Merge master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa

* master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa:
  [ALSA] echoaudio - Remove kfree_nocheck()
  [ALSA] echoaudio - Fix Makefile
  [ALSA] Add Intel D965 board support
  [ALSA] Fix/add support of Realtek ALC883 / ALC888 and ALC861 codecs
  [ALSA] Fix a typo in echoaudio/midi.c
  [ALSA] snd-aoa: enable dual-edge in GPIOs
  [ALSA] snd-aoa: support iMac G5 iSight
  [ALSA] snd-aoa: not experimental
  [ALSA] Add echoaudio sound drivers
  [ALSA] ak4xxx-adda - Code clean-up
  [ALSA] Remove CONFIG_EXPERIMENTAL from intel8x0m driver
  [ALSA] Stereo controls for M-Audio Revolution cards
  [ALSA] Fix misuse of __list_add() in seq_ports.c
  [ALSA] hda-codec - Add model entry for Samsung X60 Chane
  [ALSA] make CONFIG_SND_DYNAMIC_MINORS non-experimental
  [ALSA] Fix wrong dependencies of snd-aoa driver
  [ALSA] fix build failure due to snd-aoa
  [ALSA] AD1888 mixer controls for DC mode
  [ALSA] Suppress irq handler mismatch messages in ALSA ISA drivers
  [ALSA] usb-audio support for Turtle Beach Roadie

1 files changed, 229 insertions, 0 deletions

diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c
new file mode 100644
index 00000000000..891c7051909
--- /dev/null
+++ b/sound/pci/echoaudio/mia_dsp.c
@@ -0,0 +1,229 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock);
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int update_flags(struct echoaudio *chip);
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+			   int gain);
+static int update_vmixer_level(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+	int err;
+
+	DE_INIT(("init_hw() - Mia\n"));
+	snd_assert((subdevice_id & 0xfff0) == MIA, return -ENODEV);
+
+	if ((err = init_dsp_comm_page(chip))) {
+		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		return err;
+	}
+
+	chip->device_id = device_id;
+	chip->subdevice_id = subdevice_id;
+	chip->bad_board = TRUE;
+	chip->dsp_code_to_load = &card_fw[FW_MIA_DSP];
+	/* Since this card has no ASIC, mark it as loaded so everything
+	   works OK */
+	chip->asic_loaded = TRUE;
+	if ((subdevice_id & 0x0000f) == MIA_MIDI_REV)
+		chip->has_midi = TRUE;
+	chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL |
+		ECHO_CLOCK_BIT_SPDIF;
+
+	if ((err = load_firmware(chip)) < 0)
+		return err;
+	chip->bad_board = FALSE;
+
+	if ((err = init_line_levels(chip)))
+		return err;
+
+	/* Default routing of the virtual channels: vchannels 0-3 go to analog
+	outputs and vchannels 4-7 go to S/PDIF outputs */
+	set_vmixer_gain(chip, 0, 0, 0);
+	set_vmixer_gain(chip, 1, 1, 0);
+	set_vmixer_gain(chip, 0, 2, 0);
+	set_vmixer_gain(chip, 1, 3, 0);
+	set_vmixer_gain(chip, 2, 4, 0);
+	set_vmixer_gain(chip, 3, 5, 0);
+	set_vmixer_gain(chip, 2, 6, 0);
+	set_vmixer_gain(chip, 3, 7, 0);
+	err = update_vmixer_level(chip);
+
+	DE_INIT(("init_hw done\n"));
+	return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+	u32 clocks_from_dsp, clock_bits;
+
+	/* Map the DSP clock detect bits to the generic driver clock
+	   detect bits */
+	clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+	clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+	if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_SPDIF)
+		clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+
+	return clock_bits;
+}
+
+
+
+/* The Mia has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+	return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+	u32 control_reg;
+
+	switch (rate) {
+	case 96000:
+		control_reg = MIA_96000;
+		break;
+	case 88200:
+		control_reg = MIA_88200;
+		break;
+	case 48000:
+		control_reg = MIA_48000;
+		break;
+	case 44100:
+		control_reg = MIA_44100;
+		break;
+	case 32000:
+		control_reg = MIA_32000;
+		break;
+	default:
+		DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+		return -EINVAL;
+	}
+
+	/* Override the clock setting if this Mia is set to S/PDIF clock */
+	if (chip->input_clock == ECHO_CLOCK_SPDIF)
+		control_reg |= MIA_SPDIF;
+
+	/* Set the control register if it has changed */
+	if (control_reg != le32_to_cpu(chip->comm_page->control_register)) {
+		if (wait_handshake(chip))
+			return -EIO;
+
+		chip->comm_page->sample_rate = cpu_to_le32(rate);	/* ignored by the DSP */
+		chip->comm_page->control_register = cpu_to_le32(control_reg);
+		chip->sample_rate = rate;
+
+		clear_handshake(chip);
+		return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+	}
+	return 0;
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+	DE_ACT(("set_input_clock(%d)\n", clock));
+	snd_assert(clock == ECHO_CLOCK_INTERNAL || clock == ECHO_CLOCK_SPDIF,
+		   return -EINVAL);
+
+	chip->input_clock = clock;
+	return set_sample_rate(chip, chip->sample_rate);
+}
+
+
+
+/* This function routes the sound from a virtual channel to a real output */
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+			   int gain)
+{
+	int index;
+
+	snd_assert(pipe < num_pipes_out(chip) &&
+		   output < num_busses_out(chip), return -EINVAL);
+
+	if (wait_handshake(chip))
+		return -EIO;
+
+	chip->vmixer_gain[output][pipe] = gain;
+	index = output * num_pipes_out(chip) + pipe;
+	chip->comm_page->vmixer[index] = gain;
+
+	DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+	return 0;
+}
+
+
+
+/* Tell the DSP to read and update virtual mixer levels in comm page. */
+static int update_vmixer_level(struct echoaudio *chip)
+{
+	if (wait_handshake(chip))
+		return -EIO;
+	clear_handshake(chip);
+	return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
+}
+
+
+
+/* Tell the DSP to reread the flags from the comm page */
+static int update_flags(struct echoaudio *chip)
+{
+	if (wait_handshake(chip))
+		return -EIO;
+	clear_handshake(chip);
+	return send_vector(chip, DSP_VC_UPDATE_FLAGS);
+}
+
+
+
+static int set_professional_spdif(struct echoaudio *chip, char prof)
+{
+	DE_ACT(("set_professional_spdif %d\n", prof));
+	if (prof)
+		chip->comm_page->flags |=
+			__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+	else
+		chip->comm_page->flags &=
+			~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+	chip->professional_spdif = prof;
+	return update_flags(chip);
+}
+