From 5e42336a461a2354b640001323cd07cebd9ade6e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 27 Apr 2009 19:18:22 +0100 Subject: ASoC: Fix logic in WM8350 master clocking check We need to check only if the WM8350 is master and only when starting the stream so if either is not true then we can skip the check. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3b1d0993bed9..0275321ff8ab 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream, * required for LRC in master mode. The DACs or ADCs need a * valid audio path i.e. pin -> ADC or DAC -> pin before * the LRC will be enabled in master mode. */ - if (!master && cmd != SNDRV_PCM_TRIGGER_START) + if (!master || cmd != SNDRV_PCM_TRIGGER_START) return 0; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { -- cgit v1.2.2 From 0c95de73a711d376dc17afe484f919bd5b66c016 Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Mon, 27 Apr 2009 12:44:41 -0400 Subject: ASoC: Set the MPC5200 i2s driver to BROKEN status. Signed-off-by: Jon Smirl Acked-by: Grant Likely Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 9fc908283371..e7dd79a1d8cd 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -21,7 +21,7 @@ config SND_SOC_MPC8610_HPCD config SND_SOC_MPC5200_I2S tristate "Freescale MPC5200 PSC in I2S mode driver" - depends on PPC_MPC52xx && PPC_BESTCOMM + depends on PPC_MPC52xx && PPC_BESTCOMM && BROKEN select SND_SOC_OF_SIMPLE select PPC_BESTCOMM_GEN_BD help -- cgit v1.2.2 From 3f68165e234233255a789c827c5d3d6fa965ddce Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Mon, 27 Apr 2009 23:23:29 +0200 Subject: ALSA: indigo-express: add missing 64KHz flags Indigo-express cards also support 64KHz sampling rate: this patch adds missing SNDRV_PCM_RATE_64000 flags. Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/indigodjx.c | 1 + sound/pci/echoaudio/indigoiox.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c index 3482ef69f491..2e44316530a2 100644 --- a/sound/pci/echoaudio/indigodjx.c +++ b/sound/pci/echoaudio/indigodjx.c @@ -88,6 +88,7 @@ static struct snd_pcm_hardware pcm_hardware_skel = { .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 32000, diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c index aebee27a40ff..eb3819f9654a 100644 --- a/sound/pci/echoaudio/indigoiox.c +++ b/sound/pci/echoaudio/indigoiox.c @@ -89,6 +89,7 @@ static struct snd_pcm_hardware pcm_hardware_skel = { .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 32000, -- cgit v1.2.2 From 2008f137e92220b98120c4803499cdddb2b0fb06 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Apr 2009 12:25:59 +0200 Subject: ALSA: Add missing SNDRV_PCM_INFO_BATCH flag to some drivers Added SNDRV_PCM_INFO_BATCH flag to PCM info field of some drivers that really don't give the precise pointer value. Signed-off-by: Takashi Iwai --- sound/isa/msnd/msnd.c | 6 ++++-- sound/pci/bt87x.c | 6 ++++-- sound/pci/korg1212/korg1212.c | 6 ++++-- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 3 ++- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 3 ++- sound/soc/sh/dma-sh7760.c | 3 ++- sound/sparc/dbri.c | 3 ++- sound/usb/usx2y/usbusx2yaudio.c | 3 ++- 9 files changed, 23 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c index 906454413ed2..3a1526ae1729 100644 --- a/sound/isa/msnd/msnd.c +++ b/sound/isa/msnd/msnd.c @@ -438,7 +438,8 @@ static void snd_msnd_capture_reset_queue(struct snd_msnd *chip, static struct snd_pcm_hardware snd_msnd_playback = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_8000_48000, .rate_min = 8000, @@ -456,7 +457,8 @@ static struct snd_pcm_hardware snd_msnd_playback = { static struct snd_pcm_hardware snd_msnd_capture = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_8000_48000, .rate_min = 8000, diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index a299340519df..ce3f2e90f4d7 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -349,7 +349,8 @@ static struct snd_pcm_hardware snd_bt87x_digital_hw = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = 0, /* set at runtime */ .channels_min = 2, @@ -365,7 +366,8 @@ static struct snd_pcm_hardware snd_bt87x_analog_hw = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, .rates = SNDRV_PCM_RATE_KNOT, .rate_min = ANALOG_CLOCK / CLOCK_DIV_MAX, diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 8b79969034be..7cc38a11e997 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -1238,7 +1238,8 @@ static struct snd_pcm_hardware snd_korg1212_playback_info = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED), + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BATCH), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000), @@ -1258,7 +1259,8 @@ static struct snd_pcm_hardware snd_korg1212_capture_info = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED), + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BATCH), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000), diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 01066c95580e..d057e6489643 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -240,7 +240,8 @@ static int pdacf_pcm_prepare(struct snd_pcm_substream *subs) static struct snd_pcm_hardware pdacf_pcm_capture_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_MMAP_VALID), + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH), .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE, diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 30490a259148..594c6c5b7838 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -82,7 +82,7 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2]; /* PCM hardware DMA capabilities - platform specific */ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED, + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH, .formats = AU1XPSC_PCM_FMTS, .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES, .period_bytes_max = 4096 * 1024 - 1, diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 3aa729df27b5..1111c710118a 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -504,7 +504,8 @@ static struct snd_soc_dai psc_i2s_dai_template = { static const struct snd_pcm_hardware psc_i2s_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, .rate_min = 8000, diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 0dad3a0bb920..baddb1242c71 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -103,7 +103,8 @@ static struct snd_pcm_hardware camelot_pcm_hardware = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH), .formats = DMABRG_FMTS, .rates = DMABRG_RATES, .rate_min = 8000, diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index af95ff1e126c..1d2e51b3f918 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -1975,7 +1975,8 @@ static struct snd_pcm_hardware snd_dbri_pcm_hw = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_U8 | diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 9a608fa85155..dd1ab6177840 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -870,7 +870,8 @@ static struct snd_pcm_hardware snd_usX2Y_2c = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH), .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, .rate_min = 44100, -- cgit v1.2.2 From 3e5b50165fd0be080044586f43fcdd460ed27610 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Apr 2009 12:07:08 +0200 Subject: ALSA: pcm core - Avoid jiffies check for devices with BATCH flag The hardware devices with SNDRV_PCM_INFO_BATCH flag can't give the precise current position. And such hardwares have often big FIFO in addition to the ring buffer, and it screws up the jiffies check in pcm_lib.c. This patch adds a simple check of info flag so that the driver skips the jiffies check in snd_pcm_period_elapsed() when BATCH flag is set. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 63d088f2265f..a2a792c18c40 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -249,6 +249,12 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) new_hw_ptr = hw_base + pos; } } + /* Skip the jiffies check for hardwares with BATCH flag. + * Such hardware usually just increases the position at each IRQ, + * thus it can't give any strange position. + */ + if (runtime->hw.info & SNDRV_PCM_INFO_BATCH) + goto no_jiffies_check; hdelta = new_hw_ptr - old_hw_ptr; jdelta = jiffies - runtime->hw_ptr_jiffies; if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) { @@ -272,6 +278,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) hw_base -= hw_base % runtime->buffer_size; delta = 0; } + no_jiffies_check: if (delta > runtime->period_size + runtime->period_size / 2) { hw_ptr_error(substream, "Lost interrupts? " -- cgit v1.2.2 From 18cc8d8d9b74c446832336d8f6e1afb145f9431b Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Tue, 28 Apr 2009 18:18:05 +0900 Subject: ASoC: TWL4030: Fix gain control for earpiece amplifier The gain control for earpiece amplifier uses 0dB ~ 12dB according to the TRM, but the present code is implemented to -6dB ~ 6dB. Signed-off-by: Joonyoung Shim Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 921b205de28a..df7c8c281d2f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -835,6 +835,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0); */ static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1); +/* + * Gain control for earpiece amplifier + * 0 dB to 12 dB in 6 dB steps (mute instead of -6) + */ +static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1); + /* * Capture gain after the ADCs * from 0 dB to 31 dB in 1 dB steps @@ -900,7 +906,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { 4, 3, 0, output_tvl), SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume", - TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl), + TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl), /* Common capture gain controls */ SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume", -- cgit v1.2.2 From 6574612fbb34c63117581e68f2231ddce027e41e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 May 2009 16:03:21 +0200 Subject: ASoC: Remove BROKEN from mpc5200 kconfig The regression was fixed by commit 3e5b50165fd0be080044586f43fcdd460ed27610, so no need to mark this driver as BROKEN. Signed-off-by: Takashi Iwai --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index e7dd79a1d8cd..9fc908283371 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -21,7 +21,7 @@ config SND_SOC_MPC8610_HPCD config SND_SOC_MPC5200_I2S tristate "Freescale MPC5200 PSC in I2S mode driver" - depends on PPC_MPC52xx && PPC_BESTCOMM && BROKEN + depends on PPC_MPC52xx && PPC_BESTCOMM select SND_SOC_OF_SIMPLE select PPC_BESTCOMM_GEN_BD help -- cgit v1.2.2 From 97a775c49c7e1b47b016a492463486a5b86da479 Mon Sep 17 00:00:00 2001 From: Jinyoung Park Date: Fri, 1 May 2009 12:54:31 +0100 Subject: ASoC: Fix errors in WM8990 The mis-typing exist in dapm controller definitions and dapm route definitions, so happen mis-matched error when snd_soc_dapm_add_routes(). Cc: stable@kernel.org Signed-off-by: Jinyoung Park Signed-off-by: Mark Brown Date: Tue, 5 May 2009 15:39:39 +0200 Subject: sound: serial-u16550: fix buffer overflow Remove most of the serial port parameters from the card longname string because it was way too long and overflowed into the mixername string. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/drivers/serial-u16550.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index b2b6d50c9425..a25fb7b1f441 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -963,16 +963,11 @@ static int __devinit snd_serial_probe(struct platform_device *devptr) if (err < 0) goto _err; - sprintf(card->longname, "%s at 0x%lx, irq %d speed %d div %d outs %d ins %d adaptor %s droponfull %d", + sprintf(card->longname, "%s [%s] at %#lx, irq %d", card->shortname, - uart->base, - uart->irq, - uart->speed, - (int)uart->divisor, - outs[dev], - ins[dev], adaptor_names[uart->adaptor], - uart->drop_on_full); + uart->base, + uart->irq); snd_card_set_dev(card, &devptr->dev); -- cgit v1.2.2 From b452e08e73c0e3dbb0be82130217be4b7084299e Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 5 May 2009 15:40:12 +0200 Subject: sound: via82xx: fix DXS volume range With 5 bits and 1.5 dB per step, the DXS volume range is only 48 dB. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 809b233dd4a3..1ef58c51c213 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1687,7 +1687,7 @@ static int snd_via8233_pcmdxs_volume_put(struct snd_kcontrol *kcontrol, return change; } -static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -9450, 150, 1); +static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -4650, 150, 1); static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = { .name = "PCM Playback Volume", -- cgit v1.2.2 From 5dd17cb992ef4c1ebb1a2d60cbef4b6967974673 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 May 2009 16:22:53 +0200 Subject: ALSA: hda - Fix line-in on Mac Mini Core2 Duo BIOS on Mac Mini Core2 Duo sets both INPUT and OUTPUT pinctl bits to the line-in jack, and it confuses the driver as if it's a valid input. This patch adds the check of OUTPUT bit so that the driver fixes the invalid pin setup. Tested-by: Tino Keitel Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 917bc5d3ac2c..03b3646018a1 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4079,7 +4079,12 @@ static int stac92xx_init(struct hda_codec *codec) pinctl = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); /* if PINCTL already set then skip */ - if (!(pinctl & AC_PINCTL_IN_EN)) { + /* Also, if both INPUT and OUTPUT are set, + * it must be a BIOS bug; need to override, too + */ + if (!(pinctl & AC_PINCTL_IN_EN) || + (pinctl & AC_PINCTL_OUT_EN)) { + pinctl &= ~AC_PINCTL_OUT_EN; pinctl |= AC_PINCTL_IN_EN; stac92xx_auto_set_pinctl(codec, nid, pinctl); -- cgit v1.2.2 From 914dc18255e430ceabb10b57394e01814c69c5cd Mon Sep 17 00:00:00 2001 From: Mike Rapoport Date: Mon, 11 May 2009 13:04:55 +0300 Subject: ASoC: soc-core: fix crash when removing not instantiated card If the card was not instantiated in snd_soc_instantiate_card, calling soc-remove will crash because some of codec, cpu_dai and card .remove methods are called twice. Fix this by returning from soc_remove immediately. Signed-off-by: Mike Rapoport Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 99712f652d0d..1cd149b9ce69 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -954,6 +954,9 @@ static int soc_remove(struct platform_device *pdev) struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; + if (!card->instantiated) + return 0; + run_delayed_work(&card->delayed_work); if (platform->remove) -- cgit v1.2.2 From 1ffafeb556d50de8039e14f1cbbe58e9e4549915 Mon Sep 17 00:00:00 2001 From: Mike Rapoport Date: Mon, 11 May 2009 13:11:38 +0300 Subject: pxa2xx-ac97: fix reset gpio mode setting Signed-off-by: Mike Rapoport Acked-by: Eric Miao Signed-off-by: Mark Brown --- sound/arm/pxa2xx-ac97-lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index a2c12d105c9a..6fdca97186e7 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -65,7 +65,7 @@ static void set_resetgpio_mode(int resetgpio_action) switch (resetgpio_action) { case RESETGPIO_NORMAL_ALTFUNC: if (reset_gpio == 113) - mode = 113 | GPIO_OUT | GPIO_DFLT_LOW; + mode = 113 | GPIO_ALT_FN_2_OUT; if (reset_gpio == 95) mode = 95 | GPIO_ALT_FN_1_OUT; break; -- cgit v1.2.2 From 1b1cc7f21c51cc81992a547b59e174dd8c44d1bd Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Wed, 13 May 2009 20:44:07 +0200 Subject: ALSA: riptide: postfix increment and off by one With a postfix increment these variables are incremented beyond CMDIF_TIMEOUT / MAX_WRITE_RETRY. Signed-off-by: Roel Kluin Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 6f1034417a02..e51a5ef1954d 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -889,7 +889,7 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm, spin_lock_irqsave(&cif->lock, irqflags); while (i++ < CMDIF_TIMEOUT && !IS_READY(cif->hwport)) udelay(10); - if (i >= CMDIF_TIMEOUT) { + if (i > CMDIF_TIMEOUT) { err = -EBUSY; goto errout; } @@ -907,8 +907,10 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm, WRITE_PORT_ULONG(cmdport->data1, cmd); /* write cmd */ if ((flags & RESP) && ret) { while (!IS_DATF(cmdport) && - time++ < CMDIF_TIMEOUT) + time < CMDIF_TIMEOUT) { udelay(10); + time++; + } if (time < CMDIF_TIMEOUT) { /* read response */ ret->retlongs[0] = READ_PORT_ULONG(cmdport->data1); @@ -1454,7 +1456,7 @@ static int snd_riptide_trigger(struct snd_pcm_substream *substream, int cmd) SEND_GPOS(cif, 0, data->id, &rptr); udelay(1); } while (i != rptr.retlongs[1] && j++ < MAX_WRITE_RETRY); - if (j >= MAX_WRITE_RETRY) + if (j > MAX_WRITE_RETRY) snd_printk(KERN_ERR "Riptide: Could not stop stream!"); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: @@ -1783,7 +1785,7 @@ snd_riptide_codec_write(struct snd_ac97 *ac97, unsigned short reg, SEND_SACR(cif, val, reg); SEND_RACR(cif, reg, &rptr); } while (rptr.retwords[1] != val && i++ < MAX_WRITE_RETRY); - if (i == MAX_WRITE_RETRY) + if (i > MAX_WRITE_RETRY) snd_printdd("Write AC97 reg failed\n"); } -- cgit v1.2.2 From 5a641bcd6398841cc4606b0a732d41a09256fd94 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Thu, 14 May 2009 08:49:13 -0700 Subject: ALSA: pcsp: fix printk format warning Fix printk format warning: sound/drivers/pcsp/pcsp_mixer.c:54: warning: format '%d' expects type 'int', but argument 3 has type 'long unsigned int' Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp_mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c index caeb0f57fcca..771955a9be71 100644 --- a/sound/drivers/pcsp/pcsp_mixer.c +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -50,7 +50,7 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol, uinfo->value.enumerated.items = chip->max_treble + 1; if (uinfo->value.enumerated.item > chip->max_treble) uinfo->value.enumerated.item = chip->max_treble; - sprintf(uinfo->value.enumerated.name, "%d", + sprintf(uinfo->value.enumerated.name, "%lu", PCSP_CALC_RATE(uinfo->value.enumerated.item)); return 0; } -- cgit v1.2.2 From 82075af6cb9b4918ab52a7100425b09fae6aafe3 Mon Sep 17 00:00:00 2001 From: David Brownell Date: Thu, 14 May 2009 12:41:22 -0700 Subject: ASoC: davinci-pcm buildfixes This is a buildfix for the DaVinci PCM code, resyncing it with the version in the DaVinci tree. The notable change is using current EDMA interfaces, which recently merged to mainline. (The older interfaces never made it into mainline.) NOTE: open issue, the DMA should be to/from SRAM; see chip errata for more info. The artifacts are extremely easy to hear on DM355 hardware (not yet supported in mainline), but don't seem as audible on DM6446 hardwaare (which does have mainline support). Signed-off-by: David Brownell Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-pcm.c | 71 ++++++++++++++++++++++++----------------- 1 file changed, 42 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 7af3b5b3a53d..a05996588489 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -22,6 +22,7 @@ #include #include +#include #include "davinci-pcm.h" @@ -51,7 +52,7 @@ struct davinci_runtime_data { spinlock_t lock; int period; /* current DMA period */ int master_lch; /* Master DMA channel */ - int slave_lch; /* Slave DMA channel */ + int slave_lch; /* linked parameter RAM reload slot */ struct davinci_pcm_dma_params *params; /* DMA params */ }; @@ -90,18 +91,18 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dst_bidx = data_type; } - davinci_set_dma_src_params(lch, src, INCR, W8BIT); - davinci_set_dma_dest_params(lch, dst, INCR, W8BIT); - davinci_set_dma_src_index(lch, src_bidx, 0); - davinci_set_dma_dest_index(lch, dst_bidx, 0); - davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC); + edma_set_src(lch, src, INCR, W8BIT); + edma_set_dest(lch, dst, INCR, W8BIT); + edma_set_src_index(lch, src_bidx, 0); + edma_set_dest_index(lch, dst_bidx, 0); + edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) prtd->period = 0; } -static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data) +static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data) { struct snd_pcm_substream *substream = data; struct davinci_runtime_data *prtd = substream->runtime->private_data; @@ -125,7 +126,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd = substream->runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; - int tcc = TCC_ANY; + struct edmacc_param p_ram; int ret; if (!dma_data) @@ -134,22 +135,34 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) prtd->params = dma_data; /* Request master DMA channel */ - ret = davinci_request_dma(prtd->params->channel, prtd->params->name, + ret = edma_alloc_channel(prtd->params->channel, davinci_pcm_dma_irq, substream, - &prtd->master_lch, &tcc, EVENTQ_0); - if (ret) + EVENTQ_0); + if (ret < 0) return ret; + prtd->master_lch = ret; - /* Request slave DMA channel */ - ret = davinci_request_dma(PARAM_ANY, "Link", - NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0); - if (ret) { - davinci_free_dma(prtd->master_lch); + /* Request parameter RAM reload slot */ + ret = edma_alloc_slot(EDMA_SLOT_ANY); + if (ret < 0) { + edma_free_channel(prtd->master_lch); return ret; } - - /* Link slave DMA channel in loopback */ - davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch); + prtd->slave_lch = ret; + + /* Issue transfer completion IRQ when the channel completes a + * transfer, then always reload from the same slot (by a kind + * of loopback link). The completion IRQ handler will update + * the reload slot with a new buffer. + * + * REVISIT save p_ram here after setting up everything except + * the buffer and its length (ccnt) ... use it as a template + * so davinci_pcm_enqueue_dma() takes less time in IRQ. + */ + edma_read_slot(prtd->slave_lch, &p_ram); + p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch); + p_ram.link_bcntrld = prtd->slave_lch << 5; + edma_write_slot(prtd->slave_lch, &p_ram); return 0; } @@ -165,12 +178,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - davinci_start_dma(prtd->master_lch); + edma_start(prtd->master_lch); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - davinci_stop_dma(prtd->master_lch); + edma_stop(prtd->master_lch); break; default: ret = -EINVAL; @@ -185,14 +198,14 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static int davinci_pcm_prepare(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct paramentry_descriptor temp; + struct edmacc_param temp; prtd->period = 0; davinci_pcm_enqueue_dma(substream); - /* Get slave channel dma params for master channel startup */ - davinci_get_dma_params(prtd->slave_lch, &temp); - davinci_set_dma_params(prtd->master_lch, &temp); + /* Copy self-linked parameter RAM entry into master channel */ + edma_read_slot(prtd->slave_lch, &temp); + edma_write_slot(prtd->master_lch, &temp); return 0; } @@ -208,7 +221,7 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream) spin_lock(&prtd->lock); - davinci_dma_getposition(prtd->master_lch, &src, &dst); + edma_get_position(prtd->master_lch, &src, &dst); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) count = src - runtime->dma_addr; else @@ -253,10 +266,10 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd = runtime->private_data; - davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch); + edma_unlink(prtd->slave_lch); - davinci_free_dma(prtd->slave_lch); - davinci_free_dma(prtd->master_lch); + edma_free_slot(prtd->slave_lch); + edma_free_channel(prtd->master_lch); kfree(prtd); -- cgit v1.2.2 From a62114cb90a351016121bca02e69d6a9e24afa0e Mon Sep 17 00:00:00 2001 From: David Brownell Date: Thu, 14 May 2009 12:47:42 -0700 Subject: ASoC: DaVinci I2S updates This resyncs the DaVinci I2S code with the version in the DaVinci tree. The behavioral change uses updated clock interfaces which recently merged to mainline. Two other changes include adding a comment on the ASP/McBSP/McASP confusion, and dropping pdev->id in order to support more boards than just the DM644x EVM. Signed-off-by: David Brownell Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 26 +++++++++++++++++++++++--- 1 file changed, 23 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index ffdb9439d3d8..b1ea52fc83c7 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -24,6 +24,26 @@ #include "davinci-pcm.h" + +/* + * NOTE: terminology here is confusing. + * + * - This driver supports the "Audio Serial Port" (ASP), + * found on dm6446, dm355, and other DaVinci chips. + * + * - But it labels it a "Multi-channel Buffered Serial Port" + * (McBSP) as on older chips like the dm642 ... which was + * backward-compatible, possibly explaining that confusion. + * + * - OMAP chips have a controller called McBSP, which is + * incompatible with the DaVinci flavor of McBSP. + * + * - Newer DaVinci chips have a controller called McASP, + * incompatible with ASP and with either McBSP. + * + * In short: this uses ASP to implement I2S, not McBSP. + * And it won't be the only DaVinci implemention of I2S. + */ #define DAVINCI_MCBSP_DRR_REG 0x00 #define DAVINCI_MCBSP_DXR_REG 0x04 #define DAVINCI_MCBSP_SPCR_REG 0x08 @@ -421,7 +441,7 @@ static int davinci_i2s_probe(struct platform_device *pdev, { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea; struct evm_snd_platform_data *pdata; @@ -448,7 +468,7 @@ static int davinci_i2s_probe(struct platform_device *pdev, cpu_dai->private_data = dev; - dev->clk = clk_get(&pdev->dev, "McBSPCLK"); + dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; goto err_free_mem; @@ -483,7 +503,7 @@ static void davinci_i2s_remove(struct platform_device *pdev, { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; struct davinci_mcbsp_dev *dev = cpu_dai->private_data; struct resource *mem; -- cgit v1.2.2 From f492ec9f02908579353e31949855f86909a5af14 Mon Sep 17 00:00:00 2001 From: David Brownell Date: Thu, 14 May 2009 13:01:59 -0700 Subject: ASoC: DaVinci EVM board support buildfixes This is a build fix, resyncing the DaVinci EVM ASoC board code with the version in the DaVinci tree. That resync includes support for the DM355 EVM, although that board isn't yet in mainline. (NOTE: also includes a bugfix to the platform_add_resources call, recently sent by Chaithrika U S but not yet merged into the DaVinci tree.) Signed-off-by: David Brownell Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 7 +++-- sound/soc/davinci/davinci-evm.c | 63 ++++++++++++++++++++++++++++++++++------- 2 files changed, 56 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index bd7392c9657e..411a710be660 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -10,13 +10,14 @@ config SND_DAVINCI_SOC_I2S tristate config SND_DAVINCI_SOC_EVM - tristate "SoC Audio support for DaVinci EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM + tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" + depends on SND_DAVINCI_SOC + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DaVinci EVM platform. + DaVinci DM6446 or DM355 EVM platforms. config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 9b90b347007c..58fd1cbedd88 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -20,7 +20,11 @@ #include #include -#include +#include + +#include +#include +#include #include "../codecs/tlv320aic3x.h" #include "davinci-pcm.h" @@ -150,7 +154,7 @@ static struct snd_soc_card snd_soc_card_evm = { /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { - .i2c_bus = 0, + .i2c_bus = 1, .i2c_address = 0x1b, }; @@ -161,36 +165,73 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; +/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ static struct resource evm_snd_resources[] = { { - .start = DAVINCI_MCBSP_BASE, - .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, + .start = DAVINCI_ASP0_BASE, + .end = DAVINCI_ASP0_BASE + SZ_8K - 1, .flags = IORESOURCE_MEM, }, }; static struct evm_snd_platform_data evm_snd_data = { - .tx_dma_ch = DM644X_DMACH_MCBSP_TX, - .rx_dma_ch = DM644X_DMACH_MCBSP_RX, + .tx_dma_ch = DAVINCI_DMA_ASP0_TX, + .rx_dma_ch = DAVINCI_DMA_ASP0_RX, +}; + +/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ +static struct resource dm335evm_snd_resources[] = { + { + .start = DAVINCI_ASP1_BASE, + .end = DAVINCI_ASP1_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data dm335evm_snd_data = { + .tx_dma_ch = DAVINCI_DMA_ASP1_TX, + .rx_dma_ch = DAVINCI_DMA_ASP1_RX, }; static struct platform_device *evm_snd_device; static int __init evm_init(void) { + struct resource *resources; + unsigned num_resources; + struct evm_snd_platform_data *data; + int index; int ret; - evm_snd_device = platform_device_alloc("soc-audio", 0); + if (machine_is_davinci_evm()) { + davinci_cfg_reg(DM644X_MCBSP); + + resources = evm_snd_resources; + num_resources = ARRAY_SIZE(evm_snd_resources); + data = &evm_snd_data; + index = 0; + } else if (machine_is_davinci_dm355_evm()) { + /* we don't use ASP1 IRQs, or we'd need to mux them ... */ + davinci_cfg_reg(DM355_EVT8_ASP1_TX); + davinci_cfg_reg(DM355_EVT9_ASP1_RX); + + resources = dm335evm_snd_resources; + num_resources = ARRAY_SIZE(dm335evm_snd_resources); + data = &dm335evm_snd_data; + index = 1; + } else + return -EINVAL; + + evm_snd_device = platform_device_alloc("soc-audio", index); if (!evm_snd_device) return -ENOMEM; platform_set_drvdata(evm_snd_device, &evm_snd_devdata); evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, &evm_snd_data, - sizeof(evm_snd_data)); + platform_device_add_data(evm_snd_device, data, sizeof(*data)); - ret = platform_device_add_resources(evm_snd_device, evm_snd_resources, - ARRAY_SIZE(evm_snd_resources)); + ret = platform_device_add_resources(evm_snd_device, resources, + num_resources); if (ret) { platform_device_put(evm_snd_device); return ret; -- cgit v1.2.2 From b3b778b387ed3849ebc4a51baf8617be90df6625 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 May 2009 17:05:52 +0200 Subject: ALSA: pcsp - fix printk format warning again The commit 5a641bcd6398841cc4606b0a732d41a09256fd94 changed the printk format to '%lu', but the value passed seems to be dependent on the architecture. On x86-64, I got a new warning now because an int value is passed actaully. As a workaround, just cast the value always to unsigned long. Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp_mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c index 771955a9be71..199b03377142 100644 --- a/sound/drivers/pcsp/pcsp_mixer.c +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -51,7 +51,7 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol, if (uinfo->value.enumerated.item > chip->max_treble) uinfo->value.enumerated.item = chip->max_treble; sprintf(uinfo->value.enumerated.name, "%lu", - PCSP_CALC_RATE(uinfo->value.enumerated.item)); + (unsigned long)PCSP_CALC_RATE(uinfo->value.enumerated.item)); return 0; } -- cgit v1.2.2 From 03fbdb15c14e9746c63168e3ff2c64b9c8336d33 Mon Sep 17 00:00:00 2001 From: Alessandro Rubini Date: Wed, 20 May 2009 22:39:08 +0100 Subject: [ARM] 5519/1: amba probe: pass "struct amba_id *" instead of void * The second argument of the probe method points to the amba_id structure, so it's better passed with the correct type. None of the current in-tree drivers uses the pointer, so they have only been checked for a clean compile. Change suggested by Russell King. Signed-off-by: Alessandro Rubini Signed-off-by: Russell King --- sound/arm/aaci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 7fbd68fab944..5c48e36038f2 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -1074,7 +1074,7 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci) return i; } -static int __devinit aaci_probe(struct amba_device *dev, void *id) +static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id) { struct aaci *aaci; int ret, i; -- cgit v1.2.2 From 87488957a68293357a94c8142de7d0ae17914912 Mon Sep 17 00:00:00 2001 From: Adam Williamson Date: Thu, 21 May 2009 18:32:59 -0400 Subject: ALSA: hda - fix audio on HP TX25xx series notebooks Fixes https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4121 Taken from https://bugzilla.redhat.com/show_bug.cgi?id=498060 Signed-off-by: Adam Williamson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b8a0d3e79272..bcbb736f94f0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12058,6 +12058,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), + SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA), -- cgit v1.2.2 From afe6d7e3c4a9aba020637f4ae15527a89ba31f21 Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Fri, 22 May 2009 17:48:58 +0200 Subject: ALSA: Kill truncate warning by shortening Sigmatel-specific AC97 control name ALSA sound/core/control.c:232: Control name 'Sigmatel Surround Phase Inversion Playback Switch' truncated to 'Sigmatel Surround Phase Inversion Playback ' bootup message by omitting weird Sigmatel prefix in this case; also fix up the related ca0106 mixer control removal part by using identical naming there. Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 7 +++++-- sound/pci/ca0106/ca0106_mixer.c | 2 +- 2 files changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 81bc93e5f1e3..7337abdbe4e3 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -958,10 +958,13 @@ static int patch_sigmatel_stac9708_3d(struct snd_ac97 * ac97) } static const struct snd_kcontrol_new snd_ac97_sigmatel_4speaker = -AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch", AC97_SIGMATEL_DAC2INVERT, 2, 1, 0); +AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch", + AC97_SIGMATEL_DAC2INVERT, 2, 1, 0); +/* "Sigmatel " removed due to excessive name length: */ static const struct snd_kcontrol_new snd_ac97_sigmatel_phaseinvert = -AC97_SINGLE("Sigmatel Surround Phase Inversion Playback Switch", AC97_SIGMATEL_DAC2INVERT, 3, 1, 0); +AC97_SINGLE("Surround Phase Inversion Playback Switch", + AC97_SIGMATEL_DAC2INVERT, 3, 1, 0); static const struct snd_kcontrol_new snd_ac97_sigmatel_controls[] = { AC97_SINGLE("Sigmatel DAC 6dB Attenuate", AC97_SIGMATEL_ANALOG, 1, 1, 0), diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index ad2888705d2a..c111efe61c3c 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -800,7 +800,7 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) "Capture Volume", "External Amplifier", "Sigmatel 4-Speaker Stereo Playback Switch", - "Sigmatel Surround Phase Inversion Playback ", + "Surround Phase Inversion Playback Switch", NULL }; static char *ca0106_rename_ctls[] = { -- cgit v1.2.2 From 6af3fb72d2437239e5eb13a59e95dc43ccab3e8f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 May 2009 10:49:26 +0200 Subject: ALSA: Fix invalid jiffies check after pause The hw_ptr_jiffies has to be reset properly to avoid the invalid check of jiffies delta in snd_pcm_update_hw_ptr*() functions. Especailly this patch fixes the bogus jiffies check after the puase and resume. This patch is a modified version of the original patch by Jaroslav. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 1 - sound/core/pcm_native.c | 6 ++++++ 2 files changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a2a792c18c40..3eea98a4e65a 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1478,7 +1478,6 @@ static int snd_pcm_lib_ioctl_reset(struct snd_pcm_substream *substream, runtime->status->hw_ptr %= runtime->buffer_size; else runtime->status->hw_ptr = 0; - runtime->hw_ptr_jiffies = jiffies; snd_pcm_stream_unlock_irqrestore(substream, flags); return 0; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index fc6f98e257df..b5da656d1ece 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -848,6 +848,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); + runtime->hw_ptr_jiffies = jiffies; runtime->status->state = state; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) @@ -961,6 +962,11 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push) { if (substream->runtime->trigger_master != substream) return 0; + /* The jiffies check in snd_pcm_update_hw_ptr*() is done by + * a delta betwen the current jiffies, this gives a large enough + * delta, effectively to skip the check once. + */ + substream->runtime->hw_ptr_jiffies = jiffies - HZ * 1000; return substream->ops->trigger(substream, push ? SNDRV_PCM_TRIGGER_PAUSE_PUSH : SNDRV_PCM_TRIGGER_PAUSE_RELEASE); -- cgit v1.2.2 From c87d9732004b3f8fd82d729f12ccfb96c0df279e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 May 2009 10:53:33 +0200 Subject: ALSA: Enable PCM hw_ptr_jiffies check only in xrun_debug mode The PCM hw_ptr jiffies check results sometimes in problems when a hardware doesn't give smooth hw_ptr updates. So far, au88x0 and some other drivers appear not working due to this strict check. However, this check is a nice debug tool, and the capability should be still kept. Hence, we disable this check now as default unless the user enables it by setting the xrun_debug mode to the specific stream via a proc file. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 3eea98a4e65a..d659995ac3ac 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -249,6 +249,11 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) new_hw_ptr = hw_base + pos; } } + + /* Do jiffies check only in xrun_debug mode */ + if (!xrun_debug(substream)) + goto no_jiffies_check; + /* Skip the jiffies check for hardwares with BATCH flag. * Such hardware usually just increases the position at each IRQ, * thus it can't give any strange position. @@ -336,7 +341,9 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) hw_base = 0; new_hw_ptr = hw_base + pos; } - if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) { + /* Do jiffies check only in xrun_debug mode */ + if (xrun_debug(substream) && + ((delta * HZ) / runtime->rate) > jdelta + HZ/100) { hw_ptr_error(substream, "hw_ptr skipping! " "(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n", -- cgit v1.2.2