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* audio HAL: added resampler on input streamEric Laurent2011-07-221-47/+129
| | | | | | Now supports capture at 8000, 11025, 16000, 22050, 24000, 32000, 44100 and 48000 Hz. Change-Id: I61526e94b8f0d315a1bf8d7587363a44c7d643ae
* audio: disable output stage when going into standbySimon Wilson2011-07-211-26/+33
| | | | | | | | | This should eliminate random clicks and pops heard when audio enters standby since the output stage is disabled before closing the PCM. In addition, this should provide a power savings in standby. Change-Id: Ief0a193e0b31e9ee2f03a58641eaebd2a0d344cb
* audio: reorganize route paths and add BT DL pathChris Kelly2011-07-201-40/+101
| | | | | | | | | | | | | | To better support multiple paths and prepare for input routing, the mixer controls were reorganized into front end and back end paths for the supported routes. BT-SCO downlink was also added. This allows more flexibiity in setting controls but does sacrifice some amunt of abstraction of the underliying ABE design. Change-Id: Ie225ae5bf90b1727178093a5f06636e6b17a737b Signed-off-by: Chris Kelly <c-kelly@ti.com>
* audio: correct low power port idChris Kelly2011-07-201-1/+1
| | | | | Change-Id: I48ffc54219360fbb5f22c695dea63ca269e6fb68 Signed-off-by: Chris Kelly <c-kelly@ti.com>
* audio: set in-call volume when user starts callUK KIM2011-07-191-1/+7
| | | | | | | | The RIL needs to have its voice volume reset every time a call is made, not every time the volume is changed from the upper layer. Change-Id: Id042da241de65f9dfb8d5c52e1b4bb910c7c0219
* audio: check for ril presence before calling into itKim Uk2011-07-193-56/+93
| | | | | | This prevents a seg fault if no ril is present. Change-Id: I5f9443e31bdcab07df21d9f12ed2dd92807300f8
* Fix issue 5048624: Native crash in video chat.Eric Laurent2011-07-191-1/+1
| | | | | | | | The problem is that the audio HAL returns a NULL string when get_parameters() is called from AudioFlinger. It should return return an empty string. Change-Id: I99365b54eb5f3c3b6694cb3e122842dff1799bfd
* audio: reselect output device when changing modeSimon Wilson2011-07-171-4/+8
| | | | | | | | | | | | | It's necessary to reselect the output device when changing from the AUDIO_MODE_IN_CALL state to AUDIO_MODE_NORMAL, otherwise the ASoC driver cannot find a route from the PCM to the endpoint. This is possibly a workaround for a bug in the ASoC driver. Also, this change means the audio HAL will no longer disable the multimedia mixer routes when in a call. This is because audio can be played during a call. Change-Id: I58c4b23289f8d6f9ad53b436215ec9a5d48f4fe1
* audio: use MM_UL2 port for 48kHz audio captureEric Laurent2011-07-141-9/+72
| | | | Change-Id: I0b45cbe4b1007c621c1c61491f94b7e3355eeb64
* audio: fix latency and buffer size calculationEric Laurent2011-07-141-11/+14
| | | | | | | | | | | | out_get_buffer_size() was returning the total buffer size in frames instead of the period size in bytes. It should also take the resampling into account so that the audio flinger buffer duration somehow matches the period duration. The calculation in out_get_latency() was assuming the period size in the pcm config structure is in bytes whereas it is in frames. Change-Id: I2025a89e753355bd321865faa726013e0a97912f
* audio: add support for multiple output devicesSimon Wilson2011-07-121-128/+51
| | | | Change-Id: I3cc1d6bd414301e82002153c08fa530c31527e48
* audio: put pcm into standby before changing output deviceSimon Wilson2011-07-121-2/+1
| | | | | | | | | | This avoids the high-pitched whine when changing the route from speaker to headset by ensuring that no bytes are being transferred through the ABE while the route is altered. It also means the workaround to avoid the kernel panic when exiting standby in out_write() can be removed. Change-Id: I67d391d003bd90892622a212b45a394e2d15ff70
* Audio HAL: added interface for audio preprocessingEric Laurent2011-07-111-0/+24
| | | | Change-Id: Ic784c7fa0e82f6ef398548741b603b55d902ae5c
* audio: output standby supportSimon Wilson2011-07-101-11/+36
| | | | Change-Id: I34894e039311e82feda05b0ac58b93518072244c
* audio: capture supportSimon Wilson2011-07-101-12/+130
| | | | | | | | - Enable start and standby of input stream - Not tested sample rates other than 8/16 - Capture causes a kernel panic if a sound is not played first Change-Id: I44ec338c7fb77c43b12f4d0ee19b9f12c7cc4ad6
* Fix audio output to play audio other than 44k stereoJean-Michel Trivi2011-06-281-0/+4
| | | | | | | | | | | | | | | When opening an output stream, the driver needs to return its format, sample rate and channel mask. Failure to do so will cause AudioFlinger to create the ouput as DIRECT, regardless of the output flags. The consequence of the output thread being in DIRECT mode is that no mixing is done (one sound at a time), and only audio that plays at the HAL's expected audio characteristics (44k 16bit stereo) can be played. The fix consists in returning the format, chanel mask and sample rate values when opening the output stream. Change-Id: Ib26e3337fe199efdba7a70b40df93518aceec04a
* audio: workaround ABE microphone disable problemSimon Wilson2011-06-271-6/+10
| | | | | | | | | The analog microphone uplink stops working if any downlink route is changed when the modem PCMs are open, so as a workaround, only modify the earpiece route when the modem PCMs are closed. Change-Id: Ib725a28da5130546015a9e05da4fca4955ce90bd
* audio: add headset output deviceSimon Wilson2011-06-271-11/+20
| | | | Change-Id: I0f5cb58d0f1fc0372f459a6fb55f30683da414d9
* audio: improve audio routing in voice callSimon Wilson2011-06-271-123/+192
| | | | | | | | With this change, the microphone now works during a voicecall. The handsfree (speaker) route also works but it currently uses the main mic instead of the sub mic. Change-Id: I37aaaefc523b5a6ebc88058c58ccb5443428e3fa
* audio: voice call supportSimon Wilson2011-06-244-31/+269
| | | | | | | - Also ensured that PRODUCT_PACKAGES is set so that the HAL is copied into the filesystem. Change-Id: I89790e5aec1d6beb7d4650316ec070503a35c436
* Enable audio supportSimon Wilson2011-06-202-0/+850
- PCM output works via music player - Sometimes ringtones fail to play - Modem routing is untested but present - PCM input needs to be implemented Change-Id: Ib58bef9674e1c9bb896be521c3d95c4e07e0442b