September 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: davinci-pcm: trivial: replace link with actual chan/link
by Ben Gardiner 16 Sep '11

16 Sep '11

The ambiguously named variable 'link' is used as a temporary throughout davinci-pcm -- its presence makes grepping (and groking) the code difficult. Replace link with the value of link in almost all sites. The exception is a couple places where the last-assigned link/chan needs to be returned by a function -- in these cases, rename to last_link. Signed-off-by: Ben Gardiner <bengardiner(a)nanometrics.ca> --- sound/soc/davinci/davinci-pcm.c | 123 +++++++++++++++++++-------------------- 1 files changed, 59 insertions(+), 64 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index a49e667..d5fe08c 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -180,7 +180,6 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - int link = prtd->asp_link[0]; unsigned int period_size; unsigned int dma_offset; dma_addr_t dma_pos; @@ -198,7 +197,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) fifo_level = prtd->params->fifo_level; pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " - "dma_ptr = %x period_size=%x\n", link, dma_pos, period_size); + "dma_ptr = %x period_size=%x\n", prtd->asp_link[0], dma_pos, + period_size); data_type = prtd->params->data_type; count = period_size / data_type; @@ -222,17 +222,19 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) } acnt = prtd->params->acnt; - edma_set_src(link, src, INCR, W8BIT); - edma_set_dest(link, dst, INCR, W8BIT); + edma_set_src(prtd->asp_link[0], src, INCR, W8BIT); + edma_set_dest(prtd->asp_link[0], dst, INCR, W8BIT); - edma_set_src_index(link, src_bidx, src_cidx); - edma_set_dest_index(link, dst_bidx, dst_cidx); + edma_set_src_index(prtd->asp_link[0], src_bidx, src_cidx); + edma_set_dest_index(prtd->asp_link[0], dst_bidx, dst_cidx); if (!fifo_level) - edma_set_transfer_params(link, acnt, count, 1, 0, ASYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, + ASYNC); else - edma_set_transfer_params(link, acnt, fifo_level, count, - fifo_level, ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, + count, fifo_level, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) @@ -305,7 +307,6 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) unsigned int acnt = params->acnt; /* divide by 2 for ping/pong */ unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1; - int link = prtd->asp_link[1]; unsigned int fifo_level = prtd->params->fifo_level; unsigned int count; if ((data_type == 0) || (data_type > 4)) { @@ -316,28 +317,26 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) dma_addr_t asp_src_pong = iram_dma->addr + ping_size; ram_src_cidx = ping_size; ram_dst_cidx = -ping_size; - edma_set_src(link, asp_src_pong, INCR, W8BIT); + edma_set_src(prtd->asp_link[1], asp_src_pong, INCR, W8BIT); - link = prtd->asp_link[0]; - edma_set_src_index(link, data_type, data_type * fifo_level); - link = prtd->asp_link[1]; - edma_set_src_index(link, data_type, data_type * fifo_level); + edma_set_src_index(prtd->asp_link[0], data_type, + data_type * fifo_level); + edma_set_src_index(prtd->asp_link[1], data_type, + data_type * fifo_level); - link = prtd->ram_link; - edma_set_src(link, runtime->dma_addr, INCR, W32BIT); + edma_set_src(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); } else { dma_addr_t asp_dst_pong = iram_dma->addr + ping_size; ram_src_cidx = -ping_size; ram_dst_cidx = ping_size; - edma_set_dest(link, asp_dst_pong, INCR, W8BIT); + edma_set_dest(prtd->asp_link[1], asp_dst_pong, INCR, W8BIT); - link = prtd->asp_link[0]; - edma_set_dest_index(link, data_type, data_type * fifo_level); - link = prtd->asp_link[1]; - edma_set_dest_index(link, data_type, data_type * fifo_level); + edma_set_dest_index(prtd->asp_link[0], data_type, + data_type * fifo_level); + edma_set_dest_index(prtd->asp_link[1], data_type, + data_type * fifo_level); - link = prtd->ram_link; - edma_set_dest(link, runtime->dma_addr, INCR, W32BIT); + edma_set_dest(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); } if (!fifo_level) { @@ -354,10 +353,9 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) count, fifo_level, ABSYNC); } - link = prtd->ram_link; - edma_set_src_index(link, ping_size, ram_src_cidx); - edma_set_dest_index(link, ping_size, ram_dst_cidx); - edma_set_transfer_params(link, ping_size, 2, + edma_set_src_index(prtd->ram_link, ping_size, ram_src_cidx); + edma_set_dest_index(prtd->ram_link, ping_size, ram_dst_cidx); + edma_set_transfer_params(prtd->ram_link, ping_size, 2, runtime->periods, 2, ASYNC); /* init master params */ @@ -406,32 +404,32 @@ static int request_ping_pong(struct snd_pcm_substream *substream, { dma_addr_t asp_src_ping; dma_addr_t asp_dst_ping; - int link; + int ret; struct davinci_pcm_dma_params *params = prtd->params; /* Request ram master channel */ - link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, + ret = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, davinci_pcm_dma_irq, substream, prtd->params->ram_chan_q); - if (link < 0) + if (ret < 0) goto exit1; /* Request ram link channel */ - link = prtd->ram_link = edma_alloc_slot( + ret = prtd->ram_link = edma_alloc_slot( EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit2; - link = prtd->asp_link[1] = edma_alloc_slot( + ret = prtd->asp_link[1] = edma_alloc_slot( EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit3; prtd->ram_link2 = -1; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - link = prtd->ram_link2 = edma_alloc_slot( + ret = prtd->ram_link2 = edma_alloc_slot( EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit4; } /* circle ping-pong buffers */ @@ -448,36 +446,33 @@ static int request_ping_pong(struct snd_pcm_substream *substream, asp_dst_ping = iram_dma->addr; } /* ping */ - link = prtd->asp_link[0]; - edma_set_src(link, asp_src_ping, INCR, W16BIT); - edma_set_dest(link, asp_dst_ping, INCR, W16BIT); - edma_set_src_index(link, 0, 0); - edma_set_dest_index(link, 0, 0); + edma_set_src(prtd->asp_link[0], asp_src_ping, INCR, W16BIT); + edma_set_dest(prtd->asp_link[0], asp_dst_ping, INCR, W16BIT); + edma_set_src_index(prtd->asp_link[0], 0, 0); + edma_set_dest_index(prtd->asp_link[0], 0, 0); - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); prtd->asp_params.opt |= TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(link, &prtd->asp_params); + edma_write_slot(prtd->asp_link[0], &prtd->asp_params); /* pong */ - link = prtd->asp_link[1]; - edma_set_src(link, asp_src_ping, INCR, W16BIT); - edma_set_dest(link, asp_dst_ping, INCR, W16BIT); - edma_set_src_index(link, 0, 0); - edma_set_dest_index(link, 0, 0); + edma_set_src(prtd->asp_link[1], asp_src_ping, INCR, W16BIT); + edma_set_dest(prtd->asp_link[1], asp_dst_ping, INCR, W16BIT); + edma_set_src_index(prtd->asp_link[1], 0, 0); + edma_set_dest_index(prtd->asp_link[1], 0, 0); - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[1], &prtd->asp_params); prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); /* interrupt after every pong completion */ prtd->asp_params.opt |= TCINTEN | TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(link, &prtd->asp_params); + edma_write_slot(prtd->asp_link[1], &prtd->asp_params); /* ram */ - link = prtd->ram_link; - edma_set_src(link, iram_dma->addr, INCR, W32BIT); - edma_set_dest(link, iram_dma->addr, INCR, W32BIT); + edma_set_src(prtd->ram_link, iram_dma->addr, INCR, W32BIT); + edma_set_dest(prtd->ram_link, iram_dma->addr, INCR, W32BIT); pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u," "for asp:%u %u %u\n", __func__, prtd->ram_channel, prtd->ram_link, prtd->ram_link2, @@ -494,7 +489,7 @@ exit2: edma_free_channel(prtd->ram_channel); prtd->ram_channel = -1; exit1: - return link; + return ret; } static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) @@ -502,22 +497,22 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) struct snd_dma_buffer *iram_dma; struct davinci_runtime_data *prtd = substream->runtime->private_data; struct davinci_pcm_dma_params *params = prtd->params; - int link; + int ret; if (!params) return -ENODEV; /* Request asp master DMA channel */ - link = prtd->asp_channel = edma_alloc_channel(params->channel, + ret = prtd->asp_channel = edma_alloc_channel(params->channel, davinci_pcm_dma_irq, substream, prtd->params->asp_chan_q); - if (link < 0) + if (ret < 0) goto exit1; /* Request asp link channels */ - link = prtd->asp_link[0] = edma_alloc_slot( + ret = prtd->asp_link[0] = edma_alloc_slot( EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit2; iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; @@ -537,17 +532,17 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) * the buffer and its length (ccnt) ... use it as a template * so davinci_pcm_enqueue_dma() takes less time in IRQ. */ - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); prtd->asp_params.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); - prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(link) << 5; - edma_write_slot(link, &prtd->asp_params); + prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5; + edma_write_slot(prtd->asp_link[0], &prtd->asp_params); return 0; exit2: edma_free_channel(prtd->asp_channel); prtd->asp_channel = -1; exit1: - return link; + return ret; } static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -- 1.7.4.1

3 3

[alsa-devel] [PATCH] pulse: Install a PulseAudio config snippet into alsa.conf.d
by gmane＠colin.guthr.ie 16 Sep '11

16 Sep '11

From: Colin Guthrie <colin(a)mageia.org> --- configure.in | 19 ++++++++++++++++++- pulse/Makefile.am | 2 ++ pulse/pulseaudio-default.conf.example | 13 +++++++++++++ pulse/pulseaudio.conf | 13 +++++++++++++ 4 files changed, 46 insertions(+), 1 deletions(-) create mode 100644 pulse/pulseaudio-default.conf.example create mode 100644 pulse/pulseaudio.conf diff --git a/configure.in b/configure.in index ccf59ba..3839dd5 100644 --- a/configure.in +++ b/configure.in @@ -131,10 +131,10 @@ fi AM_CONDITIONAL(HAVE_PPH, test "$PPH" = "builtin" -o "$PPH" = "lib") AM_CONDITIONAL(USE_LIBSPEEX, test "$PPH" = "lib") -dnl ALSA plugin directory test "x$prefix" = xNONE && prefix=$ac_default_prefix test "x$exec_prefix" = xNONE && exec_prefix=$prefix +dnl ALSA plugin directory AC_ARG_WITH(plugindir, AS_HELP_STRING([--with-plugindir=dir], [path where ALSA plugin files are stored]), @@ -151,6 +151,23 @@ AC_DEFINE_UNQUOTED(ALSA_PLUGIN_DIR, "$plugindir", [directory containing ALSA add ALSA_PLUGIN_DIR="$plugindir" AC_SUBST(ALSA_PLUGIN_DIR) +dnl ALSA data directory +AC_ARG_WITH(alsadatadir, + AS_HELP_STRING([--with-alsadatadir=dir], + [path where ALSA data files are stored]), + alsadatadir="$withval", alsadatadir="") +if test -z "$alsadatadir"; then + eval dir="$datadir" + case "$dir" in + /*) ;; + *) dir="$dir" + esac + alsadatadir="$dir/alsa" +fi +AC_DEFINE_UNQUOTED(ALSA_DATA_DIR, "$alsadatadir", [directory containing ALSA data files]) +ALSA_DATA_DIR="$alsadatadir" +AC_SUBST(ALSA_DATA_DIR) + SAVE_PLUGINS_VERSION AC_OUTPUT([ diff --git a/pulse/Makefile.am b/pulse/Makefile.am index e89e8ee..cca49ac 100644 --- a/pulse/Makefile.am +++ b/pulse/Makefile.am @@ -1,10 +1,12 @@ asound_module_pcm_LTLIBRARIES = libasound_module_pcm_pulse.la asound_module_ctl_LTLIBRARIES = libasound_module_ctl_pulse.la asound_module_conf_LTLIBRARIES = libasound_module_conf_pulse.la +asound_module_data_DATA = pulseaudio.conf pulseaudio-default.conf.example asound_module_pcmdir = @ALSA_PLUGIN_DIR@ asound_module_ctldir = @ALSA_PLUGIN_DIR@ asound_module_confdir = @ALSA_PLUGIN_DIR@ +asound_module_datadir = @ALSA_DATA_DIR@/alsa.conf.d AM_CFLAGS = -Wall -g @ALSA_CFLAGS@ $(PTHREAD_CFLAGS) $(pulseaudio_CFLAGS) -D_GNU_SOURCE AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED) diff --git a/pulse/pulseaudio-default.conf.example b/pulse/pulseaudio-default.conf.example new file mode 100644 index 0000000..4f58858 --- /dev/null +++ b/pulse/pulseaudio-default.conf.example @@ -0,0 +1,13 @@ +# Default to PulseAudio + +pcm.!default { + type pulse + hint { + show on + description "Default ALSA Output (currently PulseAudio Sound Server)" + } +} + +ctl.!default { + type pulse +} diff --git a/pulse/pulseaudio.conf b/pulse/pulseaudio.conf new file mode 100644 index 0000000..dd85dab --- /dev/null +++ b/pulse/pulseaudio.conf @@ -0,0 +1,13 @@ +# Add a specific named PulseAudio pcm and ctl (typically useful for testing) + +pcm.pulse { + type pulse + hint { + show on + description "PulseAudio Sound Server" + } +} + +ctl.pulse { + type pulse +} -- 1.7.6

3 2

[PATCH] ALSA: HDA: Fix Realtek automute for Headphone to Line Out
by David Henningsson 16 Sep '11

16 Sep '11

If the machine has "Headphone out" and "Line Out", but no "Speaker out", the Realtek autoparser currently creates an Automute kcontrol, but it isn't working. This patch enables automuting of "Line out" when headphones are plugged, provided that the automute kcontrol is set to "enabled". BugLink: http://bugs.launchpad.net/bugs/851697 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 92 ++++++++++++++++++++++++----------------- 1 files changed, 54 insertions(+), 38 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 70ba45e..8dff8d2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -167,10 +167,12 @@ struct alc_spec { unsigned int master_mute:1; unsigned int auto_mic:1; unsigned int auto_mic_valid_imux:1; /* valid imux for auto-mic */ - unsigned int automute:1; /* HP automute enabled */ + unsigned int automute:1; /* Speaker automute enabled */ + unsigned int detect_hp:1; /* Headphones detection enabled */ unsigned int detect_line:1; /* Line-out detection enabled */ unsigned int automute_lines:1; /* automute line-out as well */ - unsigned int automute_hp_lo:1; /* both HP and LO available */ + unsigned int automute_speaker_possible:1; /* there are speakers and either LO or HP */ + unsigned int automute_lo_possible:1; /* there are line outs and HP */ /* other flags */ unsigned int no_analog :1; /* digital I/O only */ @@ -556,7 +558,7 @@ static void update_speakers(struct hda_codec *codec) if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] || spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0]) return; - if (!spec->automute_lines || !spec->automute) + if (!spec->automute_lines) on = 0; else on = spec->jack_present; @@ -582,7 +584,7 @@ static void alc_hp_automute(struct hda_codec *codec) spec->jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins); - if (!spec->automute) + if (!spec->detect_hp || (!spec->automute && !spec->automute_lines)) return; call_update_speakers(codec); } @@ -795,7 +797,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - if (spec->automute_hp_lo) { + if (spec->automute_speaker_possible && spec->automute_lo_possible) { uinfo->value.enumerated.items = 3; texts = texts3; } else { @@ -814,13 +816,12 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - unsigned int val; - if (!spec->automute) - val = 0; - else if (!spec->automute_lines) - val = 1; - else - val = 2; + unsigned int val = 0; + if (spec->automute) + val++; + if (spec->automute_lines) + val++; + ucontrol->value.enumerated.item[0] = val; return 0; } @@ -833,18 +834,29 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol, switch (ucontrol->value.enumerated.item[0]) { case 0: - if (!spec->automute) + if (!spec->automute && !spec->automute_lines) return 0; spec->automute = 0; + spec->automute_lines = 0; break; case 1: - if (spec->automute && !spec->automute_lines) - return 0; - spec->automute = 1; - spec->automute_lines = 0; + if (spec->automute_speaker_possible) { + if (!spec->automute_lines && spec->automute) + return 0; + spec->automute = 1; + spec->automute_lines = 0; + } + else if (spec->automute_lo_possible) { + if (spec->automute_lines) + return 0; + spec->automute_lines = 1; + } + else + return -EINVAL; + break; case 2: - if (!spec->automute_hp_lo) + if (!spec->automute_lo_possible || !spec->automute_speaker_possible) return -EINVAL; if (spec->automute && spec->automute_lines) return 0; @@ -906,8 +918,6 @@ static void alc_init_auto_hp(struct hda_codec *codec) present++; if (present < 2) /* need two different output types */ return; - if (present == 3) - spec->automute_hp_lo = 1; /* both HP and LO automute */ if (!cfg->speaker_pins[0] && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { @@ -923,6 +933,8 @@ static void alc_init_auto_hp(struct hda_codec *codec) cfg->hp_outs = cfg->line_outs; } + spec->automute_mode = ALC_AUTOMUTE_PIN; + for (i = 0; i < cfg->hp_outs; i++) { hda_nid_t nid = cfg->hp_pins[i]; if (!is_jack_detectable(codec, nid)) @@ -932,28 +944,32 @@ static void alc_init_auto_hp(struct hda_codec *codec) snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT); - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; - } - if (spec->automute && cfg->line_out_pins[0] && - cfg->speaker_pins[0] && - cfg->line_out_pins[0] != cfg->hp_pins[0] && - cfg->line_out_pins[0] != cfg->speaker_pins[0]) { - for (i = 0; i < cfg->line_outs; i++) { - hda_nid_t nid = cfg->line_out_pins[i]; - if (!is_jack_detectable(codec, nid)) - continue; - snd_printdd("realtek: Enable Line-Out auto-muting " - "on NID 0x%x\n", nid); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC_FRONT_EVENT); - spec->detect_line = 1; + spec->detect_hp = 1; + } + + if (cfg->line_out_type == AUTO_PIN_LINE_OUT && cfg->line_outs) { + if (cfg->speaker_outs) + for (i = 0; i < cfg->line_outs; i++) { + hda_nid_t nid = cfg->line_out_pins[i]; + if (!is_jack_detectable(codec, nid)) + continue; + snd_printdd("realtek: Enable Line-Out " + "auto-muting on NID 0x%x\n", nid); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | ALC_FRONT_EVENT); + spec->detect_line = 1; } - spec->automute_lines = spec->detect_line; + spec->automute_lo_possible = spec->detect_hp; } - if (spec->automute) { + spec->automute_speaker_possible = cfg->speaker_outs && + (spec->detect_hp || spec->detect_line); + + spec->automute_lines = spec->automute_lo_possible; + spec->automute = spec->automute_speaker_possible; + + if (spec->automute_speaker_possible || spec->automute_lo_possible) { /* create a control for automute mode */ alc_add_automute_mode_enum(codec); spec->unsol_event = alc_sku_unsol_event; -- 1.7.5.4 --------------050207030404040202010807--

1 0

[alsa-devel] [PATCH v3 1/1] ASoC: mxs-saif: add record function
by Dong Aisheng 16 Sep '11

16 Sep '11

1. add different clkmux mode handling SAIF can use two instances to implement full duplex (playback & recording) and record saif may work on EXTMASTER mode which is using other saif's BITCLK&LRCLK. The clkmux mode could be set in pdata->init() in mach-specific code. For generic saif driver, it only needs to know who is his master and the master id is also provided in mach-specific code. 2. support playback and capture simutaneously however the sample rates can not be different due to hw limitation. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Wolfram Sang <w.sang(a)pengutronix.de> --- Changes since v2: * remove mach-specific code(clkmux in DIGCTL) out of saif driver For supporting EXTMASTER mode, SAIF only nees to know who's its master and the master id is provided by mach layer according to different clkmux setting. So we need to add a pdata->get_master_id(); Changes since v1: * calc the delay based on the rate * change the saif.h #ifndef micro to match the directory --- include/sound/saif.h | 16 +++++ sound/soc/mxs/mxs-saif.c | 145 +++++++++++++++++++++++++++++++++++++++++----- sound/soc/mxs/mxs-saif.h | 4 + 3 files changed, 151 insertions(+), 14 deletions(-) diff --git a/include/sound/saif.h b/include/sound/saif.h new file mode 100644 index 0000000..d0e0de7 --- /dev/null +++ b/include/sound/saif.h @@ -0,0 +1,16 @@ +/* + * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SAIF_H__ +#define __SOUND_SAIF_H__ + +struct mxs_saif_platform_data { + int (*init) (void); + int (*get_master_id) (unsigned int saif_id); +}; +#endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index af5734f..401944c 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -23,10 +23,12 @@ #include <linux/dma-mapping.h> #include <linux/clk.h> #include <linux/delay.h> +#include <linux/time.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/saif.h> #include <mach/dma.h> #include <asm/mach-types.h> #include <mach/hardware.h> @@ -36,6 +38,24 @@ static struct mxs_saif *mxs_saif[2]; +/* + * SAIF is a little different with other normal SOC DAIs on clock using. + * + * For MXS, two SAIF modules are instantiated on-chip. + * Each SAIF has a set of clock pins and can be operating in master + * mode simultaneously if they are connected to different off-chip codecs. + * Also, one of the two SAIFs can master or drive the clock pins while the + * other SAIF, in slave mode, receives clocking from the master SAIF. + * This also means that both SAIFs must operate at the same sample rate. + * + * We abstract this as each saif has a master, the master could be + * himself or other saifs. In the generic saif driver, saif does not need + * to know the different clkmux. Saif only needs to know who is his master + * and operating his master to generate the proper clock rate for him. + * The master id is provided in mach-specific layer according to different + * clkmux setting. + */ + static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -52,6 +72,17 @@ static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, } /* + * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK + * is provided by other SAIF, we provide a interface here to get its master + * from its master_id. + * Note that the master could be himself. + */ +static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif) +{ + return mxs_saif[saif->master_id]; +} + +/* * Set SAIF clock and MCLK */ static int mxs_saif_set_clk(struct mxs_saif *saif, @@ -60,8 +91,26 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, { u32 scr; int ret; + struct mxs_saif *master_saif; - scr = __raw_readl(saif->base + SAIF_CTRL); + dev_dbg(saif->dev, "mclk %d rate %d\n", mclk, rate); + + /* Set master saif to generate proper clock */ + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + + dev_dbg(saif->dev, "master saif%d\n", master_saif->id); + + /* Checking if can playback and capture simutaneously */ + if (master_saif->ongoing && rate != master_saif->cur_rate) { + dev_err(saif->dev, + "can not change clock, master saif%d(rate %d) is ongoing\n", + master_saif->id, master_saif->cur_rate); + return -EINVAL; + } + + scr = __raw_readl(master_saif->base + SAIF_CTRL); scr &= ~BM_SAIF_CTRL_BITCLK_MULT_RATE; scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; @@ -75,27 +124,29 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, * * If MCLK is not used, we just set saif clk to 512*fs. */ - if (saif->mclk_in_use) { + if (master_saif->mclk_in_use) { if (mclk % 32 == 0) { scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; - ret = clk_set_rate(saif->clk, 512 * rate); + ret = clk_set_rate(master_saif->clk, 512 * rate); } else if (mclk % 48 == 0) { scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE; - ret = clk_set_rate(saif->clk, 384 * rate); + ret = clk_set_rate(master_saif->clk, 384 * rate); } else { /* SAIF MCLK should be either 32x or 48x */ return -EINVAL; } } else { - ret = clk_set_rate(saif->clk, 512 * rate); + ret = clk_set_rate(master_saif->clk, 512 * rate); scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; } if (ret) return ret; - if (!saif->mclk_in_use) { - __raw_writel(scr, saif->base + SAIF_CTRL); + master_saif->cur_rate = rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(scr, master_saif->base + SAIF_CTRL); return 0; } @@ -137,7 +188,7 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, return -EINVAL; } - __raw_writel(scr, saif->base + SAIF_CTRL); + __raw_writel(scr, master_saif->base + SAIF_CTRL); return 0; } @@ -183,6 +234,7 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, struct mxs_saif *saif = mxs_saif[saif_id]; u32 stat; int ret; + struct mxs_saif *master_saif; if (!saif) return -EINVAL; @@ -195,6 +247,12 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, __raw_writel(BM_SAIF_CTRL_CLKGATE, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + master_saif = mxs_saif_get_master(saif); + if (saif != master_saif) { + dev_err(saif->dev, "can not get mclk from a non-master saif\n"); + return -EINVAL; + } + stat = __raw_readl(saif->base + SAIF_STAT); if (stat & BM_SAIF_STAT_BUSY) { dev_err(saif->dev, "error: busy\n"); @@ -278,10 +336,17 @@ static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) /* * Note: We simply just support master mode since SAIF TX can only * work as master. + * Here the master is relative to codec side. + * Saif internally could be slave when working on EXTMASTER mode. + * We just hide this to machine driver. */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + if (saif->id == saif->master_id) + scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + else + scr |= BM_SAIF_CTRL_SLAVE_MODE; + __raw_writel(scr | scr0, saif->base + SAIF_CTRL); break; default: @@ -396,6 +461,12 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + struct mxs_saif *master_saif; + u32 delay; + + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -403,10 +474,20 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: dev_dbg(cpu_dai->dev, "start\n"); - clk_enable(saif->clk); - if (!saif->mclk_in_use) + clk_enable(master_saif->clk); + if (!master_saif->mclk_in_use) + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_SET_ADDR); + + /* + * If the saif's master is not himself, we also need to enable + * itself clk for its internal basic logic to work. + */ + if (saif != master_saif) { + clk_enable(saif->clk); __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_SET_ADDR); + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* @@ -422,20 +503,39 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, __raw_readl(saif->base + SAIF_DATA); } - dev_dbg(cpu_dai->dev, "CTRL 0x%x STAT 0x%x\n", + master_saif->ongoing = 1; + + dev_dbg(saif->dev, "CTRL 0x%x STAT 0x%x\n", __raw_readl(saif->base + SAIF_CTRL), __raw_readl(saif->base + SAIF_STAT)); + dev_dbg(master_saif->dev, "CTRL 0x%x STAT 0x%x\n", + __raw_readl(master_saif->base + SAIF_CTRL), + __raw_readl(master_saif->base + SAIF_STAT)); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: dev_dbg(cpu_dai->dev, "stop\n"); - clk_disable(saif->clk); - if (!saif->mclk_in_use) + /* wait a while for the current sample to complete */ + delay = USEC_PER_SEC / master_saif->cur_rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(delay); + } + clk_disable(master_saif->clk); + + if (saif != master_saif) { __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(delay); + clk_disable(saif->clk); + } + + master_saif->ongoing = 0; break; default: @@ -519,16 +619,33 @@ static int mxs_saif_probe(struct platform_device *pdev) { struct resource *res; struct mxs_saif *saif; + struct mxs_saif_platform_data *pdata; int ret = 0; if (pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; + pdata = pdev->dev.platform_data; + if (pdata && pdata->init) { + ret = pdata->init(); + if (ret) + return ret; + } + saif = kzalloc(sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; mxs_saif[pdev->id] = saif; + saif->id = pdev->id; + + saif->master_id = saif->id; + if (pdata && pdata->get_master_id) { + saif->master_id = pdata->get_master_id(saif->id); + if (saif->master_id < 0 || + saif->master_id >= ARRAY_SIZE(mxs_saif)) + return -EINVAL; + } saif->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 0e2ff8c..12c91e4 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -118,6 +118,10 @@ struct mxs_saif { void __iomem *base; int irq; struct mxs_pcm_dma_params dma_param; + unsigned int id; + unsigned int master_id; + unsigned int cur_rate; + unsigned int ongoing; struct platform_device *soc_platform_pdev; u32 fifo_underrun; -- 1.7.0.4

6 7

[alsa-devel] snd_pcm_pause kills plaback handle on OMAP4
by john＠johnandjuliet.com 16 Sep '11

16 Sep '11

I'm trying to control the McBSP2 that is used for the modem in an android environment (BLAZE OMAP4) What I am trying to do is control the audio paths for capture and playback seperatly like in a SIMPLEX (half duplex) instead of full DUPLEX mode audio .. I also want to stop the audio without total shutdown of the audio system. From what I can see that would be to call snd_pcm_pause(handle,bool) But here is what happens. With the handle forthe plaback path everything works OK... But if I call that then the capture path stops working... even if I try to turn on (unpause) both paths. This implies some sort of bug in the omap soc driver? Any help on how to figure this out? Thnkx JBP

2 1

[alsa-devel] Allow alsa configuration to be loaded from a alsa.conf.d folder
by gmane＠colin.guthr.ie 16 Sep '11

16 Sep '11

Hi, This is a small patch for alsa-lib that supports the common ".d" folder configs. The idea here is to make it easier for packagers and distros to supply plugin configuration without having to patch alsa-lib to do so (or by managing a /etc/asound.conf file which may want to be tweaked by multiple files) If this patch is accepted, I'll modify alsa-plugins to inlcude a pulseaudio.conf file in /usr/share/alsa/alsa.conf.d/ which defines a pcm.pulse anc ctl.pulse. If distros want to default to PA, they should include their own file in there that defines pcm|ctl.!default appropriately. Similar configuration could be added for third party alsa plugins such as the DTS encoder plugin being developed by Alexander Patrikov which are shipped outside of alsa. I don't think there is anything too controversial in here, so hopefully it'll be accepted. Cheers Col

4 9

[alsa-devel] snd-usb: endpoint code rework
by Daniel Mack 16 Sep '11

16 Sep '11

With major parts of code shuffling out of the way, it might get clear where the actual rework of the endpoint code goes. I've attached my local diff (basically the same code base as some weeks ago) for an review of the rough idea. Thanks, Daniel

2 3

[alsa-devel] [PATCH] snd-usb-6fire: volume control for individual analog channels
by Torsten Schenk 16 Sep '11

16 Sep '11

Added a stereo volume control for each analog output channel pair. Master volume modified so that it acts as a software weight for the individual channel faders. Added mute switches for master and individual analog channels. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> Cc: stable(a)kernel.org [3.0+] --- diff -Nur a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c --- a/sound/usb/6fire/chip.c 2011-09-15 16:11:08.000000000 +0000 +++ b/sound/usb/6fire/chip.c 2011-09-15 16:11:46.000000000 +0000 @@ -5,7 +5,7 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 + * Version: 0.3.6 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify @@ -29,7 +29,7 @@ #include <sound/initval.h> MODULE_AUTHOR("Torsten Schenk <torsten.schenk(a)zoho.com>"); -MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver, version 0.3.0"); +MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver, version 0.3.6"); MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{TerraTec, DMX 6Fire USB}}"); diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2011-09-02 13:48:10.000000000 +0000 +++ b/sound/usb/6fire/control.c 2011-09-02 13:50:35.000000000 +0000 @@ -5,9 +5,13 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 + * Version: 0.3.6 * Copyright: (C) Torsten Schenk * + * Thanks to: + * - Holger Ruckdeschel: he found out how to control individual analog + * channel volumes and introduced mute switches + * * 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 @@ -59,7 +63,7 @@ { 0x22, 0x03, 0x00 }, { 0x20, 0x03, 0x08 }, { 0x22, 0x04, 0x00 }, { 0x20, 0x04, 0x08 }, { 0x22, 0x05, 0x01 }, { 0x20, 0x05, 0x08 }, { 0x22, 0x04, 0x01 }, { 0x12, 0x04, 0x00 }, { 0x12, 0x05, 0x00 }, - { 0x12, 0x0d, 0x78 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, + { 0x12, 0x0d, 0x38 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, { 0x12, 0x23, 0x00 }, { 0x12, 0x06, 0x02 }, { 0x12, 0x03, 0x00 }, { 0x12, 0x02, 0x00 }, { 0x22, 0x03, 0x01 }, { 0 } /* TERMINATING ENTRY */ @@ -74,15 +78,35 @@ DIGITAL_THRU_ONLY_SAMPLERATE = 3 }; -static void usb6fire_control_master_vol_update(struct control_runtime *rt) +static void usb6fire_control_output_vol_update(struct control_runtime *rt) +{ + struct comm_runtime *comm_rt = rt->chip->comm; + u8 value; + + if (comm_rt) { + int i; + for (i = 0; i < 6; i++) { + /* set volume */ + value = rt->output_vol[i] * rt->master_vol / 0x7f; + comm_rt->write8(comm_rt, 0x12, 0x0f + i, 0x7f - + log_volume_table[value]); + } + } +} + +static void usb6fire_control_output_switch_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; + int i; + u8 value = 0x00; + if (comm_rt) { - /* set volume */ - comm_rt->write8(comm_rt, 0x12, 0x0f, 0x7f - - log_volume_table[rt->master_vol]); - /* unmute */ - comm_rt->write8(comm_rt, 0x12, 0x0e, 0x00); + if (rt->master_switch) /* mute/unmute single outputs */ + for (i = 0; i < 6; i++) + value |= !rt->output_switch[i] << i; + else /* mute all outputs */ + value = 0x3f; + comm_rt->write8(comm_rt, 0x12, 0x0e, value); } } @@ -171,7 +195,17 @@ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 127; + uinfo->value.integer.max = 0x7f; + return 0; +} + +static int usb6fire_control_output_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0x7f; return 0; } @@ -182,7 +216,12 @@ int changed = 0; if (rt->master_vol != ucontrol->value.integer.value[0]) { rt->master_vol = ucontrol->value.integer.value[0]; - usb6fire_control_master_vol_update(rt); + usb6fire_control_output_vol_update(rt); + changed = 1; + } + if (rt->master_vol != ucontrol->value.integer.value[0]) { + rt->master_vol = ucontrol->value.integer.value[0]; + usb6fire_control_output_vol_update(rt); changed = 1; } return changed; @@ -196,6 +235,127 @@ return 0; } +static int usb6fire_control_master_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + int changed = 0; + if (rt->master_switch != ucontrol->value.integer.value[0]) { + rt->master_switch = ucontrol->value.integer.value[0]; + usb6fire_control_output_switch_update(rt); + changed = 1; + } + return changed; +} + +static int usb6fire_control_master_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = rt->master_switch; + return 0; +} + +static int usb6fire_control_output_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; + int changed = 0; + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX + "Invalid channel in volume control."); + return 0; + } + + if (rt->output_vol[ch0] != ucontrol->value.integer.value[0]) { + rt->output_vol[ch0] = ucontrol->value.integer.value[0]; + changed = 1; + } + if (rt->output_vol[ch1] != ucontrol->value.integer.value[1]) { + rt->output_vol[ch1] = ucontrol->value.integer.value[1]; + changed = 1; + } + + if (changed) + usb6fire_control_output_vol_update(rt); + + return changed; +} + +static int usb6fire_control_output_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX + "Invalid channel in volume control."); + return 0; + } + + ucontrol->value.integer.value[0] = rt->output_vol[ch0]; + ucontrol->value.integer.value[1] = rt->output_vol[ch1]; + + return 0; +} + +static int usb6fire_control_output_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; + int changed = 0; + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX + "Invalid channel in volume control."); + return 0; + } + + if (rt->output_switch[ch0] != ucontrol->value.integer.value[0]) { + rt->output_switch[ch0] = ucontrol->value.integer.value[0]; + changed = 1; + } + if (rt->output_switch[ch1] != ucontrol->value.integer.value[1]) { + rt->output_switch[ch1] = ucontrol->value.integer.value[1]; + changed = 1; + } + + if (changed) + usb6fire_control_output_switch_update(rt); + + return changed; +} + +static int usb6fire_control_output_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX + "Invalid channel in volume control."); + return 0; + } + + ucontrol->value.integer.value[0] = rt->output_switch[ch0]; + ucontrol->value.integer.value[1] = rt->output_switch[ch1]; + + return 0; +} + static int usb6fire_control_line_phono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -299,6 +459,75 @@ }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_mono_info, + .get = usb6fire_control_master_switch_get, + .put = usb6fire_control_master_switch_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 1/2 Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .private_value = 0 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 1/2 Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_switch_get, + .put = usb6fire_control_output_switch_put, + .private_value = 0 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 3/4 Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .private_value = 1 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 3/4 Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_switch_get, + .put = usb6fire_control_output_switch_put, + .private_value = 1 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 5/6 Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .private_value = 2 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 5/6 Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_switch_get, + .put = usb6fire_control_output_switch_put, + .private_value = 2 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line/Phono Capture Route", .index = 0, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -352,7 +581,8 @@ usb6fire_control_opt_coax_update(rt); usb6fire_control_line_phono_update(rt); - usb6fire_control_master_vol_update(rt); + usb6fire_control_output_vol_update(rt); + usb6fire_control_output_switch_update(rt); usb6fire_control_streaming_update(rt); i = 0; @@ -378,3 +608,4 @@ kfree(chip->control); chip->control = NULL; } + diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2011-09-02 13:48:10.000000000 +0000 +++ b/sound/usb/6fire/control.h 2011-09-02 13:50:30.000000000 +0000 @@ -3,9 +3,13 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 + * Version: 0.3.6 * Copyright: (C) Torsten Schenk * + * Thanks to: + * - Holger Ruckdeschel: he found out how to control individual analog + * channel volumes and introduced mute switches + * * 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 @@ -45,6 +49,9 @@ bool digital_thru_switch; bool usb_streaming; u8 master_vol; + bool master_switch; + u8 output_vol[6]; + bool output_switch[6]; }; int __devinit usb6fire_control_init(struct sfire_chip *chip);

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[alsa-devel] Temporary locations of sound git and snapshot tarballs
by Takashi Iwai 16 Sep '11

16 Sep '11

Hi, as kernel.org downtime continues unexpectedly long, I've set up a tentative git repo in below as already mentioned in another post: git://github.com/tiwai/sound.git Also, the corresponding alsa-driver-build tree is found in git://github.com/tiwai/alsa-driver-build.git The snapshot tarballs are found in: ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-snapshot.tar.gz No daily snapshots are found there, and only *.gz is present. thanks, Takashi

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[alsa-devel] [PATCH] ASoC: bf5xx-ad73311: Fix prototype for bf5xx_probe
by Axel Lin 16 Sep '11

16 Sep '11

Fix below build warning: sound/soc/blackfin/bf5xx-ad73311.c: warning: initialization from incompatible pointer type Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/blackfin/bf5xx-ad73311.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 732a247..b94eb7e 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -128,7 +128,7 @@ static int snd_ad73311_configure(void) return 0; } -static int bf5xx_probe(struct platform_device *pdev) +static int bf5xx_probe(struct snd_soc_card *card) { int err; if (gpio_request(GPIO_SE, "AD73311_SE")) { -- 1.7.4.1

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