012-01-201-9/+9
| | | | | | | | Change-Id: I2e2a5f625956dc5d09dbdc3f6f2d9a010ecc7bad
* | Turn off execute bitGlenn Kasten2012-01-183-0/+0
| | | | | | | | Change-Id: I711920dde1560ca202ef878ee93a2af61545524b
* | Use audio_mode_t consistentlyGlenn Kasten2012-01-121-2/+2
| | | | | | | | Change-Id: I7a30fe3f66933aed8b5a6185553112575b4de1a7
* | Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGESteve Block2012-01-082-16/+16
| | | | | | | | | | | | | | See https://android-git.corp.google.com/g/#/c/157220 Bug: 5449033 Change-Id: I2e1c43800c19b718cc7ee94ec299c62bc14873b4
* | Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGESteve Block2012-01-061-2/+2
| | | | | | | | | | | | | | See https://android-git.corp.google.com/g/157065 Bug: 5449033 Change-Id: I02cfaca251935e4a50ad4302a72c4273be41db22
* | am 31688e73: am 7a170e19: audio HAL: release audio pre processing buffers.Eric Laurent2011-12-131-0/+4
|\ \ | |/ | | | | | | * commit '620c8ad611fee5de98f778514c0418c1f48903e5': audio HAL: release audio pre processing buffers.
| * audio HAL: release audio pre processing buffers.Eric Laurent2011-12-131-0/+4
| | | | | | | | | | | | | | | | | | Buffers allocated for audio pre processing are not released when an input stream is closed. Issue 5753047. Change-Id: Ie8fd5f49d97e9bebc70fc38de0844a79074ac526
* | audio: delete unused ril-client API.UK KIM2011-11-103-14/+1
| | | | | | | | | | | | The clock sync func is unused for both HSPA and LTE device. Change-Id: Ia9f369a0151cb3bb15242544e5f5442b893253bc
* | am ec429c13: Merge "audio: force speaker route for call when docked" into ↵Simon Wilson2011-11-021-3/+6
|\ \ | |/ | | | | | | | | | | ics-mr1 * commit '8e4929d7f9501e499853bd51ad0ce7cc8b586906': audio: force speaker route for call when docked
| * audio: force speaker route for call when dockedSimon Wilson2011-10-311-3/+6
| | | | | | | | | | | | | | | | | | | | | | 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
* | am 56e8b292: am e6f399a5: audio: decrease headset gain by 14dB for ringtone modeUK KIM2011-10-261-0/+3
|\ \ | |/ | | | | | | * commit '361998818f2d0f5008c24eeeb1c4ad013a01a862': audio: decrease headset gain by 14dB for ringtone mode
| * audio: decrease headset gain by 14dB for ringtone modeUK KIM2011-10-261-0/+3
| | | | | | | | | | | | This is to prevent audio shock in AUDIO_MODE_RINGTONE. Change-Id: Ic21c347a64ee0e2668dbff49dc6addcb93e4d82f
* | Rename LOGV(_IF) to ALOGV(_IF) DO NOT MERGESteve Block2011-10-261-5/+5
|/ | | | | | | See https://android-git.corp.google.com/g/#/c/143865 Bug: 5449033 Change-Id: Iab0aa050fba57491f5cb7ed928f44a0fda7d1ea4
* audio: Fix pop noises during call switch to the modemvenkappa mala2011-10-201-2/+22
| | | | | | | | | 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>
* audio: final audio gains following tuningSimon Wilson2011-10-201-45/+87
| | | | | | | | - new gains for toro and maguro devices for various use cases. - use of DL2 digital gains to compensate for lack of range in codec speaker volume. Change-Id: I4ff1ebe79aa53934720389fbef5f60b9c0cc2138
* audio: enable DL2 mono mixer only for speaker/mediaSimon Wilson2011-10-201-12/+11
| | | | | | | | | | Mono mixer is only strictly required for downmixing stereo media content to the mono speaker, so only enable it then. This works around an issue with modem rx mute when using handsfree. Fixes bug 5481245 Change-Id: I8e4c5400241a0d8bb8d74966b6f612b7bab56301
* audio: increased low power playback buffer size.Eric Laurent2011-10-201-10/+12
| | | | | | | | | Defined new audio buffer sizes to help increase periods of idle CPU with new scaling governor settings. Related to issue 5486806: mp3 playback power re-regressed... Change-Id: I5f0f54d0ef8e189c2e3ac84bf8eed4bafece9111
* audio: use 4Khz LPF in DL1 while in voicecallChangoh.Heo2011-10-191-7/+30
| | | | | | | | | Some metalic noise is happened on headset, earpiece voicecall. Especially, The noise can be felt easily in woman voice. If we use 4Khz LPF, the noise is gone. Change-Id: I106efd89af2b84fad40314c8c07b5f0aa7901c8b Signed-off-by: Changoh.Heo <changoh.heo@samsung.com>
* audio HAL: low power playback off when capturingEric Laurent2011-10-181-1/+1
| | | | | | | Disable low power audio playback when audio capture is active even if screen is off to avoid high latency during SIP calls. Change-Id: Ib559bf2877b0cf89731e039b1bfab2bc3806f56a
* audio: enable DL2 mono mixer for speakerSimon Wilson2011-10-171-0/+7
| | | | | | | | Since the speaker is only connected to the DL2 left channel, downmix all DL2 audio from stereo to mono to avoid losing information. Change-Id: I8f536d3373b5517682722422df648d9d8050b840
* Merge "audio: get wb amr status when ril is connected" into ics-mr0Simon Wilson2011-10-141-0/+9
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| * audio: get wb amr status when ril is connectedgaon.yoon2011-10-141-0/+9
| | | | | | | | | | | | | | | | | | | | | | At the first incoming call, wb amr callback time is faster than ril-connecting time so wb status is not updated. To update wb amr status get it at ril-connecting time. HSPA supports getting wb amr status, but LTE does not support it. Change-Id: I477cb19f8ef72d5461c2800e09958f504ae733e5
* | audio HAL: support for low power audioEric Laurent2011-10-141-42/+96
|/ | | | | | | | | | | | | Implement a mechanism to dynamically switch between short and long buffers in kernel pcm driver. Using long buffer significantly decreases power consumption at the expense of latency. Therefore a hint is given to audio HAL by AudioService indicating when the screen is off and low latency is not required any more because neither video playback, VoIP/video chat or any user interaction is expected. This mechanism relies on the support for MMAP and NO IRQ write modes in tinyalsa. Change-Id: Ida9216a141750137a0592187e24a68f263ef3fbe
* audio HAL: change ALSA period sizeEric Laurent2011-10-131-2/+2
| | | | | | | ALSA period sizes must be a multiple of 24 frames to match ABE requirement. Change-Id: I52ac1d5d4a2588a1b66100bfecab6d35339fc718
* audio: bypass resampler for HDMI audioSimon Wilson2011-10-061-22/+32
| | | | | | | Native 44.1kHz will be used for HDMI audio since the output device supports it. Change-Id: I60eebf2556c0384e2a4c21150bee2fbbbd5ca6fd
* audio: add locks, only tear down PCMs when needed for WB AMRSimon Wilson2011-10-061-5/+10
| | | | Change-Id: I03ba325b613aef21dba8d16187aaccca08d2a328
* am fcb204e9: Merge "Fix issue 5415809: increase HP volume for TTY." into ↵Simon Wilson2011-10-061-9/+18
|\ | | | | | | | | | | | | ics-factoryrom * commit '44bd1775d04b3fd62825ce6cebcb107db939fc71': Fix issue 5415809: increase HP volume for TTY.
| * Fix issue 5415809: increase HP volume for TTY.Eric Laurent2011-10-061-9/+18
| | | | | | | | | | | | | | Increase headphones volume to -2dB when TTY mode is full or VCO as per Samsung's request. Change-Id: I92da179b487c87d07bc363f7344c20cc8779abd6
* | audio: route to S/PDIF when digital dock detectedSimon Wilson2011-10-051-6/+16
| | | | | | | | Change-Id: Ia571fca8e0ce384283a15024b6b271231bf86479
* | audio HAL: fix echo reference.Eric Laurent2011-10-051-1/+1
| | | | | | | | | | | | | | | | The number of frames written to the echo reference buffer in out_write() was wrong. As we write frames at the audioflinger sampling rate we should write the number of frames passed to out_write(), not the number of frames passed to tynialsa after resamopling. Change-Id: Ia6a1c7e090c73e1566634a17b720e1e6049b22fe
* | audio HAL: fix start_call() error handling.Eric Laurent2011-10-051-3/+3
|/ | | | | | | | In case of an error when opening the modem pcm driver in start_call(), the order in which the tinyalsa pcm streams were relased was wrong and could cause calling pcm_close() on a null pcm stream. Change-Id: Iad7149997d3993561f4a3ed4b2005f5867b51c56
* audio: support wideband call audioSimon Wilson2011-09-303-1/+60
| | | | | | | | Some networks support wideband AMR for voice calls. To support this, implement a callback that the RIL uses to set the wideband config. Change-Id: Ifa75ff189cc300728f560b77fd4fb3f1798e776d
* audio: adjust gains based on level tuningSimon Wilson2011-09-301-11/+20
| | | | Change-Id: I1e7e7738dad3643bd006d19708895f9f5815f429
* audio HAL: different heaphone volume for EuropeEric Laurent2011-09-291-4/+24
| | | | | | | | | Added the possibility to set difference headphones volume to comply to European regulation. Set conservative gains for headphones and headset. Change-Id: I77af0325baca8d5d5a8ebbec2431918cf2bff3a0
* audio: use-case gain adjustmentsSimon Wilson2011-09-291-18/+49
| | | | | | | | | - allow a 6dB higher volume for headphones without mics - increase voice call speaker volume by 6dB - increase voice call sub mic gain for toro by 2dB - turn off headset DAC when only earpiece is active Change-Id: I344b0fc5ec97a6c9ce14a7db7602a4700a2c765e
* Revert "audio: defer ril acoustic call until after modem PCM is open"Simon Wilson2011-09-281-3/+2
| | | | | | | | | | | | Now that the modem PCMs are never closed for route changes, we don't need to defer the call of set_incall_device() any more. This also fixes a bug where the acoustic property is not sent to the modem upon an output device change now that we don't close/open the modem PCMs for every route change. This reverts commit 56c8d101b1a7b6660ce4f2504ee24a7c78eb19b1. Change-Id: I63bc4e25a602d99cd335b7b2a1db4ece45df93e1
* audio: don't tear down modem PCMs for route changeSimon Wilson2011-09-211-17/+0
| | | | | | Fixes bug 5278856 Change-Id: I25bdae020241c2388db298637d111fba1c3acecd
* audio: use right capture path for sub micSimon Wilson2011-09-201-5/+21
| | | | | | | | | | The sub mic is on the right capture path, so when the front end portion of the route is selected, the mic choice must be taken into account. Fixes the lack of sound in camcorder. Fixes bug 5350006 Change-Id: I347922af04a0114a8e269b9edea3eec260175f79
* audio: force initial non-speaker output for callSimon Wilson2011-09-201-6/+26
| | | | | | | | | | | | | | | | | | When the phone is ringing the active output device is always the speaker, perhaps with a secondary output device such as headset. When we answer the call the active output device is still speaker, and set_mode() causes the modem PCM to be opened for this route. However, we never use the speaker as our initial audio route for call audio. This change forces speaker off when we set up the initial in-call state so we don't have to change it immediately when out_set_parameters() is called with a different route. This works for earpiece, headset and headphones. It doesn't help bluetooth because the SCO connection is only begun after the call is started. Change-Id: Ie9f411c61570749fc26ab2ffa18cd1477e68a7e6
* audio: route mono mics through both muxesSimon Wilson2011-09-201-5/+5
| | | | | | | Mono mics were previously only routed to a single channel on each mux. Route through both instead. Change-Id: Ie954a436ec24e377e6821b85b994ed5294a6c4d8
* Revert "audio: change mixer name for earpiece control"Simon Wilson2011-09-191-9/+7
| | | | This reverts commit a6b80f563286d38e7c9210bac9c549b09110d075.
* audio: add support for HDMI ouputEric Laurent2011-09-181-2/+20
| | | | | | | | | | Add basic support of HDMI output during playback. Note that if multiple output devices are set, if HDMI is one of them, only HDMI will be used. Change-Id: I0a3ccdd6824a73553649e63b2d6ccde6aa99310e Signed-off-by: Chris Kelly <c-kelly@ti.com> Signed-off-by: Eric Laurent <elaurent@google.com>
generated by cgit v1.1 at 2024-05-11 15:42:27 +0000