diff options
| -rw-r--r-- | Documentation/sound/alsa/ALSA-Configuration.txt | 9 | ||||
| -rw-r--r-- | sound/firewire/dice/dice-midi.c | 23 | ||||
| -rw-r--r-- | sound/firewire/dice/dice-pcm.c | 199 | ||||
| -rw-r--r-- | sound/firewire/dice/dice-stream.c | 69 | ||||
| -rw-r--r-- | sound/firewire/dice/dice-transaction.c | 54 | ||||
| -rw-r--r-- | sound/firewire/dice/dice.c | 67 | ||||
| -rw-r--r-- | sound/firewire/dice/dice.h | 8 | ||||
| -rw-r--r-- | sound/mips/Kconfig | 12 | ||||
| -rw-r--r-- | sound/mips/Makefile | 2 | ||||
| -rw-r--r-- | sound/mips/au1x00.c | 734 | ||||
| -rw-r--r-- | sound/usb/card.c | 56 | ||||
| -rw-r--r-- | sound/usb/midi.c | 15 | ||||
| -rw-r--r-- | sound/usb/midi.h | 14 | ||||
| -rw-r--r-- | sound/usb/quirks.c | 36 | ||||
| -rw-r--r-- | sound/usb/quirks.h | 3 |
15 files changed, 222 insertions, 1079 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 48148d6d9307..fc53ccd9a629 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1910,6 +1910,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. - Default: 0x0000 ignore_ctl_error - Ignore any USB-controller regarding mixer interface (default: no) + autoclock - Enable auto-clock selection for UAC2 devices + (default: yes) + quirk_alias - Quirk alias list, pass strings like + "0123abcd:5678beef", which applies the existing + quirk for the device 5678:beef to a new device + 0123:abcd. This module supports multiple devices, autoprobe and hotplugging. @@ -1919,6 +1925,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. NB: ignore_ctl_error=1 may help when you get an error at accessing the mixer element such as URB error -22. This happens on some buggy USB device or the controller. + NB: quirk_alias option is provided only for testing / development. + If you want to have a proper support, contact to upstream for + adding the matching quirk in the driver code statically. Module snd-usb-caiaq -------------------- diff --git a/sound/firewire/dice/dice-midi.c b/sound/firewire/dice/dice-midi.c index 151b09f240f2..2461311e695a 100644 --- a/sound/firewire/dice/dice-midi.c +++ b/sound/firewire/dice/dice-midi.c @@ -103,16 +103,27 @@ static void set_midi_substream_names(struct snd_dice *dice, int snd_dice_create_midi(struct snd_dice *dice) { + __be32 reg; struct snd_rawmidi *rmidi; struct snd_rawmidi_str *str; - unsigned int i, midi_in_ports, midi_out_ports; + unsigned int midi_in_ports, midi_out_ports; int err; - midi_in_ports = midi_out_ports = 0; - for (i = 0; i < 3; i++) { - midi_in_ports = max(dice->tx_midi_ports[i], midi_in_ports); - midi_out_ports = max(dice->rx_midi_ports[i], midi_out_ports); - } + /* + * Use the number of MIDI conformant data channel at current sampling + * transfer frequency. + */ + err = snd_dice_transaction_read_tx(dice, TX_NUMBER_MIDI, + ®, sizeof(reg)); + if (err < 0) + return err; + midi_in_ports = be32_to_cpu(reg); + + err = snd_dice_transaction_read_rx(dice, RX_NUMBER_MIDI, + ®, sizeof(reg)); + if (err < 0) + return err; + midi_out_ports = be32_to_cpu(reg); if (midi_in_ports + midi_out_ports == 0) return 0; diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 9b3431999fc8..a5c9b58655ef 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -9,99 +9,40 @@ #include "dice.h" -static int dice_rate_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_pcm_substream *substream = rule->private; - struct snd_dice *dice = substream->private_data; - - const struct snd_interval *c = - hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_interval *r = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval rates = { - .min = UINT_MAX, .max = 0, .integer = 1 - }; - unsigned int i, rate, mode, *pcm_channels; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - pcm_channels = dice->tx_channels; - else - pcm_channels = dice->rx_channels; - - for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { - rate = snd_dice_rates[i]; - if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) - continue; - - if (!snd_interval_test(c, pcm_channels[mode])) - continue; - - rates.min = min(rates.min, rate); - rates.max = max(rates.max, rate); - } - - return snd_interval_refine(r, &rates); -} - -static int dice_channels_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_pcm_substream *substream = rule->private; - struct snd_dice *dice = substream->private_data; - - const struct snd_interval *r = - hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *c = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_interval channels = { - .min = UINT_MAX, .max = 0, .integer = 1 - }; - unsigned int i, rate, mode, *pcm_channels; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - pcm_channels = dice->tx_channels; - else - pcm_channels = dice->rx_channels; - - for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { - rate = snd_dice_rates[i]; - if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) - continue; - - if (!snd_interval_test(r, rate)) - continue; - - channels.min = min(channels.min, pcm_channels[mode]); - channels.max = max(channels.max, pcm_channels[mode]); - } - - return snd_interval_refine(c, &channels); -} - -static void limit_channels_and_rates(struct snd_dice *dice, - struct snd_pcm_runtime *runtime, - unsigned int *pcm_channels) +static int limit_channels_and_rates(struct snd_dice *dice, + struct snd_pcm_runtime *runtime, + struct amdtp_stream *stream) { struct snd_pcm_hardware *hw = &runtime->hw; - unsigned int i, rate, mode; + unsigned int rate; + __be32 reg[2]; + int err; - hw->channels_min = UINT_MAX; - hw->channels_max = 0; + /* + * Retrieve current Multi Bit Linear Audio data channel and limit to + * it. + */ + if (stream == &dice->tx_stream) { + err = snd_dice_transaction_read_tx(dice, TX_NUMBER_AUDIO, + reg, sizeof(reg)); + } else { + err = snd_dice_transaction_read_rx(dice, RX_NUMBER_AUDIO, + reg, sizeof(reg)); + } + if (err < 0) + return err; - for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) { - rate = snd_dice_rates[i]; - if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0) - continue; - hw->rates |= snd_pcm_rate_to_rate_bit(rate); + hw->channels_min = hw->channels_max = be32_to_cpu(reg[0]); - if (pcm_channels[mode] == 0) - continue; - hw->channels_min = min(hw->channels_min, pcm_channels[mode]); - hw->channels_max = max(hw->channels_max, pcm_channels[mode]); - } + /* Retrieve current sampling transfer frequency and limit to it. */ + err = snd_dice_transaction_get_rate(dice, &rate); + if (err < 0) + return err; + hw->rates = snd_pcm_rate_to_rate_bit(rate); snd_pcm_limit_hw_rates(runtime); + + return 0; } static void limit_period_and_buffer(struct snd_pcm_hardware *hw) @@ -122,7 +63,6 @@ static int init_hw_info(struct snd_dice *dice, struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hardware *hw = &runtime->hw; struct amdtp_stream *stream; - unsigned int *pcm_channels; int err; hw->info = SNDRV_PCM_INFO_MMAP | @@ -135,37 +75,22 @@ static int init_hw_info(struct snd_dice *dice, if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { hw->formats = AM824_IN_PCM_FORMAT_BITS; stream = &dice->tx_stream; - pcm_channels = dice->tx_channels; } else { hw->formats = AM824_OUT_PCM_FORMAT_BITS; stream = &dice->rx_stream; - pcm_channels = dice->rx_channels; } - limit_channels_and_rates(dice, runtime, pcm_channels); - limit_period_and_buffer(hw); - - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - dice_rate_constraint, substream, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - if (err < 0) - goto end; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - dice_channels_constraint, substream, - SNDRV_PCM_HW_PARAM_RATE, -1); + err = limit_channels_and_rates(dice, runtime, stream); if (err < 0) - goto end; + return err; + limit_period_and_buffer(hw); - err = amdtp_am824_add_pcm_hw_constraints(stream, runtime); -end: - return err; + return amdtp_am824_add_pcm_hw_constraints(stream, runtime); } static int pcm_open(struct snd_pcm_substream *substream) { struct snd_dice *dice = substream->private_data; - unsigned int source, rate; - bool internal; int err; err = snd_dice_stream_lock_try(dice); @@ -176,39 +101,6 @@ static int pcm_open(struct snd_pcm_substream *substream) if (err < 0) goto err_locked; - err = snd_dice_transaction_get_clock_source(dice, &source); - if (err < 0) - goto err_locked; - switch (source) { - case CLOCK_SOURCE_AES1: - case CLOCK_SOURCE_AES2: - case CLOCK_SOURCE_AES3: - case CLOCK_SOURCE_AES4: - case CLOCK_SOURCE_AES_ANY: - case CLOCK_SOURCE_ADAT: - case CLOCK_SOURCE_TDIF: - case CLOCK_SOURCE_WC: - internal = false; - break; - default: - internal = true; - break; - } - - /* - * When source of clock is not internal or any PCM streams are running, - * available sampling rate is limited at current sampling rate. - */ - if (!internal || - amdtp_stream_pcm_running(&dice->tx_stream) || - amdtp_stream_pcm_running(&dice->rx_stream)) { - err = snd_dice_transaction_get_rate(dice, &rate); - if (err < 0) - goto err_locked; - substream->runtime->hw.rate_min = rate; - substream->runtime->hw.rate_max = rate; - } - snd_pcm_set_sync(substream); end: return err; @@ -402,17 +294,30 @@ int snd_dice_create_pcm(struct snd_dice *dice) .page = snd_pcm_lib_get_vmalloc_page, .mmap = snd_pcm_lib_mmap_vmalloc, }; + __be32 reg; struct snd_pcm *pcm; - unsigned int i, capture, playback; + unsigned int capture, playback; int err; - capture = playback = 0; - for (i = 0; i < 3; i++) { - if (dice->tx_channels[i] > 0) - capture = 1; - if (dice->rx_channels[i] > 0) - playback = 1; - } + /* + * Check whether PCM substreams are required. + * + * TODO: in the case that any PCM substreams are not avail at a certain + * sampling transfer frequency? + */ + err = snd_dice_transaction_read_tx(dice, TX_NUMBER_AUDIO, + ®, sizeof(reg)); + if (err < 0) + return err; + if (be32_to_cpu(reg) > 0) + capture = 1; + + err = snd_dice_transaction_read_rx(dice, RX_NUMBER_AUDIO, + ®, sizeof(reg)); + if (err < 0) + return err; + if (be32_to_cpu(reg) > 0) + playback = 1; err = snd_pcm_new(dice->card, "DICE", 0, playback, capture, &pcm); if (err < 0) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index a6a39f7ef58d..e4938b0cddbe 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -10,6 +10,7 @@ #include "dice.h" #define CALLBACK_TIMEOUT 200 +#define NOTIFICATION_TIMEOUT_MS (2 * MSEC_PER_SEC) const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT] = { /* mode 0 */ @@ -24,21 +25,33 @@ const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT] = { [6] = 192000, }; -int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate, - unsigned int *mode) +/* + * This operation has an effect to synchronize GLOBAL_STATUS/GLOBAL_SAMPLE_RATE + * to GLOBAL_STATUS. Especially, just after powering on, these are different. + */ +static int ensure_phase_lock(struct snd_dice *dice) { - int i; + __be32 reg; + int err; - for (i = 0; i < ARRAY_SIZE(snd_dice_rates); i++) { - if (!(dice->clock_caps & BIT(i))) - continue; - if (snd_dice_rates[i] != rate) - continue; + err = snd_dice_transaction_read_global(dice, GLOBAL_CLOCK_SELECT, + ®, sizeof(reg)); + if (err < 0) + return err; - *mode = (i - 1) / 2; - return 0; - } - return -EINVAL; + if (completion_done(&dice->clock_accepted)) + reinit_completion(&dice->clock_accepted); + + err = snd_dice_transaction_write_global(dice, GLOBAL_CLOCK_SELECT, + ®, sizeof(reg)); + if (err < 0) + return err; + + if (wait_for_completion_timeout(&dice->clock_accepted, + msecs_to_jiffies(NOTIFICATION_TIMEOUT_MS)) == 0) + return -ETIMEDOUT; + + return 0; } static void release_resources(struct snd_dice *dice, @@ -99,23 +112,27 @@ static int start_stream(struct snd_dice *dice, struct amdtp_stream *stream, unsigned int rate) { struct fw_iso_resources *resources; - unsigned int i, mode, pcm_chs, midi_ports; + __be32 reg[2]; + unsigned int i, pcm_chs, midi_ports; bool double_pcm_frames; int err; - err = snd_dice_stream_get_rate_mode(dice, rate, &mode); - if (err < 0) - goto end; if (stream == &dice->tx_stream) { resources = &dice->tx_resources; - pcm_chs = dice->tx_channels[mode]; - midi_ports = dice->tx_midi_ports[mode]; + err = snd_dice_transaction_read_tx(dice, TX_NUMBER_AUDIO, + reg, sizeof(reg)); } else { resources = &dice->rx_resources; - pcm_chs = dice->rx_channels[mode]; - midi_ports = dice->rx_midi_ports[mode]; + err = snd_dice_transaction_read_rx(dice, RX_NUMBER_AUDIO, + reg, sizeof(reg)); } + if (err < 0) + goto end; + + pcm_chs = be32_to_cpu(reg[0]); + midi_ports = be32_to_cpu(reg[1]); + /* * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in * one data block of AMDTP packet. Thus sampling transfer frequency is @@ -126,7 +143,7 @@ static int start_stream(struct snd_dice *dice, struct amdtp_stream *stream, * For this quirk, blocking mode is required and PCM buffer size should * be aligned to SYT_INTERVAL. */ - double_pcm_frames = mode > 1; + double_pcm_frames = rate > 96000; if (double_pcm_frames) { rate /= 2; pcm_chs *= 2; @@ -224,8 +241,10 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) } if (rate == 0) rate = curr_rate; - if (rate != curr_rate) - stop_stream(dice, master); + if (rate != curr_rate) { + err = -EINVAL; + goto end; + } if (!amdtp_stream_running(master)) { stop_stream(dice, slave); @@ -233,10 +252,10 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) amdtp_stream_set_sync(sync_mode, master, slave); - err = snd_dice_transaction_set_rate(dice, rate); + err = ensure_phase_lock(dice); if (err < 0) { dev_err(&dice->unit->device, - "fail to set sampling rate\n"); + "fail to ensure phase lock\n"); goto end; } diff --git a/sound/firewire/dice/dice-transaction.c b/sound/firewire/dice/dice-transaction.c index a4ff4e0bc0af..76f9f72df2a9 100644 --- a/sound/firewire/dice/dice-transaction.c +++ b/sound/firewire/dice/dice-transaction.c @@ -9,8 +9,6 @@ #include "dice.h" -#define NOTIFICATION_TIMEOUT_MS (2 * MSEC_PER_SEC) - static u64 get_subaddr(struct snd_dice *dice, enum snd_dice_addr_type type, u64 offset) { @@ -62,54 +60,6 @@ static unsigned int get_clock_info(struct snd_dice *dice, __be32 *info) info, 4); } -static int set_clock_info(struct snd_dice *dice, - unsigned int rate, unsigned int source) -{ - unsigned int i; - __be32 info; - u32 mask; - u32 clock; - int err; - - err = get_clock_info(dice, &info); - if (err < 0) - return err; - - clock = be32_to_cpu(info); - if (source != UINT_MAX) { - mask = CLOCK_SOURCE_MASK; - clock &= ~mask; - clock |= source; - } - if (rate != UINT_MAX) { - for (i = 0; i < ARRAY_SIZE(snd_dice_rates); i++) { - if (snd_dice_rates[i] == rate) - break; - } - if (i == ARRAY_SIZE(snd_dice_rates)) - return -EINVAL; - - mask = CLOCK_RATE_MASK; - clock &= ~mask; - clock |= i << CLOCK_RATE_SHIFT; - } - info = cpu_to_be32(clock); - - if (completion_done(&dice->clock_accepted)) - reinit_completion(&dice->clock_accepted); - - err = snd_dice_transaction_write_global(dice, GLOBAL_CLOCK_SELECT, - &info, 4); - if (err < 0) - return err; - - if (wait_for_completion_timeout(&dice->clock_accepted, - msecs_to_jiffies(NOTIFICATION_TIMEOUT_MS)) == 0) - return -ETIMEDOUT; - - return 0; -} - int snd_dice_transaction_get_clock_source(struct snd_dice *dice, unsigned int *source) { @@ -143,10 +93,6 @@ int snd_dice_transaction_get_rate(struct snd_dice *dice, unsigned int *rate) end: return err; } -int snd_dice_transaction_set_rate(struct snd_dice *dice, unsigned int rate) -{ - return set_clock_info(dice, rate, UINT_MAX); -} int snd_dice_transaction_set_enable(struct snd_dice *dice) { diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index b91b3739c810..f7303a650ac2 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -57,65 +57,10 @@ static int check_dice_category(struct fw_unit *unit) return 0; } -static int highest_supported_mode_rate(struct snd_dice *dice, - unsigned int mode, unsigned int *rate) -{ - unsigned int i, m; - - for (i = ARRAY_SIZE(snd_dice_rates); i > 0; i--) { - *rate = snd_dice_rates[i - 1]; - if (snd_dice_stream_get_rate_mode(dice, *rate, &m) < 0) - continue; - if (mode == m) - break; - } - if (i == 0) - return -EINVAL; - - return 0; -} - -static int dice_read_mode_params(struct snd_dice *dice, unsigned int mode) -{ - __be32 values[2]; - unsigned int rate; - int err; - - if (highest_supported_mode_rate(dice, mode, &rate) < 0) { - dice->tx_channels[mode] = 0; - dice->tx_midi_ports[mode] = 0; - dice->rx_channels[mode] = 0; - dice->rx_midi_ports[mode] = 0; - return 0; - } - - err = snd_dice_transaction_set_rate(dice, rate); - if (err < 0) - return err; - - err = snd_dice_transaction_read_tx(dice, TX_NUMBER_AUDIO, - values, sizeof(values)); - if (err < 0) - return err; - - dice->tx_channels[mode] = be32_to_cpu(values[0]); - dice->tx_midi_ports[mode] = be32_to_cpu(values[1]); - - err = snd_dice_transaction_read_rx(dice, RX_NUMBER_AUDIO, - values, sizeof(values)); - if (err < 0) - return err; - - dice->rx_channels[mode] = be32_to_cpu(values[0]); - dice->rx_midi_ports[mode] = be32_to_cpu(values[1]); - - return 0; -} - -static int dice_read_params(struct snd_dice *dice) +static int check_clock_caps(struct snd_dice *dice) { __be32 value; - int mode, err; + int err; /* some very old firmwares don't tell about their clock support */ if (dice->clock_caps > 0) { @@ -133,12 +78,6 @@ static int dice_read_params(struct snd_dice *dice) CLOCK_CAP_SOURCE_INTERNAL; } - for (mode = 2; mode >= 0; --mode) { - err = dice_read_mode_params(dice, mode); - if (err < 0) - return err; - } - return 0; } @@ -215,7 +154,7 @@ static void do_registration(struct work_struct *work) if (err < 0) goto error; - err = dice_read_params(dice); + err = check_clock_caps(dice); if (err < 0) goto error; diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index 3d5ebebe61ea..423cdba99726 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -56,10 +56,6 @@ struct snd_dice { unsigned int rsrv_offset; unsigned int clock_caps; - unsigned int tx_channels[3]; - unsigned int rx_channels[3]; - unsigned int tx_midi_ports[3]; - unsigned int rx_midi_ports[3]; struct fw_address_handler notification_handler; int owner_generation; @@ -158,7 +154,6 @@ static inline int snd_dice_transaction_read_sync(struct snd_dice *dice, int snd_dice_transaction_get_clock_source(struct snd_dice *dice, unsigned int *source); -int snd_dice_transaction_set_rate(struct snd_dice *dice, unsigned int rate); int snd_dice_transaction_get_rate(struct snd_dice *dice, unsigned int *rate); int snd_dice_transaction_set_enable(struct snd_dice *dice); void snd_dice_transaction_clear_enable(struct snd_dice *dice); @@ -169,9 +164,6 @@ void snd_dice_transaction_destroy(struct snd_dice *dice); #define SND_DICE_RATES_COUNT 7 extern const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT]; -int snd_dice_stream_get_rate_mode(struct snd_dice *dice, - unsigned int rate, unsigned int *mode); - int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate); void snd_dice_stream_stop_duplex(struct snd_dice *dice); int snd_dice_stream_init_duplex(struct snd_dice *dice); diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 2153d31fb663..4a4705031cb9 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -23,17 +23,5 @@ config SND_SGI_HAL2 help Sound support for the SGI Indy and Indigo2 Workstation. - -config SND_AU1X00 - tristate "Au1x00 AC97 Port Driver (DEPRECATED)" - depends on MIPS_ALCHEMY - select SND_PCM - select SND_AC97_CODEC - help - ALSA Sound driver for the Au1x00's AC97 port. - - Newer drivers for ASoC are available, please do not use - this driver as it will be removed in the future. - endif # SND_MIPS diff --git a/sound/mips/Makefile b/sound/mips/Makefile index 861ec0a574b4..b977c44330d6 100644 --- a/sound/mips/Makefile +++ b/sound/mips/Makefile @@ -2,11 +2,9 @@ # Makefile for ALSA # -snd-au1x00-objs := au1x00.o snd-sgi-o2-objs := sgio2audio.o ad1843.o snd-sgi-hal2-objs := hal2.o # Toplevel Module Dependency -obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o obj-$(CONFIG_SND_SGI_HAL2) += snd-sgi-hal2.o diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c deleted file mode 100644 index 1e30e8475431..000000000000 --- a/sound/mips/au1x00.c +++ /dev/null @@ -1,734 +0,0 @@ -/* - * BRIEF MODULE DESCRIPTION - * Driver for AMD Au1000 MIPS Processor, AC'97 Sound Port - * - * Copyright 2004 Cooper Street Innovations Inc. - * Author: Charles Eidsness <charles@cooper-street.com> - * - * This program 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; either version 2 of the License, or (at your - * option) any later version. - * - * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN - * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, - * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT - * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF - * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON - * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT - * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF - * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - * - * 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., - * 675 Mass Ave, Cambridge, MA 02139, USA. - * - * History: - * - * 2004-09-09 Charles Eidsness -- Original verion -- based on - * sa11xx-uda1341.c ALSA driver and the - * au1000.c OSS driver. - * 2004-09-09 Matt Porter -- Added support for ALSA 1.0.6 - * - */ - -#include <linux/ioport.h> -#include <linux/interrupt.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/module.h> -#include <sound/core.h> -#include <sound/initval.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/ac97_codec.h> -#include <asm/mach-au1x00/au1000.h> -#include <asm/mach-au1x00/au1000_dma.h> - -MODULE_AUTHOR("Charles Eidsness <charles@cooper-street.com>"); -MODULE_DESCRIPTION("Au1000 AC'97 ALSA Driver"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{AMD,Au1000 AC'97}}"); - -#define PLAYBACK 0 -#define CAPTURE 1 -#define AC97_SLOT_3 0x01 -#define AC97_SLOT_4 0x02 -#define AC97_SLOT_6 0x08 -#define AC97_CMD_IRQ 31 -#define READ 0 -#define WRITE 1 -#define READ_WAIT 2 -#define RW_DONE 3 - -struct au1000_period -{ - u32 start; - u32 relative_end; /*realtive to start of buffer*/ - struct au1000_period * next; -}; - -/*Au1000 AC97 Port Control Reisters*/ -struct au1000_ac97_reg { - u32 volatile config; - u32 volatile status; - u32 volatile data; - u32 volatile cmd; - u32 volatile cntrl; -}; - -struct audio_stream { - struct snd_pcm_substream *substream; - int dma; - spinlock_t dma_lock; - struct au1000_period * buffer; - unsigned int period_size; - unsigned int periods; -}; - -struct snd_au1000 { - struct snd_card *card; - struct au1000_ac97_reg volatile *ac97_ioport; - - struct resource *ac97_res_port; - spinlock_t ac97_lock; - struct snd_ac97 *ac97; - - struct snd_pcm *pcm; - struct audio_stream *stream[2]; /* playback & capture */ - int dmaid[2]; /* tx(0)/rx(1) DMA ids */ -}; - -/*--------------------------- Local Functions --------------------------------*/ -static void -au1000_set_ac97_xmit_slots(struct snd_au1000 *au1000, long xmit_slots) -{ - u32 volatile ac97_config; - - spin_lock(&au1000->ac97_lock); - ac97_config = au1000->ac97_ioport->config; - ac97_config = ac97_config & ~AC97C_XMIT_SLOTS_MASK; - ac97_config |= (xmit_slots << AC97C_XMIT_SLOTS_BIT); - au1000->ac97_ioport->config = ac97_config; - spin_unlock(&au1000->ac97_lock); -} - -static void -au1000_set_ac97_recv_slots(struct snd_au1000 *au1000, long recv_slots) -{ - u32 volatile ac97_config; - - spin_lock(&au1000->ac97_lock); - ac97_config = au1000->ac97_ioport->config; - ac97_config = ac97_config & ~AC97C_RECV_SLOTS_MASK; - ac97_config |= (recv_slots << AC97C_RECV_SLOTS_BIT); - au1000->ac97_ioport->config = ac97_config; - spin_unlock(&au1000->ac97_lock); -} - - -static void -au1000_release_dma_link(struct audio_stream *stream) -{ - struct au1000_period * pointer; - struct au1000_period * pointer_next; - - stream->period_size = 0; - stream->periods = 0; - pointer = stream->buffer; - if (! pointer) - return; - do { - pointer_next = pointer->next; - kfree(pointer); - pointer = pointer_next; - } while (pointer != stream->buffer); - stream->buffer = NULL; -} - -static int -au1000_setup_dma_link(struct audio_stream *stream, unsigned int period_bytes, - unsigned int periods) -{ - struct snd_pcm_substream *substream = stream->substream; - struct snd_pcm_runtime *runtime = substream->runtime; - struct au1000_period *pointer; - unsigned long dma_start; - int i; - - dma_start = virt_to_phys(runtime->dma_area); - - if (stream->period_size == period_bytes && - stream->periods == periods) - return 0; /* not changed */ - - au1000_release_dma_link(stream); - - stream->period_size = period_bytes; - stream->periods = periods; - - stream->buffer = kmalloc(sizeof(struct au1000_period), GFP_KERNEL); - if (! stream->buffer) - return -ENOMEM; - pointer = stream->buffer; - for (i = 0; i < periods; i++) { - pointer->start = (u32)(dma_start + (i * period_bytes)); - pointer->relative_end = (u32) (((i+1) * period_bytes) - 0x1); - if (i < periods - 1) { - pointer->next = kmalloc(sizeof(struct au1000_period), GFP_KERNEL); - if (! pointer->next) { - au1000_release_dma_link(stream); - return -ENOMEM; - } - pointer = pointer->next; - } - } - pointer->next = stream->buffer; - return 0; -} - -static void -au1000_dma_stop(struct audio_stream *stream) -{ - if (snd_BUG_ON(!stream->buffer)) - return; - disable_dma(stream->dma); -} - -static void -au1000_dma_start(struct audio_stream *stream) -{ - if (snd_BUG_ON(!stream->buffer)) - return; - - init_dma(stream->dma); - if (get_dma_active_buffer(stream->dma) == 0) { - clear_dma_done0(stream->dma); - set_dma_addr0(stream->dma, stream->buffer->start); - set_dma_count0(stream->dma, stream->period_size >> 1); - set_dma_addr1(stream->dma, stream->buffer->next->start); - set_dma_count1(stream->dma, stream->period_size >> 1); - } else { - clear_dma_done1(stream->dma); - set_dma_addr1(stream->dma, stream->buffer->start); - set_dma_count1(stream->dma, stream->period_size >> 1); - set_dma_addr0(stream->dma, stream->buffer->next->start); - set_dma_count0(stream->dma, stream->period_size >> 1); - } - enable_dma_buffers(stream->dma); - start_dma(stream->dma); -} - -static irqreturn_t -au1000_dma_interrupt(int irq, void *dev_id) -{ - struct audio_stream *stream = (struct audio_stream *) dev_id; - struct snd_pcm_substream *substream = stream->substream; - - spin_lock(&stream->dma_lock); - switch (get_dma_buffer_done(stream->dma)) { - case DMA_D0: - stream->buffer = stream->buffer->next; - clear_dma_done0(stream->dma); - set_dma_addr0(stream->dma, stream->buffer->next->start); - set_dma_count0(stream->dma, stream->period_size >> 1); - enable_dma_buffer0(stream->dma); - break; - case DMA_D1: - stream->buffer = stream->buffer->next; - clear_dma_done1(stream->dma); - set_dma_addr1(stream->dma, stream->buffer->next->start); - set_dma_count1(stream->dma, stream->period_size >> 1); - enable_dma_buffer1(stream->dma); - break; - case (DMA_D0 | DMA_D1): - printk(KERN_ERR "DMA %d missed interrupt.\n",stream->dma); - au1000_dma_stop(stream); - au1000_dma_start(stream); - break; - case (~DMA_D0 & ~DMA_D1): - printk(KERN_ERR "DMA %d empty irq.\n",stream->dma); - } - spin_unlock(&stream->dma_lock); - snd_pcm_period_elapsed(substream); - return IRQ_HANDLED; -} - -/*-------------------------- PCM Audio Streams -------------------------------*/ - -static unsigned int rates[] = {8000, 11025, 16000, 22050}; -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static struct snd_pcm_hardware snd_au1000_hw = -{ - .info = (SNDRV_PCM_INFO_INTERLEAVED | \ - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050), - .rate_min = 8000, - .rate_max = 22050, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = 128*1024, - .period_bytes_min = 32, - .period_bytes_max = 16*1024, - .periods_min = 8, - .periods_max = 255, - .fifo_size = 16, -}; - -static int -snd_au1000_playback_open(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - - au1000->stream[PLAYBACK]->substream = substream; - au1000->stream[PLAYBACK]->buffer = NULL; - substream->private_data = au1000->stream[PLAYBACK]; - substream->runtime->hw = snd_au1000_hw; - return (snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates) < 0); -} - -static int -snd_au1000_capture_open(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - - au1000->stream[CAPTURE]->substream = substream; - au1000->stream[CAPTURE]->buffer = NULL; - substream->private_data = au1000->stream[CAPTURE]; - substream->runtime->hw = snd_au1000_hw; - return (snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates) < 0); -} - -static int -snd_au1000_playback_close(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - - au1000->stream[PLAYBACK]->substream = NULL; - return 0; -} - -static int -snd_au1000_capture_close(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - - au1000->stream[CAPTURE]->substream = NULL; - return 0; -} - -static int -snd_au1000_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct audio_stream *stream = substream->private_data; - int err; - - err = snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); - if (err < 0) - return err; - return au1000_setup_dma_link(stream, - params_period_bytes(hw_params), - params_periods(hw_params)); -} - -static int -snd_au1000_hw_free(struct snd_pcm_substream *substream) -{ - struct audio_stream *stream = substream->private_data; - au1000_release_dma_link(stream); - return snd_pcm_lib_free_pages(substream); -} - -static int -snd_au1000_playback_prepare(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - - if (runtime->channels == 1) - au1000_set_ac97_xmit_slots(au1000, AC97_SLOT_4); - else - au1000_set_ac97_xmit_slots(au1000, AC97_SLOT_3 | AC97_SLOT_4); - snd_ac97_set_rate(au1000->ac97, AC97_PCM_FRONT_DAC_RATE, runtime->rate); - return 0; -} - -static int -snd_au1000_capture_prepare(struct snd_pcm_substream *substream) -{ - struct snd_au1000 *au1000 = substream->pcm->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - - if (runtime->channels == 1) - au1000_set_ac97_recv_slots(au1000, AC97_SLOT_4); - else - au1000_set_ac97_recv_slots(au1000, AC97_SLOT_3 | AC97_SLOT_4); - snd_ac97_set_rate(au1000->ac97, AC97_PCM_LR_ADC_RATE, runtime->rate); - return 0; -} - -static int -snd_au1000_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct audio_stream *stream = substream->private_data; - int err = 0; - - spin_lock(&stream->dma_lock); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - au1000_dma_start(stream); - break; - case SNDRV_PCM_TRIGGER_STOP: - au1000_dma_stop(stream); - break; - default: - err = -EINVAL; - break; - } - spin_unlock(&stream->dma_lock); - return err; -} - -static snd_pcm_uframes_t -snd_au1000_pointer(struct snd_pcm_substream *substream) -{ - struct audio_stream *stream = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - long location; - - spin_lock(&stream->dma_lock); - location = get_dma_residue(stream->dma); - spin_unlock(&stream->dma_lock); - location = stream->buffer->relative_end - location; - if (location == -1) - location = 0; - return bytes_to_frames(runtime,location); -} - -static struct snd_pcm_ops snd_card_au1000_playback_ops = { - .open = snd_au1000_playback_open, - .close = snd_au1000_playback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_au1000_hw_params, - .hw_free = snd_au1000_hw_free, - .prepare = snd_au1000_playback_prepare, - .trigger = snd_au1000_trigger, - .pointer = snd_au1000_pointer, -}; - -static struct snd_pcm_ops snd_card_au1000_capture_ops = { - .open = snd_au1000_capture_open, - .close = snd_au1000_capture_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_au1000_hw_params, - .hw_free = snd_au1000_hw_free, - .prepare = snd_au1000_capture_prepare, - .trigger = snd_au1000_trigger, - .pointer = snd_au1000_pointer, -}; - -static int -snd_au1000_pcm_new(struct snd_au1000 *au1000) -{ - struct snd_pcm *pcm; - int err; - unsigned long flags; - - if ((err = snd_pcm_new(au1000->card, "AU1000 AC97 PCM", 0, 1, 1, &pcm)) < 0) - return err; - - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), 128*1024, 128*1024); - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, - &snd_card_au1000_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &snd_card_au1000_capture_ops); - - pcm->private_data = au1000; - pcm->info_flags = 0; - strcpy(pcm->name, "Au1000 AC97 PCM"); - - spin_lock_init(&au1000->stream[PLAYBACK]->dma_lock); - spin_lock_init(&au1000->stream[CAPTURE]->dma_lock); - - flags = claim_dma_lock(); - au1000->stream[PLAYBACK]->dma = request_au1000_dma(au1000->dmaid[0], - "AC97 TX", au1000_dma_interrupt, 0, - au1000->stream[PLAYBACK]); - if (au1000->stream[PLAYBACK]->dma < 0) { - release_dma_lock(flags); - return -EBUSY; - } - au1000->stream[CAPTURE]->dma = request_au1000_dma(au1000->dmaid[1], - "AC97 RX", au1000_dma_interrupt, 0, - au1000->stream[CAPTURE]); - if (au1000->stream[CAPTURE]->dma < 0){ - release_dma_lock(flags); - return -EBUSY; - } - /* enable DMA coherency in read/write DMA channels */ - set_dma_mode(au1000->stream[PLAYBACK]->dma, - get_dma_mode(au1000->stream[PLAYBACK]->dma) & ~DMA_NC); - set_dma_mode(au1000->stream[CAPTURE]->dma, - get_dma_mode(au1000->stream[CAPTURE]->dma) & ~DMA_NC); - release_dma_lock(flags); - au1000->pcm = pcm; - return 0; -} - - -/*-------------------------- AC97 CODEC Control ------------------------------*/ - -static unsigned short -snd_au1000_ac97_read(struct snd_ac97 *ac97, unsigned short reg) -{ - struct snd_au1000 *au1000 = ac97->private_data; - u32 volatile cmd; - u16 volatile data; - int i; - - spin_lock(&au1000->ac97_lock); -/* would rather use the interrupt than this polling but it works and I can't -get the interrupt driven case to work efficiently */ - for (i = 0; i < 0x5000; i++) - if (!(au1000->ac97_ioport->status & AC97C_CP)) - break; - if (i == 0x5000) - printk(KERN_ERR "au1000 AC97: AC97 command read timeout\n"); - - cmd = (u32) reg & AC97C_INDEX_MASK; - cmd |= AC97C_READ; - au1000->ac97_ioport->cmd = cmd; - - /* now wait for the data */ - for (i = 0; i < 0x5000; i++) - if (!(au1000->ac97_ioport->status & AC97C_CP)) - break; - if (i == 0x5000) { - printk(KERN_ERR "au1000 AC97: AC97 command read timeout\n"); - spin_unlock(&au1000->ac97_lock); - return 0; - } - - data = au1000->ac97_ioport->cmd & 0xffff; - spin_unlock(&au1000->ac97_lock); - - return data; - -} - - -static void -snd_au1000_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) -{ - struct snd_au1000 *au1000 = ac97->private_data; - u32 cmd; - int i; - - spin_lock(&au1000->ac97_lock); -/* would rather use the interrupt than this polling but it works and I can't -get the interrupt driven case to work efficiently */ - for (i = 0; i < 0x5000; i++) - if (!(au1000->ac97_ioport->status & AC97C_CP)) - break; - if (i == 0x5000) - printk(KERN_ERR "au1000 AC97: AC97 command write timeout\n"); - - cmd = (u32) reg & AC97C_INDEX_MASK; - cmd &= ~AC97C_READ; - cmd |= ((u32) val << AC97C_WD_BIT); - au1000->ac97_ioport->cmd = cmd; - spin_unlock(&au1000->ac97_lock); -} - -/*------------------------------ Setup / Destroy ----------------------------*/ - -static void snd_au1000_free(struct snd_card *card) -{ - struct snd_au1000 *au1000 = card->private_data; - - if (au1000->stream[PLAYBACK]) { - if (au1000->stream[PLAYBACK]->dma >= 0) - free_au1000_dma(au1000->stream[PLAYBACK]->dma); - kfree(au1000->stream[PLAYBACK]); - } - - if (au1000->stream[CAPTURE]) { - if (au1000->stream[CAPTURE]->dma >= 0) - free_au1000_dma(au1000->stream[CAPTURE]->dma); - kfree(au1000->stream[CAPTURE]); - } - - if (au1000->ac97_res_port) { - /* put internal AC97 block into reset */ - if (au1000->ac97_ioport) { - au1000->ac97_ioport->cntrl = AC97C_RS; - iounmap(au1000->ac97_ioport); - au1000->ac97_ioport = NULL; - } - release_and_free_resource(au1000->ac97_res_port); - au1000->ac97_res_port = NULL; - } -} - -static struct snd_ac97_bus_ops ops = { - .write = snd_au1000_ac97_write, - .read = snd_au1000_ac97_read, -}; - -static int au1000_ac97_probe(struct platform_device *pdev) -{ - int err; - void __iomem *io; - struct resource *r; - struct snd_card *card; - struct snd_au1000 *au1000; - struct snd_ac97_bus *pbus; - struct snd_ac97_template ac97; - - err = snd_card_new(&pdev->dev, -1, "AC97", THIS_MODULE, - sizeof(struct snd_au1000), &card); - if (err < 0) - return err; - - au1000 = card->private_data; - au1000->card = card; - spin_lock_init(&au1000->ac97_lock); - - /* from here on let ALSA call the special freeing function */ - card->private_free = snd_au1000_free; - - /* TX DMA ID */ - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) { - err = -ENODEV; - snd_printk(KERN_INFO "no TX DMA platform resource!\n"); - goto out; - } - au1000->dmaid[0] = r->start; - - /* RX DMA ID */ - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) { - err = -ENODEV; - snd_printk(KERN_INFO "no RX DMA platform resource!\n"); - goto out; - } - au1000->dmaid[1] = r->start; - - au1000->stream[PLAYBACK] = kmalloc(sizeof(struct audio_stream), - GFP_KERNEL); - if (!au1000->stream[PLAYBACK]) { - err = -ENOMEM; - goto out; - } - au1000->stream[PLAYBACK]->dma = -1; - - au1000->stream[CAPTURE] = kmalloc(sizeof(struct audio_stream), - GFP_KERNEL); - if (!au1000->stream[CAPTURE]) { - err = -ENOMEM; - goto out; - } - au1000->stream[CAPTURE]->dma = -1; - - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { - err = -ENODEV; - goto out; - } - - err = -EBUSY; - au1000->ac97_res_port = request_mem_region(r->start, resource_size(r), - pdev->name); - if (!au1000->ac97_res_port) { - snd_printk(KERN_ERR "ALSA AC97: can't grab AC97 port\n"); - goto out; - } - - io = ioremap(r->start, resource_size(r)); - if (!io) - goto out; - - au1000->ac97_ioport = (struct au1000_ac97_reg *)io; - - /* configure pins for AC'97 - TODO: move to board_setup.c */ - au_writel(au_readl(SYS_PINFUNC) & ~0x02, SYS_PINFUNC); - - /* Initialise Au1000's AC'97 Control Block */ - au1000->ac97_ioport->cntrl = AC97C_RS | AC97C_CE; - udelay(10); - au1000->ac97_ioport->cntrl = AC97C_CE; - udelay(10); - - /* Initialise External CODEC -- cold reset */ - au1000->ac97_ioport->config = AC97C_RESET; - udelay(10); - au1000->ac97_ioport->config = 0x0; - mdelay(5); - - /* Initialise AC97 middle-layer */ - err = snd_ac97_bus(au1000->card, 0, &ops, au1000, &pbus); - if (err < 0) - goto out; - - memset(&ac97, 0, sizeof(ac97)); - ac97.private_data = au1000; - err = snd_ac97_mixer(pbus, &ac97, &au1000->ac97); - if (err < 0) - goto out; - - err = snd_au1000_pcm_new(au1000); - if (err < 0) - goto out; - - strcpy(card->driver, "Au1000-AC97"); - strcpy(card->shortname, "AMD Au1000-AC97"); - sprintf(card->longname, "AMD Au1000--AC97 ALSA Driver"); - - err = snd_card_register(card); - if (err < 0) - goto out; - - printk(KERN_INFO "ALSA AC97: Driver Initialized\n"); - - platform_set_drvdata(pdev, card); - - return 0; - - out: - snd_card_free(card); - return err; -} - -static int au1000_ac97_remove(struct platform_device *pdev) -{ - return snd_card_free(platform_get_drvdata(pdev)); -} - -struct platform_driver au1000_ac97c_driver = { - .driver = { - .name = "au1000-ac97c", - .owner = THIS_MODULE, - }, - .probe = au1000_ac97_probe, - .remove = au1000_ac97_remove, -}; - -module_platform_driver(au1000_ac97c_driver); diff --git a/sound/usb/card.c b/sound/usb/card.c index 1f09d9591276..3fc63583a537 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -82,6 +82,7 @@ static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; static int device_setup[SNDRV_CARDS]; /* device parameter for this card */ static bool ignore_ctl_error; static bool autoclock = true; +static char *quirk_alias[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the USB audio adapter."); @@ -100,6 +101,8 @@ MODULE_PARM_DESC(ignore_ctl_error, "Ignore errors from USB controller for mixer interfaces."); module_param(autoclock, bool, 0444); MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes)."); +module_param_array(quirk_alias, charp, NULL, 0444); +MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef."); /* * we keep the snd_usb_audio_t instances by ourselves for merging @@ -171,8 +174,9 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && altsd->bInterfaceSubClass == USB_SUBCLASS_MIDISTREAMING) { - int err = snd_usbmidi_create(chip->card, iface, - &chip->midi_list, NULL); + int err = __snd_usbmidi_create(chip->card, iface, + &chip->midi_list, NULL, + chip->usb_id); if (err < 0) { dev_err(&dev->dev, "%u:%d: cannot create sequencer device\n", @@ -311,6 +315,7 @@ static int snd_usb_audio_free(struct snd_usb_audio *chip) snd_usb_endpoint_free(ep); mutex_destroy(&chip->mutex); + dev_set_drvdata(&chip->dev->dev, NULL); kfree(chip); return 0; } @@ -455,6 +460,48 @@ static int snd_usb_audio_create(struct usb_interface *intf, return 0; } +/* look for a matching quirk alias id */ +static bool get_alias_id(struct usb_device *dev, unsigned int *id) +{ + int i; + unsigned int src, dst; + + for (i = 0; i < ARRAY_SIZE(quirk_alias); i++) { + if (!quirk_alias[i] || + sscanf(quirk_alias[i], "%x:%x", &src, &dst) != 2 || + src != *id) + continue; + dev_info(&dev->dev, + "device (%04x:%04x): applying quirk alias %04x:%04x\n", + USB_ID_VENDOR(*id), USB_ID_PRODUCT(*id), + USB_ID_VENDOR(dst), USB_ID_PRODUCT(dst)); + *id = dst; + return true; + } + + return false; +} + +static struct usb_device_id usb_audio_ids[]; /* defined below */ + +/* look for the corresponding quirk */ +static const struct snd_usb_audio_quirk * +get_alias_quirk(struct usb_device *dev, unsigned int id) +{ + const struct usb_device_id *p; + + for (p = usb_audio_ids; p->match_flags; p++) { + /* FIXME: this checks only vendor:product pair in the list */ + if ((p->match_flags & USB_DEVICE_ID_MATCH_DEVICE) == + USB_DEVICE_ID_MATCH_DEVICE && + p->idVendor == USB_ID_VENDOR(id) && + p->idProduct == USB_ID_PRODUCT(id)) + return (const struct snd_usb_audio_quirk *)p->driver_info; + } + + return NULL; +} + /* * probe the active usb device * @@ -481,10 +528,12 @@ static int usb_audio_probe(struct usb_interface *intf, ifnum = get_iface_desc(alts)->bInterfaceNumber; id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), le16_to_cpu(dev->descriptor.idProduct)); + if (get_alias_id(dev, &id)) + quirk = get_alias_quirk(dev, id); if (quirk && quirk->ifnum >= 0 && ifnum != quirk->ifnum) return -ENXIO; - err = snd_usb_apply_boot_quirk(dev, intf, quirk); + err = snd_usb_apply_boot_quirk(dev, intf, quirk, id); if (err < 0) return err; @@ -503,6 +552,7 @@ static int usb_audio_probe(struct usb_interface *intf, goto __error; } chip = usb_chip[i]; + dev_set_drvdata(&dev->dev, chip); atomic_inc(&chip->active); /* avoid autopm */ break; } diff --git a/sound/usb/midi.c b/sound/usb/midi.c index cc39f63299ef..b79875ebec1e 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -2320,10 +2320,11 @@ EXPORT_SYMBOL(snd_usbmidi_resume); /* * Creates and registers everything needed for a MIDI streaming interface. */ -int snd_usbmidi_create(struct snd_card *card, - struct usb_interface *iface, - struct list_head *midi_list, - const struct snd_usb_audio_quirk *quirk) +int __snd_usbmidi_create(struct snd_card *card, + struct usb_interface *iface, + struct list_head *midi_list, + const struct snd_usb_audio_quirk *quirk, + unsigned int usb_id) { struct snd_usb_midi *umidi; struct snd_usb_midi_endpoint_info endpoints[MIDI_MAX_ENDPOINTS]; @@ -2341,8 +2342,10 @@ int snd_usbmidi_create(struct snd_card *card, spin_lock_init(&umidi->disc_lock); init_rwsem(&umidi->disc_rwsem); mutex_init(&umidi->mutex); - umidi->usb_id = USB_ID(le16_to_cpu(umidi->dev->descriptor.idVendor), + if (!usb_id) + usb_id = USB_ID(le16_to_cpu(umidi->dev->descriptor.idVendor), le16_to_cpu(umidi->dev->descriptor.idProduct)); + umidi->usb_id = usb_id; setup_timer(&umidi->error_timer, snd_usbmidi_error_timer, (unsigned long)umidi); @@ -2464,4 +2467,4 @@ int snd_usbmidi_create(struct snd_card *card, list_add_tail(&umidi->list, midi_list); return 0; } -EXPORT_SYMBOL(snd_usbmidi_create); +EXPORT_SYMBOL(__snd_usbmidi_create); diff --git a/sound/usb/midi.h b/sound/usb/midi.h index ad8a3211f8e7..5e25a3fd6c1d 100644 --- a/sound/usb/midi.h +++ b/sound/usb/midi.h @@ -39,10 +39,20 @@ struct snd_usb_midi_endpoint_info { /* for QUIRK_MIDI_AKAI, data is NULL */ -int snd_usbmidi_create(struct snd_card *card, +int __snd_usbmidi_create(struct snd_card *card, + struct usb_interface *iface, + struct list_head *midi_list, + const struct snd_usb_audio_quirk *quirk, + unsigned int usb_id); + +static inline int snd_usbmidi_create(struct snd_card *card, struct usb_interface *iface, struct list_head *midi_list, - const struct snd_usb_audio_quirk *quirk); + const struct snd_usb_audio_quirk *quirk) +{ + return __snd_usbmidi_create(card, iface, midi_list, quirk, 0); +} + void snd_usbmidi_input_stop(struct list_head *p); void snd_usbmidi_input_start(struct list_head *p); void snd_usbmidi_disconnect(struct list_head *p); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4f6ce1cac8e2..2585c175b54e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -446,8 +446,9 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, const struct snd_usb_audio_quirk *quirk = chip->usb_id == USB_ID(0x0582, 0x002b) ? &ua700_quirk : &uaxx_quirk; - return snd_usbmidi_create(chip->card, iface, - &chip->midi_list, quirk); + return __snd_usbmidi_create(chip->card, iface, + &chip->midi_list, quirk, + chip->usb_id); } if (altsd->bNumEndpoints != 1) @@ -974,11 +975,9 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, int snd_usb_apply_boot_quirk(struct usb_device *dev, struct usb_interface *intf, - const struct snd_usb_audio_quirk *quirk) + const struct snd_usb_audio_quirk *quirk, + unsigned int id) { - u32 id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), - le16_to_cpu(dev->descriptor.idProduct)); - switch (id) { case USB_ID(0x041e, 0x3000): /* SB Extigy needs special boot-up sequence */ @@ -1183,7 +1182,7 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) * "Playback Design" products send bogus feedback data at the start * of the stream. Ignore them. */ - if ((le16_to_cpu(ep->chip->dev->descriptor.idVendor) == 0x23ba) && + if (USB_ID_VENDOR(ep->chip->usb_id) == 0x23ba && ep->type == SND_USB_ENDPOINT_TYPE_SYNC) ep->skip_packets = 4; @@ -1202,11 +1201,15 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) void snd_usb_set_interface_quirk(struct usb_device *dev) { + struct snd_usb_audio *chip = dev_get_drvdata(&dev->dev); + + if (!chip) + return; /* * "Playback Design" products need a 50ms delay after setting the * USB interface. */ - switch (le16_to_cpu(dev->descriptor.idVendor)) { + switch (USB_ID_VENDOR(chip->usb_id)) { case 0x23ba: /* Playback Design */ case 0x0644: /* TEAC Corp. */ mdelay(50); @@ -1214,15 +1217,20 @@ void snd_usb_set_interface_quirk(struct usb_device *dev) } } +/* quirk applied after snd_usb_ctl_msg(); not applied during boot quirks */ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size) { + struct snd_usb_audio *chip = dev_get_drvdata(&dev->dev); + + if (!chip) + return; /* * "Playback Design" products need a 20ms delay after each * class compliant request */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) && + if (USB_ID_VENDOR(chip->usb_id) == 0x23ba && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); @@ -1230,23 +1238,21 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, * "TEAC Corp." products need a 20ms delay after each * class compliant request */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x0644) && + if (USB_ID_VENDOR(chip->usb_id) == 0x0644 && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); /* Marantz/Denon devices with USB DAC functionality need a delay * after each class compliant request */ - if (is_marantz_denon_dac(USB_ID(le16_to_cpu(dev->descriptor.idVendor), - le16_to_cpu(dev->descriptor.idProduct))) + if (is_marantz_denon_dac(chip->usb_id) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); /* Zoom R16/24 needs a tiny delay here, otherwise requests like * get/set frequency return as failed despite actually succeeding. */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x1686) && - (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) && + if (chip->usb_id == USB_ID(0x1686, 0x00dd) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(1); } @@ -1263,7 +1269,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, unsigned int sample_bytes) { /* Playback Designs */ - if (le16_to_cpu(chip->dev->descriptor.idVendor) == 0x23ba) { + if (USB_ID_VENDOR(chip->usb_id) == 0x23ba) { switch (fp->altsetting) { case 1: fp->dsd_dop = true; diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index 2cd71ed1201f..192ff5ce9452 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -16,7 +16,8 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, int snd_usb_apply_boot_quirk(struct usb_device *dev, struct usb_interface *intf, - const struct snd_usb_audio_quirk *quirk); + const struct snd_usb_audio_quirk *quirk, + unsigned int usb_id); void snd_usb_set_format_quirk(struct snd_usb_substream *subs, struct audioformat *fmt); |