From c593b520cf70b0672680da04cc1e8c5f93bd739d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 27 Oct 2010 20:11:17 -0700 Subject: [PATCH 01/17] ASoC: Check return value of struct_strtoul() in pmdown_time_set() strict_strtoul() has just been made must check so do so. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 70d9a7394b2b..805343fe903b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -165,8 +165,11 @@ static ssize_t pmdown_time_set(struct device *dev, { struct snd_soc_pcm_runtime *rtd = container_of(dev, struct snd_soc_pcm_runtime, dev); + int ret; - strict_strtol(buf, 10, &rtd->pmdown_time); + ret = strict_strtol(buf, 10, &rtd->pmdown_time); + if (ret) + return ret; return count; } From 911a0f0bfc01750590e8ac6e7f9f4921f470b0d1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 26 Oct 2010 11:45:59 +0300 Subject: [PATCH 02/17] ASoC: tlv320dac33: Error handling for broken chip Correct/Implement handling of broken chip. Fail the soc_prope if the communication with the chip fails (can not read chip ID). Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 26 +++++++++++++++++++------- 1 file changed, 19 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index d251ff54a2d3..fed14582b498 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -200,7 +200,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg, u8 *value) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); - int val; + int val, ret = 0; *value = reg & 0xff; @@ -210,6 +210,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg, if (val < 0) { dev_err(codec->dev, "Read failed (%d)\n", val); value[0] = dac33_read_reg_cache(codec, reg); + ret = val; } else { value[0] = val; dac33_write_reg_cache(codec, reg, val); @@ -218,7 +219,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg, value[0] = dac33_read_reg_cache(codec, reg); } - return 0; + return ret; } static int dac33_write(struct snd_soc_codec *codec, unsigned int reg, @@ -329,13 +330,18 @@ static void dac33_init_chip(struct snd_soc_codec *codec) dac33_read_reg_cache(codec, DAC33_LINER_TO_RLO_VOL)); } -static inline void dac33_read_id(struct snd_soc_codec *codec) +static inline int dac33_read_id(struct snd_soc_codec *codec) { + int i, ret = 0; u8 reg; - dac33_read(codec, DAC33_DEVICE_ID_MSB, ®); - dac33_read(codec, DAC33_DEVICE_ID_LSB, ®); - dac33_read(codec, DAC33_DEVICE_REV_ID, ®); + for (i = 0; i < 3; i++) { + ret = dac33_read(codec, DAC33_DEVICE_ID_MSB + i, ®); + if (ret < 0) + break; + } + + return ret; } static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) @@ -1414,9 +1420,15 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) dev_err(codec->dev, "Failed to power up codec: %d\n", ret); goto err_power; } - dac33_read_id(codec); + ret = dac33_read_id(codec); dac33_hard_power(codec, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to read chip ID: %d\n", ret); + ret = -ENODEV; + goto err_power; + } + /* Check if the IRQ number is valid and request it */ if (dac33->irq >= 0) { ret = request_irq(dac33->irq, dac33_interrupt_handler, From d54e1f4fdf4cf9754b7220ae4cb66dcae0fc1702 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Oct 2010 14:07:25 +0300 Subject: [PATCH 03/17] ASoC: tlv320dac33: Limit the US_TO_SAMPLES macro Limit the time window to maximum 1s in the macro. The driver deals with much shorter times (<200ms). This will fix a rare division by zero bug in Mode1. This could happen, when the work is not executed in time (within mode1_latency) after the interrupt. In this case the DAC33 will not receive the needed nSample command in time, and enters to an unknown state, and won't recover. In such event the time window will increase, and eventually going to be bigger than 1s, resulting devision by zero. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index fed14582b498..c47c20d21ea5 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -58,7 +58,7 @@ (1000000000 / ((rate * 1000) / samples)) #define US_TO_SAMPLES(rate, us) \ - (rate / (1000000 / us)) + (rate / (1000000 / (us < 1000000 ? us : 1000000))) #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \ ((samples * 5000) / ((burstrate * 5000) / (burstrate - playrate))) From 1bc13b2e3518ff7856924d7c2bdf06196f605260 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 29 Oct 2010 09:49:37 +0300 Subject: [PATCH 04/17] ASoC: tlv320dac33: Mode1 FIFO auto configuration fix Do not allow invalid (too big) nSample value, when FIFO Mode1 and automatic fifo configuration has been selected. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index c47c20d21ea5..c5ab8c805771 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1082,6 +1082,9 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) /* Number of samples under i2c latency */ dac33->alarm_threshold = US_TO_SAMPLES(rate, dac33->mode1_latency); + nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - + dac33->alarm_threshold; + if (dac33->auto_fifo_config) { if (period_size <= dac33->alarm_threshold) /* @@ -1092,6 +1095,8 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) ((dac33->alarm_threshold / period_size) + (dac33->alarm_threshold % period_size ? 1 : 0)); + else if (period_size > nsample_limit) + dac33->nsample = nsample_limit; else dac33->nsample = period_size; } else { @@ -1103,8 +1108,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) */ dac33->nsample_max = substream->runtime->buffer_size - period_size; - nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - - dac33->alarm_threshold; + if (dac33->nsample_max > nsample_limit) dac33->nsample_max = nsample_limit; From 63f7526f26f0a9291ac3f7a986aa18ebfb61ec19 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 28 Oct 2010 14:05:40 +0300 Subject: [PATCH 05/17] ASoC: tpa6130a2: Fix unbalanced regulator disables This driver has unbalanced regulator_disable when doing module loading and unloading. This is because tpa6130a2_probe followed by tpa6130a2_remove calls twice tpa6130a2_power(0). Fix this by implementing a state checking in tpa6130a2_power. Signed-off-by: Jarkko Nikula Cc: Peter Ujfalusi Acked-by: Mark Brown Acked-by: Peter Ujfalusi Signed-off-by: Liam Girdwood --- sound/soc/codecs/tpa6130a2.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 329acc1a2074..83b5631b13a8 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -125,7 +125,7 @@ static int tpa6130a2_power(int power) data = i2c_get_clientdata(tpa6130a2_client); mutex_lock(&data->mutex); - if (power) { + if (power && !data->power_state) { /* Power on */ if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 1); @@ -153,7 +153,7 @@ static int tpa6130a2_power(int power) val = tpa6130a2_read(TPA6130A2_REG_CONTROL); val &= ~TPA6130A2_SWS; tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); - } else { + } else if (!power && data->power_state) { /* set SWS */ val = tpa6130a2_read(TPA6130A2_REG_CONTROL); val |= TPA6130A2_SWS; From 8a8d56b2a2f9aa423c3d8b6b1e2792c0492059ed Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Fri, 29 Oct 2010 20:40:23 +0200 Subject: [PATCH 06/17] ALSA: usb - driver neglects kmalloc return value check and may deref NULL sound/usb/pcm.c::snd_usb_pcm_check_knot() fails to check the return value from kmalloc() and may end up dereferencing a null pointer. The patch below (compile tested only) should take care of that little problem. Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index cff3a3c465d7..4132522ac90f 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -676,8 +676,10 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, if (!needs_knot) return 0; - subs->rate_list.count = count; subs->rate_list.list = kmalloc(sizeof(int) * count, GFP_KERNEL); + if (!subs->rate_list.list) + return -ENOMEM; + subs->rate_list.count = count; subs->rate_list.mask = 0; count = 0; list_for_each_entry(fp, &subs->fmt_list, list) { From bb617ee3f82ba94072c8b08043d9166bbfe397a2 Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Fri, 29 Oct 2010 22:54:45 +0200 Subject: [PATCH 07/17] ALSA: cs46xx memory management fixes for cs46xx_dsp_spos_create() When reading through sound/pci/cs46xx/dsp_spos.c I noticed a couple of things in cs46xx_dsp_spos_create(). It seems to me that we don't always free the various memory buffers we allocate and we also do some work (structure member assignment) early, that is completely pointless if some of the memory allocations fail and we end up just aborting the whole thing. I don't have hardware to test, so the patch below is compile tested only, but it makes the following changes: - Make sure we always free all allocated memory on failures. - Don't do pointless work assigning to structure members before we know all memory allocations, that may abort progress, have completed successfully. - Remove some trailing whitespace. Signed-off-by: Jesper Juhl Tested-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/pci/cs46xx/dsp_spos.c | 33 +++++++++++---------------------- 1 file changed, 11 insertions(+), 22 deletions(-) diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 3e5ca8fb519f..e377287192aa 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -225,39 +225,25 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip) { struct dsp_spos_instance * ins = kzalloc(sizeof(struct dsp_spos_instance), GFP_KERNEL); - if (ins == NULL) + if (ins == NULL) return NULL; /* better to use vmalloc for this big table */ - ins->symbol_table.nsymbols = 0; ins->symbol_table.symbols = vmalloc(sizeof(struct dsp_symbol_entry) * DSP_MAX_SYMBOLS); - ins->symbol_table.highest_frag_index = 0; - - if (ins->symbol_table.symbols == NULL) { + ins->code.data = kmalloc(DSP_CODE_BYTE_SIZE, GFP_KERNEL); + ins->modules = kmalloc(sizeof(struct dsp_module_desc) * DSP_MAX_MODULES, GFP_KERNEL); + if (!ins->symbol_table.symbols || !ins->code.data || !ins->modules) { cs46xx_dsp_spos_destroy(chip); goto error; } - + ins->symbol_table.nsymbols = 0; + ins->symbol_table.highest_frag_index = 0; ins->code.offset = 0; ins->code.size = 0; - ins->code.data = kmalloc(DSP_CODE_BYTE_SIZE, GFP_KERNEL); - - if (ins->code.data == NULL) { - cs46xx_dsp_spos_destroy(chip); - goto error; - } - ins->nscb = 0; ins->ntask = 0; - ins->nmodules = 0; - ins->modules = kmalloc(sizeof(struct dsp_module_desc) * DSP_MAX_MODULES, GFP_KERNEL); - - if (ins->modules == NULL) { - cs46xx_dsp_spos_destroy(chip); - goto error; - } /* default SPDIF input sample rate to 48000 khz */ @@ -271,8 +257,8 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip) /* set left and right validity bits and default channel status */ - ins->spdif_csuv_default = - ins->spdif_csuv_stream = + ins->spdif_csuv_default = + ins->spdif_csuv_stream = /* byte 0 */ ((unsigned int)_wrap_all_bits( (SNDRV_PCM_DEFAULT_CON_SPDIF & 0xff)) << 24) | /* byte 1 */ ((unsigned int)_wrap_all_bits( ((SNDRV_PCM_DEFAULT_CON_SPDIF >> 8) & 0xff)) << 16) | /* byte 3 */ (unsigned int)_wrap_all_bits( (SNDRV_PCM_DEFAULT_CON_SPDIF >> 24) & 0xff) | @@ -281,6 +267,9 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip) return ins; error: + kfree(ins->modules); + kfree(ins->code.data); + vfree(ins->symbol_table.symbols); kfree(ins); return NULL; } From f7467452291f7c9e5e1271e8c8e45b77f34b1257 Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Sun, 31 Oct 2010 19:46:19 +0100 Subject: [PATCH 08/17] ALSA: lx6464es - make 1 bit signed bitfield unsigned converts a 1 bit signed bitfield to an unsigned. Reported-by: Dr. David Alan Gilbert Signed-off-by: Tim Blechmann Signed-off-by: Takashi Iwai --- sound/pci/lx6464es/lx6464es.c | 4 ++-- sound/pci/lx6464es/lx6464es.h | 2 +- sound/pci/lx6464es/lx_core.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index ef9af3f4ace2..1bd7a540fd49 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -425,7 +425,7 @@ static int lx_pcm_hw_free(struct snd_pcm_substream *substream) static void lx_trigger_start(struct lx6464es *chip, struct lx_stream *lx_stream) { struct snd_pcm_substream *substream = lx_stream->stream; - const int is_capture = lx_stream->is_capture; + const unsigned int is_capture = lx_stream->is_capture; int err; @@ -473,7 +473,7 @@ static void lx_trigger_start(struct lx6464es *chip, struct lx_stream *lx_stream) static void lx_trigger_stop(struct lx6464es *chip, struct lx_stream *lx_stream) { - const int is_capture = lx_stream->is_capture; + const unsigned int is_capture = lx_stream->is_capture; int err; snd_printd(LXP "stopping: stopping stream\n"); diff --git a/sound/pci/lx6464es/lx6464es.h b/sound/pci/lx6464es/lx6464es.h index 51afc048961d..aea621eafbb5 100644 --- a/sound/pci/lx6464es/lx6464es.h +++ b/sound/pci/lx6464es/lx6464es.h @@ -60,7 +60,7 @@ struct lx_stream { snd_pcm_uframes_t frame_pos; enum lx_stream_status status; /* free, open, running, draining * pause */ - int is_capture:1; + unsigned int is_capture:1; }; diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 3086b751da4a..617f98b0cbae 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -1152,7 +1152,7 @@ static int lx_interrupt_request_new_buffer(struct lx6464es *chip, struct lx_stream *lx_stream) { struct snd_pcm_substream *substream = lx_stream->stream; - int is_capture = lx_stream->is_capture; + const unsigned int is_capture = lx_stream->is_capture; int err; unsigned long flags; From 6d212d8e86fb4221bd91b9266b7567ee2b83bd01 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 29 Oct 2010 15:41:17 -0700 Subject: [PATCH 09/17] ASoC: Remove volatility from WM8900 POWER1 register Not all bits can be read back from POWER1 so avoid corruption when using a read/modify/write cycle by marking it non-volatile - the only thing we read back from it is the chip revision which has diagnostic value only. We can re-add later but that's a more invasive change than is suitable for a bugfix. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8900.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index b4f11724a63f..aca4b1ea10bb 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -186,7 +186,6 @@ static int wm8900_volatile_register(unsigned int reg) { switch (reg) { case WM8900_REG_ID: - case WM8900_REG_POWER1: return 1; default: return 0; @@ -1200,11 +1199,6 @@ static int wm8900_probe(struct snd_soc_codec *codec) return -ENODEV; } - /* Read back from the chip */ - reg = snd_soc_read(codec, WM8900_REG_POWER1); - reg = (reg >> 12) & 0xf; - dev_info(codec->dev, "WM8900 revision %d\n", reg); - wm8900_reset(codec); /* Turn the chip on */ From 703dde6219346bc3b7d41d4fa2c36846d728e52c Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 29 Oct 2010 16:47:44 +0300 Subject: [PATCH 10/17] ASoC: Fix SND_SOC_ALL_CODECS typo for jz4740 Include jz4740.c to SND_SOC_ALL_CODECS when the dependencies are met. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 94a9d06b9027..02a9751bf149 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,7 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_DA7210 if I2C - select SND_SOC_JZ4740 if SOC_JZ4740 + select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 From 76a6106f124e375df0ea6ba6bcf204b8caff786a Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 29 Oct 2010 16:47:45 +0300 Subject: [PATCH 11/17] ASoC: Include cx20442 to SND_SOC_ALL_CODECS Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 02a9751bf149..3b5690d28b8b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -25,6 +25,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_MAX98088 if I2C From 5a0b07433ddd808ecbb5f4287b61be6fa7af1b57 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Sat, 30 Oct 2010 14:08:56 -0700 Subject: [PATCH 12/17] ASoC: Update WARN uses in wm_hubs Add missing newlines. Signed-off-by: Joe Perches Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2cb81538cd91..19ca782ac970 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -123,7 +123,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; break; default: - WARN(1, "Unknown DCS readback method"); + WARN(1, "Unknown DCS readback method\n"); break; } From fd0977d0f42d3e73121b88f57c7d48ca9b861a58 Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Fri, 29 Oct 2010 21:35:25 +0200 Subject: [PATCH 13/17] ALSA: asihpi - Unsafe memory management when allocating control cache I noticed that sound/pci/asihpi/hpicmn.c::hpi_alloc_control_cache() does not check the return value from kmalloc(), which may fail. If kmalloc() fails we'll dereference a null pointer and things will go bad fast. There are two memory allocations in that function and there's also the problem that the first may succeed and the second may fail and nothing is done about that either which will also go wrong down the line. Signed-off-by: Jesper Juhl Acked-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi6000.c | 2 ++ sound/pci/asihpi/hpi6205.c | 2 ++ sound/pci/asihpi/hpicmn.c | 12 +++++++++--- 3 files changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c index f7e374ec4414..1b9bf9395cfe 100644 --- a/sound/pci/asihpi/hpi6000.c +++ b/sound/pci/asihpi/hpi6000.c @@ -625,6 +625,8 @@ static short create_adapter_obj(struct hpi_adapter_obj *pao, control_cache_size, (struct hpi_control_cache_info *) &phw->control_cache[0] ); + if (!phw->p_cache) + pao->has_control_cache = 0; } else pao->has_control_cache = 0; diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 22c5fc625533..2672f6591ceb 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -644,6 +644,8 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao, interface->control_cache.size_in_bytes, (struct hpi_control_cache_info *) p_control_cache_virtual); + if (!phw->p_cache) + err = HPI_ERROR_MEMORY_ALLOC; } if (!err) { err = hpios_locked_mem_get_phys_addr(&phw-> diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c index dda4f1c6f658..d67f4d3db911 100644 --- a/sound/pci/asihpi/hpicmn.c +++ b/sound/pci/asihpi/hpicmn.c @@ -571,14 +571,20 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 { struct hpi_control_cache *p_cache = kmalloc(sizeof(*p_cache), GFP_KERNEL); + if (!p_cache) + return NULL; + p_cache->p_info = + kmalloc(sizeof(*p_cache->p_info) * number_of_controls, + GFP_KERNEL); + if (!p_cache->p_info) { + kfree(p_cache); + return NULL; + } p_cache->cache_size_in_bytes = size_in_bytes; p_cache->control_count = number_of_controls; p_cache->p_cache = (struct hpi_control_cache_single *)pDSP_control_buffer; p_cache->init = 0; - p_cache->p_info = - kmalloc(sizeof(*p_cache->p_info) * p_cache->control_count, - GFP_KERNEL); return p_cache; } From ca8dc34eaf7a1db7daa604495ed2c143af32f1ed Mon Sep 17 00:00:00 2001 From: Mandar Joshi Date: Tue, 2 Nov 2010 14:43:19 +0000 Subject: [PATCH 14/17] ALSA: usb-audio - Support for Power/Status LED on Creative USB X-Fi S51 This patch adds support for Power/Status LED on Creative USB X-Fi S51. There is just one LED on the device. The LED can either be On or it can be set to Blink. There doesn't seem to be a way to switch it off. The control message to change LED status is similar to that of audigy2nx except that the index is to be set to 0 and value is 1 for Blink and 0 for On. The 'Power LED' control in alsamixer when muted will cause the LED to Blink continuously. When unmuted the LED will stay On. The Creative driver under Windows sets the LED to blink whenever audio is muted. This LED can be treated as the CMSS LED but I figured since there is just one LED, it should be treated as the Power LED. Is that alright? I've also changed the comment "Usb X-Fi" to "Usb X-Fi S51" as there are other external X-Fi devices from Creative like Usb X-Fi Go and Xmod. The volume knob and LED support patch doesn't apply to them. Signed-off-by: Mandar Joshi Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 7dae05d8783e..782f741cd00a 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -60,7 +60,7 @@ static const struct rc_config { { USB_ID(0x041e, 0x3000), 0, 1, 2, 1, 18, 0x0013 }, /* Extigy */ { USB_ID(0x041e, 0x3020), 2, 1, 6, 6, 18, 0x0013 }, /* Audigy 2 NX */ { USB_ID(0x041e, 0x3040), 2, 2, 6, 6, 2, 0x6e91 }, /* Live! 24-bit */ - { USB_ID(0x041e, 0x3042), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi */ + { USB_ID(0x041e, 0x3042), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 */ { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; @@ -183,7 +183,13 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e if (value > 1) return -EINVAL; changed = value != mixer->audigy2nx_leds[index]; - err = snd_usb_ctl_msg(mixer->chip->dev, + if (mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + !value, 0, NULL, 0, 100); + else + err = snd_usb_ctl_msg(mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, value, index + 2, NULL, 0, 100); @@ -225,8 +231,12 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) int i, err; for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) { + /* USB X-Fi S51 doesn't have a CMSS LED */ + if ((mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) && i == 0) + continue; if (i > 1 && /* Live24ext has 2 LEDs only */ (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3042) || mixer->chip->usb_id == USB_ID(0x041e, 0x3048))) break; err = snd_ctl_add(mixer->chip->card, @@ -365,6 +375,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) || mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3042) || mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) { if ((err = snd_audigy2nx_controls_create(mixer)) < 0) return err; From cb9906229595941d632fc4022b05da4f9533856a Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Tue, 2 Nov 2010 05:10:07 +0800 Subject: [PATCH 15/17] ASoC: fix the building issue of missing codec field in 'struct snd_soc_card' Signed-off-by: Mark Brown --- sound/soc/pxa/tosa.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index a3bfb2e8b70f..73d0edd8ded9 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -79,7 +79,7 @@ static void tosa_ext_control(struct snd_soc_codec *codec) static int tosa_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->card->codec; + struct snd_soc_codec *codec = rtd->codec; /* check the jack status at stream startup */ tosa_ext_control(codec); From 87232dd49aeb6b7d1af291edca8bd129a82ef4b5 Mon Sep 17 00:00:00 2001 From: "Edgar (gimli) Hucek" Date: Wed, 3 Nov 2010 08:14:10 +0100 Subject: [PATCH 16/17] ALSA: hda - MacBookAir3,1(3,2) alsa support This patch add support for the MacBookAir3,1 and MacBookAir3,2 to the alsa sound system. Signed-off-by: Edgar (gimli) Hucek Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 460fb2ef7e39..18af38ebf757 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1166,6 +1166,7 @@ static const char *cs420x_models[CS420X_MODELS] = { static struct snd_pci_quirk cs420x_cfg_tbl[] = { SND_PCI_QUIRK(0x10de, 0x0ac0, "MacBookPro 5,3", CS420X_MBP53), + SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb89, "MacBookPro 7,1", CS420X_MBP55), SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27), From 75e3f3137cb570661c2ad3035a139dda671fbb63 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 3 Nov 2010 16:39:00 +0200 Subject: [PATCH 17/17] ASoC: tpa6130a2: Get rid of compile warning from tpa6130a2_power MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Patch "ASoC: tpa6130a2: Fix unbalanced regulator disables" introduced a compiler warning "‘ret’ may be used uninitialized in this function". Initialize ret to zero to get rid of it and making sure that the function does not return any random error code when the code is falling through. Signed-off-by: Jarkko Nikula Signed-off-by: Takashi Iwai --- sound/soc/codecs/tpa6130a2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 83b5631b13a8..ee4fb201de60 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -119,7 +119,7 @@ static int tpa6130a2_power(int power) { struct tpa6130a2_data *data; u8 val; - int ret; + int ret = 0; BUG_ON(tpa6130a2_client == NULL); data = i2c_get_clientdata(tpa6130a2_client);