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author | Takashi Sakamoto <o-takashi@sakamocchi.jp> | 2014-12-09 00:10:42 +0900 |
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committer | Takashi Iwai <tiwai@suse.de> | 2014-12-10 10:47:37 +0100 |
commit | 5cd1d3f47a6321612a51ab88ffe8ef65120fcbe0 (patch) | |
tree | f2f537e9ec9185754441c37f7896788a76795312 /sound/firewire/oxfw/oxfw-stream.c | |
parent | 5b59d8098d2a3fa8ea4ad07b96f62c00c3b3e8d3 (diff) | |
download | linux-5cd1d3f47a6321612a51ab88ffe8ef65120fcbe0.tar.bz2 |
ALSA: oxfw: Change the way to make PCM rules/constraints
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>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Diffstat (limited to 'sound/firewire/oxfw/oxfw-stream.c')
-rw-r--r-- | sound/firewire/oxfw/oxfw-stream.c | 268 |
1 files changed, 268 insertions, 0 deletions
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index ebd156f3e29d..17e3802e6ac2 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; +} |