ALSA: oxfw: Change the way to make PCM rules/constraints (5cd1d3f4) · Commits · 戴 / test

sound/firewire/oxfw/oxfw-pcm.c

+116 −81

Original line number	Diff line number	Diff line
		@@ -7,117 +7,152 @@

		#include "oxfw.h"

		static int firewave_rate_constraint(struct snd_pcm_hw_params *params,
		static int hw_rule_rate(struct snd_pcm_hw_params *params,
		struct snd_pcm_hw_rule *rule)
		{
		static unsigned int stereo_rates[] = { 48000, 96000 };
		struct snd_interval *channels =
		hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
		struct snd_interval *rate =
		u8 **formats = rule->private;
		struct snd_interval *r =
		hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
		const struct snd_interval *c =
		hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS);
		struct snd_interval t = {
		.min = UINT_MAX, .max = 0, .integer = 1
		};
		struct snd_oxfw_stream_formation formation;
		unsigned int i, err;

		for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
		if (formats[i] == NULL)
		continue;

		err = snd_oxfw_stream_parse_format(formats[i], &formation);
		if (err < 0)
		continue;
		if (!snd_interval_test(c, formation.pcm))
		continue;

		t.min = min(t.min, formation.rate);
		t.max = max(t.max, formation.rate);

		/* two channels work only at 48/96 kHz */
		if (snd_interval_max(channels) < 6)
		return snd_interval_list(rate, 2, stereo_rates, 0);
		return 0;
		}
		return snd_interval_refine(r, &t);
		}

		static int firewave_channels_constraint(struct snd_pcm_hw_params *params,
		static int hw_rule_channels(struct snd_pcm_hw_params *params,
		struct snd_pcm_hw_rule *rule)
		{
		static const struct snd_interval all_channels = { .min = 6, .max = 6 };
		struct snd_interval *rate =
		hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
		struct snd_interval *channels =
		u8 **formats = rule->private;
		struct snd_interval *c =
		hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
		const struct snd_interval *r =
		hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE);
		struct snd_oxfw_stream_formation formation;
		unsigned int i, j, err;
		unsigned int count, list[SND_OXFW_STREAM_FORMAT_ENTRIES] = {0};

		count = 0;
		for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
		if (formats[i] == NULL)
		break;

		/* 32/44.1 kHz work only with all six channels */
		if (snd_interval_max(rate) < 48000)
		return snd_interval_refine(channels, &all_channels);
		return 0;
		err = snd_oxfw_stream_parse_format(formats[i], &formation);
		if (err < 0)
		continue;
		if (!snd_interval_test(r, formation.rate))
		continue;
		if (list[count] == formation.pcm)
		continue;

		for (j = 0; j < ARRAY_SIZE(list); j++) {
		if (list[j] == formation.pcm)
		break;
		}
		if (j == ARRAY_SIZE(list)) {
		list[count] = formation.pcm;
		if (++count == ARRAY_SIZE(list))
		break;
		}
		}

		int firewave_constraints(struct snd_pcm_runtime *runtime)
		return snd_interval_list(c, count, list, 0);
		}

		static void limit_channels_and_rates(struct snd_pcm_hardware hw, u8 *formats)
		{
		static unsigned int channels_list[] = { 2, 6 };
		static struct snd_pcm_hw_constraint_list channels_list_constraint = {
		.count = 2,
		.list = channels_list,
		};
		int err;
		struct snd_oxfw_stream_formation formation;
		unsigned int i, err;

		runtime->hw.rates = SNDRV_PCM_RATE_32000 \|
		SNDRV_PCM_RATE_44100 \|
		SNDRV_PCM_RATE_48000 \|
		SNDRV_PCM_RATE_96000;
		runtime->hw.channels_max = 6;
		hw->channels_min = UINT_MAX;
		hw->channels_max = 0;

		err = snd_pcm_hw_constraint_list(runtime, 0,
		SNDRV_PCM_HW_PARAM_CHANNELS,
		&channels_list_constraint);
		if (err < 0)
		return err;
		err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
		firewave_rate_constraint, NULL,
		SNDRV_PCM_HW_PARAM_CHANNELS, -1);
		if (err < 0)
		return err;
		err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
		firewave_channels_constraint, NULL,
		SNDRV_PCM_HW_PARAM_RATE, -1);
		hw->rate_min = UINT_MAX;
		hw->rate_max = 0;
		hw->rates = 0;

		for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
		if (formats[i] == NULL)
		break;

		err = snd_oxfw_stream_parse_format(formats[i], &formation);
		if (err < 0)
		return err;
		continue;

		return 0;
		hw->channels_min = min(hw->channels_min, formation.pcm);
		hw->channels_max = max(hw->channels_max, formation.pcm);

		hw->rate_min = min(hw->rate_min, formation.rate);
		hw->rate_max = max(hw->rate_max, formation.rate);
		hw->rates \|= snd_pcm_rate_to_rate_bit(formation.rate);
		}
		}

		int lacie_speakers_constraints(struct snd_pcm_runtime *runtime)
		static void limit_period_and_buffer(struct snd_pcm_hardware *hw)
		{
		runtime->hw.rates = SNDRV_PCM_RATE_32000 \|
		SNDRV_PCM_RATE_44100 \|
		SNDRV_PCM_RATE_48000 \|
		SNDRV_PCM_RATE_88200 \|
		SNDRV_PCM_RATE_96000;
		hw->periods_min = 2; /* SNDRV_PCM_INFO_BATCH */
		hw->periods_max = UINT_MAX;

		return 0;
		hw->period_bytes_min = 4 * hw->channels_max; /* bytes for a frame */

		/* Just to prevent from allocating much pages. */
		hw->period_bytes_max = hw->period_bytes_min * 2048;
		hw->buffer_bytes_max = hw->period_bytes_max * hw->periods_min;
		}

		static int pcm_open(struct snd_pcm_substream *substream)
		{
		static const struct snd_pcm_hardware hardware = {
		.info = SNDRV_PCM_INFO_MMAP \|
		SNDRV_PCM_INFO_MMAP_VALID \|
		SNDRV_PCM_INFO_BATCH \|
		SNDRV_PCM_INFO_INTERLEAVED \|
		SNDRV_PCM_INFO_BLOCK_TRANSFER,
		.formats = AMDTP_OUT_PCM_FORMAT_BITS,
		.channels_min = 2,
		.channels_max = 2,
		.buffer_bytes_max = 4 * 1024 * 1024,
		.period_bytes_min = 1,
		.period_bytes_max = UINT_MAX,
		.periods_min = 1,
		.periods_max = UINT_MAX,
		};
		struct snd_oxfw *oxfw = substream->private_data;
		struct snd_pcm_runtime *runtime = substream->runtime;
		bool used;
		u8 **formats;
		int err;

		err = cmp_connection_check_used(&oxfw->in_conn, &used);
		if ((err < 0) \|\| used)
		goto end;
		formats = oxfw->rx_stream_formats;

		runtime->hw.info = SNDRV_PCM_INFO_BATCH \|
		SNDRV_PCM_INFO_BLOCK_TRANSFER \|
		SNDRV_PCM_INFO_INTERLEAVED \|
		SNDRV_PCM_INFO_MMAP \|
		SNDRV_PCM_INFO_MMAP_VALID;

		runtime->hw = hardware;
		limit_channels_and_rates(&runtime->hw, formats);
		limit_period_and_buffer(&runtime->hw);

		err = oxfw->device_info->pcm_constraints(runtime);
		err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
		hw_rule_channels, formats,
		SNDRV_PCM_HW_PARAM_RATE, -1);
		if (err < 0)
		goto end;
		err = snd_pcm_limit_hw_rates(runtime);

		err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
		hw_rule_rate, formats,
		SNDRV_PCM_HW_PARAM_CHANNELS, -1);
		if (err < 0)
		goto end;

		err = amdtp_stream_add_pcm_hw_constraints(&oxfw->rx_stream, runtime);
		if (err < 0)
		goto end;

		snd_pcm_set_sync(substream);
		end:
		return err;
		}

sound/firewire/oxfw/oxfw-stream.c

+268 −0

Original line number	Diff line number	Diff line
		@@ -8,6 +8,35 @@

		#include "oxfw.h"

		#define AVC_GENERIC_FRAME_MAXIMUM_BYTES 512

		/*
		* According to datasheet of Oxford Semiconductor:
		* OXFW970: 32.0/44.1/48.0/96.0 Khz, 8 audio channels I/O
		* OXFW971: 32.0/44.1/48.0/88.2/96.0/192.0 kHz, 16 audio channels I/O, MIDI I/O
		*/
		static const unsigned int oxfw_rate_table[] = {
		[0] = 32000,
		[1] = 44100,
		[2] = 48000,
		[3] = 88200,
		[4] = 96000,
		[5] = 192000,
		};

		/*
		* See Table 5.7 – Sampling frequency for Multi-bit Audio
		* in AV/C Stream Format Information Specification 1.1 (Apr 2005, 1394TA)
		*/
		static const unsigned int avc_stream_rate_table[] = {
		[0] = 0x02,
		[1] = 0x03,
		[2] = 0x04,
		[3] = 0x0a,
		[4] = 0x05,
		[5] = 0x07,
		};

		int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw)
		{
		int err;
		@@ -78,3 +107,242 @@ void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw)
		else
		amdtp_stream_update(&oxfw->rx_stream);
		}

		/*
		* See Table 6.16 - AM824 Stream Format
		* Figure 6.19 - format_information field for AM824 Compound
		* in AV/C Stream Format Information Specification 1.1 (Apr 2005, 1394TA)
		* Also 'Clause 12 AM824 sequence adaption layers' in IEC 61883-6:2005
		*/
		int snd_oxfw_stream_parse_format(u8 *format,
		struct snd_oxfw_stream_formation *formation)
		{
		unsigned int i, e, channels, type;

		memset(formation, 0, sizeof(struct snd_oxfw_stream_formation));

		/*
		* this module can support a hierarchy combination that:
		* Root: Audio and Music (0x90)
		* Level 1: AM824 Compound (0x40)
		*/
		if ((format[0] != 0x90) \|\| (format[1] != 0x40))
		return -ENOSYS;

		/* check the sampling rate */
		for (i = 0; i < ARRAY_SIZE(avc_stream_rate_table); i++) {
		if (format[2] == avc_stream_rate_table[i])
		break;
		}
		if (i == ARRAY_SIZE(avc_stream_rate_table))
		return -ENOSYS;

		formation->rate = oxfw_rate_table[i];

		for (e = 0; e < format[4]; e++) {
		channels = format[5 + e * 2];
		type = format[6 + e * 2];

		switch (type) {
		/* IEC 60958 Conformant, currently handled as MBLA */
		case 0x00:
		/* Multi Bit Linear Audio (Raw) */
		case 0x06:
		formation->pcm += channels;
		break;
		/* MIDI Conformant */
		case 0x0d:
		formation->midi = channels;
		break;
		/* IEC 61937-3 to 7 */
		case 0x01:
		case 0x02:
		case 0x03:
		case 0x04:
		case 0x05:
		/* Multi Bit Linear Audio */
		case 0x07: /* DVD-Audio */
		case 0x0c: /* High Precision */
		/* One Bit Audio */
		case 0x08: /* (Plain) Raw */
		case 0x09: /* (Plain) SACD */
		case 0x0a: /* (Encoded) Raw */
		case 0x0b: /* (Encoded) SACD */
		/* SMPTE Time-Code conformant */
		case 0x0e:
		/* Sample Count */
		case 0x0f:
		/* Anciliary Data */
		case 0x10:
		/* Synchronization Stream (Stereo Raw audio) */
		case 0x40:
		/* Don't care */
		case 0xff:
		default:
		return -ENOSYS; /* not supported */
		}
		}

		if (formation->pcm > AMDTP_MAX_CHANNELS_FOR_PCM \|\|
		formation->midi > AMDTP_MAX_CHANNELS_FOR_MIDI)
		return -ENOSYS;

		return 0;
		}

		static int
		assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir,
		unsigned int pid, u8 buf, unsigned int len,
		u8 **formats)
		{
		struct snd_oxfw_stream_formation formation;
		unsigned int i, eid;
		int err;

		/* get format at current sampling rate */
		err = avc_stream_get_format_single(oxfw->unit, dir, pid, buf, len);
		if (err < 0) {
		dev_err(&oxfw->unit->device,
		"fail to get current stream format for isoc %s plug %d:%d\n",
		(dir == AVC_GENERAL_PLUG_DIR_IN) ? "in" : "out",
		pid, err);
		goto end;
		}

		/* parse and set stream format */
		eid = 0;
		err = snd_oxfw_stream_parse_format(buf, &formation);
		if (err < 0)
		goto end;

		formats[eid] = kmalloc(*len, GFP_KERNEL);
		if (formats[eid] == NULL) {
		err = -ENOMEM;
		goto end;
		}
		memcpy(formats[eid], buf, *len);

		/* apply the format for each available sampling rate */
		for (i = 0; i < ARRAY_SIZE(oxfw_rate_table); i++) {
		if (formation.rate == oxfw_rate_table[i])
		continue;

		err = avc_general_inquiry_sig_fmt(oxfw->unit,
		oxfw_rate_table[i],
		dir, pid);
		if (err < 0)
		continue;

		eid++;
		formats[eid] = kmalloc(*len, GFP_KERNEL);
		if (formats[eid] == NULL) {
		err = -ENOMEM;
		goto end;
		}
		memcpy(formats[eid], buf, *len);
		formats[eid][2] = avc_stream_rate_table[i];
		}

		err = 0;
		oxfw->assumed = true;
		end:
		return err;
		}

		static int fill_stream_formats(struct snd_oxfw *oxfw,
		enum avc_general_plug_dir dir,
		unsigned short pid)
		{
		u8 buf, *formats;
		unsigned int len, eid = 0;
		struct snd_oxfw_stream_formation dummy;
		int err;

		buf = kmalloc(AVC_GENERIC_FRAME_MAXIMUM_BYTES, GFP_KERNEL);
		if (buf == NULL)
		return -ENOMEM;

		formats = oxfw->rx_stream_formats;

		/* get first entry */
		len = AVC_GENERIC_FRAME_MAXIMUM_BYTES;
		err = avc_stream_get_format_list(oxfw->unit, dir, 0, buf, &len, 0);
		if (err == -ENOSYS) {
		/* LIST subfunction is not implemented */
		len = AVC_GENERIC_FRAME_MAXIMUM_BYTES;
		err = assume_stream_formats(oxfw, dir, pid, buf, &len,
		formats);
		goto end;
		} else if (err < 0) {
		dev_err(&oxfw->unit->device,
		"fail to get stream format %d for isoc %s plug %d:%d\n",
		eid, (dir == AVC_GENERAL_PLUG_DIR_IN) ? "in" : "out",
		pid, err);
		goto end;
		}

		/* LIST subfunction is implemented */
		while (eid < SND_OXFW_STREAM_FORMAT_ENTRIES) {
		/* The format is too short. */
		if (len < 3) {
		err = -EIO;
		break;
		}

		/* parse and set stream format */
		err = snd_oxfw_stream_parse_format(buf, &dummy);
		if (err < 0)
		break;

		formats[eid] = kmalloc(len, GFP_KERNEL);
		if (formats[eid] == NULL) {
		err = -ENOMEM;
		break;
		}
		memcpy(formats[eid], buf, len);

		/* get next entry */
		len = AVC_GENERIC_FRAME_MAXIMUM_BYTES;
		err = avc_stream_get_format_list(oxfw->unit, dir, 0,
		buf, &len, ++eid);
		/* No entries remained. */
		if (err == -EINVAL) {
		err = 0;
		break;
		} else if (err < 0) {
		dev_err(&oxfw->unit->device,
		"fail to get stream format %d for isoc %s plug %d:%d\n",
		eid, (dir == AVC_GENERAL_PLUG_DIR_IN) ? "in" :
		"out",
		pid, err);
		break;
		}
		}
		end:
		kfree(buf);
		return err;
		}

		int snd_oxfw_stream_discover(struct snd_oxfw *oxfw)
		{
		u8 plugs[AVC_PLUG_INFO_BUF_BYTES];
		int err;

		/* the number of plugs for isoc in/out, ext in/out */
		err = avc_general_get_plug_info(oxfw->unit, 0x1f, 0x07, 0x00, plugs);
		if (err < 0) {
		dev_err(&oxfw->unit->device,
		"fail to get info for isoc/external in/out plugs: %d\n",
		err);
		goto end;
		} else if (plugs[0] == 0) {
		err = -ENOSYS;
		goto end;
		}

		/* use iPCR[0] if exists */
		if (plugs[0] > 0)
		err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_IN, 0);
		end:
		return err;
		}

sound/firewire/oxfw/oxfw.c

+8 −2

Original line number	Diff line number	Diff line
		@@ -58,6 +58,10 @@ end:
		static void oxfw_card_free(struct snd_card *card)
		{
		struct snd_oxfw *oxfw = card->private_data;
		unsigned int i;

		for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++)
		kfree(oxfw->rx_stream_formats[i]);

		mutex_destroy(&oxfw->mutex);
		}
		@@ -81,6 +85,10 @@ static int oxfw_probe(struct fw_unit *unit,
		oxfw->unit = unit;
		oxfw->device_info = (const struct device_info *)id->driver_data;

		err = snd_oxfw_stream_discover(oxfw);
		if (err < 0)
		goto error;

		err = name_card(oxfw);
		if (err < 0)
		goto error;
		@@ -136,7 +144,6 @@ static const struct device_info griffin_firewave = {
		.driver_name = "FireWave",
		.vendor_name = "Griffin",
		.model_name = "FireWave",
		.pcm_constraints = firewave_constraints,
		.mixer_channels = 6,
		.mute_fb_id = 0x01,
		.volume_fb_id = 0x02,
		@@ -146,7 +153,6 @@ static const struct device_info lacie_speakers = {
		.driver_name = "FWSpeakers",
		.vendor_name = "LaCie",
		.model_name = "FireWire Speakers",
		.pcm_constraints = lacie_speakers_constraints,
		.mixer_channels = 1,
		.mute_fb_id = 0x01,
		.volume_fb_id = 0x01,

sound/firewire/oxfw/oxfw.h

+18 −3

Original line number	Diff line number	Diff line
		@@ -30,19 +30,24 @@ struct device_info {
		const char *driver_name;
		const char *vendor_name;
		const char *model_name;
		int (pcm_constraints)(struct snd_pcm_runtime runtime);
		unsigned int mixer_channels;
		u8 mute_fb_id;
		u8 volume_fb_id;
		};

		/* This is an arbitrary number for convinience. */
		#define SND_OXFW_STREAM_FORMAT_ENTRIES 10
		struct snd_oxfw {
		struct snd_card *card;
		struct fw_unit *unit;
		const struct device_info *device_info;
		struct mutex mutex;

		u8 *rx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES];
		bool assumed;
		struct cmp_connection in_conn;
		struct amdtp_stream rx_stream;

		bool mute;
		s16 volume[6];
		s16 volume_min;
		@@ -88,8 +93,18 @@ void snd_oxfw_stream_stop_simplex(struct snd_oxfw *oxfw);
		void snd_oxfw_stream_destroy_simplex(struct snd_oxfw *oxfw);
		void snd_oxfw_stream_update_simplex(struct snd_oxfw *oxfw);

		int firewave_constraints(struct snd_pcm_runtime *runtime);
		int lacie_speakers_constraints(struct snd_pcm_runtime *runtime);
		struct snd_oxfw_stream_formation {
		unsigned int rate;
		unsigned int pcm;
		unsigned int midi;
		};
		int snd_oxfw_stream_parse_format(u8 *format,
		struct snd_oxfw_stream_formation *formation);
		int snd_oxfw_stream_get_current_formation(struct snd_oxfw *oxfw,
		enum avc_general_plug_dir dir,
		struct snd_oxfw_stream_formation *formation);
		int snd_oxfw_stream_discover(struct snd_oxfw *oxfw);

		int snd_oxfw_create_pcm(struct snd_oxfw *oxfw);

		int snd_oxfw_create_mixer(struct snd_oxfw *oxfw);

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