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[alsa-devel] [PATCH 3/4] pulse: unify destruction of snd_pulse_t
by Lennart Poettering 31 Jul '09

31 Jul '09

--- pulse/pulse.c | 31 ++++++++++++------------------- 1 files changed, 12 insertions(+), 19 deletions(-) diff --git a/pulse/pulse.c b/pulse/pulse.c index dd17384..ae66b0c 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -217,36 +217,29 @@ snd_pulse_t *pulse_new(void) return p; fail: + pulse_free(p); + + return NULL; +} + +void pulse_free(snd_pulse_t * p) +{ if (p->mainloop) pa_threaded_mainloop_stop(p->mainloop); - if (p->context) + if (p->context) { + pa_context_disconnect(p->context); pa_context_unref(p->context); + } if (p->mainloop) pa_threaded_mainloop_free(p->mainloop); - if (p->main_fd >= 0) - close(p->main_fd); - if (p->thread_fd >= 0) close(p->thread_fd); - free(p); - - return NULL; -} - -void pulse_free(snd_pulse_t * p) -{ - pa_threaded_mainloop_stop(p->mainloop); - - pa_context_disconnect(p->context); - pa_context_unref(p->context); - pa_threaded_mainloop_free(p->mainloop); - - close(p->thread_fd); - close(p->main_fd); + if (p->main_fd >= 0) + close(p->main_fd); free(p); } -- 1.6.3.3 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH 2/4] pulse: use PA_CONTEXT_IS_GOOD where applicable
by Lennart Poettering 31 Jul '09

31 Jul '09

PA_CONTEXT_IS_GOOD is a safer way to check whether a context is still valid. This patch also bumps the version requirement of libpulse to 0.9.11. --- configure.in | 2 +- pulse/pulse.c | 5 ++--- 2 files changed, 3 insertions(+), 4 deletions(-) diff --git a/configure.in b/configure.in index 36740e9..f8bc669 100644 --- a/configure.in +++ b/configure.in @@ -30,7 +30,7 @@ AC_ARG_ENABLE([pulseaudio], AS_HELP_STRING([--disable-pulseaudio], [Disable building of pulseaudio plugin])) if test "x$enable_pulseaudio" != "xno"; then - PKG_CHECK_MODULES(pulseaudio, [libpulse >= 0.9.2], [HAVE_PULSE=yes], [HAVE_PULSE=no]) + PKG_CHECK_MODULES(pulseaudio, [libpulse >= 0.9.11], [HAVE_PULSE=yes], [HAVE_PULSE=no]) fi AM_CONDITIONAL(HAVE_PULSE, test x$HAVE_PULSE = xyes) diff --git a/pulse/pulse.c b/pulse/pulse.c index 95d8dde..dd17384 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -38,7 +38,7 @@ int pulse_check_connection(snd_pulse_t * p) state = pa_context_get_state(p->context); - if (state != PA_CONTEXT_READY) + if (!PA_CONTEXT_IS_GOOD(state)) return -EIO; return 0; @@ -127,8 +127,7 @@ int pulse_wait_stream_state(snd_pulse_t * p, pa_stream * stream, if (state == target) break; - if (state == PA_STREAM_FAILED || - state == PA_STREAM_TERMINATED) + if (!PA_STREAM_IS_GOOD(state)) return -EIO; pa_threaded_mainloop_wait(p->mainloop); -- 1.6.3.3 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[PATCH] pulse: get rid of a number of assert()s
by Lennart Poettering 31 Jul '09

31 Jul '09

Instead of hitting an assert when any of the plugin functions is called in an invalid context we should return a clean error to make sure programs are not unnecessarily aborted. This should fix issues such as http://pulseaudio.org/ticket/595 --- pulse/ctl_pulse.c | 35 ++++++++++++++++++----- pulse/pcm_pulse.c | 79 +++++++++++++++++++++++++++++++++++++++++------------ pulse/pulse.c | 34 ++++++++++++++++------- 3 files changed, 112 insertions(+), 36 deletions(-) diff --git a/pulse/ctl_pulse.c b/pulse/ctl_pulse.c index c6cf9e2..2caa29b 100644 --- a/pulse/ctl_pulse.c +++ b/pulse/ctl_pulse.c @@ -125,8 +125,9 @@ static void event_cb(pa_context * c, pa_subscription_event_type_t t, pa_operation *o; assert(ctl); - assert(ctl->p); - assert(ctl->p->context); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return; o = pa_context_get_sink_info_by_name(ctl->p->context, ctl->sink, sink_info_cb, ctl); @@ -148,8 +149,9 @@ static int pulse_update_volume(snd_ctl_pulse_t * ctl) pa_operation *o; assert(ctl); - assert(ctl->p); - assert(ctl->p->context); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; o = pa_context_get_sink_info_by_name(ctl->p->context, ctl->sink, sink_info_cb, ctl); @@ -203,6 +205,9 @@ static int pulse_elem_list(snd_ctl_ext_t * ext, unsigned int offset, assert(ctl); + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; + snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -260,7 +265,9 @@ static int pulse_get_attribute(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, return -EINVAL; assert(ctl); - assert(ctl->p); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -311,7 +318,9 @@ static int pulse_read_integer(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, pa_cvolume *vol = NULL; assert(ctl); - assert(ctl->p); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -361,7 +370,9 @@ static int pulse_write_integer(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, pa_cvolume *vol = NULL; assert(ctl); - assert(ctl->p && ctl->p->context); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -465,6 +476,9 @@ static void pulse_subscribe_events(snd_ctl_ext_t * ext, int subscribe) assert(ctl); + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return; + pa_threaded_mainloop_lock(ctl->p->mainloop); ctl->subscribed = !!(subscribe & SND_CTL_EVENT_MASK_VALUE); @@ -481,6 +495,9 @@ static int pulse_read_event(snd_ctl_ext_t * ext, snd_ctl_elem_id_t * id, assert(ctl); + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; + pa_threaded_mainloop_lock(ctl->p->mainloop); if (!ctl->updated || !ctl->subscribed) @@ -526,7 +543,9 @@ static int pulse_ctl_poll_revents(snd_ctl_ext_t * ext, struct pollfd *pfd, int err = 0; assert(ctl); - assert(ctl->p); + + if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index c276839..24347f9 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -106,6 +106,9 @@ static int update_active(snd_pcm_pulse_t *pcm) { assert(pcm); + if (!pcm->p) + return -EBADFD; + ret = check_active(pcm); if (ret < 0) return ret; @@ -125,7 +128,9 @@ static int pulse_start(snd_pcm_ioplug_t * io) int err = 0, err_o = 0, err_u = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -174,7 +179,9 @@ static int pulse_stop(snd_pcm_ioplug_t * io) int err = 0, err_o = 0, err_u = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -224,7 +231,9 @@ static int pulse_drain(snd_pcm_ioplug_t * io) int err = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -259,7 +268,9 @@ static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) snd_pcm_sframes_t ret = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; if (io->state == SND_PCM_STATE_XRUN) return -EPIPE; @@ -269,7 +280,10 @@ static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + ret = -EBADFD; + goto finish; + } ret = pulse_check_connection(pcm->p); if (ret < 0) @@ -305,11 +319,16 @@ static int pulse_delay(snd_pcm_ioplug_t * io, snd_pcm_sframes_t * delayp) pa_usec_t lat = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + err = -EBADFD; + goto finish; + } for (;;) { err = pulse_check_connection(pcm->p); @@ -354,11 +373,16 @@ static snd_pcm_sframes_t pulse_write(snd_pcm_ioplug_t * io, snd_pcm_sframes_t ret = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + ret = -EBADFD; + goto finish; + } ret = pulse_check_connection(pcm->p); if (ret < 0) @@ -409,11 +433,16 @@ static snd_pcm_sframes_t pulse_read(snd_pcm_ioplug_t * io, snd_pcm_sframes_t ret = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - assert(pcm->stream); + if (!pcm->stream) { + ret = -EBADFD; + goto finish; + } ret = pulse_check_connection(pcm->p); if (ret < 0) @@ -480,7 +509,9 @@ static void stream_request_cb(pa_stream * p, size_t length, void *userdata) snd_pcm_pulse_t *pcm = userdata; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return; update_active(pcm); } @@ -490,7 +521,9 @@ static void stream_underrun_cb(pa_stream * p, void *userdata) snd_pcm_pulse_t *pcm = userdata; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return; pcm->underrun = 1; } @@ -499,7 +532,9 @@ static void stream_latency_cb(pa_stream *p, void *userdata) { snd_pcm_pulse_t *pcm = userdata; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return; pa_threaded_mainloop_signal(pcm->p->mainloop, 0); } @@ -512,7 +547,9 @@ static int pulse_pcm_poll_revents(snd_pcm_ioplug_t * io, snd_pcm_pulse_t *pcm = io->private_data; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -541,7 +578,9 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) unsigned c, d; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -645,7 +684,9 @@ static int pulse_hw_params(snd_pcm_ioplug_t * io, int err = 0; assert(pcm); - assert(pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -745,7 +786,9 @@ static int pulse_pause(snd_pcm_ioplug_t * io, int enable) int err = 0; assert (pcm); - assert (pcm->p); + + if (!pcm->p) + return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); diff --git a/pulse/pulse.c b/pulse/pulse.c index 3940238..95d8dde 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -32,8 +32,9 @@ int pulse_check_connection(snd_pulse_t * p) pa_context_state_t state; assert(p); - assert(p->context); - assert(p->mainloop); + + if (!p->context || !p->mainloop) + return -EBADFD; state = pa_context_get_state(p->context); @@ -77,8 +78,12 @@ int pulse_wait_operation(snd_pulse_t * p, pa_operation * o) { assert(p); assert(o); - assert(p->state == PULSE_STATE_READY); - assert(p->mainloop); + + if (p->state != PULSE_STATE_READY) + return -EBADFD; + + if (!p->mainloop) + return -EBADFD; for (;;) { int err; @@ -103,8 +108,12 @@ int pulse_wait_stream_state(snd_pulse_t * p, pa_stream * stream, assert(p); assert(stream); - assert(p->state == PULSE_STATE_READY); - assert(p->mainloop); + + if (p->state != PULSE_STATE_READY) + return -EBADFD; + + if (!p->mainloop) + return -EBADFD; for (;;) { int err; @@ -197,7 +206,9 @@ snd_pulse_t *pulse_new(void) p->context = pa_context_new(pa_threaded_mainloop_get_api(p->mainloop), buf); - assert(p->context); + + if (!p->context) + goto fail; pa_context_set_state_callback(p->context, context_state_cb, p); @@ -246,9 +257,12 @@ int pulse_connect(snd_pulse_t * p, const char *server) int err; assert(p); - assert(p->context); - assert(p->mainloop); - assert(p->state == PULSE_STATE_INIT); + + if (!p->context || !p->mainloop) + return -EBADFD; + + if (p->state != PULSE_STATE_INIT) + return -EBADFD; pa_threaded_mainloop_lock(p->mainloop); -- 1.6.3.3 --ikeVEW9yuYc//A+q--

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[alsa-devel] DMA HW bug in MX2x platforms?
by javier Martin 31 Jul '09

31 Jul '09

I am currently working on an ASoC driver for mx2x platforms. I am porting an old working ASoC driver for 2.6.22 + Freescale BSP kernel to current 2.6.30 kernel. Among other things, I had to translate old Freescale's DMA api to the one provided in 2.6.30 for i.mx2x platforms. Here I will describe the problem that I am having: in "aplay" I am using (buffer_size/period_size=4), each (2*period_size) samples I have some erroneous ones that should come (buffer_size) samples later, and this relation is valid for several values with (buffer_size/period_size=4). I have a trace log with all details for DMA transfers that I have compared for both 2.6.22 and new 2.6.30 and they are identical. Some people has also told me that there could be a problem with DMA in mx2x platforms. Do you have knowledge of this kind of problem? Is there a new DMA version for mx2x in any git repository? -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

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[alsa-devel] [PATCH 5/5] sound/aoa: Add kmalloc NULL tests
by Julia Lawall 31 Jul '09

31 Jul '09

From: Julia Lawall <julia(a)diku.dk> Check that the result of kzalloc is not NULL before a dereference. The semantic match that finds this problem is as follows: (http://www.emn.fr/x-info/coccinelle/) // <smpl> @@ expression *x; identifier f; constant char *C; @@ x = $kmalloc\|kcalloc\|kzalloc$(...); ... when != x == NULL when != x != NULL when != (x || ...) ( kfree(x) | f(...,C,...,x,...) | *f(...,x,...) | *x->f ) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/aoa/core/gpio-pmf.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c index 5ca2220..b4439ce 100644 --- a/sound/aoa/core/gpio-pmf.c +++ b/sound/aoa/core/gpio-pmf.c @@ -182,6 +182,12 @@ static int pmf_set_notify(struct gpio_runtime *rt, if (!old && notify) { irq_client = kzalloc(sizeof(struct pmf_irq_client), GFP_KERNEL); + if (!irq_client) { + err = -ENOMEM; + printk(KERN_ERR "snd-aoa: gpio layer failed to" + " register %s irq (%d)\n", name, err); + goto out_unlock; + } irq_client->data = notif; irq_client->handler = pmf_handle_notify_irq; irq_client->owner = THIS_MODULE;

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[alsa-devel] Huawei K3715 UMTS stick and alsa
by Artem Makhutov 31 Jul '09

31 Jul '09

Hello, i am wondering if it is possible to write a quirk in usbquirks.h for a Huawei K3715 UMTS stick to make the audio interfaces available in linux. This two interfaces are used for audio (I have sniffed in Windows): bEndpointAddress 0x83 EP 3 IN bEndpointAddress 0x02 EP 2 OUT I have attached a lsusb -vv. The payload of this interfaces is 320 bytes. Here is an example payload data. What kind of audio could this be? 93 06 1D 01 BD F6 17 EA 94 E3 DA DB 1C E2 E1 FA 23 0C 13 0B FD 08 78 11 51 17 02 13 C4 04 F5 F2 CA EC 03 F9 5F 09 1E 15 4D 1B B2 0A B7 F5 3D EE A4 E2 62 E5 D2 FA 23 07 26 05 B2 09 6A 20 D8 22 97 0B 77 F7 3F EE C0 E8 48 E1 11 DB 55 E7 E9 05 AC 24 57 28 FF 1B D5 0E 45 07 DC 06 26 07 FE FC 17 ED C8 E0 3F D4 3D E0 8B 03 75 13 0F 04 D3 F2 FF F4 31 08 A2 09 A6 01 FF FA 14 01 49 10 D3 1A 63 25 27 1F C5 00 4C E8 15 E0 0A D8 E3 DE 53 F0 D5 F2 BD F9 F5 10 8B 26 F1 20 55 0C 3E 00 56 FA C8 F5 E8 F0 39 ED F5 F1 F7 00 66 13 7F 1A 67 0F 70 01 D1 FC 71 01 63 09 61 05 0B F7 7F EB 6E E8 CA F6 D2 07 BF 0B 75 FC 70 F0 10 F5 44 06 ED 08 0B 05 92 FE 2F FD 95 0B 40 16 05 22 A7 16 A5 FA 08 E9 52 E5 B5 D9 1F E3 F7 FB 9C 02 6E 03 EF 09 76 1C E1 19 50 06 02 F6 20 F0 8A F0 99 EE 7E E6 42 EB EF 01 AA 1C FF 20 73 14 F0 06 D4 01 41 05 F4 07 EF 00 2E EF D8 E4 12 E2 55 F5 DE 03 C3 09 EF 04 A4 FB 93 F9 77 FE F9 03 CC 07 14 07 BF 0B CF 16 26 1A 6B 16 CC 05 CC F7 FA ED D9 E1 DA D5 Thanks, Artem

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[alsa-devel] [PATCH] Enable HP output on nVidia Corporation MCP79 HDA (rev b1) (Macbook Pro 5, 5)
by Stelian Pop 30 Jul '09

30 Jul '09

Hi, The patch below, to be applied on the latest sound-unstable-2.6.git, enables headphones output on my MacBookPro 5,5, together with the automuting feature. Here is the exact soundcard id: Vendor Id: 0x10134206 Subsystem Id: 0x106b4d00 Revision Id: 0x100301 Thanks for the previous efforts to make this work ! Stelian. diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index f552738..09be633 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -832,6 +832,12 @@ static void cs_automute(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, hp_present ? 0 : PIN_OUT); } + if (spec->board_config == CS420X_MBP55) { + if (hp_present) + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0x2); + else + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0x8); + } } static void cs_automic(struct hda_codec *codec) @@ -1133,10 +1139,10 @@ static int patch_cs420x(struct hda_codec *codec) switch (spec->board_config) { case CS420X_MBP55: - /* GPIO3 = EAPD? */ - spec->gpio_mask = 0x08; - spec->gpio_dir = 0x08; - spec->gpio_data = 0x08; + /* GPIO1 = headphones */ + /* GPIO3 = speakers */ + spec->gpio_mask = 0x0a; + spec->gpio_dir = 0x0a; break; } -- Stelian Pop <stelian(a)popies.net>

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[PATCH] [ALSA] hda: Correct EAPD for Dell Inspiron 1525
by Chengu Wang 30 Jul '09

30 Jul '09

The commit 24918b61b55c21e09a3e07cd82e1b3a8154782dc statically changes the model from dell-bios to dell-3stack to solve the sound decreasing regression (http://lkml.org/lkml/2008/9/12/203), however it leads to another problem that the 2nd headphone jack doesn't work (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3987). So I think the commit 249**2dc is just a workaround. I would like to give a true solution here. The datasheet for STAC9228 says, GPIO2 is the same pin as VOL DOWN, and the EAPD pin is GPIO0. This is why the sound decreases if we set EAPD as GPIO2. This patch changes EAPD to GPIO0 to solve the problem. Signed-off-by: Chengu Wang <wangchengu(a)gmail.com> --- sound/pci/hda/patch_sigmatel.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8c48f86..3857add 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2264,7 +2264,7 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0227, "Dell Vostro 1400 ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022e, "Dell ", STAC_DELL_BIOS), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell Inspiron 1525", STAC_DELL_3ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell Inspiron 1525", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0242, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0243, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ff, "Dell ", STAC_DELL_BIOS), @@ -5659,6 +5659,13 @@ static int patch_stac927x(struct hda_codec *codec) /* GPIO2 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x04; spec->gpio_data = 0x04; + switch (codec->subsystem_id) { + case 0x1028022f: + /* correct EAPD to be GPIO0 */ + spec->eapd_mask = spec->gpio_mask = 0x01; + spec->gpio_dir = spec->gpio_data = 0x01; + break; + }; spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; -- 1.6.0.4 --001636457ed0b1bcc70470129999 Content-Type: text/x-patch; charset=US-ASCII; name="0001--ALSA-hda-Correct-EAPD-for-Dell-Inspiron-1525.patch" Content-Disposition: attachment; filename="0001--ALSA-hda-Correct-EAPD-for-Dell-Inspiron-1525.patch" Content-Transfer-Encoding: base64 X-Attachment-Id: f_fxu8s8l10 RnJvbSAzZGZiNTQ2NjZkMDk0MGJmZjQ3NGJlOWRiNzkxNjAzZjgxYzQ2MzAzIE1vbiBTZXAgMTcg MDA6MDA6MDAgMjAwMQpGcm9tOiBDaGVuZ3UgV2FuZyA8d2FuZ2NoZW5ndUBnbWFpbC5jb20+CkRh dGU6IFRodSwgMzAgSnVsIDIwMDkgMTk6NDM6NTUgKzA4MDAKU3ViamVjdDogW1BBVENIXSBbQUxT QV0gaGRhOiBDb3JyZWN0IEVBUEQgZm9yIERlbGwgSW5zcGlyb24gMTUyNQoKVGhlIGNvbW1pdCAy NDkxOGI2MWI1NWMyMWUwOWEzZTA3Y2Q4MmUxYjNhODE1NDc4MmRjIHN0YXRpY2FsbHkgY2hhbmdl cwp0aGUgbW9kZWwgZnJvbSBkZWxsLWJpb3MgdG8gZGVsbC0zc3RhY2sgdG8gc29sdmUgdGhlIHNv dW5kIGRlY3JlYXNpbmcKcmVncmVzc2lvbiAoaHR0cDovL2xrbWwub3JnL2xrbWwvMjAwOC85LzEy LzIwMyksIGhvd2V2ZXIgaXQgbGVhZHMgdG8gYW5vdGhlcgpwcm9ibGVtIHRoYXQgdGhlIDJuZCBo ZWFkcGhvbmUgamFjayBkb2Vzbid0IHdvcmsKKGh0dHBzOi8vYnVndHJhY2suYWxzYS1wcm9qZWN0 Lm9yZy9hbHNhLWJ1Zy92aWV3LnBocD9pZD0zOTg3KS4gU28gSSB0aGluawp0aGUgY29tbWl0IDI0 OSoqMmRjIGlzIGp1c3QgYSB3b3JrYXJvdW5kLiBJIHdvdWxkIGxpa2UgdG8gZ2l2ZSBhIHRydWUg c29sdXRpb24KaGVyZS4KClRoZSBkYXRhc2hlZXQgZm9yIFNUQUM5MjI4IHNheXMsIEdQSU8yIGlz IHRoZSBzYW1lIHBpbiBhcyBWT0wgRE9XTiwgYW5kCnRoZSBFQVBEIHBpbiBpcyBHUElPMC4gVGhp cyBpcyB3aHkgdGhlIHNvdW5kIGRlY3JlYXNlcyBpZiB3ZSBzZXQgRUFQRCBhcwpHUElPMi4gVGhp cyBwYXRjaCBjaGFuZ2VzIEVBUEQgdG8gR1BJTzAgdG8gc29sdmUgdGhlIHByb2JsZW0uCgpTaWdu ZWQtb2ZmLWJ5OiBDaGVuZ3UgV2FuZyA8d2FuZ2NoZW5ndUBnbWFpbC5jb20+Ci0tLQogc291bmQv cGNpL2hkYS9wYXRjaF9zaWdtYXRlbC5jIHwgICAgOSArKysrKysrKy0KIDEgZmlsZXMgY2hhbmdl ZCwgOCBpbnNlcnRpb25zKCspLCAxIGRlbGV0aW9ucygtKQoKZGlmZiAtLWdpdCBhL3NvdW5kL3Bj aS9oZGEvcGF0Y2hfc2lnbWF0ZWwuYyBiL3NvdW5kL3BjaS9oZGEvcGF0Y2hfc2lnbWF0ZWwuYwpp bmRleCA4YzQ4Zjg2Li4zODU3YWRkIDEwMDY0NAotLS0gYS9zb3VuZC9wY2kvaGRhL3BhdGNoX3Np Z21hdGVsLmMKKysrIGIvc291bmQvcGNpL2hkYS9wYXRjaF9zaWdtYXRlbC5jCkBAIC0yMjY0LDcg KzIyNjQsNyBAQCBzdGF0aWMgc3RydWN0IHNuZF9wY2lfcXVpcmsgc3RhYzkyN3hfY2ZnX3RibFtd ID0gewogCVNORF9QQ0lfUVVJUksoUENJX1ZFTkRPUl9JRF9ERUxMLCAgMHgwMWYzLCAiRGVsbCBJ bnNwaXJvbiAxNDIwIiwgU1RBQ19ERUxMX0JJT1MpLAogCVNORF9QQ0lfUVVJUksoUENJX1ZFTkRP Ul9JRF9ERUxMLCAgMHgwMjI3LCAiRGVsbCBWb3N0cm8gMTQwMCAgIiwgU1RBQ19ERUxMX0JJT1Mp LAogCVNORF9QQ0lfUVVJUksoUENJX1ZFTkRPUl9JRF9ERUxMLCAgMHgwMjJlLCAiRGVsbCAgICAg IiwgU1RBQ19ERUxMX0JJT1MpLAotCVNORF9QQ0lfUVVJUksoUENJX1ZFTkRPUl9JRF9ERUxMLCAg MHgwMjJmLCAiRGVsbCBJbnNwaXJvbiAxNTI1IiwgU1RBQ19ERUxMXzNTVCksCisJU05EX1BDSV9R VUlSSyhQQ0lfVkVORE9SX0lEX0RFTEwsICAweDAyMmYsICJEZWxsIEluc3Bpcm9uIDE1MjUiLCBT VEFDX0RFTExfQklPUyksCiAJU05EX1BDSV9RVUlSSyhQQ0lfVkVORE9SX0lEX0RFTEwsICAweDAy NDIsICJEZWxsICAgICAiLCBTVEFDX0RFTExfQklPUyksCiAJU05EX1BDSV9RVUlSSyhQQ0lfVkVO RE9SX0lEX0RFTEwsICAweDAyNDMsICJEZWxsICAgICAiLCBTVEFDX0RFTExfQklPUyksCiAJU05E X1BDSV9RVUlSSyhQQ0lfVkVORE9SX0lEX0RFTEwsICAweDAyZmYsICJEZWxsICAgICAiLCBTVEFD X0RFTExfQklPUyksCkBAIC01NjU5LDYgKzU2NTksMTMgQEAgc3RhdGljIGludCBwYXRjaF9zdGFj OTI3eChzdHJ1Y3QgaGRhX2NvZGVjICpjb2RlYykKIAkJLyogR1BJTzIgSGlnaCA9IEVuYWJsZSBF QVBEICovCiAJCXNwZWMtPmVhcGRfbWFzayA9IHNwZWMtPmdwaW9fbWFzayA9IHNwZWMtPmdwaW9f ZGlyID0gMHgwNDsKIAkJc3BlYy0+Z3Bpb19kYXRhID0gMHgwNDsKKwkJc3dpdGNoIChjb2RlYy0+ c3Vic3lzdGVtX2lkKSB7CisJCWNhc2UgMHgxMDI4MDIyZjoKKwkJCS8qIGNvcnJlY3QgRUFQRCB0 byBiZSBHUElPMCAqLworCQkJc3BlYy0+ZWFwZF9tYXNrID0gc3BlYy0+Z3Bpb19tYXNrID0gMHgw MTsKKwkJCXNwZWMtPmdwaW9fZGlyID0gc3BlYy0+Z3Bpb19kYXRhID0gMHgwMTsKKwkJCWJyZWFr OworCQl9OwogCQlzcGVjLT5kbWljX25pZHMgPSBzdGFjOTI3eF9kbWljX25pZHM7CiAJCXNwZWMt Pm51bV9kbWljcyA9IFNUQUM5MjdYX05VTV9ETUlDUzsKIAotLSAKMS42LjAuNAoK --001636457ed0b1bcc70470129999--

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[alsa-devel] [PATCH] ALSA: Write outside array bounds
by Roel Kluin 30 Jul '09

30 Jul '09

e->sad[] is declared with size ELD_MAX_SAD (16), but the guard allows the range 0-31 Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found with Parfait, http://research.sun.com/projects/parfait/ diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index fcad5ec..ec04e58 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -539,7 +539,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry, sname++; n = 10 * n + name[4] - '0'; } - if (n < 0 || n > 31) /* double the CEA limit */ + if (n < 0 || n > ELD_MAX_SAD) continue; if (!strcmp(sname, "_coding_type")) e->sad[n].format = val;

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[alsa-devel] [PATCH] hda: add HP automute support to Intel ALC889/ALC889A models
by Wu Fengguang 30 Jul '09

30 Jul '09

It auto mutes all 8-channel outputs at rear panel when the front panel headphone is connected. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_realtek.c | 42 ++++++++++++++++++++++++-------- 1 file changed, 32 insertions(+), 10 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_realtek.c +++ sound-2.6/sound/pci/hda/patch_realtek.c @@ -1436,6 +1436,25 @@ static void alc_automute_amp_unsol_event alc_automute_amp(codec); } +static void alc889_automute_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x17; + spec->autocfg.speaker_pins[3] = 0x19; + spec->autocfg.speaker_pins[4] = 0x1a; + alc_automute_amp(codec); +} + +static void alc889_intel_init_hook(struct hda_codec *codec) +{ + alc889_coef_init(codec); + alc889_automute_init(codec); +} + static void alc888_fujitsu_xa3530_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -6994,6 +7013,11 @@ static struct hda_verb alc889_eapd_verbs { } }; +static struct hda_verb alc262_hippo_unsol_verbs[] = { + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} +}; static struct hda_verb alc885_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ @@ -7019,7 +7043,7 @@ static struct hda_verb alc885_init_verbs {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* Front HP Pin: output 0 (0x0c) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front Pin: output 0 (0x0c) */ @@ -8861,7 +8885,8 @@ static struct alc_config_preset alc882_p }, [ALC889A_INTEL] = { .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer }, - .init_verbs = { alc885_init_verbs, alc885_init_input_verbs }, + .init_verbs = { alc885_init_verbs, alc885_init_input_verbs, + alc262_hippo_unsol_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), @@ -8873,12 +8898,14 @@ static struct alc_config_preset alc882_p .channel_mode = alc889_8ch_intel_modes, .capsrc_nids = alc889_capsrc_nids, .input_mux = &alc889_capture_source, + .init_hook = alc889_automute_init, + .unsol_event = alc_automute_amp_unsol_event, .need_dac_fix = 1, }, [ALC889_INTEL] = { .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer }, .init_verbs = { alc885_init_verbs, alc889_init_input_verbs, - alc889_eapd_verbs }, + alc889_eapd_verbs, alc262_hippo_unsol_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), @@ -8890,7 +8917,8 @@ static struct alc_config_preset alc882_p .channel_mode = alc889_8ch_intel_modes, .capsrc_nids = alc889_capsrc_nids, .input_mux = &alc889_capture_source, - .init_hook = alc889_coef_init, + .init_hook = alc889_intel_init_hook, + .unsol_event = alc_automute_amp_unsol_event, .need_dac_fix = 1, }, [ALC883_6ST_DIG] = { @@ -10123,12 +10151,6 @@ static struct hda_verb alc262_eapd_verbs { } }; -static struct hda_verb alc262_hippo_unsol_verbs[] = { - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {} -}; - static struct hda_verb alc262_hippo1_unsol_verbs[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},

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[alsa-devel] [PATCH] hda: add 2-channel mode to Intel ALC889/ALC889A models
by Wu Fengguang 30 Jul '09

30 Jul '09

This 2-channel mode is useful in that it will broadcast a 2-channel audio stream to all front/side/... ports. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_realtek.c | 44 ++++++++++++++++++-------------- 1 file changed, 25 insertions(+), 19 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_realtek.c +++ sound-2.6/sound/pci/hda/patch_realtek.c @@ -6682,18 +6682,27 @@ static struct hda_channel_mode alc883_3S }; /* + * 2ch mode + */ +static struct hda_verb alc889_ch2_intel_init[] = { + { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* * 6ch mode */ static struct hda_verb alc889_ch6_intel_init[] = { - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { } /* end */ }; @@ -6702,21 +6711,18 @@ static struct hda_verb alc889_ch6_intel_ * 8ch mode */ static struct hda_verb alc889_ch8_intel_init[] = { - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 }, { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 }, { } /* end */ }; -static struct hda_channel_mode alc889_8ch_intel_modes[2] = { +static struct hda_channel_mode alc889_8ch_intel_modes[3] = { + { 2, alc889_ch2_intel_init }, { 6, alc889_ch6_intel_init }, { 8, alc889_ch8_intel_init }, };

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[alsa-devel] Chat 50 at usb-0000:00:1a.7-4.3 audio issues
by Jason Bell 30 Jul '09

30 Jul '09

G'day ASLA Developers I am hoping that someone might be able to assist me with an audio issue I am having. Recently I have begun using the Dell T3500 precisions as an Video Conferencing machine (using the Access Grid and EVO Software). Information on hardware specs, etc can be found at http://www.alsa-project.org/db/?f=7f33770c89802b0ab3132f21914cc6ca7f6626 37 Unfortunately, I have issues when using either a clearone Chat50 or Chat150 (these are essentially speaker phone devices [See http://www.clearone.com/products/product.php?cat=9&prod=94 for device description]) with the speaker audio being distorted and static, cracks and popping type sounds being introduced. The microphone audio quality is fine. These issues were seen using Fedora 10 and Fedora 11. I tried upgrading to the latest ALSA drivers and "testing" kernels without any success (on a previous Fedora 10 install). It was found (still in the initial stages of testings) that if I plugged the Chat50 device into a USB hub, rather than directly into the Machine, the audio issues disappeared (at least for some of the software). It should also be noted that I have a number of Dell T3400 series machines that don't experience these issues. Additionally is should be mentioned that the T3400 uses the ICH9 chipset family whereas the T3500 uses the ICH10 chipset family which includes the USB controller. Just wondering if there is anything I can do to resolve this issue? Many thanks in advance, Jason. -------------------------------------------- Jason Bell, B.I.T. (Honours) Research Systems Support Officer Information Technology Division CQ University Australia Australian Research Collaboration Service http://www.arcs.org.au/ E-mail : j.bell(a)cqu.edu.au jason.bell(a)arcs.org.au Work : +61 7 4930 9229 Mobile : 0409 630897 Postal : Building 19 Central Queensland University Bruce Highway Rockhampton, Queensland, Australia, 4702 -------------------------------------------- Patience is a virtue. But if I wanted Patience, I would have become a Doctor. --------------------------------------------

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[alsa-devel] [PATCH] Fix checkpatch issues and typos of ad1938 codec and bf5xx-tdm dai
by Barry Song 29 Jul '09

29 Jul '09

1. fix "line over 80 characters" checkpatch warnings 2. ‘DMA_nnBIT_MASK’ is deprecated, use DMA_BIT_MASK instead 3. fix typos Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/bf5xx-tdm-pcm.c | 4 +- sound/soc/blackfin/bf5xx-tdm-pcm.h | 2 +- sound/soc/blackfin/bf5xx-tdm.h | 2 +- sound/soc/codecs/ad1938.c | 50 +++++++++++++++++++++++------------ 4 files changed, 37 insertions(+), 21 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index af7fa8d..ccb5e82 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -276,7 +276,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) sport_done(sport_handle); } -static u64 bf5xx_pcm_dmamask = DMA_32BIT_MASK; +static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) @@ -286,7 +286,7 @@ static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->dma_mask) card->dev->dma_mask = &bf5xx_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_32BIT_MASK; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (dai->playback.channels_min) { ret = bf5xx_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.h b/sound/soc/blackfin/bf5xx-tdm-pcm.h index 9e8473d..ddc5047 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.h +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.h @@ -1,5 +1,5 @@ /* - * linux/sound/arm/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin + * sound/soc/blackfin/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin * * Copyright 2009 Analog Device Inc. * diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h index e5157cd..618ec3d 100644 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -1,5 +1,5 @@ /* - * linux/sound/arm/bf5xx-i2s.h + * sound/soc/blackfin/bf5xx-tdm.h * * 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 diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index ffc97c9..aafda74 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -72,14 +72,20 @@ static const struct snd_kcontrol_new ad1938_snd_controls[] = { AD1938_DAC_R4_VOL, 0, 0xFF, 1), /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD1938_ADC_CTRL0, AD1938_ADCL1_MUTE, AD1938_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD1938_ADC_CTRL0, AD1938_ADCL2_MUTE, AD1938_ADCR2_MUTE, 1, 1), + SOC_DOUBLE("ADC1 Switch", AD1938_ADC_CTRL0, AD1938_ADCL1_MUTE, + AD1938_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD1938_ADC_CTRL0, AD1938_ADCL2_MUTE, + AD1938_ADCR2_MUTE, 1, 1), /* DAC switch control */ - SOC_DOUBLE("DAC1 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL1_MUTE, AD1938_DACR1_MUTE, 1, 1), - SOC_DOUBLE("DAC2 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL2_MUTE, AD1938_DACR2_MUTE, 1, 1), - SOC_DOUBLE("DAC3 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL3_MUTE, AD1938_DACR3_MUTE, 1, 1), - SOC_DOUBLE("DAC4 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL4_MUTE, AD1938_DACR4_MUTE, 1, 1), + SOC_DOUBLE("DAC1 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL1_MUTE, + AD1938_DACR1_MUTE, 1, 1), + SOC_DOUBLE("DAC2 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL2_MUTE, + AD1938_DACR2_MUTE, 1, 1), + SOC_DOUBLE("DAC3 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL3_MUTE, + AD1938_DACR3_MUTE, 1, 1), + SOC_DOUBLE("DAC4 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL4_MUTE, + AD1938_DACR4_MUTE, 1, 1), /* ADC high-pass filter */ SOC_SINGLE("ADC High Pass Filter Switch", AD1938_ADC_CTRL0, @@ -110,7 +116,8 @@ static int ad1938_mute(struct snd_soc_dai *dai, int mute) int reg; reg = codec->read(codec, AD1938_DAC_CTRL2); - reg = (mute > 0) ? reg | AD1938_DAC_MASTER_MUTE : reg & (~AD1938_DAC_MASTER_MUTE); + reg = (mute > 0) ? reg | AD1938_DAC_MASTER_MUTE : reg & + (~AD1938_DAC_MASTER_MUTE); codec->write(codec, AD1938_DAC_CTRL2, reg); return 0; @@ -119,7 +126,8 @@ static int ad1938_mute(struct snd_soc_dai *dai, int mute) static inline int ad1938_pll_powerctrl(struct snd_soc_codec *codec, int cmd) { int reg = codec->read(codec, AD1938_PLL_CLK_CTRL0); - reg = (cmd > 0) ? reg & (~AD1938_PLL_POWERDOWN) : reg | AD1938_PLL_POWERDOWN; + reg = (cmd > 0) ? reg & (~AD1938_PLL_POWERDOWN) : reg | + AD1938_PLL_POWERDOWN; codec->write(codec, AD1938_PLL_CLK_CTRL0, reg); return 0; @@ -171,8 +179,8 @@ static int ad1938_set_dai_fmt(struct snd_soc_dai *codec_dai, adc_reg = codec->read(codec, AD1938_ADC_CTRL2); dac_reg = codec->read(codec, AD1938_DAC_CTRL1); - /* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S with TDM) - * and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A) + /* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S + * with TDM) and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A) */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -363,7 +371,8 @@ static unsigned int ad1938_read_reg_cache(struct snd_soc_codec *codec, * read from the ad1938 register space */ -static unsigned int ad1938_read_reg(struct snd_soc_codec *codec, unsigned int reg) +static unsigned int ad1938_read_reg(struct snd_soc_codec *codec, + unsigned int reg) { char w_buf[AD1938_SPI_BUFLEN]; char r_buf[AD1938_SPI_BUFLEN]; @@ -502,12 +511,19 @@ static int ad1938_register(struct ad1938_priv *ad1938) ad1938_codec = codec; /* default setting for ad1938 */ - codec->write(codec, AD1938_DAC_CHNL_MUTE, 0x0); /* unmute dac channels */ - codec->write(codec, AD1938_DAC_CTRL2, 0x1A); /* de-emphasis: 48kHz, powedown dac */ - codec->write(codec, AD1938_DAC_CTRL0, 0x21); /* powerdown dac, dac tdm mode */ - codec->write(codec, AD1938_ADC_CTRL0, 0x3); /* high-pass filter enable */ - codec->write(codec, AD1938_ADC_CTRL1, 0x43); /* sata delay=1, adc aux mode */ - codec->write(codec, AD1938_PLL_CLK_CTRL0, 0x9D); /* pll input:mclki/xi */ + + /* unmute dac channels */ + codec->write(codec, AD1938_DAC_CHNL_MUTE, 0x0); + /* de-emphasis: 48kHz, powedown dac */ + codec->write(codec, AD1938_DAC_CTRL2, 0x1A); + /* powerdown dac, dac in tdm mode */ + codec->write(codec, AD1938_DAC_CTRL0, 0x41); + /* high-pass filter enable */ + codec->write(codec, AD1938_ADC_CTRL0, 0x3); + /* sata delay=1, adc aux mode */ + codec->write(codec, AD1938_ADC_CTRL1, 0x43); + /* pll input: mclki/xi */ + codec->write(codec, AD1938_PLL_CLK_CTRL0, 0x9D); codec->write(codec, AD1938_PLL_CLK_CTRL1, 0x04); ad1938_fill_cache(codec); -- 1.5.6.3

2 1

[alsa-devel] how to configure .asoundrc to make arecord use a default device
by Jelle de Jong 29 Jul '09

29 Jul '09

Hello everybody, I am having some issues making arecord use the default device for recording. Could somebody help me get a working .asoundrc configuration? I attached all information in the attachment and also made temporary hyperlink: http://debian.pastebin.com/m52601230 It works with: arecord -v -f S16_LE -r 44100 -D audiodevice0 ~/record0.wav but it does not work with: arecord -v -f S16_LE -r 44100 ~/record0.wav background: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-November/012366.h… Thanks in advance, Best regards, Jelle de Jong

2 6

Re: [alsa-devel] clicking noise with Sigmatel STAC9202
by Christian Steffen 29 Jul '09

29 Jul '09

Here I send the two also-info files as attachments. Initial problem was: Under kernel 2.6.30, Sound is working, but has constant clicking noise in the background, also if the sound is completely muted. First I thought this sound would come from some hdd or whatever, but the device has no mechanical parts. But this is what it sounds like - Irregular clicking noise. This noise immediately starts during boot up as soon as the module is loaded and before X11 has started. I already tried options like "position_fix=1 or 2" and different "model="-options with no success. "position_fix" didn't do anything at all and different model-options have muted the device completely. Kernel 2.6.24 with the same files has no issues. >> Then give as attachments later (and don't forget Cc to alsa-devel ML) >> these two files, not URL. It's important because ML is archived >> while pastebin can be removed at any time. Thanks, Christian Steffen -----Ursprüngliche Nachricht----- Von: Takashi Iwai [mailto:tiwai@suse.de] Gesendet: Mittwoch, 29. Juli 2009 17:32 An: Christian Steffen Cc: 'Takashi Iwai' Betreff: Re: AW: [alsa-devel] clicking noise with Sigmatel STAC9202 At Wed, 29 Jul 2009 17:11:18 +0200, Christian Steffen wrote: > > Thanks, > > >> Well, then at least, the contents of /proc/asound/card0/codec#*, > >> /sys/class/sound/hwC0D*/*_pin_configs files (if present), and the file > >> generated via "alsactl -f somefile store". These are the most > >> important information. > > There is no pinconfig file and there is also no alsactl tool present. > I could obtain the output from the codec stuff: > > http://pastebin.com/m1b81f47d (old kernel 2.6.24, only the codec info) > http://pastebin.com/m5baeca9a (new 2.6.30) > > that's all I could get for the older kernel as I only have access using > telnet. Then give as attachments later (and don't forget Cc to alsa-devel ML) these two files, not URL. It's important because ML is archived while pastebin can be removed at any time. Takashi

1 0

[alsa-devel] backward/forward binary compatibility checker
by Andrey Ponomarenko 29 Jul '09

29 Jul '09

Colleagues, I'm software engineer from Institute for System Programing of Russian Academy of Sciences and we are developing a free lightweight tool for checking backward/forward binary compatibility of shared C/C++ libraries in OS Linux. It checks interface signatures and data type definitions in two library versions (headers and shared objects) and searches differences that may lead to incompatibility according to ABI standards. We have released beta version of this tool and we'd like you to consider its usefulness for your project. The package with the latest version of tool is in the attachment. The wiki-page with the latest release of binary compatibility checker is http://ispras.linux-foundation.org/index.php/ABI_compliance_checker Andrey Ponomarenko <susanin(a)ispras.ru>

1 0

[alsa-devel] Master channel is only mono
by Sune Mølgaard 29 Jul '09

29 Jul '09

As per gnubien's advice on #alsa on irc.freenode.net, I have run alsa-info.sh - the URL is http://www.alsa-project.org/db/?f=fbe93f301ca214007e66c521cd6c41a17a604c79 . My problem is that the master channel for my card is shown to be mono only, which I don't think seems right... Also, sound in general is rather low. My system is a very new Dell Studio laptop - I have loaded snd-hda-intel with mode=dell-m6 as per the kernel documentation. Best regards, Sune Mølgaard

2 2

[alsa-devel] pulseaudio workaround for alsa-driver bug
by jack craig 29 Jul '09

29 Jul '09

Hi alsa-developers, I wanted to report my recent issues with audio on fc11. after an update from fc10, my audio started failing. not the failure to year any audio, but instead, it would start off fine and then tail spin. i am using mplayer to play an internet asf stream to the local audio. the stream processing failure is described in https://bugzilla.redhat.com/show_bug.cgi?id=514213 i was able to get a workaround from the #pulseaudio guys, but! they encouraged me to report my experience and its fix to you to encourage the fix of this snd_pcm_delay() defect. if you get a driver fix for this, i am happy to try it out when its ready. thx, jackc... -- jack craig jackc(a)LinuxLightHouse.com 831-684-1375 (Office) 831-596-6924 (cell) IM: jackcraigaptos (AIM) _________________________________ This email has been ClamScanned ! www.LinuxLightHouse.com

3 3

Re: [alsa-devel] clicking noise with Sigmatel STAC9202
by Takashi Iwai 29 Jul '09

29 Jul '09

At Wed, 29 Jul 2009 16:46:34 +0200, Christian Steffen wrote: > > >>>Is the older kernel as is even with the same user-space stuff? > > Probaly yes, I mean I can start the on-device linux with the newer kernel > and also get clicking noise. > But this way I can't run alsa-info.sh as this system has not all the apps it > needs. > > So I started another ubuntu linux on the device to run alsa-info.sh on the > 2.6.30 kernel. > > I've uploaded the result to: > http://pastebin.com/m5baeca9a Please run it with --no-upload on both kernels, and attach both generated files. > Does this already help? > It's difficult to install apps to the linux which is on the device as it is > very very limited. Well, then at least, the contents of /proc/asound/card0/codec#*, /sys/class/sound/hwC0D*/*_pin_configs files (if present), and the file generated via "alsactl -f somefile store". These are the most important information. Takashi

1 0

[alsa-devel] clicking noise with Sigmatel STAC9202
by Christian Steffen 29 Jul '09

29 Jul '09

Hello to all subscribers, I'm new to this list and will start directly with my problem: I'm running some ubuntu linux on a device called "o2 Joggler", which is a digital picture frame with touchscreen control. Its chipsets are from intel, the cpu is an intel atom, and the other hardware devices are controlled by an Intel System Controller Hub "Poulsbo". I'm running kernel 2.6.30 and the newest alsa version 1.0.20. This is lscpi output: >>00:1b.0 Audio Device: Intel Corporation System Controller Hub (SCH Poulsbo) HD Audio Controller (rev 07) cat proc/asound/cards: >>0 [MID ]: HDA-Intel - HDA Intel MID >> HDA Intel MID at 0xd01c000 irq 16 The Codec is a >>Sigmatel STAC9202 My problem is, that sound is working, but has constant clicking noise in the background, also if the sound is completely muted. First I thought this sound would come from some hdd or whatever, but the device has no mechanical parts. But this is what it sounds like - Irregular clicking noise. This noise immediately starts during boot up as soon as the module is loaded and before X11 has started. I already tried options like "position_fix=1 or 2" and different "model="-options with no success. "position_fix" didn't do anything at all and different model-options have muted the device completely. Has someone any idea to get this working? The device comes with an own (hardly) closed linux which runs a 2.6.24 kernel - here sound is working with no problems at all. If you need further info, let me know. Thanks in advance, best! Christian

2 1

[alsa-devel] [PATCH] ASoC: MPC5200: Changed how appl_ptr is initialized
by John Bonesio 29 Jul '09

29 Jul '09

Hello, I've been working in the ALSA SOC area for the MPC5200. I've noticed that in the routine psc_dma_trigger() in sound/soc/fsl/mpc5200_dma.c there are times when the computed value for s->appl_ptr ends up wrapping negative. This occurs when s->runtime->control->appl_ptr is less than runtime->period_size * runtime->periods It seemed like this would occur at then of the data stream or when the data stream is small (smaller than runtime->period_size * runtime->periods). As far as I can tell, s->appl_ptr is supposed to be the previous value of s->runtime->control->appl_ptr, so the driver can send out the right number of bytes (from previous appl_ptr to current appl_ptr) to avoid pops and clicks at the end of the audio. So it appears the assumption in this code is wrong, that the previous s->runtime->control->appl_ptr is always rumtime->period_size * runtime->periods bytes prior to the current s->runtime->control->appl_ptr. I took out this code and instead added code to intialize s->appl_ptr to 0 in psc_dma_open(). Does this fix seem right? - John The computation for appl_ptr in the trigger() routine made a bad assumption when the stream is at the end of the file and there is not (periods * period_size) bytes remaining in the data. This causes playback issues when the total stream size is less than this amount, as the s->appl_ptr wrapps negative and the data is never played. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index b2eba27..cfe0ea4 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -181,8 +181,6 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) * end of stream and not over running it. */ s->runtime = runtime; - s->appl_ptr = s->runtime->control->appl_ptr - - (runtime->period_size * runtime->periods); /* Fill up the bestcomm bd queue and enable DMA. * This will begin filling the PSC's fifo. @@ -287,6 +285,7 @@ static int psc_dma_open(struct snd_pcm_substream *substream) } s->stream = substream; + s->appl_ptr = 0; return 0; }

2 1

[alsa-devel] [PATCH] aedsp16: Buffer overflow
by Roel Kluin 29 Jul '09

29 Jul '09

DSPVersion is declared as char[3], but the sprintf writes at least 4 bytes including terminating null. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found with Parfait, http://research.sun.com/projects/parfait/ on line 498: static char DSPVersion[CARDVERLEN + 1] __initdata = {0, }; diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c index 3ee9900..35b5912 100644 --- a/sound/oss/aedsp16.c +++ b/sound/oss/aedsp16.c @@ -325,8 +325,9 @@ /* * Size of character arrays that store name and version of sound card */ -#define CARDNAMELEN 15 /* Size of the card's name in chars */ -#define CARDVERLEN 2 /* Size of the card's version in chars */ +#define CARDNAMELEN 15 /* Size of the card's name in chars */ +#define CARDVERLEN 10 /* Size of the card's version in chars */ +#define CARDVERDIGITS 2 /* Number of digits in the version */ #if defined(CONFIG_SC6600) /* @@ -410,7 +411,7 @@ static int soft_cfg __initdata = 0; /* bitmapped config */ static int soft_cfg_mss __initdata = 0; /* bitmapped mss config */ -static int ver[CARDVERLEN] __initdata = {0, 0}; /* DSP Ver: +static int ver[CARDVERDIGITS] __initdata = {0, 0}; /* DSP Ver: hi->ver[0] lo->ver[1] */ #if defined(CONFIG_SC6600) @@ -957,7 +958,7 @@ static int __init aedsp16_dsp_version(int port) * string is finished. */ ver[len++] = ret; - } while (len < CARDVERLEN); + } while (len < CARDVERDIGITS); sprintf(DSPVersion, "%d.%d", ver[0], ver[1]); DBG(("success.\n"));

2 1

[alsa-devel] [PATCH] mpu401.c: Buffer overflow
by Roel Kluin 29 Jul '09

29 Jul '09

mpu_synth_info[m].name is a char[30], and the minimum length of the data written by sprintf is 31 bytes including terminating null. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found with Parfait, http://research.sun.com/projects/parfait/ diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 1b2316f..734b8f9 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -1074,7 +1074,7 @@ int attach_mpu401(struct address_info *hw_config, struct module *owner) sprintf(mpu_synth_info[m].name, "%s (MPU401)", hw_config->name); else sprintf(mpu_synth_info[m].name, - "MPU-401 %d.%d%c Midi interface #%d", + "MPU-401 %d.%d%c MIDI #%d", (int) (devc->version & 0xf0) >> 4, devc->version & 0x0f, revision_char,

2 1

[alsa-devel] my udev rules are breaking my dmixer setup why?
by Jelle de Jong 29 Jul '09

29 Jul '09

Hello everybody, I am trying to use udev to get fixed device names depending on the location of the device on the usb bus. The udev rules seems to be working fine, however i cant get dmix to work with the udev rules using only hw dies work. If i remove the udev rules the dmixer works fine. Can somebody look at the attachment, and maybe do some testing, i would highly appreciate it. I am also on chat.freenode.org and #alsa as tuxcrafter Thanks in advance, Jelle ls -hal /dev/snd/ total 0 drwxr-xr-x 2 root root 320 2008-11-09 15:44 . drwxr-xr-x 12 root root 4.8K 2008-11-09 15:50 .. crw-rw---- 1 root user0 116, 0 2008-11-09 15:44 controlC0 crw-rw---- 1 root user1 116, 32 2008-11-09 15:44 controlC1 crw-rw---- 1 root user2 116, 64 2008-11-09 15:44 controlC2 crw-rw---- 1 root user3 116, 96 2008-11-09 15:44 controlC3 crw-rw---- 1 root user0 116, 24 2008-11-09 15:44 pcmC0D0c crw-rw---- 1 root user0 116, 16 2008-11-09 15:49 pcmC0D0p crw-rw---- 1 root user1 116, 56 2008-11-09 15:44 pcmC1D0c crw-rw---- 1 root user1 116, 48 2008-11-09 15:44 pcmC1D0p crw-rw---- 1 root user2 116, 88 2008-11-09 15:44 pcmC2D0c crw-rw---- 1 root user2 116, 80 2008-11-09 15:44 pcmC2D0p crw-rw---- 1 root user3 116, 120 2008-11-09 15:44 pcmC3D0c crw-rw---- 1 root user3 116, 112 2008-11-09 15:44 pcmC3D0p crw-rw---- 1 root audio 116, 1 2008-11-09 15:44 seq crw-rw---- 1 root audio 116, 33 2008-11-09 15:44 timer ------------------------------------------------------------------------ udevinfo info --attribute-walk --name=/dev/snd/controlC0 ------------------------------------------------------------------------ echo 'SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="controlC[0-9]*", NAME="snd/controlC4" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="controlC[0-9]", NAME="snd/controlC5" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="controlC[0-9]", NAME="snd/controlC6" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="controlC[0-9]", NAME="snd/controlC7" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC4D0c" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC5D0c" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC6D0c" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC7D0c" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC4D0c" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC5D0c" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC6D0c" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC7D0c" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC4D0p" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC5D0p" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC6D0p" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC7D0p"' | tee /etc/udev/rules.d/10-persistent-sound.rules chmod 600 /etc/udev/rules.d/10-persistent-sound.rules ------------------------------------------------------------------------ # without dmixer # this works .... echo 'pcm.softvol { type softvol slave.pcm hw:1 control.name Master control.card 1 } pcm.!default { type plug slave.pcm softvol } ctl.!default { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------ # with dmixer # this does not work echo 'pcm.softvol { type softvol slave.pcm dmix:1 control.name Master control.card 1 } pcm.!default { type plug slave.pcm softvol } ctl.!default { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------ # with dmixer # this does not work echo 'pcm.!default { type plug slave.pcm dmixer } pcm.dmixer { type dmix ipc_key 2048 slave.pcm hw:1 } ctl.dmixer { type hw card 1 } pcm.dsp { type plug slave.pcm dmixer } ctl.mixer { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------

3 13

[alsa-devel] [RFC] [PATCH v2] ASoC: CX20442: push down machine independent line discipline bits
by Janusz Krzysztofik 29 Jul '09

29 Jul '09

This corrected patch adds machine independent line discipline code, prevoiusly exsiting inside my Amstrad Delta ASoC machine dirver, to the Conexant CX20442 codec driver. The code can be used as a standalone line discipline, or as a set of codec specific functions called from machine's line discipline callbacks. Anyway, the line discipline itself must be registered by a machine driver. Applies on top of the followup to my initial driver version: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019757.html Suggested by ASoC manintainer Mark Brown <broonie(a)opensource.wolfsonmicro.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Corrections: sound/soc/codecs/cx20442.c line 300: -static struct tty_ldisc_ops v253_ops = { +struct tty_ldisc_ops v253_ops = { line 420: - dev_err(&codec->dev, "Failed to register codec: %d\n", ret); + dev_err(codec->dev, "Failed to register codec: %d\n", ret); line 425: - dev_err(&codec->dev, "Failed to register DAI: %d\n", ret); + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); Corrected patch: --- linux-2.6.31-rc3/sound/soc/codecs/cx20442.h.orig 2009-07-22 04:56:45.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/codecs/cx20442.h 2009-07-28 21:23:17.000000000 +0200 @@ -15,5 +15,6 @@ extern struct snd_soc_dai cx20442_dai; extern struct snd_soc_codec_device cx20442_codec_dev; +extern struct tty_ldisc_ops v253_ops; #endif --- linux-2.6.31-rc3/sound/soc/codecs/cx20442.c.orig 2009-07-28 20:26:18.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/codecs/cx20442.c 2009-07-29 02:16:05.000000000 +0200 @@ -13,6 +13,8 @@ * option) any later version. */ +#include <linux/tty.h> + #include <sound/core.h> #include <sound/initval.h> #include <sound/soc-dapm.h> @@ -172,8 +174,7 @@ static int cx20442_write(struct snd_soc_ return -EINVAL; /* hw_write and control_data pointers required for talking to the modem - * are expected to be set by the machine driver's line discipline - * initialization code */ + * are expected to be set by the line discipline initialization code */ if (!codec->hw_write || !codec->control_data) return -EIO; @@ -208,6 +209,111 @@ static int cx20442_write(struct snd_soc_ return 0; } + +/* Moved up here as line discipline referres it during initialization */ +static struct snd_soc_codec *cx20442_codec; + + +/* + * Line discpline related code + * + * Any of the callback functions below can be used in two ways: + * 1) registerd by a machine driver as one of line discipline operations, + * 2) called from a machine's provided line discipline callback function + * in case when extra machine specific code must be run as well. + */ + +/* Modem init: echo off, digital speaker off, quiet off, voice mode */ +static const char *v253_init = "ate0m0q0+fclass=8\r"; + +/* Line discipline .open() */ +static int v253_open(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = cx20442_codec; + int ret, len = strlen(v253_init); + + /* Doesn't make sense without write callback */ + if (!tty->ops->write) + return -EINVAL; + + /* Pass the codec structure address for use by other ldisc callbacks */ + tty->disc_data = codec; + + if (tty->ops->write(tty, v253_init, len) != len) { + ret = -EIO; + goto err; + } + /* Actual setup will be performed after the modem responds. */ + return 0; +err: + tty->disc_data = NULL; + return ret; +} + +/* Line discipline .close() */ +static void v253_close(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = tty->disc_data; + + tty->disc_data = NULL; + + if (!codec) + return; + + /* Prevent the codec driver from further accessing the modem */ + codec->hw_write = NULL; + codec->control_data = NULL; + codec->pop_time = 0; +} + +/* Line discipline .hangup() */ +static int v253_hangup(struct tty_struct *tty) +{ + v253_close(tty); + return 0; +} + +/* Line discipline .receive_buf() */ +static void v253_receive(struct tty_struct *tty, + const unsigned char *cp, char *fp, int count) +{ + struct snd_soc_codec *codec = tty->disc_data; + + if (!codec) + return; + + if (!codec->control_data) { + /* First modem response, complete setup procedure */ + + /* Set up codec driver access to modem controls */ + codec->control_data = tty; + codec->hw_write = (hw_write_t)tty->ops->write; + codec->pop_time = 1; + } +} + +/* Line discipline .write_wakeup() */ +static void v253_wakeup(struct tty_struct *tty) +{ +} + +struct tty_ldisc_ops v253_ops = { + .magic = TTY_LDISC_MAGIC, + .name = "cx20442", + .owner = THIS_MODULE, + .open = v253_open, + .close = v253_close, + .hangup = v253_hangup, + .receive_buf = v253_receive, + .write_wakeup = v253_wakeup, +}; +EXPORT_SYMBOL_GPL(v253_ops); + + +/* + * Codec DAI + */ + struct snd_soc_dai cx20442_dai = { .name = "CX20442", .playback = { @@ -227,8 +333,6 @@ struct snd_soc_dai cx20442_dai = { }; EXPORT_SYMBOL_GPL(cx20442_dai); -static struct snd_soc_codec *cx20442_codec; - static int cx20442_codec_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -313,13 +417,13 @@ static int cx20442_register(struct cx204 ret = snd_soc_register_codec(codec); if (ret != 0) { - dev_err(&codec->dev, "Failed to register codec: %d\n", ret); + dev_err(codec->dev, "Failed to register codec: %d\n", ret); goto err; } ret = snd_soc_register_dai(&cx20442_dai); if (ret != 0) { - dev_err(&codec->dev, "Failed to register DAI: %d\n", ret); + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); goto err_codec; }

1 0

[alsa-devel] Strange behaviour with varying frame sizes on read
by Metathronius Galabant 29 Jul '09

29 Jul '09

Hi all, please consider the following: Period size is 80 frames, buffer size is 320 frames. For all 3 tests I have 240 frames ready to read. snd_pcm_avail_update() confirms this. Outcome of my tests: 1) I want to read 80 frames, and get 80 frames. - OK, this is really basic. 2) I want to read 81 frames, and get 80 frames. - FAIL. 3) I want to read 80 frames, and get 80 frames. Then I read 1 frame, and get 1 frame. - HUH? I've both tested snd_pcm_readi() and snd_pcm_mmap_readi() - both have the same semantic. Is there any explanation for this kind of illogical behaviour (I have version 1.0.14, which might be a bit old, but still)? I should be able to read rather arbitrary amounts from a software buffer, don't I? Cheers, M.

1 0

[alsa-devel] Using Alsa for AD1981b on 9263
by Shilpa Kedar Walvekar 29 Jul '09

29 Jul '09

Hi, Addition to my earlier post: We are using AT91SAM9263 evaluation kit with on-board AD1981b codec. We are using Linux kernel version 2.6.27 with patches available on http://linux4sam.com. We found from the forum that we need to use ALSA for playing and recording audio using this codec. So we downloaded Alsa-lib-1.0.20 Alsa-oss-1.0.20 Alsa-utils-1.0.20 We installed above using: ./configure && make && make install We then downloaded alsa-driver-1.0.20 and installed it using below: ./configure --with-cards=atmel-ac97c --with-kernel=../../linux-2.6.27 --with-build=../../linux-2.6.27 && make ARCH=arm && make install-modules Driver was compiled and installed successfully. We found following: # cat /proc/asound/cards 0 [AC97 ]: ac97c - Atmel AC97 Atmel AC97 Controller at 0xfffa0000, irq 18 # lsmod Module Size Used by Not tainted snd_at91_ac97 8420 0 snd_ac97_codec 99904 1 snd_at91_ac97 snd_soc_atmel_pcm 4484 0 snd_soc_core 30852 1 snd_soc_atmel_pcm snd_pcm 60772 4 snd_at91_ac97,snd_ac97_codec,snd_soc_atmel_pcm,s nd_soc_core snd_page_alloc 6056 1 snd_pcm snd_timer 19588 1 snd_pcm snd 47700 5 snd_at91_ac97,snd_ac97_codec,snd_soc_core,snd_pc m,snd_timer soundcore 6052 1 snd ac97_bus 1568 1 snd_ac97_codec # cat /home/.asoundrc pcm.!default { type hw card 0 } ctl.!default { type hw card 0 } We have following devices in /dev/snd # cat /dev/snd/ /dev/snd/controlC0 /dev/snd/controlC3 /dev/snd/hwC0D2 /dev/snd/timer /dev/snd/controlC1 /dev/snd/hwC0D0 /dev/snd/hwC0D3 /dev/snd/controlC2 /dev/snd/hwC0D1 /dev/snd/seq # aplay -l **** List of PLAYBACK Hardware Devices **** ALSA lib conf.c:2700:(snd_config_hooks_call) Cannot open shared library (null) ALSA lib conf.c:3079:(snd_config_update_r) hooks failed, removing configuration aplay: device_list:232: control open (0): No such file or directory We are not able to get if module is getting loaded, then why aplay doesn't list the device? Please help us to get the audio codec working. Regards, --Shilpa.

3 3

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 29 Jul '09

29 Jul '09

The following changes since commit 4ce2f2fe61002ab35c9333d52c95b5ff9e949f50: Marek Vasut (1): ASoC: Switch palm27x-asoc to jack detection api are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (2): ASoC: blackfin I2S(TDM mode) CPU DAI driver ASoC: board driver to connect bf5xx with ad1938 Janusz Krzysztofik (1): ASoC: CX20442: fix issues pointed out by subsystem maintainer sound/soc/blackfin/Kconfig | 23 +++- sound/soc/blackfin/Makefile | 6 + sound/soc/blackfin/bf5xx-ad1938.c | 142 +++++++++++++++ sound/soc/blackfin/bf5xx-tdm-pcm.c | 330 ++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-tdm-pcm.h | 21 +++ sound/soc/blackfin/bf5xx-tdm.c | 343 ++++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-tdm.h | 14 ++ sound/soc/codecs/cx20442.c | 17 +- 8 files changed, 888 insertions(+), 8 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad1938.c create mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.c create mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h create mode 100644 sound/soc/blackfin/bf5xx-tdm.c create mode 100644 sound/soc/blackfin/bf5xx-tdm.h

2 1

Re: [alsa-devel] Testers wanted: New SB X-Fi driver
by Takashi Iwai 29 Jul '09

29 Jul '09

[Please don't do top-posting] At Tue, 28 Jul 2009 14:50:10 -0500, Ted T. Logan wrote: > > Just got the latest as of today, > > click > click > click > click > click > > :( > mplayer yesterday.mp3 For testing this kind of problem, use aplay and try different --period-size and --buffer-size options. mplayer isn't suitable for primary testing. If the click period doesn't change regardless of the period size option value, it implies that the noise doesn't come from the buffer set up but rather from other routes. If the noise period changes with different period size options, the click noise is a buffer setup issue. In addition, it'd be better to use aplay with -Dplughw to avoid the automatic routing to PA. Also, your description was unclear. It it only click without other tone, or repeated sound, or the click noise appear on the normal playback, or what? The repeated sound bug with emu20k2 (it didn't appear with emu20k1) was fixed as I mentioned. But, if it's click noise on the normal playback, then I don't know. Better to ask Creative guys... thanks, Takashi

1 0

[alsa-devel] Assert when attempting to use pulseaudio git master with the a52 plugin.
by Luke Yelavich 29 Jul '09

29 Jul '09

Hi all I would like to bring to your attention a bug in the a52 plugin when working with pulseaudio. The full details are found in the ALSA bug tracker: https://bugtrack.alsa-project.org/alsa-bug/bug_view_advanced_page.php?bug_i… I am happy for follow ups to be in the bug or on this list. Thanks Luke

1 0

[PATCH] ASoC: MPC5200: Support for buffer wrap around
by John Bonesio 29 Jul '09

29 Jul '09

The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that s->runtime->control->appl_ptr can wrap around to the beginning of the buffer. This change fixes this problem. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 17 +++++++++++++++++ 1 files changed, 17 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index cfe0ea4..2551c58 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -70,6 +70,23 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) { + if (s->appl_ptr > s->runtime->control->appl_ptr) { + /* + * In this case s->runtime->control->appl_ptr has wrapped around. + * Play the data to the end of the boundary, then wrap our own + * appl_ptr back around. + */ + while (s->appl_ptr < s->runtime->boundary) { + if (bcom_queue_full(s->bcom_task)) + return; + + s->appl_ptr += s->period_size; + + psc_dma_bcom_enqueue_next_buffer(s); + } + s->appl_ptr -= s->runtime->boundary; + } + while (s->appl_ptr < s->runtime->control->appl_ptr) { if (bcom_queue_full(s->bcom_task)) -- 1.6.0.4 --=-D7TcNx246dz2g/fqLr/C--

1 0

[alsa-devel] [PATCH] ASoC: CX20442: fix issues pointed out by subsystem maintainer
by Janusz Krzysztofik 28 Jul '09

28 Jul '09

The patch fixes some checkpatch identified issues and adds a comment about line discipline interaction to my driver code, as requested by Mark on my inital submission (thank you Mark for applying my imperfect patch anyway). It also fixes MODULE_ALIAS mismatch as used in my machine driver. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- In my next attempt, I'll try to satisfy Mark's request for pushing down the line discipline code, now existing in my machine driver, to the codec driver. --- linux-2.6.31-rc3/sound/soc/codecs/cx20442.c.orig 2009-07-22 04:56:45.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/codecs/cx20442.c 2009-07-22 21:46:41.000000000 +0200 @@ -132,7 +132,7 @@ enum v253_vls { static int cx20442_pm_to_v253_vls(u8 value) { - switch(value & ~(1 << CX20442_AGC)) { + switch (value & ~(1 << CX20442_AGC)) { case 0: return V253_VLS_T; case (1 << CX20442_SPKOUT): @@ -152,7 +152,7 @@ static int cx20442_pm_to_v253_vls(u8 val } static int cx20442_pm_to_v253_vsp(u8 value) { - switch(value & ~(1 << CX20442_AGC)) { + switch (value & ~(1 << CX20442_AGC)) { case (1 << CX20442_SPKOUT): case (1 << CX20442_MIC): case (1 << CX20442_SPKOUT) | (1 << CX20442_MIC): @@ -171,6 +171,9 @@ static int cx20442_write(struct snd_soc_ if (reg >= codec->reg_cache_size) return -EINVAL; + /* hw_write and control_data pointers required for talking to the modem + * are expected to be set by the machine driver's line discipline + * initialization code */ if (!codec->hw_write || !codec->control_data) return -EIO; @@ -182,7 +185,7 @@ static int cx20442_write(struct snd_soc_ return vls; vsp = cx20442_pm_to_v253_vsp(value); - if (vsp < 0 ) + if (vsp < 0) return vsp; if ((vls == V253_VLS_T) || @@ -232,7 +235,7 @@ static int cx20442_codec_probe(struct pl struct snd_soc_codec *codec; int ret; - if(!cx20442_codec) { + if (!cx20442_codec) { dev_err(&pdev->dev, "cx20442 not yet discovered\n"); return -ENODEV; } @@ -310,13 +313,13 @@ static int cx20442_register(struct cx204 ret = snd_soc_register_codec(codec); if (ret != 0) { - //dev_err(&dev->dev, "Failed to register codec: %d\n", ret); + dev_err(&codec->dev, "Failed to register codec: %d\n", ret); goto err; } ret = snd_soc_register_dai(&cx20442_dai); if (ret != 0) { - //dev_err(&dev->dev, "Failed to register DAI: %d\n", ret); + dev_err(&codec->dev, "Failed to register DAI: %d\n", ret); goto err_codec; } @@ -392,4 +395,4 @@ module_exit(cx20442_exit); MODULE_DESCRIPTION("ASoC CX20442-11 voice modem codec driver"); MODULE_AUTHOR("Janusz Krzysztofik"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:cx20442-codec"); +MODULE_ALIAS("platform:cx20442");

2 1

[alsa-devel] [RFC] [PATCH] ASoC: CX20442: push down machine independent line discipline bits
by Janusz Krzysztofik 28 Jul '09

28 Jul '09

This patch adds machine independent line discipline code, prevoiusly exsiting inside my Amstrad Delta ASoC machine dirver, to the Conexant CX20442 codec driver. The code can be used as a standalone line discipline, or as a set of codec specific functions called from machine's line discipline callbacks. Anyway, the line discipline itself must be registered by a machine driver. Suggested by ASoC manintainer Mark Brown <broonie(a)opensource.wolfsonmicro.com> (Hardly understood, imperfectly implemented and) Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Mark, does it follow your idea closely enough? --- linux-2.6.31-rc3/sound/soc/codecs/cx20442.h.orig 2009-07-22 04:56:45.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/codecs/cx20442.h 2009-07-28 21:23:17.000000000 +0200 @@ -15,5 +15,6 @@ extern struct snd_soc_dai cx20442_dai; extern struct snd_soc_codec_device cx20442_codec_dev; +extern struct tty_ldisc_ops v253_ops; #endif --- linux-2.6.31-rc3/sound/soc/codecs/cx20442.c.orig 2009-07-28 20:26:18.000000000 +0200 +++ linux-2.6.31-rc3/sound/soc/codecs/cx20442.c 2009-07-28 21:19:43.000000000 +0200 @@ -13,6 +13,8 @@ * option) any later version. */ +#include <linux/tty.h> + #include <sound/core.h> #include <sound/initval.h> #include <sound/soc-dapm.h> @@ -172,8 +174,7 @@ static int cx20442_write(struct snd_soc_ return -EINVAL; /* hw_write and control_data pointers required for talking to the modem - * are expected to be set by the machine driver's line discipline - * initialization code */ + * are expected to be set by a line discipline initialization code */ if (!codec->hw_write || !codec->control_data) return -EIO; @@ -208,6 +209,111 @@ static int cx20442_write(struct snd_soc_ return 0; } + +/* Moved up here as line discipline referres it during initialization */ +static struct snd_soc_codec *cx20442_codec; + + +/* + * Line discpline related code + * + * Any of the callback functions below can be used in two ways: + * 1) registerd by a machine driver as one of line discipline operations, + * 2) called from inside a machine specific line discipline callback function + * in case when extra machine specific code must be run as well. + */ + +/* Modem init: echo off, digital speaker off, quiet off, voice mode */ +static const char *v253_init = "ate0m0q0+fclass=8\r"; + +/* Line discipline .open() */ +static int v253_open(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = cx20442_codec; + int ret, len = strlen(v253_init); + + /* Doesn't make sense without write callback */ + if (!tty->ops->write) + return -EINVAL; + + /* Pass the codec structure address for use by other ldisc callbacks */ + tty->disc_data = codec; + + if (tty->ops->write(tty, v253_init, len) != len) { + ret = -EIO; + goto err; + } + /* Actual setup will be performed after the modem responds. */ + return 0; +err: + tty->disc_data = NULL; + return ret; +} + +/* Line discipline .close() */ +static void v253_close(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = tty->disc_data; + + tty->disc_data = NULL; + + if (!codec) + return; + + /* Prevent the codec driver from further accessing the modem */ + codec->hw_write = NULL; + codec->control_data = NULL; + codec->pop_time = 0; +} + +/* Line discipline .hangup() */ +static int v253_hangup(struct tty_struct *tty) +{ + v253_close(tty); + return 0; +} + +/* Line discipline .receive_buf() */ +static void v253_receive(struct tty_struct *tty, + const unsigned char *cp, char *fp, int count) +{ + struct snd_soc_codec *codec = tty->disc_data; + + if (!codec) + return; + + if (!codec->control_data) { + /* First modem response, complete setup procedure */ + + /* Set up codec driver access to modem controls */ + codec->control_data = tty; + codec->hw_write = (hw_write_t)tty->ops->write; + codec->pop_time = 1; + } +} + +/* Line discipline .write_wakeup() */ +static void v253_wakeup(struct tty_struct *tty) +{ +} + +static struct tty_ldisc_ops v253_ops = { + .magic = TTY_LDISC_MAGIC, + .name = "cx20442", + .owner = THIS_MODULE, + .open = v253_open, + .close = v253_close, + .hangup = v253_hangup, + .receive_buf = v253_receive, + .write_wakeup = v253_wakeup, +}; +EXPORT_SYMBOL_GPL(v253_ops); + + +/* + * Codec DAI + */ + struct snd_soc_dai cx20442_dai = { .name = "CX20442", .playback = { @@ -227,8 +333,6 @@ struct snd_soc_dai cx20442_dai = { }; EXPORT_SYMBOL_GPL(cx20442_dai); -static struct snd_soc_codec *cx20442_codec; - static int cx20442_codec_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev);

2 1

[alsa-devel] [PATCH 0/2]blackfin TDM dai and blackfin-ad1938 machine driver
by Barry Song 28 Jul '09

28 Jul '09

Hi Mark, The following two patches are the 2nd version for the "new support for ad1938 codec" patch sequence. Since you have applied the codec driver patch([PATCH 1/3] new ad1938 codec driver based on asoc), here I only send the DAI and Board driver patches again according to your feedback. Patch 1: blackfin I2S(TDM mode) CPU DAI driver Patch 2: board driver to connect bf5xx with ad1938 Detailed description about every patch and changes is included in the corresponding patch. Thanks Barry

2 9

[alsa-devel] [PATCH 1/2] blackfin I2S(TDM mode) CPU DAI driver
by Barry Song 28 Jul '09

28 Jul '09

The I2S DAI driver for blackfin SPORT, but works in TDM mode. I2S is not a special case of TDM with only left and right two slots for SPORT interface. I2S coordinates with TDM in SPORT, but not a part of TDM. TDM require different hardware configuration with I2S, not only different slot number. One is "Stereo Serial Operation" mode of SPORT, the other one is "Multichannel Operation" mode. They are incompatible at the same time. Hardware and DMA description and data transfer flow are much different for I2S and TDM. Merging them as a whole will be very ugly and difficult to maintain. So we don't define a new DAI type, but give two DAI instances for standard I2S and TDM, both in I2S-family DAI type. The TDM instance still uses the I2S-family DAI type. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/Kconfig | 15 ++- sound/soc/blackfin/Makefile | 4 + sound/soc/blackfin/bf5xx-tdm-pcm.c | 330 ++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-tdm-pcm.h | 21 +++ sound/soc/blackfin/bf5xx-tdm.c | 343 ++++++++++++++++++++++++++++++++++++ sound/soc/blackfin/bf5xx-tdm.h | 14 ++ 6 files changed, 726 insertions(+), 1 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.c create mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h create mode 100644 sound/soc/blackfin/bf5xx-tdm.c create mode 100644 sound/soc/blackfin/bf5xx-tdm.h diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 811596f..2ab6d2f 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -7,6 +7,15 @@ config SND_BF5XX_I2S mode (supports single stereo In/Out). You will also need to select the audio interfaces to support below. +config SND_BF5XX_TDM + tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" + depends on (BLACKFIN && SND_SOC) + help + Say Y or M if you want to add support for codecs attached to + the Blackfin SPORT (synchronous serial ports) interface in TDM + mode. + You will also need to select the audio interfaces to support below. + config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio support for BF52x ezkit" depends on SND_BF5XX_I2S @@ -69,6 +78,10 @@ config SND_BF5XX_SOC_I2S tristate select SND_BF5XX_SOC_SPORT +config SND_BF5XX_SOC_TDM + tristate + select SND_BF5XX_SOC_SPORT + config SND_BF5XX_SOC_AC97 tristate select AC97_BUS @@ -85,7 +98,7 @@ config SND_BF5XX_SOC_AD1980 config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" - depends on (SND_BF5XX_I2S || SND_BF5XX_AC97) + depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM) range 0 3 if BF54x range 0 1 if !BF54x default 0 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 97bb37a..f861926 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -1,15 +1,19 @@ # Blackfin Platform Support snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o +snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o snd-soc-bf5xx-sport-objs := bf5xx-sport.o snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o +snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o +obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o +obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support snd-ad1980-objs := bf5xx-ad1980.o diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c new file mode 100644 index 0000000..af7fa8d --- /dev/null +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -0,0 +1,330 @@ +/* + * File: sound/soc/blackfin/bf5xx-tdm-pcm.c + * Author: Barry Song <Barry.Song(a)analog.com> + * + * Created: Tue June 06 2009 + * Description: DMA driver for tdm codec + * + * Modified: + * Copyright 2009 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * 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 program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> + +#include "bf5xx-tdm-pcm.h" +#include "bf5xx-tdm.h" +#include "bf5xx-sport.h" + +#define PCM_BUFFER_MAX 0x10000 +#define FRAGMENT_SIZE_MIN (4*1024) +#define FRAGMENTS_MIN 2 +#define FRAGMENTS_MAX 32 + +static void bf5xx_dma_irq(void *data) +{ + struct snd_pcm_substream *pcm = data; + snd_pcm_period_elapsed(pcm); +} + +static const struct snd_pcm_hardware bf5xx_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_RESUME), + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_48000, + .channels_min = 2, + .channels_max = 8, + .buffer_bytes_max = PCM_BUFFER_MAX, + .period_bytes_min = FRAGMENT_SIZE_MIN, + .period_bytes_max = PCM_BUFFER_MAX/2, + .periods_min = FRAGMENTS_MIN, + .periods_max = FRAGMENTS_MAX, +}; + +static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + size_t size = bf5xx_pcm_hardware.buffer_bytes_max; + snd_pcm_lib_malloc_pages(substream, size * 4); + + return 0; +} + +static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + int fragsize_bytes = frames_to_bytes(runtime, runtime->period_size); + + fragsize_bytes /= runtime->channels; + /* inflate the fragsize to match the dma width of SPORT */ + fragsize_bytes *= 8; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sport_set_tx_callback(sport, bf5xx_dma_irq, substream); + sport_config_tx_dma(sport, runtime->dma_area, + runtime->periods, fragsize_bytes); + } else { + sport_set_rx_callback(sport, bf5xx_dma_irq, substream); + sport_config_rx_dma(sport, runtime->dma_area, + runtime->periods, fragsize_bytes); + } + + return 0; +} + +static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_tx_start(sport); + else + sport_rx_start(sport); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_tx_stop(sport); + else + sport_rx_stop(sport); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + unsigned int diff; + snd_pcm_uframes_t frames; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + diff = sport_curr_offset_tx(sport); + frames = diff / (8*4); /* 32 bytes per frame */ + } else { + diff = sport_curr_offset_rx(sport); + frames = diff / (8*4); + } + return frames; +} + +static int bf5xx_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + if (sport_handle != NULL) + runtime->private_data = sport_handle; + else { + pr_err("sport_handle is NULL\n"); + ret = -ENODEV; + } +out: + return ret; +} + +static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) +{ + unsigned int *src; + unsigned int *dst; + int i; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = buf; + dst = (unsigned int *)substream->runtime->dma_area; + + dst += pos * 8; + while (count--) { + for (i = 0; i < substream->runtime->channels; i++) + *(dst + i) = *src++; + dst += 8; + } + } else { + src = (unsigned int *)substream->runtime->dma_area; + dst = buf; + + src += pos * 8; + while (count--) { + for (i = 0; i < substream->runtime->channels; i++) + *dst++ = *(src+i); + src += 8; + } + } + + return 0; +} + +static int bf5xx_pcm_silence(struct snd_pcm_substream *substream, + int channel, snd_pcm_uframes_t pos, snd_pcm_uframes_t count) +{ + unsigned char *buf = substream->runtime->dma_area; + buf += pos * 8 * 4; + memset(buf, '\0', count * 8 * 4); + + return 0; +} + + +struct snd_pcm_ops bf5xx_pcm_tdm_ops = { + .open = bf5xx_pcm_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = bf5xx_pcm_hw_params, + .hw_free = bf5xx_pcm_hw_free, + .prepare = bf5xx_pcm_prepare, + .trigger = bf5xx_pcm_trigger, + .pointer = bf5xx_pcm_pointer, + .copy = bf5xx_pcm_copy, + .silence = bf5xx_pcm_silence, +}; + +static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = bf5xx_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size * 4, + &buf->addr, GFP_KERNEL); + if (!buf->area) { + pr_err("Failed to allocate dma memory \ + Please increase uncached DMA memory region\n"); + return -ENOMEM; + } + buf->bytes = size; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_handle->tx_buf = buf->area; + else + sport_handle->rx_buf = buf->area; + + return 0; +} + +static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(NULL, buf->bytes, buf->area, 0); + buf->area = NULL; + } + if (sport_handle) + sport_done(sport_handle); +} + +static u64 bf5xx_pcm_dmamask = DMA_32BIT_MASK; + +static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &bf5xx_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_32BIT_MASK; + + if (dai->playback.channels_min) { + ret = bf5xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = bf5xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } +out: + return ret; +} + +struct snd_soc_platform bf5xx_tdm_soc_platform = { + .name = "bf5xx-audio", + .pcm_ops = &bf5xx_pcm_tdm_ops, + .pcm_new = bf5xx_pcm_tdm_new, + .pcm_free = bf5xx_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(bf5xx_tdm_soc_platform); + +static int __init bfin_pcm_tdm_init(void) +{ + return snd_soc_register_platform(&bf5xx_tdm_soc_platform); +} +module_init(bfin_pcm_tdm_init); + +static void __exit bfin_pcm_tdm_exit(void) +{ + snd_soc_unregister_platform(&bf5xx_tdm_soc_platform); +} +module_exit(bfin_pcm_tdm_exit); + +MODULE_AUTHOR("Barry Song"); +MODULE_DESCRIPTION("ADI Blackfin TDM PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.h b/sound/soc/blackfin/bf5xx-tdm-pcm.h new file mode 100644 index 0000000..9e8473d --- /dev/null +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.h @@ -0,0 +1,21 @@ +/* + * linux/sound/arm/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin + * + * Copyright 2009 Analog Device Inc. + * + * 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 _BF5XX_TDM_PCM_H +#define _BF5XX_TDM_PCM_H + +struct bf5xx_pcm_dma_params { + char *name; /* stream identifier */ +}; + +/* platform data */ +extern struct snd_soc_platform bf5xx_tdm_soc_platform; + +#endif diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c new file mode 100644 index 0000000..3096bad --- /dev/null +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -0,0 +1,343 @@ +/* + * File: sound/soc/blackfin/bf5xx-tdm.c + * Author: Barry Song <Barry.Song(a)analog.com> + * + * Created: Thurs June 04 2009 + * Description: Blackfin I2S(TDM) CPU DAI driver + * Even though TDM mode can be as part of I2S DAI, but there + * are so much difference in configuration and data flow, + * it's very ugly to integrate I2S and TDM into a module + * + * Modified: + * Copyright 2009 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * 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 program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <asm/irq.h> +#include <asm/portmux.h> +#include <linux/mutex.h> +#include <linux/gpio.h> + +#include "bf5xx-sport.h" +#include "bf5xx-tdm.h" + +struct bf5xx_tdm_port { + u16 tcr1; + u16 rcr1; + u16 tcr2; + u16 rcr2; + int configured; +}; + +static struct bf5xx_tdm_port bf5xx_tdm; +static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; + +static struct sport_param sport_params[2] = { + { + .dma_rx_chan = CH_SPORT0_RX, + .dma_tx_chan = CH_SPORT0_TX, + .err_irq = IRQ_SPORT0_ERROR, + .regs = (struct sport_register *)SPORT0_TCR1, + }, + { + .dma_rx_chan = CH_SPORT1_RX, + .dma_tx_chan = CH_SPORT1_TX, + .err_irq = IRQ_SPORT1_ERROR, + .regs = (struct sport_register *)SPORT1_TCR1, + } +}; + +/* + * Setting the TFS pin selector for SPORT 0 based on whether the selected + * port id F or G. If the port is F then no conflict should exist for the + * TFS. When Port G is selected and EMAC then there is a conflict between + * the PHY interrupt line and TFS. Current settings prevent the conflict + * by ignoring the TFS pin when Port G is selected. This allows both + * ssm2602 using Port G and EMAC concurrently. + */ +#ifdef CONFIG_BF527_SPORT0_PORTF +#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) +#else +#define LOCAL_SPORT0_TFS (0) +#endif + +static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, + P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0}, + {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI, + P_SPORT1_RSCLK, P_SPORT1_TFS, 0} }; + +static int bf5xx_tdm_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + int ret = 0; + + /* interface format:support TDM,slave mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + break; + default: + printk(KERN_ERR "%s: Unknown DAI format type\n", __func__); + ret = -EINVAL; + break; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + ret = -EINVAL; + break; + default: + printk(KERN_ERR "%s: Unknown DAI master type\n", __func__); + ret = -EINVAL; + break; + } + + return ret; +} + +static int bf5xx_tdm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int ret = 0; + + bf5xx_tdm.tcr2 &= ~0x1f; + bf5xx_tdm.rcr2 &= ~0x1f; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + bf5xx_tdm.tcr2 |= 31; + bf5xx_tdm.rcr2 |= 31; + sport_handle->wdsize = 4; + break; + /* at present, we only support 32bit transfer */ + default: + pr_err("not supported PCM format yet\n"); + return -EINVAL; + break; + } + + if (!bf5xx_tdm.configured) { + /* + * TX and RX are not independent,they are enabled at the + * same time, even if only one side is running. So, we + * need to configure both of them at the time when the first + * stream is opened. + * + * CPU DAI:slave mode. + */ + ret = sport_config_rx(sport_handle, bf5xx_tdm.rcr1, + bf5xx_tdm.rcr2, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + return -EBUSY; + } + + ret = sport_config_tx(sport_handle, bf5xx_tdm.tcr1, + bf5xx_tdm.tcr2, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + return -EBUSY; + } + + bf5xx_tdm.configured = 1; + } + + return 0; +} + +static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + /* No active stream, SPORT is allowed to be configured again. */ + if (!dai->active) + bf5xx_tdm.configured = 0; +} + +#ifdef CONFIG_PM +static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) +{ + struct sport_device *sport = + (struct sport_device *)dai->private_data; + + if (!dai->active) + return 0; + if (dai->capture.active) + sport_rx_stop(sport); + if (dai->playback.active) + sport_tx_stop(sport); + return 0; +} + +static int bf5xx_tdm_resume(struct snd_soc_dai *dai) +{ + int ret; + struct sport_device *sport = + (struct sport_device *)dai->private_data; + + if (!dai->active) + return 0; + + ret = sport_set_multichannel(sport, 8, 0xFF, 1); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + } + + ret = sport_config_rx(sport, IRFS, 0x1F, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + } + + ret = sport_config_tx(sport, ITFS, 0x1F, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + } + + return 0; +} + +#else +#define bf5xx_tdm_suspend NULL +#define bf5xx_tdm_resume NULL +#endif + +static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { + .hw_params = bf5xx_tdm_hw_params, + .set_fmt = bf5xx_tdm_set_dai_fmt, + .shutdown = bf5xx_tdm_shutdown, +}; + +struct snd_soc_dai bf5xx_tdm_dai = { + .name = "bf5xx-tdm", + .id = 0, + .suspend = bf5xx_tdm_suspend, + .resume = bf5xx_tdm_resume, + .playback = { + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .ops = &bf5xx_tdm_dai_ops, +}; +EXPORT_SYMBOL_GPL(bf5xx_tdm_dai); + +static int __devinit bfin_tdm_probe(struct platform_device *pdev) +{ + int ret = 0; + + if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { + pr_err("Requesting Peripherals failed\n"); + return -EFAULT; + } + + /* request DMA for SPORT */ + sport_handle = sport_init(&sport_params[sport_num], 4, \ + 8 * sizeof(u32), NULL); + if (!sport_handle) { + peripheral_free_list(&sport_req[sport_num][0]); + return -ENODEV; + } + + /* SPORT works in TDM mode */ + ret = sport_set_multichannel(sport_handle, 8, 0xFF, 1); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + goto sport_config_err; + } + + ret = sport_config_rx(sport_handle, IRFS, 0x1F, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + goto sport_config_err; + } + + ret = sport_config_tx(sport_handle, ITFS, 0x1F, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + goto sport_config_err; + } + + ret = snd_soc_register_dai(&bf5xx_tdm_dai); + if (ret) { + pr_err("Failed to register DAI: %d\n", ret); + goto sport_config_err; + } + return 0; + +sport_config_err: + peripheral_free_list(&sport_req[sport_num][0]); + return ret; +} + +static int __devexit bfin_tdm_remove(struct platform_device *pdev) +{ + peripheral_free_list(&sport_req[sport_num][0]); + snd_soc_unregister_dai(&bf5xx_tdm_dai); + + return 0; +} + +static struct platform_driver bfin_tdm_driver = { + .probe = bfin_tdm_probe, + .remove = __devexit_p(bfin_tdm_remove), + .driver = { + .name = "bfin-tdm", + .owner = THIS_MODULE, + }, +}; + +static int __init bfin_tdm_init(void) +{ + return platform_driver_register(&bfin_tdm_driver); +} +module_init(bfin_tdm_init); + +static void __exit bfin_tdm_exit(void) +{ + platform_driver_unregister(&bfin_tdm_driver); +} +module_exit(bfin_tdm_exit); + +/* Module information */ +MODULE_AUTHOR("Barry Song"); +MODULE_DESCRIPTION("TDM driver for ADI Blackfin"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h new file mode 100644 index 0000000..e5157cd --- /dev/null +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -0,0 +1,14 @@ +/* + * linux/sound/arm/bf5xx-i2s.h + * + * 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 _BF5XX_TDM_H +#define _BF5XX_TDM_H + +extern struct snd_soc_dai bf5xx_tdm_dai; + +#endif -- 1.5.6.3

2 1

[alsa-devel] [RFC] [PATCH 1/3] ASoC: Add support for Conexant CX20442-11 voice modem codec
by Janusz Krzysztofik 28 Jul '09

28 Jul '09

This patch adds support for Conexant CX20442-11 voice modem codec, suitable for use by the ASoC board driver for Amstrad E3 (Delta) videophone. Related sound card driver will follow. Created and tested against linux-2.6.31-rc3. Applies and works with linux-omap-2.6 commit 7c5cb7862d32cb344be7831d466535d5255e35ac as well. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- This codec is an optional part of the Conexant SmartV three chip modem design. As such, documentation for its proprietary digital audio interface is not available. However, on Amstrad Delta board, thanks to Mark Underwood who created an initial, omap-alsa based sound driver a few years ago[1], the codec has been discovered to be accessible not only from the modem side, but also over the OMAP McBSP based CPU DAI. Thus, the driver can be used by any sound card that can access the codec DAI directly. The DAI configuration parameters (sample rate and format, number of channels) has been selected out empirically for best user experience. The codec analogue interface consists of two pairs of analogue I/O pins: speakerphone interface or telephone handset/headset interface. Furthermore, it seams to provide two operation modes for speakerphone I/O: standard and advanced, with automatic gain control and echo cancelation. Even if the codec control interface is unknown and not available, all those interfaces and modes can be selected over the modem chip using V.253 commands. The driver is able to issue necessary commands over a suitable hw_write function if provided by a sound card driver. Otherwise, the codec can be controlled over the modem from userspace while inactive. Even if nothig is known about the codec internal power management capabilities, DAPM widgets has been used to model the codec audio map. Automatically performed powering up/down of those virtual widgets results in corresponding V.253 commands being issued. Some driver features/oddities may be board specific, but I have no way to verify that with any board other than Amstrad Delta. [1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.html --- linux-2.6.31-rc1/sound/soc/codecs/Kconfig.orig 2009-07-04 16:40:00.000000000 +0200 +++ linux-2.6.31-rc1/sound/soc/codecs/Kconfig 2009-07-22 01:01:08.000000000 +0200 @@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_CX20442 select SND_SOC_PCM3008 select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C @@ -86,6 +87,9 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 +config SND_SOC_CX20442 + tristate + config SND_SOC_L3 tristate --- linux-2.6.31-rc1/sound/soc/codecs/Makefile.orig 2009-07-04 16:40:00.000000000 +0200 +++ linux-2.6.31-rc1/sound/soc/codecs/Makefile 2009-07-22 01:01:08.000000000 +0200 @@ -4,6 +4,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o +snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-spdif-objs := spdif_transciever.o @@ -41,6 +42,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o --- /dev/null 2009-06-24 10:33:03.437003511 +0200 +++ linux-2.6.31-rc1/sound/soc/codecs/cx20442.h 2009-07-22 01:01:08.000000000 +0200 @@ -0,0 +1,19 @@ +/* + * cx20442.h -- audio driver for CX20442 + * + * Copyright 2009 Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> + * + * 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. + * + */ + +#ifndef _CX20442_CODEC_H +#define _CX20442_CODEC_H + +extern struct snd_soc_dai cx20442_dai; +extern struct snd_soc_codec_device cx20442_codec_dev; + +#endif --- /dev/null 2009-06-24 10:33:03.437003511 +0200 +++ linux-2.6.31-rc1/sound/soc/codecs/cx20442.c 2009-07-22 01:01:08.000000000 +0200 @@ -0,0 +1,395 @@ +/* + * cx20442.c -- CX20442 ALSA Soc Audio driver + * + * Copyright 2009 Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> + * + * Initially based on sound/soc/codecs/wm8400.c + * Copyright 2008, 2009 Wolfson Microelectronics PLC. + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.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. + */ + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/soc-dapm.h> + +#include "cx20442.h" + + +struct cx20442_priv { + struct snd_soc_codec codec; + u8 reg_cache[1]; +}; + +#define CX20442_PM 0x0 + +#define CX20442_TELIN 0 +#define CX20442_TELOUT 1 +#define CX20442_MIC 2 +#define CX20442_SPKOUT 3 +#define CX20442_AGC 4 + +static const struct snd_soc_dapm_widget cx20442_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("TELOUT"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + SND_SOC_DAPM_OUTPUT("AGCOUT"), + + SND_SOC_DAPM_MIXER("SPKOUT Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_PGA("TELOUT Amp", CX20442_PM, CX20442_TELOUT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPKOUT Amp", CX20442_PM, CX20442_SPKOUT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPKOUT AGC", CX20442_PM, CX20442_AGC, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MICBIAS("TELIN Bias", CX20442_PM, CX20442_TELIN, 0), + SND_SOC_DAPM_MICBIAS("MIC Bias", CX20442_PM, CX20442_MIC, 0), + + SND_SOC_DAPM_PGA("MIC AGC", CX20442_PM, CX20442_AGC, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("TELIN"), + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_INPUT("AGCIN"), +}; + +static const struct snd_soc_dapm_route cx20442_audio_map[] = { + {"TELOUT", NULL, "TELOUT Amp"}, + + {"SPKOUT", NULL, "SPKOUT Mixer"}, + {"SPKOUT Mixer", NULL, "SPKOUT Amp"}, + + {"TELOUT Amp", NULL, "DAC"}, + {"SPKOUT Amp", NULL, "DAC"}, + + {"SPKOUT Mixer", NULL, "SPKOUT AGC"}, + {"SPKOUT AGC", NULL, "AGCIN"}, + + {"AGCOUT", NULL, "MIC AGC"}, + {"MIC AGC", NULL, "MIC"}, + + {"MIC Bias", NULL, "MIC"}, + {"Input Mixer", NULL, "MIC Bias"}, + + {"TELIN Bias", NULL, "TELIN"}, + {"Input Mixer", NULL, "TELIN Bias"}, + + {"ADC", NULL, "Input Mixer"}, +}; + +static int cx20442_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, cx20442_dapm_widgets, + ARRAY_SIZE(cx20442_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, cx20442_audio_map, + ARRAY_SIZE(cx20442_audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static unsigned int cx20442_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *reg_cache = codec->reg_cache; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + return reg_cache[reg]; +} + +enum v253_vls { + V253_VLS_NONE = 0, + V253_VLS_T, + V253_VLS_L, + V253_VLS_LT, + V253_VLS_S, + V253_VLS_ST, + V253_VLS_M, + V253_VLS_MST, + V253_VLS_S1, + V253_VLS_S1T, + V253_VLS_MS1T, + V253_VLS_M1, + V253_VLS_M1ST, + V253_VLS_M1S1T, + V253_VLS_H, + V253_VLS_HT, + V253_VLS_MS, + V253_VLS_MS1, + V253_VLS_M1S, + V253_VLS_M1S1, + V253_VLS_TEST, +}; + +static int cx20442_pm_to_v253_vls(u8 value) +{ + switch(value & ~(1 << CX20442_AGC)) { + case 0: + return V253_VLS_T; + case (1 << CX20442_SPKOUT): + return V253_VLS_S1; + case (1 << CX20442_MIC): + return V253_VLS_M1; + case (1 << CX20442_SPKOUT) | (1 << CX20442_MIC): + return V253_VLS_M1S1; + case (1 << CX20442_TELOUT): + case (1 << CX20442_TELIN): + case (1 << CX20442_TELOUT) | (1 << CX20442_TELIN): + return V253_VLS_L; + case (1 << CX20442_TELOUT) | (1 << CX20442_MIC): + return V253_VLS_NONE; + } + return -EINVAL; +} +static int cx20442_pm_to_v253_vsp(u8 value) +{ + switch(value & ~(1 << CX20442_AGC)) { + case (1 << CX20442_SPKOUT): + case (1 << CX20442_MIC): + case (1 << CX20442_SPKOUT) | (1 << CX20442_MIC): + return (bool)(value & (1 << CX20442_AGC)); + } + return (value & (1 << CX20442_AGC)) ? -EINVAL : 0; +} + +static int cx20442_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 *reg_cache = codec->reg_cache; + int vls, vsp, old, len; + char buf[18]; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + if (!codec->hw_write || !codec->control_data) + return -EIO; + + old = reg_cache[reg]; + reg_cache[reg] = value; + + vls = cx20442_pm_to_v253_vls(value); + if (vls < 0) + return vls; + + vsp = cx20442_pm_to_v253_vsp(value); + if (vsp < 0 ) + return vsp; + + if ((vls == V253_VLS_T) || + (vls == cx20442_pm_to_v253_vls(old))) { + if (vsp == cx20442_pm_to_v253_vsp(old)) + return 0; + len = snprintf(buf, ARRAY_SIZE(buf), "at+vsp=%d\r", vsp); + } else if (vsp == cx20442_pm_to_v253_vsp(old)) + len = snprintf(buf, ARRAY_SIZE(buf), "at+vls=%d\r", vls); + else + len = snprintf(buf, ARRAY_SIZE(buf), + "at+vls=%d;+vsp=%d\r", vls, vsp); + + if (unlikely(len > (ARRAY_SIZE(buf) - 1))) + return -ENOMEM; + + if (codec->hw_write(codec->control_data, buf, len) != len) + return -EIO; + + return 0; +} + +struct snd_soc_dai cx20442_dai = { + .name = "CX20442", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; +EXPORT_SYMBOL_GPL(cx20442_dai); + +static struct snd_soc_codec *cx20442_codec; + +static int cx20442_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + if(!cx20442_codec) { + dev_err(&pdev->dev, "cx20442 not yet discovered\n"); + return -ENODEV; + } + codec = cx20442_codec; + + socdev->card->codec = codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + goto pcm_err; + } + + cx20442_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int cx20442_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device cx20442_codec_dev = { + .probe = cx20442_codec_probe, + .remove = cx20442_codec_remove, +}; +EXPORT_SYMBOL_GPL(cx20442_codec_dev); + +static int cx20442_register(struct cx20442_priv *cx20442) +{ + struct snd_soc_codec *codec = &cx20442->codec; + int ret; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "CX20442"; + codec->owner = THIS_MODULE; + codec->private_data = cx20442; + + codec->dai = &cx20442_dai; + codec->num_dai = 1; + + codec->reg_cache = &cx20442->reg_cache; + codec->reg_cache_size = ARRAY_SIZE(cx20442->reg_cache); + codec->read = cx20442_read_reg_cache; + codec->write = cx20442_write; + + codec->bias_level = SND_SOC_BIAS_OFF; + + cx20442_dai.dev = codec->dev; + + cx20442_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + //dev_err(&dev->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&cx20442_dai); + if (ret != 0) { + //dev_err(&dev->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + cx20442_codec = NULL; + kfree(cx20442); + return ret; +} + +static void cx20442_unregister(struct cx20442_priv *cx20442) +{ + snd_soc_unregister_dai(&cx20442_dai); + snd_soc_unregister_codec(&cx20442->codec); + + cx20442_codec = NULL; + kfree(cx20442); +} + +static int cx20442_platform_probe(struct platform_device *pdev) +{ + struct cx20442_priv *cx20442; + struct snd_soc_codec *codec; + + cx20442 = kzalloc(sizeof(struct cx20442_priv), GFP_KERNEL); + if (cx20442 == NULL) + return -ENOMEM; + + codec = &cx20442->codec; + + codec->control_data = NULL; + codec->hw_write = NULL; + codec->pop_time = 0; + + codec->dev = &pdev->dev; + platform_set_drvdata(pdev, cx20442); + + return cx20442_register(cx20442); +} + +static int __exit cx20442_platform_remove(struct platform_device *pdev) +{ + struct cx20442_priv *cx20442 = platform_get_drvdata(pdev); + + cx20442_unregister(cx20442); + return 0; +} + +static struct platform_driver cx20442_platform_driver = { + .driver = { + .name = "cx20442", + .owner = THIS_MODULE, + }, + .probe = cx20442_platform_probe, + .remove = __exit_p(cx20442_platform_remove), +}; + +static int __init cx20442_init(void) +{ + return platform_driver_register(&cx20442_platform_driver); +} +module_init(cx20442_init); + +static void __exit cx20442_exit(void) +{ + platform_driver_unregister(&cx20442_platform_driver); +} +module_exit(cx20442_exit); + +MODULE_DESCRIPTION("ASoC CX20442-11 voice modem codec driver"); +MODULE_AUTHOR("Janusz Krzysztofik"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:cx20442-codec");

2 8

[alsa-devel] Using Alsa for AD1981b on 9263
by Shilpa Kedar Walvekar 28 Jul '09

28 Jul '09

Hi, I am using AT91SAM9263 evaluation kit with AD1981b codec. I want to use alsa to play and record audio on this kit. I tried to install the alsa library, alsa-oss, alsa-util. Alsa driver was part of kernel (2.6.27). I did, ./configure && make && make install for library, oss and util. Now when I try to do aplay -l following happens: aplay: device_list:223: no soundcards found... Don't know what is happening. Am I missing any step? What else should be done? Please help me asap. Regards, --Shilpa.

2 1

Re: [alsa-devel] get two mono channel from stereo line
by Clemens Ladisch 28 Jul '09

28 Jul '09

arif setiawan wrote: > It worked now. thank you. Apparently we changed something in your > plugin sample. Sorry, that was my error; I wrote a wrong number for the right dshare plugin. > The ipc_key for dshare in both left and right is supposed to be same > right?, also for dsnoop. Yes; the two processes that are to share the device use this to communicate with each other. Best regards, Clemens

1 0

Re: [alsa-devel] HDA Intel IDT 92HD71B7X - HP Pavilion DV7-1299EF - Subwoofer not Working
by Wael Nasreddine (a.k.a eMxyzptlk) 27 Jul '09

27 Jul '09

I still got no solution for this problem, guys please HELP!! I removed completely Windows, and I can't even enjoy music nor a movie!! On Thu, Jun 11, 2009 at 8:31 PM, Wael Nasreddine (a.k.a eMxyzptlk) < mla(a)nasreddine.com> wrote: > Yes it does mute the speakers once I plug in headphones.. > > > On Thu, Jun 11, 2009 at 8:12 PM, Gustavo Vieira <gustavo(a)sagui.org> wrote: > >> Em Dom, 2009-06-07 às 16:57 +0200, Wael Nasreddine (a.k.a eMxyzptlk) >> escreveu: >> > Last night I bought myself a new laptop, the HP Pavilion DV7-1299EF. >> > >> >> Sorry for hijacking your thread, but does this laptop mute the speaker >> if you plug in the headphones? >> >> I'm experiencing problems with jack sensing in a Pavilion DV4 with the >> same codec and what to know if this problem is general to the codec or >> specific to my model. >> >> Abraços, >> Gustavo >> >> >> > > > -- > Wael Nasreddine > > Blog : http://wael.nasreddine.com > E-mail : wael.nasreddine(a)gmail.com > gTalk : wael.nasreddine(a)gmail.com > Tel : +33.6.32.94.70.13 > Skype : eMxyzptlk > Twitter : @eMxyzptlk > > Sabayon Linux Chief Development Officer - http://www.sabayonlinux.org > > PGP: 1024D/C8DD18A2 06F6 1622 4BC8 4CEB D724 DE12 5565 3945 C8DD 18A2 > > .: An infinite number of monkeys typing into GNU emacs, > would never make a good program. (L. Torvalds 1995) :. > > -- Wael Nasreddine Blog : http://wael.nasreddine.com E-mail : wael.nasreddine(a)gmail.com gTalk : wael.nasreddine(a)gmail.com Tel : +33.6.32.94.70.13 Skype : eMxyzptlk Twitter : @eMxyzptlk Sabayon Linux Chief Development Officer - http://www.sabayonlinux.org PGP: 1024D/C8DD18A2 06F6 1622 4BC8 4CEB D724 DE12 5565 3945 C8DD 18A2 .: An infinite number of monkeys typing into GNU emacs, would never make a good program. (L. Torvalds 1995) :.

2 1

[alsa-devel] my driver can't reproduce continuously
by Kuninori Morimoto 27 Jul '09

27 Jul '09

Hi Now I'm creating sound driver. But can't reproduce continuously. So, I would like to ask. I use module of SH to play sound. Player is aplay, sound data is wav file. Q1) When I play sound, aplay said underrun!!! (at least 0.000 ms long) underrun!!! (at least 0.000 ms long) underrun!!! (at least 0.000 ms long) underrun!!! (at least 0.000 ms long) I checked source of aplay, but can not understand about snd_pcm_status_get_state. why snd_pcm_status_get_state return SND_PCM_STATE_XRUN ? What should my driver do (or where should I check) ? Q2) The reason that it can not reproduce continuously seems soc-core.c :: soc_pcm_trigger timing. Call timing of soc_pcm_trigger is slow. What should my driver do to make call timing more fast ? Best regards -- Kuninori Morimoto

3 11

[alsa-devel] get two mono channel from stereo line
by arif setiawan 27 Jul '09

27 Jul '09

Thank you. It works as we expected, however we found that we can't run two lines (left-capture - left-playback) and (right-capture - right-playback) simulatenously. The error message is : ALSA lib pcm_dshare.c:706:(snd_pcm_dshare_open) unable to open slave cannot open output audio device right: Device or resource busy Is there any workaround for this?

1 0

Re: [alsa-devel] Getting around initial sample loss with intel HDMI
by Takashi Iwai 27 Jul '09

27 Jul '09

At Sun, 26 Jul 2009 23:21:50 -0700, Shane W wrote: > > On Mon, Jul 27, 2009 at 08:07:40AM +0200, Takashi Iwai wrote: > > At Sun, 26 Jul 2009 10:32:10 -0700, > > Shane W wrote: > > > I did some digging in the drivers to try and find the > > > source of this problem. Note that I am really not familiar > > > with this stuff. It appears to be here: > > > hda_codec.c > > > void snd_hda_codec_setup_stream(...) > > > ... > > > snd_hda_codec_write(codec, nid, 0, > > > AC_VERB_SET_CHANNEL_STREAMID, > > > > > > Introducing a delay after this point fixes the issue but > > > it's around a full second so that's obviously not a > > > solution. Could we not cache the current format and only do > > > this when the format changes rather than on every device > > > open? > > > > The format itself is kept. But the stream tag and channel assignment > > are assigned dynamically at each open. Thus, this call has to be > > reset at each open, at least. > > > > > For testing, I tried doing this though and disabling > > > the set_streamid, 0 call in snd_codec_cleanup() but it > > > still missed those first samples. Is there somewhere else I > > > need to be looking? > > > > Does it happen without dmix, too? > > Yeah, with and without dmix it still misses the first bit. OK. I just thought of a possibility of dmix bug. Good to know that it's not :) > The only way it doesn't is if I have another program open > the device and sleep, then have the samples play. Actually, > an open isn't sufficient, it has to go and set the playback > parameters as well. It's logical since the stream and format ids are assigned dynamically first after the stream is set up. One another problem is that we currently call azx_stream_reset() in each prepare callback in hda_intel.c. This was newly added because keeping the format id without resetting caused some other serious errors. Maybe this can be optimized for specific hardwares, e.g. skipping the unneeded reset for HDMI/SPDIF. Takashi

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Re: [alsa-devel] Getting around initial sample loss with intel HDMI
by Takashi Iwai 27 Jul '09

27 Jul '09

At Sun, 26 Jul 2009 10:32:10 -0700, Shane W wrote: > > On Sun, Jul 26, 2009 at 10:52:01AM +0200, Takashi Iwai wrote: > > > It's a known issue that Intel HDMI audio loses the first > > > half second or so of audio when opening the device. Using > > > the dmix interface, I've been able to get around that by > > > writing a quick app to hold the device open so the HDMI > > > stays live. However, with the recent asound.conf posted > > > here which uses the route plugin to fix the multichannel > > > order, I can't do that as dmix isn't in the picture. I'd > > > rather not use dmix anyway since it can bugger with the > > > sample rate so I am wondering if there's another way to get > > > around this. > > > > It's possible to combine with dmix. > > You set up dmix with 8 channels, put route plugin over it. > > This has a few problems. > > It doesn't seem to handle varying sample rates. IE if I > play a 96khz stream at 6 channels, sound is terrible. This is rather a problem of Nvidia HDMI codec driver... > Also, > streaming DTS or AC3 no longer works since the streams are > being upmixed to 6ch. This has to be sent anyway exclusively without dmix, so you'll need another PCM definition. > I did some digging in the drivers to try and find the > source of this problem. Note that I am really not familiar > with this stuff. It appears to be here: > hda_codec.c > void snd_hda_codec_setup_stream(...) > ... > snd_hda_codec_write(codec, nid, 0, > AC_VERB_SET_CHANNEL_STREAMID, > > Introducing a delay after this point fixes the issue but > it's around a full second so that's obviously not a > solution. Could we not cache the current format and only do > this when the format changes rather than on every device > open? The format itself is kept. But the stream tag and channel assignment are assigned dynamically at each open. Thus, this call has to be reset at each open, at least. > For testing, I tried doing this though and disabling > the set_streamid, 0 call in snd_codec_cleanup() but it > still missed those first samples. Is there somewhere else I > need to be looking? Does it happen without dmix, too? The reset part can be a bit more optimized. But, this is really a sensitive part, which can easily break the I/O status without knowing... Takashi

1 0

Re: [alsa-devel] Getting around initial sample loss with intel HDMI
by Takashi Iwai 26 Jul '09

26 Jul '09

At Sat, 25 Jul 2009 13:35:18 -0700, Shane W wrote: > > Hi, > > It's a known issue that Intel HDMI audio loses the first > half second or so of audio when opening the device. Using > the dmix interface, I've been able to get around that by > writing a quick app to hold the device open so the HDMI > stays live. However, with the recent asound.conf posted > here which uses the route plugin to fix the multichannel > order, I can't do that as dmix isn't in the picture. I'd > rather not use dmix anyway since it can bugger with the > sample rate so I am wondering if there's another way to get > around this. It's possible to combine with dmix. You set up dmix with 8 channels, put route plugin over it. Takashi

1 0

[alsa-devel] [snd_via82xx] EAGAIN on snd_pcm_prepare after EPIPE on mmap_commit
by rom1dep 25 Jul '09

25 Jul '09

Hello ! I am encountering some issues with pulseaudio : after some time and for no evident reason PA stops routing the sound. That's what I get with $pulseaudio -vvv : http://pastebin.mandriva.com/13648 (got from a 2.6.29.6_mdv kernel) running $pacmd list-sinks when stuck gives : http://pastebin.mandriva.com/13649 Our pulseaudio integrator Colin Guthrie concluded : "OK, so what is happening is that some alsa level problem is occurring (due to mmap). Pulse tries to recover from this but cannot. The alsa sink (and the whole card) is subsequently unloaded due to this fatal error." And now.. a quick paste from irc : "[00:44:20] <coling> Jul 22 14:04:29 <mezcalero> coling: driver issue i am pretty sure [00:44:20] <coling> Jul 22 14:04:33 <mezcalero> i saw that before once [00:44:20] <coling> Jul 22 14:04:55 <mezcalero> coling: i.e. EAGAIN on snd_pcm_prepare after EPIPE on mmap_commit [00:44:23] <coling> Jul 22 14:05:08 <mezcalero> coling: needs to be brought to attention of alsa folks i guess [00:44:30] <coling> That's what I was told." I'm now on 2.6.30 and I can reproduce that bug, then I'm bringing that to you ! :) Thanks !

3 4

[alsa-devel] Fw: ALSA Volume control with Creative X
by yes work 25 Jul '09

25 Jul '09

----- Forwarded Message ---- From: yes work <yeswork(a)yahoo.com> To: clemens(a)ladisch.de Sent: Wednesday, July 1, 2009 13:03:22 Subject: ALSA Volume control with Creative X Hi, I read in one of your posts about an interest info to improve ALSA. I would like to provide some information that might help improve/augment ALSA for me and others using a creative external usb sound "card." I am running Fedora 11, command line (no windowing environment) and I have a Creative Xmod external USB sound card http://us.creative.com/products/product.asp?category=209&subcategory=668&pr… This Creative device outputs sound OK through ALSA, but the Volume control/Mute knob DO NOT CHANGE the audio. Creative does not provide Linux drivers (the knob works to control volume and mute without installing any drivers on a MS Windows setup). In Fedora, turning the volume knob causes "escape characters" to be displayed on the command line. I have included a copy of an cropped lsusb -v log, below. I hope this might be a step towards getting the volume/mute knob to work with ALSA/Fedora. Let me know if I can be of more help. Thanks, Keith [vortexbox.localdomain ~]# lsusb -v Bus 002 Device 002: ID 041e:30d0 Creative Technology, Ltd Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x041e Creative Technology, Ltd idProduct 0x30d0 bcdDevice 1.00 iManufacturer 1 Creative Technology Ltd iProduct 2 Creative Xmod iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 315 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 98mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 159 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 6 wChannelConfig 0x003f Left Front (L) Right Front (R) Center Front (C) Low Freqency Enhancement (LFE) Left Surround (LS) Right Surround (RS) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 12 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 11 bNrInPins 2 baSourceID( 0) 1 baSourceID( 1) 13 bNrChannels 6 wChannelConfig 0x003f Left Front (L) Right Front (R) Center Front (C) Low Freqency Enhancement (LFE) Left Surround (LS) Right Surround (RS) iChannelNames 0 bmControls 0x00 bmControls 0x00 iMixer 0 junk at descriptor end: 00 00 00 AudioControl Interface Descriptor: bLength 18 bDescriptorType 36 bDescriptorSubtype 7 (PROCESSING_UNIT) bUnitID 10 wProcessType 1 bNrPins 1 baSourceID( 0) 11 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 bControlSize 1 bmControls( 0) 0x01 Enable Processing iProcessing 0 Process-Specific 01 03 00 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 5 bSourceID 10 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 6 bSourceID 4 bControlSize 1 bmaControls( 0) 0x00 bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 9 bSourceID 7 bControlSize 1 bmaControls( 0) 0x40 Automatic Gain bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 13 bSourceID 12 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 8 bNrInPins 2 baSource( 0) 6 baSource( 1) 9 iSelector 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 8 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 0 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 6 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 44100 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x021c 1x 540 bytes bInterval 1 bRefresh 0 bSynchAddress 129 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 5 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bDelay 0 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 44100 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x00b4 1x 180 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.11 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 45 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0001 1x 1 bytes bInterval 10 Device Status: 0x0000 (Bus Powered) [vortexbox.localdomain ~]# New Email names for you! Get the Email name you've always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/ Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/

1 0

[alsa-devel] Help with multichannel High Speed USB audio device
by Laurent . 25 Jul '09

25 Jul '09

Dear list, I made an embedded device that has a CODEC with 6 inputs connected to an ARM9 micro-controller that has an integrated high-speed USB controller. Now, I need to write the USB code on that micro-controller so that i can capture those channels in real-time at 24bits / 48 KHz with low latency (ie. sub 10ms), on a computer. I only need my device to work with Linux, and if possible, I would like to avoid writing a device driver on the Linux-side. Is there already such a USB audio driver available that I could nearly use "as is" and to which i would conform ? Would you have some general guideline on how to write the high-speed USB device code ? Would you have any comment on the following : For simplicity, my device is going to be the "clock master" of the system since it is already has the exact clock that drives the CODEC. I will maintain a circular buffer of samples (well, frames of samples) coming from the CODEC. Each time a new frame arrives, i put the samples in the buffer and increase a counter so i know how many frames of samples I have in the buffer, ready to be sent. Each USB micro-frame (125ms/8000Hz), I send all the bytes corresponding to the frames that I have in the circular buffer. In average, 6*3*48000/8000 = 108, but I presume that could jitter slightly, which should not be a problem at all. Thanks in advance, Laurent _________________________________________________________________ Windows Live™ Hotmail®: Celebrate the moment with your favorite sports pics. Check it out. http://www.windowslive.com/Online/Hotmail/Campaign/QuickAdd?ocid=TXT_TAGLM_…

2 2

[alsa-devel] ich10 usb_host vs snd_usb_audio Hercules - irq broadcast.
by Alexey Fisher 24 Jul '09

24 Jul '09

Hallo, i found some time to investigate the issue with my "Hercules USB 5.1 Cannel Audio Adapter". The issue is pretty tricky and depend to host hardware, in my situation this is Intel DG45ID board. Symptoms are: if i use this sound adapter after some time i'll get: irq 16: nobody cared ... Disabling IRQ #16. After closer look i found, it depend on usb port. And if i use one "wrong" usb port on IRQ 18 it will broadcast interrupts to two IRQs 18 and 16, until IRQ 16 will die. The problem is... this happens only with this sound card. So this is not easy to locate bug. And it's working fine on Win Vista and on Mac OS on same hardware.. so it should be software issue. Here is a copy of my post on http://bugzilla.kernel.org/show_bug.cgi?id=13816 If use any usb connector located on irq 18 for usb soundcard it will cause "irq 16: nobody cared", so every thing connected (usb mouse, keyboard ) to irq 16 will be broken. If i connect devices with less traffic (like mouse or keyboard) to irq 18, and usb soundcard to irq 16 or 21 ... every thing will be fine. irqpoll do not do any difference. / irq16|uhci-3 (mouse, keyboard) irq18|ehci-1- irq21|uhci-4 (sound work fine normal) \ irq18|uhci-5 (sound will brake uhci-3) / irq23|uhci-6 (not used) irq23|ehci-2- irq19|uhci-7 (not used) \ irq18|uhci-8 (not used) ========================================================================= [ 315.056793] irq 16: nobody cared (try booting with the "irqpoll" option) [ 315.056799] Pid: 0, comm: swapper Not tainted 2.6.31-rc3-00216-gea11951-dirty #98 [ 315.056802] Call Trace: [ 315.056804] <IRQ> [<ffffffff8109d666>] __report_bad_irq+0x26/0xa0 [ 315.056814] [<ffffffff8109d86c>] note_interrupt+0x18c/0x1d0 [ 315.056819] [<ffffffff8106a373>] ? sched_clock_tick+0x73/0xb0 [ 315.056823] [<ffffffff8109deb5>] handle_fasteoi_irq+0xb5/0xe0 [ 315.056827] [<ffffffff8100dffd>] handle_irq+0x1d/0x30 [ 315.056830] [<ffffffff8100d527>] do_IRQ+0x67/0xe0 [ 315.056834] [<ffffffff8100c013>] ret_from_intr+0x0/0xa [ 315.056837] <EOI> [<ffffffff812822e1>] ? acpi_idle_enter_bm+0x256/0x288 [ 315.056844] [<ffffffff812822de>] ? acpi_idle_enter_bm+0x253/0x288 [ 315.056849] [<ffffffff813a480b>] ? cpuidle_idle_call+0x9b/0xf0 [ 315.056853] [<ffffffff8100a59c>] ? cpu_idle+0xac/0x100 [ 315.056858] [<ffffffff81496825>] ? rest_init+0x65/0x70 [ 315.056862] [<ffffffff8172ed35>] ? start_kernel+0x3ba/0x3c5 [ 315.056866] [<ffffffff8172e315>] ? x86_64_start_reservations+0x125/0x129 [ 315.056870] [<ffffffff8172e3fd>] ? x86_64_start_kernel+0xe4/0xeb [ 315.056872] handlers: [ 315.056874] [<ffffffff8134bdb0>] (usb_hcd_irq+0x0/0x70) [ 315.056878] Disabling IRQ #16 ================================================================== I did smole script to test haw interrupts are used: $for i in $(seq 1 1000); do cat /proc/interrupts >> test; sleep 10s; done $grep usb[3,4,5] test there is no connections on usb3. on usb4 are keyboard an mouse. on usb5 is the soundcard. ====================================================================== -------after boot. there is too match interrupts on 16, -------there is nothing connected. 16: 4276 4520 IO-APIC-fasteoi uhci_hcd:usb3 18: 2732 2598 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 856 816 IO-APIC-fasteoi uhci_hcd:usb4 -------here i begin to play to usb sound card on usb5. 16: 12162 10987 IO-APIC-fasteoi uhci_hcd:usb3 18: 8070 7134 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 887 875 IO-APIC-fasteoi uhci_hcd:usb4 16: 20097 17784 IO-APIC-fasteoi uhci_hcd:usb3 18: 13539 11674 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1003 958 IO-APIC-fasteoi uhci_hcd:usb4 ------- count of interrupts on irq 16 is almost double of irq18 16: 28083 24881 IO-APIC-fasteoi uhci_hcd:usb3 18: 19091 16130 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1005 968 IO-APIC-fasteoi uhci_hcd:usb4 ...... ...... 16: 94758 81082 IO-APIC-fasteoi uhci_hcd:usb3 18: 61430 53863 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1389 1539 IO-APIC-fasteoi uhci_hcd:usb4 16: 103141 87919 IO-APIC-fasteoi uhci_hcd:usb3 18: 66678 58625 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1389 1539 IO-APIC-fasteoi uhci_hcd:usb4 ------- irq 16: nobody cared, Disabling IRQ #16. 16: 107840 92161 IO-APIC-fasteoi uhci_hcd:usb3 18: 71854 63458 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1389 1539 IO-APIC-fasteoi uhci_hcd:usb4 16: 107840 92161 IO-APIC-fasteoi uhci_hcd:usb3 18: 76498 68823 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1389 1539 IO-APIC-fasteoi uhci_hcd:usb4

2 2

[alsa-devel] snd_pcm_hw_params_set_period_time_near
by Guilherme Raymo Longo 24 Jul '09

24 Jul '09

Mates, I can´t find the real meaning of this function. snd_pcm_hw_params_set_period_time_near At first I thought that it was used to set the time between 2 consecutive samples, but it is not. Does anyone could gimme some help with that? Tks

1 0

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 24 Jul '09

24 Jul '09

The following changes since commit 416c8fe3cdcd2d46a6ca12d8ec6143c62d8569d9: Takashi Iwai (1): ASoC: Kill direct accesses to driver_data are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Candelaria Villareal, Jorge (1): ASoC: SDP3430: Add support for EXTMUTE using TWL GPIO6 Chaithrika U S (1): ASoC: tlv320aic3x: Enable PLL when not bypassed Janusz Krzysztofik (2): ASoC: Add support for Conexant CX20442-11 voice modem codec ASoC: Jack handling enhancements as suggested by subsystem maintainer John Bonesio (1): ASoC: MPC5200: Increase the delay time between resets Joonyoung Shim (2): ASoC: MAX9877: separate callback functions ASoC: MAX9877: fix write operation for register Lopez Cruz, Misael (1): ASoC: OMAP: Staticise pcm creation function of omap-pcm Marek Vasut (2): ALSA: Allow passing platform_data to devices attached to AC97 bus ASoC: Switch palm27x-asoc to jack detection api Mark Brown (4): ASoC: Bodge around GCC 4.4.0 flow analysis bug in GCC 4.4.0 ASoC: Fix FLL reference clock division setup in WM8993 ASoC: Fix checkpatch issues in AD1938 Merge branch 'for-2.6.31' into for-2.6.32 include/sound/ac97_codec.h | 6 + include/sound/soc-dai.h | 1 + sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad1938.c | 156 ++++++++-------- sound/soc/codecs/cx20442.c | 395 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cx20442.h | 19 ++ sound/soc/codecs/max9877.c | 110 +++++++---- sound/soc/codecs/tlv320aic3x.c | 11 +- sound/soc/codecs/wm8400.c | 9 +- sound/soc/codecs/wm8993.c | 2 + sound/soc/fsl/mpc5200_psc_ac97.c | 3 +- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/omap/sdp3430.c | 9 + sound/soc/pxa/palm27x.c | 204 +++++++------------- sound/soc/soc-core.c | 6 +- sound/soc/soc-jack.c | 20 ++- 17 files changed, 703 insertions(+), 255 deletions(-) create mode 100644 sound/soc/codecs/cx20442.c create mode 100644 sound/soc/codecs/cx20442.h

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