October 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] NULL pointer deref when reloading snd_soc_simple_card
by Felipe Balbi 22 Oct '15

22 Oct '15

Hi, I just triggered a NULL point deref with the following commands running on AM437x SK board. This is with v4.3-rc6: modprobe -r snd_soc_simple_card sleep 1 modprobe snd_soc_simple_card sleep 1 details below: [ 228.020921] Unable to handle kernel NULL pointer dereference at virtual address 000000f8 [ 228.029546] pgd = ed4bc000 [ 228.032375] [000000f8] *pgd=00000000 [ 228.036154] Internal error: Oops: 5 [#1] SMP ARM [ 228.040968] Modules linked in: snd_soc_simple_card(+) matrix_keypad matrix_keymap pwm_bl xhci_plat_hcd xhci_hcd usbcore joydev m25p80 spi_nor lis3lv02d_i2c lis3lv02d input_polldev cpufreq_dt thermal_sys hwmon dwc3_omap extcon tps65218_pwrbutton omap_wdt spi_ti_qspi evdev rtc_omap leds_gpio led_class dwc3 udc_core usb_common omapfb cfbfillrect cfbimgblt cfbcopyarea panel_dpi snd_soc_tlv320aic3x snd_soc_davinci_mcasp snd_soc_edma snd_soc_omap snd_soc_core omapdss snd_compress snd_pcm_dmaengine snd_pcm pwm_tiecap snd_timer snd soundcore phy_omap_usb2 autofs4 [last unloaded: snd_soc_simple_card] [ 228.096008] CPU: 0 PID: 710 Comm: modprobe Not tainted 4.3.0-rc6-00001-gada6475ae6e4 #97 [ 228.104436] Hardware name: Generic AM43 (Flattened Device Tree) [ 228.110608] task: ed4b9140 ti: ed52e000 task.ti: ed52e000 [ 228.116370] PC is at dapm_wcache_lookup+0x50/0x7c [snd_soc_core] [ 228.122664] LR is at dapm_wcache_lookup+0x38/0x7c [snd_soc_core] [ 228.128922] pc : [<bf16dd2c>] lr : [<bf16dd14>] psr: a0070013 [ 228.128922] sp : ed52fba8 ip : 00000005 fp : 00000000 [ 228.140883] r10: bf17d238 r9 : bf1f138c r8 : bf1f616c [ 228.146327] r7 : bf1f138c r6 : bf1f616c r5 : ee6f5158 r4 : 000000f4 [ 228.153126] r3 : 00000100 r2 : 00000052 r1 : ed1057c1 r0 : ffffffff [ 228.159925] Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none [ 228.167354] Control: 10c5387d Table: ad4bc059 DAC: 00000051 [ 228.173339] Process modprobe (pid: 710, stack limit = 0xed52e218) [ 228.179689] Stack: (0xed52fba8 to 0xed530000) [ 228.184233] fba0: 00000000 ee7c58c0 bf1f6050 bf16f244 00000000 ed52fc28 [ 228.192765] fbc0: ee577800 c1014a44 00000000 c06430d8 00000001 00000000 ee6c6e90 ee6f50a0 [ 228.201294] fbe0: 00000000 00000000 00000000 c09c3354 60070013 c00943bc ed4b9700 00000004 [ 228.209835] fc00: 00000004 ed4b9140 00000006 bf1f138c bf1ef834 c0091884 c063f37c ed4b9140 [ 228.218365] fc20: 00000001 c118a28c ed4b9140 c00919e8 ee6f506c 60070013 ee6f5070 c063f37c [ 228.226899] fc40: 00000001 00000000 bf16f430 00000003 00000004 ed4b9140 00000006 00000000 [ 228.235417] fc60: bf1ef834 c0091884 c0641060 bf1f138c 00000000 ee7c58c0 00000004 0000001b [ 228.243938] fc80: bf1f138c bf17d238 00000000 bf16f468 00000000 00000012 ee7c58c0 ee7c58c0 [ 228.252463] fca0: ee6c6e90 ee6c6e90 00000004 ee6c6ec0 00000000 bf1ef834 00000000 bf1efcec [ 228.260989] fcc0: ee7c5828 00000000 ee7c5810 ee6f5010 ee7c58c0 ee7c5858 ed496010 bf169ef4 [ 228.269506] fce0: ee6f5180 00000002 00000634 ee6f5010 00000000 00000000 00000000 00000000 [ 228.278029] fd00: ed440e0c bf16cb2c 00000000 ee6f5020 ee6f5180 bf17fd60 00000001 ee6f5028 [ 228.286547] fd20: ee6f5168 60070013 00000000 00000000 eeee3304 00000000 ee178410 ee6f5010 [ 228.295065] fd40: ed105c90 ee6f5010 ee178410 ee178410 ee178400 00000001 12f9f228 bf17968c [ 228.303583] fd60: ee6f5010 fffffdfb 00000001 eeee87d8 ee178410 bf0faa38 00000000 ee178410 [ 228.312108] fd80: ee178410 ee178410 ee178410 bf0fb28c fffffdfb 0000004e ed060e00 c03dff28 [ 228.320626] fda0: ee178410 c11be018 bf0fb28c 00000000 0000004e c03de5dc ee178410 bf0fb28c [ 228.329143] fdc0: ee178444 c098df20 00000000 c03de76c 00000000 bf0fb28c c03de6d8 c03dca40 [ 228.337666] fde0: ee0362a4 ee179f10 bf0fb28c ed462ec0 00000000 c03ddba4 bf0fb080 c09123a0 [ 228.346188] fe00: ed060d40 bf0fb28c c09123a0 ed060d40 bf0fd000 c03defb0 c09123a0 c09123a0 [ 228.354713] fe20: ed060d40 c0009804 60070093 00000000 00000000 00000000 0000000f 00000000 [ 228.363240] fe40: ef7c4464 40000000 0000002e c0091ccc ed060e40 000000d0 000000d0 c0162850 [ 228.371760] fe60: ed52ff58 c0091ccc c090e108 ee0000c0 a0070013 bf0fb300 bf0fb300 c09c34a8 [ 228.380285] fe80: ed060e40 bf0fb300 bf0fb348 00000001 12f9f228 c011b55c ed060e08 bf0fb300 [ 228.388815] fea0: ed52ff58 c09c34a8 ed060e08 c00cc6cc bf0fb30c 00007fff 00000000 c00c9e60 [ 228.397340] fec0: c119bfa4 bf0fb458 c090e990 bf0fb51c f07fa7bc bf0fb30c 00000000 c064c2c0 [ 228.405863] fee0: f07cd000 0002d80c 02e60649 00000000 0000000f 00000000 00000000 00000000 [ 228.414388] ff00: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 228.422911] ff20: 00000000 00000000 00000000 00000000 00000170 00000000 00000003 7f606ddc [ 228.431432] ff40: 0000017b c000f8e4 ed52e000 00000000 7f61a2e8 c00ccf24 f07cd000 0002d80c [ 228.439952] ff60: f07fa0dc f07eebe5 f07ef600 00001690 000019d0 00000000 00000000 00000000 [ 228.448475] ff80: 0000002c 0000002d 00000014 00000018 0000000f 00000000 7f607a28 00000000 [ 228.457000] ffa0: 1c7d6500 c000f740 7f607a28 00000000 00000003 7f606ddc 00000000 7f607d10 [ 228.465523] ffc0: 7f607a28 00000000 1c7d6500 0000017b 00040000 00000000 00000000 7f61a2e8 [ 228.474050] ffe0: bed179b0 bed179a0 7f5fd4bb b6f18852 80070030 00000003 185b0001 05000135 [ 228.482659] [<bf16dd2c>] (dapm_wcache_lookup [snd_soc_core]) from [<bf16f244>] (snd_soc_dapm_add_route+0x74/0x23c [snd_soc_core]) [ 228.494875] [<bf16f244>] (snd_soc_dapm_add_route [snd_soc_core]) from [<bf16f468>] (snd_soc_dapm_add_routes+0x5c/0xbc [snd_soc_core]) [ 228.507428] [<bf16f468>] (snd_soc_dapm_add_routes [snd_soc_core]) from [<bf1efcec>] (aic3x_probe+0x41c/0x57c [snd_soc_tlv320aic3x]) [ 228.519797] [<bf1efcec>] (aic3x_probe [snd_soc_tlv320aic3x]) from [<bf169ef4>] (soc_probe_component+0x208/0x348 [snd_soc_core]) [ 228.531812] [<bf169ef4>] (soc_probe_component [snd_soc_core]) from [<bf16cb2c>] (snd_soc_register_card+0x8d8/0x10f4 [snd_soc_core]) [ 228.544204] [<bf16cb2c>] (snd_soc_register_card [snd_soc_core]) from [<bf17968c>] (devm_snd_soc_register_card+0x2c/0x68 [snd_soc_core]) [ 228.556948] [<bf17968c>] (devm_snd_soc_register_card [snd_soc_core]) from [<bf0faa38>] (asoc_simple_card_probe+0x1f4/0x434 [snd_soc_simple_card]) [ 228.570587] [<bf0faa38>] (asoc_simple_card_probe [snd_soc_simple_card]) from [<c03dff28>] (platform_drv_probe+0x44/0xac) [ 228.581921] [<c03dff28>] (platform_drv_probe) from [<c03de5dc>] (driver_probe_device+0x1f4/0x2f0) [ 228.591170] [<c03de5dc>] (driver_probe_device) from [<c03de76c>] (__driver_attach+0x94/0x98) [ 228.599962] [<c03de76c>] (__driver_attach) from [<c03dca40>] (bus_for_each_dev+0x6c/0xa0) [ 228.608485] [<c03dca40>] (bus_for_each_dev) from [<c03ddba4>] (bus_add_driver+0x18c/0x214) [ 228.617100] [<c03ddba4>] (bus_add_driver) from [<c03defb0>] (driver_register+0x78/0xf8) [ 228.625451] [<c03defb0>] (driver_register) from [<c0009804>] (do_one_initcall+0x80/0x1dc) [ 228.633985] [<c0009804>] (do_one_initcall) from [<c011b55c>] (do_init_module+0x5c/0x1d0) [ 228.642421] [<c011b55c>] (do_init_module) from [<c00cc6cc>] (load_module+0x1a4c/0x20c0) [ 228.650769] [<c00cc6cc>] (load_module) from [<c00ccf24>] (SyS_finit_module+0x7c/0x90) [ 228.658936] [<c00ccf24>] (SyS_finit_module) from [<c000f740>] (ret_fast_syscall+0x0/0x1c) [ 228.667457] Code: e594300c e1550003 e243400c 0a000006 (e5131008) [ 228.673935] ---[ end trace d70ffb1b3028bdb3 ]--- -- balbi

1 0

[alsa-devel] ALSA DMA ping-pong
by Ran Shalit 22 Oct '15

22 Oct '15

Hello, I hope someone can help me understand the concept of ping-pong dma with ALSA. I am using davinci soc: /sound/soc/davinci/davinci_pcm.c But I am not sure if it using ping-pong dma or not. In the header of "davinci_pcm_enqueue_dma" function it states "Not used with ping/pong", but inside I see that it use period parameter in order to set the offset inside the dma buffer which is used. So, does it mean that ping-pong is used or not ? /* * Not used with ping/pong */ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) ..... period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; dma_pos = runtime->dma_addr + dma_offset; .... acnt = prtd->params->acnt; edma_set_src(link, src, INCR, W8BIT); edma_set_dest(link, dst, INCR, W8BIT); ...... } Regards, Ran

1 0

[alsa-devel] [PATCH v2 0/3] ASoC: rockchip: i2s: add 8 channels capture support and share lrck
by Sugar Zhang 22 Oct '15

22 Oct '15

support 8 channels capture support. share lrck tx/rx when symmetric_rates enabled. Sugar Zhang (3): ASoC: rockchip: i2s: add 8 channels capture support Documentation: DT bindings: rockchip-i2s: add capture property ASoC: rockchip: i2s: share tx/rx lrck when symmetric_rates enabled .../devicetree/bindings/sound/rockchip-i2s.txt | 2 + sound/soc/rockchip/rockchip_i2s.c | 48 +++++++++++++++++++++- sound/soc/rockchip/rockchip_i2s.h | 16 ++++++++ 3 files changed, 64 insertions(+), 2 deletions(-) -- 2.3.6

3 9

[alsa-devel] [PATCH] ALSA: Remove transfer_ack_{begin, end} callbacks from struct snd_pcm_runtime
by Lars-Peter Clausen 22 Oct '15

22 Oct '15

While there is nothing wrong with the transfer_ack_begin and transfer_ack_end callbacks per-se, the last documented user was part of the alsa-driver 0.5.12a package, which was released 14 years ago and even predates the upstream integration of the ALSA core and has subsequently been superseded by newer alsa-driver releases. This seems to indicate that there is no need for having these callbacks and they are just cruft that can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Documentation/DocBook/writing-an-alsa-driver.tmpl | 19 ++----------------- include/sound/pcm.h | 4 ---- sound/core/pcm_lib.c | 5 ----- 3 files changed, 2 insertions(+), 26 deletions(-) diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl index 84ef6a9..a27ab9f5 100644 --- a/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -2181,10 +2181,6 @@ struct _snd_pcm_runtime { struct snd_pcm_hardware hw; struct snd_pcm_hw_constraints hw_constraints; - /* -- interrupt callbacks -- */ - void (*transfer_ack_begin)(struct snd_pcm_substream *substream); - void (*transfer_ack_end)(struct snd_pcm_substream *substream); - /* -- timer -- */ unsigned int timer_resolution; /* timer resolution */ @@ -2209,9 +2205,8 @@ struct _snd_pcm_runtime { For the operators (callbacks) of each sound driver, most of these records are supposed to be read-only. Only the PCM middle-layer changes / updates them. The exceptions are - the hardware description (hw), interrupt callbacks - (transfer_ack_xxx), DMA buffer information, and the private - data. Besides, if you use the standard buffer allocation + the hardware description (hw) DMA buffer information and the + private data. Besides, if you use the standard buffer allocation method via <function>snd_pcm_lib_malloc_pages()</function>, you don't need to set the DMA buffer information by yourself. </para> @@ -2538,16 +2533,6 @@ struct _snd_pcm_runtime { </para> </section> - <section id="pcm-interface-runtime-intr"> - <title>Interrupt Callbacks</title> - <para> - The field <structfield>transfer_ack_begin</structfield> and - <structfield>transfer_ack_end</structfield> are called at - the beginning and at the end of - <function>snd_pcm_period_elapsed()</function>, respectively. - </para> - </section> - </section> <section id="pcm-interface-operators"> diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2882ddd..3e0ffd2 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -402,10 +402,6 @@ struct snd_pcm_runtime { struct snd_pcm_hardware hw; struct snd_pcm_hw_constraints hw_constraints; - /* -- interrupt callbacks -- */ - void (*transfer_ack_begin)(struct snd_pcm_substream *substream); - void (*transfer_ack_end)(struct snd_pcm_substream *substream); - /* -- timer -- */ unsigned int timer_resolution; /* timer resolution */ int tstamp_type; /* timestamp type */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6dc4277..05a3ca9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1875,9 +1875,6 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) return; runtime = substream->runtime; - if (runtime->transfer_ack_begin) - runtime->transfer_ack_begin(substream); - snd_pcm_stream_lock_irqsave(substream, flags); if (!snd_pcm_running(substream) || snd_pcm_update_hw_ptr0(substream, 1) < 0) @@ -1889,8 +1886,6 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) #endif _end: snd_pcm_stream_unlock_irqrestore(substream, flags); - if (runtime->transfer_ack_end) - runtime->transfer_ack_end(substream); kill_fasync(&runtime->fasync, SIGIO, POLL_IN); } -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: rt5645: Add the HWEQ for the customized speaker output
by Oder Chiou 22 Oct '15

22 Oct '15

The patch adds the HWEQ function for the customized speaker output that is in order to the customer's requirement. The customer can enable the HWEQ in the platform data, and set the parameters of the HWEQ in the ALSA binary control byte-by-byte. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5645.h | 2 + sound/soc/codecs/rt5645.c | 109 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 111 insertions(+) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index a5cf615..efa9f08 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -23,6 +23,8 @@ struct rt5645_platform_data { unsigned int jd_mode; /* Invert JD when jack insert */ bool jd_invert; + /* Speaker HWEQ */ + bool hweq; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4d54999..f26f08b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -42,6 +42,8 @@ #define RT5645_PR_BASE (RT5645_PR_RANGE_BASE + (0 * RT5645_PR_SPACING)) +#define RT5645_HWEQ_NUM 57 + static const struct regmap_range_cfg rt5645_ranges[] = { { .name = "PR", @@ -224,6 +226,11 @@ static const struct reg_default rt5645_reg[] = { { 0xff, 0x6308 }, }; +struct rt5645_eq_param_s { + unsigned short reg; + unsigned short val; +}; + static const char *const rt5645_supply_names[] = { "avdd", "cpvdd", @@ -240,6 +247,7 @@ struct rt5645_priv { struct snd_soc_jack *btn_jack; struct delayed_work jack_detect_work; struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; + struct rt5645_eq_param_s *eq_param; int codec_type; int sysclk; @@ -477,6 +485,80 @@ static const DECLARE_TLV_DB_RANGE(spk_clsd_tlv, 7, 7, TLV_DB_SCALE_ITEM(228, 0, 0) ); +static int rt5645_hweq_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s); + + return 0; +} + +static int rt5645_hweq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + struct rt5645_eq_param_s *eq_param = + (struct rt5645_eq_param_s *)ucontrol->value.bytes.data; + + if (rt5645->pdata.hweq) + memcpy(eq_param, rt5645->eq_param, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s)); + + return 0; +} + +static bool rt5645_validate_hweq(unsigned short reg) +{ + if ((reg >= 0x1a4 && reg <= 0x1cd) | (reg >= 0x1e5 && reg <= 0x1f8) | + (reg == RT5645_EQ_CTRL2)) + return true; + + return false; +} + +static int rt5645_hweq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + struct rt5645_eq_param_s *eq_param = + (struct rt5645_eq_param_s *)ucontrol->value.bytes.data; + int i; + + if (rt5645->pdata.hweq) { + /* The final setting of the table should be RT5645_EQ_CTRL2 */ + for (i = RT5645_HWEQ_NUM - 1; i >= 0; i--) { + if (eq_param[i].reg == 0) + continue; + else if (eq_param[i].reg != RT5645_EQ_CTRL2) + return 0; + else + break; + } + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + if (!rt5645_validate_hweq(eq_param[i].reg) && + eq_param[i].reg != 0) + return 0; + else if (eq_param[i].reg == 0) + break; + } + + memcpy(rt5645->eq_param, eq_param, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s)); + } + + return 0; +} + +#define RT5645_HWEQ(xname) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = rt5645_hweq_info, \ + .get = rt5645_hweq_get,\ + .put = rt5645_hweq_put } + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, @@ -541,6 +623,7 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* I2S2 function select */ SOC_SINGLE("I2S2 Func Switch", RT5645_GPIO_CTRL1, RT5645_I2S2_SEL_SFT, 1, 1), + RT5645_HWEQ("Speaker HWEQ"), }; /** @@ -631,6 +714,22 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +static int rt5645_enable_hweq(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + if (rt5645_validate_hweq(rt5645->eq_param[i].reg)) + regmap_write(rt5645->regmap, rt5645->eq_param[i].reg, + rt5645->eq_param[i].val); + else + break; + } + + return 0; +} + /** * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters * @codec: SoC audio codec device. @@ -1532,9 +1631,12 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: + if (rt5645->pdata.hweq) + rt5645_enable_hweq(codec); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, @@ -1543,6 +1645,8 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + if (rt5645->pdata.hweq) + snd_soc_write(codec, RT5645_EQ_CTRL2, 0); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, 0); @@ -3060,6 +3164,11 @@ static int rt5645_probe(struct snd_soc_codec *codec) snd_soc_dapm_sync(dapm); } + if (rt5645->pdata.hweq) + rt5645->eq_param = devm_kzalloc(codec->dev, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s), + GFP_KERNEL); + return 0; } -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [patch] ASoC: es8328: harmless underflow in es8328_put_deemph()
by Dan Carpenter 22 Oct '15

22 Oct '15

Valid values for "deemph" are zero and one but we accidentally allow negative values as well. It's harmless but it causes static checker warnings and we may as well clean it up. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 6a09101..969e337 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -129,7 +129,7 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; int ret; if (deemph > 1)

2 2

[alsa-devel] [patch] ASoC: wm8960: harmless underflow in wm8960_put_deemph()
by Dan Carpenter 22 Oct '15

22 Oct '15

Valid values of "deemph" are zero and one, but we accidentally allow negative values as well. It's harmless but we may as well fix it. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e3b7d0c..dc47c2f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -201,7 +201,7 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int deemph = ucontrol->value.integer.value[0]; + unsigned int deemph = ucontrol->value.integer.value[0]; if (deemph > 1) return -EINVAL;

2 1

[alsa-devel] [PATCH v4] ASoC: soc-compress: add config item for soc-compress to make it compiled only when needed
by Jie Yang 22 Oct '15

22 Oct '15

We don't always need soc-compress in soc, here add a config item SND_SOC_COMPRESS, when nobody select it, the soc-compress will not be compiled. Here also change Kconfig to 'select SND_SOC_COMPRESS' for drivers that needed soc-compress. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/soc-dai.h | 2 +- include/sound/soc.h | 4 +++- sound/soc/Kconfig | 5 ++++- sound/soc/Makefile | 3 ++- sound/soc/intel/Kconfig | 1 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/soc-compress.c | 12 ++++++++++-- sound/soc/soc-core.c | 4 ++-- 8 files changed, 24 insertions(+), 9 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1..238200f 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -214,7 +214,7 @@ struct snd_soc_dai_driver { int (*suspend)(struct snd_soc_dai *dai); int (*resume)(struct snd_soc_dai *dai); /* compress dai */ - bool compress_dai; + int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num); /* DAI is also used for the control bus */ bool bus_control; diff --git a/include/sound/soc.h b/include/sound/soc.h index 98e080b..10b7802 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -459,7 +459,9 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, int snd_soc_platform_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); -int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +#ifdef CONFIG_SND_SOC_COMPRESS +int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +#endif struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 7de792b..7ff7d88 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -9,7 +9,6 @@ menuconfig SND_SOC select SND_JACK if INPUT=y || INPUT=SND select REGMAP_I2C if I2C select REGMAP_SPI if SPI_MASTER - select SND_COMPRESS_OFFLOAD ---help--- If you want ASoC support, you should say Y here and also to the @@ -30,6 +29,10 @@ config SND_SOC_GENERIC_DMAENGINE_PCM bool select SND_DMAENGINE_PCM +config SND_SOC_COMPRESS + bool + select SND_COMPRESS_OFFLOAD + config SND_SOC_TOPOLOGY bool diff --git a/sound/soc/Makefile b/sound/soc/Makefile index af0a571..8eb06db 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,6 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o ifneq ($(CONFIG_SND_SOC_TOPOLOGY),) snd-soc-core-objs += soc-topology.o diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 664df1f..aaafb13 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -12,6 +12,7 @@ config SND_MFLD_MACHINE config SND_SST_MFLD_PLATFORM tristate + select SND_SOC_COMPRESS config SND_SST_IPC tristate diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 5e9c316..0487cfa 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -512,7 +512,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { }, { .name = "compress-cpu-dai", - .compress_dai = 1, + .compress_new = snd_soc_new_compress, .ops = &sst_compr_dai_ops, .playback = { .stream_name = "Compress Playback", diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 025c38f..12a9820 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -612,8 +612,15 @@ static struct snd_compr_ops soc_compr_dyn_ops = { .get_codec_caps = soc_compr_get_codec_caps }; -/* create a new compress */ -int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +/** + * snd_soc_new_compress - create a new compress. + * + * @rtd: The runtime for which we will create compress + * @num: the device index number (zero based - shared with normal PCMs) + * + * Return: 0 for success, else error. + */ +int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; @@ -703,3 +710,4 @@ compr_err: kfree(compr); return ret; } +EXPORT_SYMBOL_GPL(snd_soc_new_compress); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3b471f9..24b0960 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1370,9 +1370,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) soc_dpcm_debugfs_add(rtd); #endif - if (cpu_dai->driver->compress_dai) { + if (cpu_dai->driver->compress_new) { /*create compress_device"*/ - ret = soc_new_compress(rtd, num); + ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Let snd_soc_limit_volume() take a snd_soc_card
by Lars-Peter Clausen 22 Oct '15

22 Oct '15

snd_soc_limit_volume() operates on a card and the CODEC that is passed in is only used to look up the card. Let it directly take the card instead. This makes it possible to use it when no snd_soc_codec is available. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 2 +- sound/soc/omap/rx51.c | 2 +- sound/soc/soc-ops.c | 8 ++++---- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index bd85564..8b06678 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -612,7 +612,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_soc_limit_volume(struct snd_soc_codec *codec, +int snd_soc_limit_volume(struct snd_soc_card *card, const char *name, int max); int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 3bebfb1..9953890 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -297,7 +297,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) dev_err(card->dev, "Failed to add TPA6130A2 controls\n"); return err; } - snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); + snd_soc_limit_volume(card, "TPA6130A2 Headphone Playback Volume", 42); err = omap_mcbsp_st_add_controls(rtd, 2); if (err < 0) { diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 05977ae..ecd38e5 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -588,16 +588,16 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); /** * snd_soc_limit_volume - Set new limit to an existing volume control. * - * @codec: where to look for the control + * @card: where to look for the control * @name: Name of the control * @max: new maximum limit * * Return 0 for success, else error. */ -int snd_soc_limit_volume(struct snd_soc_codec *codec, +int snd_soc_limit_volume(struct snd_soc_card *card, const char *name, int max) { - struct snd_card *card = codec->component.card->snd_card; + struct snd_card *snd_card = card->snd_card; struct snd_kcontrol *kctl; struct soc_mixer_control *mc; int found = 0; @@ -607,7 +607,7 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec, if (unlikely(!name || max <= 0)) return -EINVAL; - list_for_each_entry(kctl, &card->controls, list) { + list_for_each_entry(kctl, &snd_card->controls, list) { if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { found = 1; break; -- 2.1.4

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[alsa-devel] [PATCH v2 0/2] *** wm8962 regmap related fix ***
by Jiada Wang 22 Oct '15

22 Oct '15

This patch set aims to fix issues in wm8962 codec driver related to regmap, currently any attempt to read from ALC Coefficient register will fail when wm8962 is in suspend mode. As ALC2 register is volatile register, it can't be read when cache_only flag is set. Another issue is, if wm8962's regulator is set to 'regulator-always-on' mode, then after wm8962 is resumed from suspend, wm8962 codec is reset, but cache_dirty flag isn't set, this cause difference between actual wm8962 HW and regmap cache. Changeset: -------------- v1 -> v2 * removed comment before regcache_mark_dirty Jiada Wang (2): ASoC: WM8962: mark cache_dirty flag after software reset in pm_resume ASoC: Codec: wm8962: declare ALC Coefficients as 4 separate registers sound/soc/codecs/wm8962.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) -- 2.4.5

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