In previous commit, this driver can get to know stream formations at each supported sampling rates. This commit uses it to make PCM rules/constraints and obsoletes hard-coded rules/constraints.
For this purpose, this commit adds 'struct snd_oxfw_stream_formation' and snd_oxfw_stream_parse_format() to parse data channel formation of data block.
According to datasheet of OXFW970/971, they support 32.0kHz to 196.0kHz.
As long as developers investigate, some devices are confirmed to have several formats for the same sampling rate.
Signed-off-by: Takashi Sakamoto o-takashi@sakamocchi.jp --- sound/firewire/oxfw/oxfw-pcm.c | 197 ++++++++++++++++------------ sound/firewire/oxfw/oxfw-stream.c | 268 ++++++++++++++++++++++++++++++++++++++ sound/firewire/oxfw/oxfw.c | 42 +++--- sound/firewire/oxfw/oxfw.h | 21 ++- 4 files changed, 426 insertions(+), 102 deletions(-)
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index d39f17a..0c0be98 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -7,117 +7,152 @@
#include "oxfw.h"
-static int firewave_rate_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +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 = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_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; + 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); + + } + return snd_interval_refine(r, &t); }
-static int firewave_channels_constraint(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +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 = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - - /* 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; + 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; + + 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; + } + } + + return snd_interval_list(c, count, list, 0); }
-int firewave_constraints(struct snd_pcm_runtime *runtime) +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); - if (err < 0) - return err; + hw->rate_min = UINT_MAX; + hw->rate_max = 0; + hw->rates = 0;
- return 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) + continue; + + 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; } diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index ebd156f..17e3802 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -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; +} diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 3308253..4514650 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -25,6 +25,22 @@ MODULE_AUTHOR("Clemens Ladisch clemens@ladisch.de"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("snd-firewire-speakers");
+static const struct device_info griffin_firewave = { + .vendor_name = "Griffin", + .driver_name = "FireWave", + .mixer_channels = 6, + .mute_fb_id = 0x01, + .volume_fb_id = 0x02, +}; + +static const struct device_info lacie_speakers = { + .vendor_name = "LaCie", + .driver_name = "FWSpeakers", + .mixer_channels = 1, + .mute_fb_id = 0x01, + .volume_fb_id = 0x01, +}; + static int name_card(struct snd_oxfw *oxfw) { struct fw_device *fw_dev = fw_parent_device(oxfw->unit); @@ -58,6 +74,10 @@ 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 +101,10 @@ 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; @@ -132,24 +156,6 @@ static void oxfw_remove(struct fw_unit *unit) snd_card_free_when_closed(oxfw->card); }
-static const struct device_info griffin_firewave = { - .vendor_name = "Griffin", - .driver_name = "FireWave", - .pcm_constraints = firewave_constraints, - .mixer_channels = 6, - .mute_fb_id = 0x01, - .volume_fb_id = 0x02, -}; - -static const struct device_info lacie_speakers = { - .vendor_name = "LaCie", - .driver_name = "FWSpeakers", - .pcm_constraints = lacie_speakers_constraints, - .mixer_channels = 1, - .mute_fb_id = 0x01, - .volume_fb_id = 0x01, -}; - static const struct ieee1394_device_id oxfw_id_table[] = { { .match_flags = IEEE1394_MATCH_VENDOR_ID | diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 583ee7b..c4c1acd 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -29,19 +29,24 @@ struct device_info { const char *vendor_name; const char *driver_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; @@ -87,8 +92,18 @@ 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);