November 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC/mpc5200: remove duplicate identical IRQ handler
by Grant Likely 10 Nov '09

10 Nov '09

The TX and RX irq handlers are identical. Merge them Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> --- Hi Mark. Here's one more cleanup patch. sound/soc/fsl/mpc5200_dma.c | 33 +++------------------------------ 1 files changed, 3 insertions(+), 30 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index ee5a606..440b749 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -66,32 +66,7 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) } /* Bestcomm DMA irq handler */ -static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream) -{ - struct psc_dma_stream *s = _psc_dma_stream; - - spin_lock(&s->psc_dma->lock); - /* For each finished period, dequeue the completed period buffer - * and enqueue a new one in it's place. */ - while (bcom_buffer_done(s->bcom_task)) { - bcom_retrieve_buffer(s->bcom_task, NULL, NULL); - - s->period_current = (s->period_current+1) % s->runtime->periods; - s->period_count++; - - psc_dma_bcom_enqueue_next_buffer(s); - } - spin_unlock(&s->psc_dma->lock); - - /* If the stream is active, then also inform the PCM middle layer - * of the period finished event. */ - if (s->active) - snd_pcm_period_elapsed(s->stream); - - return IRQ_HANDLED; -} - -static irqreturn_t psc_dma_bcom_irq_rx(int irq, void *_psc_dma_stream) +static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream) { struct psc_dma_stream *s = _psc_dma_stream; @@ -486,11 +461,9 @@ int mpc5200_audio_dma_create(struct of_device *op) rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED, "psc-dma-status", psc_dma); - rc |= request_irq(psc_dma->capture.irq, - &psc_dma_bcom_irq_rx, IRQF_SHARED, + rc |= request_irq(psc_dma->capture.irq, &psc_dma_bcom_irq, IRQF_SHARED, "psc-dma-capture", &psc_dma->capture); - rc |= request_irq(psc_dma->playback.irq, - &psc_dma_bcom_irq_tx, IRQF_SHARED, + rc |= request_irq(psc_dma->playback.irq, &psc_dma_bcom_irq, IRQF_SHARED, "psc-dma-playback", &psc_dma->playback); if (rc) { ret = -ENODEV;

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[alsa-devel] [PATCH 0/4 v2] ASoC/MFD/OMAP: TWL4030: APLL_CTL handling change
by Peter Ujfalusi 10 Nov '09

10 Nov '09

Hello, Changes from the first series: - Typos fixed and warning/error messages for APLL frequency mismatch now consistent - twl4030-codec MFD driver audio_mclk validity check simplified - Commit message for the twl4030-codec MFD driver has been changed also - Patch: "ASoC: TWL4030: Make sure, that the codec is powered on startup" has been removed from the series, since it is already taken. Intro text from the first series: The following series changes the way how the APLL_CTL register is handled by the twl4030 codec related drivers. Moving the configuration of APLL_CTL register from child drivers (specifically from soc codec driver) to the twl4030-codec MFD driver. Because the audio_mclk is static, board specific and can not be changed in runtime, the codec MFD is the right place to configure it early enough, so child can use the hardware correctly. Before this patch for example the APLL_INFREQ was configured first, when the user played or recorded audio via the twl4030. The digital bypass needs correct APLL_INFREQ to be configured in order to work properly. If the digital bypass enabled before any audio activity, in some cases the bypassed audio was not correct. The series also addresses the issue, that after boot-up, neither of the loopbacks was working, because the codec was actually off. Setting the codec->bias_level to _OFF, before setting the codec bias level to STANDBY fixes this issue. This series is on top of Takashi's sound-2.6 topic/asoc branch, which has the twl403-codec MFD and related patches for 2.6.33. It would be really nice, if this series would make it also to the same branch, since this fixes some real anomalies brought in by the cleanup of the twl4030 codec related drivers. These problems were hidden before because of the inconsistent handling of bits and states in the codec driver in the past. --- Peter Ujfalusi (4): MFD: TWL4030: Add audio_mclk to the codec platform data OMAP: Configure audio_mclk for twl4030-codec MFD MFD: twl4030-codec: APLL_INFREQ handling in the MFD driver ASoC: TWL4030: Do not modify the APLL_CTL register arch/arm/mach-omap2/board-3430sdp.c | 1 + arch/arm/mach-omap2/board-omap3beagle.c | 1 + arch/arm/mach-omap2/board-omap3evm.c | 1 + arch/arm/mach-omap2/board-omap3pandora.c | 1 + arch/arm/mach-omap2/board-overo.c | 1 + arch/arm/mach-omap2/board-zoom2.c | 1 + drivers/mfd/twl4030-codec.c | 35 ++++++++++++++ include/linux/i2c/twl4030.h | 1 + include/linux/mfd/twl4030-codec.h | 1 + sound/soc/codecs/twl4030.c | 76 ++++++++++++------------------ 10 files changed, 74 insertions(+), 45 deletions(-)

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[alsa-devel] [PATCH V3] ASoC: omap-mcbsp - add support for upto 16 channels.
by Liam Girdwood 10 Nov '09

10 Nov '09

This time we use wlen * channels when CODEC masters BCLK and FRAME, otherwise same as V2. From: Graeme Gregory <gg(a)slimlogic.co.uk> This patch increases the number of supported audio channels from 4 to 16 and has been sponsored by Shotspotter Inc. It also fixes a FSYNC rate calculation bug when McBSP is FSYNC master. Signed-off-by: Graeme Gregory <gg(a)slimlogic.co.uk> Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/omap/omap-mcbsp.c | 63 +++++++++++++++++++++++++----------------- 1 files changed, 37 insertions(+), 26 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 3341f49..45be942 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -49,6 +49,8 @@ struct omap_mcbsp_data { */ int active; int configured; + unsigned int in_freq; + int clk_div; }; #define to_mcbsp(priv) container_of((priv), struct omap_mcbsp_data, bus_id) @@ -257,7 +259,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT; unsigned long port; - unsigned int format; + unsigned int format, div, framesize, master; if (cpu_class_is_omap1()) { dma = omap1_dma_reqs[bus_id][substream->stream]; @@ -294,28 +296,19 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); - switch (channels) { - case 2: - if (format == SND_SOC_DAIFMT_I2S) { - /* Use dual-phase frames */ - regs->rcr2 |= RPHASE; - regs->xcr2 |= XPHASE; - /* Set 1 word per (McBSP) frame for phase1 and phase2 */ - wpf--; - regs->rcr2 |= RFRLEN2(wpf - 1); - regs->xcr2 |= XFRLEN2(wpf - 1); - } - case 1: - case 4: - /* Set word per (McBSP) frame for phase1 */ - regs->rcr1 |= RFRLEN1(wpf - 1); - regs->xcr1 |= XFRLEN1(wpf - 1); - break; - default: - /* Unsupported number of channels */ - return -EINVAL; + if (channels == 2 && format == SND_SOC_DAIFMT_I2S) { + /* Use dual-phase frames */ + regs->rcr2 |= RPHASE; + regs->xcr2 |= XPHASE; + /* Set 1 word per (McBSP) frame for phase1 and phase2 */ + wpf--; + regs->rcr2 |= RFRLEN2(wpf - 1); + regs->xcr2 |= XFRLEN2(wpf - 1); } + regs->rcr1 |= RFRLEN1(wpf - 1); + regs->xcr1 |= XFRLEN1(wpf - 1); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: /* Set word lengths */ @@ -330,15 +323,30 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + /* In McBSP master modes, FRAME (i.e. sample rate) is generated + * by _counting_ BCLKs. Calculate frame size in BCLKs */ + master = mcbsp_data->fmt & SND_SOC_DAIFMT_MASTER_MASK; + if (master == SND_SOC_DAIFMT_CBS_CFS) { + div = mcbsp_data->clk_div ? mcbsp_data->clk_div : 1; + framesize = (mcbsp_data->in_freq / div) / params_rate(params); + + if (framesize < wlen * channels) { + printk(KERN_ERR "%s: not enough bandwidth for desired rate and " + "channels\n", __func__); + return -EINVAL; + } + } else + framesize = wlen * channels; + /* Set FS period and length in terms of bit clock periods */ switch (format) { case SND_SOC_DAIFMT_I2S: - regs->srgr2 |= FPER(wlen * channels - 1); - regs->srgr1 |= FWID(wlen - 1); + regs->srgr2 |= FPER(framesize - 1); + regs->srgr1 |= FWID((framesize >> 1) - 1); break; case SND_SOC_DAIFMT_DSP_A: case SND_SOC_DAIFMT_DSP_B: - regs->srgr2 |= FPER(wlen * channels - 1); + regs->srgr2 |= FPER(framesize - 1); regs->srgr1 |= FWID(0); break; } @@ -454,6 +462,7 @@ static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, if (div_id != OMAP_MCBSP_CLKGDV) return -ENODEV; + mcbsp_data->clk_div = div; regs->srgr1 |= CLKGDV(div - 1); return 0; @@ -554,6 +563,8 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; + mcbsp_data->in_freq = freq; + switch (clk_id) { case OMAP_MCBSP_SYSCLK_CLK: regs->srgr2 |= CLKSM; @@ -598,13 +609,13 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = { .id = (link_id), \ .playback = { \ .channels_min = 1, \ - .channels_max = 4, \ + .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ .capture = { \ .channels_min = 1, \ - .channels_max = 4, \ + .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ -- 1.6.3.3

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Re: [alsa-devel] RME HDSPe AIO
by Demian Martin 10 Nov '09

10 Nov '09

Flo, Why do you say "No. This is a pro audio card and it's full potential cannot be used with ALSA."? What is there in Alsa that will restrict the use of the card (or any card) for pro audio use? The Trinnov room correction system which is a pro market product uses Hammerfall cards and Alsa to do some very ambitious pro audio functions for example. Demian Martin Product Design Services -----Original Message----- Date: Sun, 08 Nov 2009 20:44:04 +0100 From: Florian Faber <faber(a)faberman.de> Subject: Re: [alsa-devel] RME HDSPe AIO To: Jonathan Leonard <jonathan.adams.leonard(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Message-ID: <4AF71F84.6090405(a)faberman.de> Content-Type: text/plain; charset=ISO-8859-1 Jonathan, > The AIO is purported to be very similar to the 9632 and should be > using the snd-hdsp module. No. It has the hdspe core and should be used as such. > I would like to use this card in ALSA, That's nice, but you should have checked before buying the card. The AIO is not supported. > Besides my individual circumstance I wonder if ALSA should not be > accomodating this great card somehow. No. This is a pro audio card and it's full potential cannot be used with ALSA. Flo -- Machines can do the work, so people have time to think. public key DA43FEF4 x-hkp://wwwkeys.eu.pgp.net

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[alsa-devel] Crackly sound with dmix rate 48000
by Daniel Drake 10 Nov '09

10 Nov '09

Hi, In our musical activities, the OLPC/Sugar platform uses csound which in turn uses ALSA for playing sounds. On our latest XO hardware model, where we now use HDA instead of cs5536, we are experiencing some crackly sounds from our csound-based applications. Other apps are fine. We discovered that changing the dmix rate from 48000 to 44100 solves the cracklyness. What does this indicate? Might csound be producing sounds at rate 44100, which dmix previously had to convert to 48000, which was too taxing to the system? Any suggestions for how to diagnose this further? (or would changing the rate be seen as a reasonable thing to do in this situation?) Thanks, Daniel

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[alsa-devel] [PATCH] ALSA: hda - Tweak OLPC XO-1.5 microphone bias
by Daniel Drake 10 Nov '09

10 Nov '09

Our contacts at Conexant suggested that we reduce the external microphone bias to 50% in order to center the input signal with the DC input range of the codec. This is because the microphone port is DC coupled for potential use with sensors. Signed-off-by: Daniel Drake <dsd(a)laptop.org> --- sound/pci/hda/patch_conexant.c | 15 ++++++++++++--- 1 files changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5d3a25d..ace8898 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -110,6 +110,7 @@ struct conexant_spec { unsigned int dell_automute; unsigned int port_d_mode; + unsigned char ext_mic_bias; }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -1917,6 +1918,11 @@ static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; #define CXT5066_SPDIF_OUT 0x21 +/* OLPC's microphone port is DC coupled for use with external sensors, + * therefore we use a 50% mic bias in order to center the input signal with + * the DC input range of the codec. */ +#define CXT5066_OLPC_EXT_MIC_BIAS PIN_VREF50 + static struct hda_channel_mode cxt5066_modes[1] = { { 2, NULL }, }; @@ -1970,9 +1976,10 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, /* toggle input of built-in and mic jack appropriately */ static void cxt5066_automic(struct hda_codec *codec) { - static struct hda_verb ext_mic_present[] = { + struct conexant_spec *spec = codec->spec; + struct hda_verb ext_mic_present[] = { /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias}, /* switch to external mic input */ {0x17, AC_VERB_SET_CONNECT_SEL, 0}, @@ -2225,7 +2232,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ /* Port B: external microphone */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS}, /* Port C: internal microphone */ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, @@ -2343,6 +2350,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->input_mux = &cxt5066_capture_source; spec->port_d_mode = PIN_HP; + spec->ext_mic_bias = PIN_VREF80; spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5066_init_verbs; @@ -2374,6 +2382,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->port_d_mode = 0; + spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS; /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; -- 1.6.2.5

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[alsa-devel] HDA timeouts
by Maxim Levitsky 10 Nov '09

10 Nov '09

Hi, Sometimes (1 in 5 boots) I see the following: azx_get_response timeout, switching to polling mode: last cmd=0x014f0900 I initially blamed MSI that I enabled explicitly. If it is enabled, there is another timeout just before this line. Now I explicitly disable msi, and still see this. It seems that codec hangs or so. my codec#0 attached. Best regards, Maxim Levitsky

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[alsa-devel] [PATCH 2/2] sh: ms7724se: Add runtime PM support for FSI
by Kuninori Morimoto 10 Nov '09

10 Nov '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- arch/sh/boards/mach-se/7724/setup.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index e78c3be..0894bba 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -313,6 +313,9 @@ static struct platform_device fsi_device = { .dev = { .platform_data = &fsi_info, }, + .archdata = { + .hwblk_id = HWBLK_SPU, /* FSI needs SPU hwblk */ + }, }; /* KEYSC in SoC (Needs SW33-2 set to ON) */ -- 1.6.3.3

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[alsa-devel] [PATCH] ALSA: hda: Use model=auto quirk for Sony VAIO VGN-FW170J using ALC262
by Dan Chen 09 Nov '09

09 Nov '09

Hi, Attached please find a patch against sound-2.6 HEAD using ALC262_AUTO for the Sony VAIO VGN-FW170J. Without using model=auto, the internal mic does not function. Thanks, Dan

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[alsa-devel] [PATCH] ASoC: Pandora: Pass SRG input clock frequency to the OMAP McBSP DAI
by Jarkko Nikula 09 Nov '09

09 Nov '09

Upcoming change to omap-mcbsp.c require that machine drivers using OMAP as a DAI master to pass sample rate generator input clock frequency to the omap-mcbsp.c DAI driver. Pandora is using 256*Fs output from the TWL4030 codec as an input clock to the McBSP sample rate generator. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Tested-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/omap/omap3pandora.c | 24 ++++++++++-------------- 1 files changed, 10 insertions(+), 14 deletions(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index ad219aa..cace5f1 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -40,9 +40,12 @@ #define PREFIX "ASoC omap3pandora: " -static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, - struct snd_soc_dai *cpu_dai, unsigned int fmt) +static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, unsigned int fmt) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* Set codec DAI configuration */ @@ -68,8 +71,9 @@ static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, } /* Set McBSP clock to external */ - ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, - SND_SOC_CLOCK_IN); + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, + 256 * params_rate(params), + SND_SOC_CLOCK_IN); if (ret < 0) { pr_err(PREFIX "can't set cpu system clock\n"); return ret; @@ -87,11 +91,7 @@ static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + return omap3pandora_cmn_hw_params(substream, params, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -100,11 +100,7 @@ static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + return omap3pandora_cmn_hw_params(substream, params, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); -- 1.6.5

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