msm8660: only use voip stream if sample rate is 16k or less

Change-Id: If735eee450fa9f9c99aa83c9cf1a81a955b12c57
msm8660: add routing for VOIP_CALL input stream

Change-Id: I5556f308670c6fe94dd8a82e2cb9484affd5e791
msm8660: use defines instead of "magic" numbers

* use AUDIO_HW_VOIP_SAMPLERATE_8K and AUDIO_HW_VOIP_SAMPLERATE_16K
  for sample rates for voip
* use AUDIO_HW_VOIP_BUFFERSIZE_8K and AUDIO_HW_VOIP_BUFFERSIZE_16K
  for buffer sizes for voip

Change-Id: Iba62c778711a5a9258a8492719dcabed834513dc
msm8660: fix getInputBufferSize to calculate buffer sizes

* dirived from alsa driver

Change-Id: Ife6dc216230617c2bc11bf6a51abe0488075acc5