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

[alsa-devel] EMU 0404USB: Odd sample rate shown during playback.
by Klaus Schulz 10 Jun '09

10 Jun '09

Hi there. The EMU 0404 is running in async USB mode as you know. When looking at the stream parameters during playback the sample rate shown is always slightly below 44100 -- something like 44098 is shown. As you write in the driver, the 0404 is returning number of samples per packet instead of the number of samples per microframes. Has this something to do with it? Question: Are we looking at a wrong output or is the samplerate really lower than 44100? THX Klaus

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[alsa-devel] [PATCH] ASoC: Automatically set TDM if needed on pxa-ssp and allow frame width override when using TDM.
by Daniel Ribeiro 10 Jun '09

10 Jun '09

* Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. * Reset SSCR0_EDSS and SSCR0_DSS on pxa_ssp_set_dai_fmt. * Makes SSCR0_MOD optional. * Automatically sets network mode when needed if set_tdm_slot was never called. * Clears SSCR1_RWOT case SSCR0_MOD is set. * Updates magician.c and wm9081.c for the new set_tdm_slot() (Patch is based on Mark's for-2.6.32 branch) Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- include/sound/soc-dai.h | 5 + sound/soc/codecs/wm9081.c | 4 - sound/soc/pxa/magician.c | 2 sound/soc/pxa/pxa-ssp.c | 119 ++++++++++++++++++++++++++++------------------ sound/soc/soc-core.c | 9 ++- 5 files changed, 85 insertions(+), 54 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7ee..f96cc36 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -106,7 +106,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, int slots, int frame_width); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); @@ -140,7 +140,8 @@ struct snd_soc_dai_ops { */ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, + int slots, int frame_width); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 86fc57e..85c720a 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1207,7 +1207,7 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, } static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int frame_width) { struct snd_soc_codec *codec = dai->codec; unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); @@ -1219,7 +1219,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; - switch (mask) { + switch (tx_mask) { case 1: break; case 2: diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index c89a3cd..2345869 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -188,7 +188,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1, 1, 16); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 46d14f3..f0931e7 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -375,21 +375,34 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, * Set the active slots in TDM/Network mode */ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int frame_width) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(7) | SSCR0_EDSS | SSCR0_DSS); + + /* set frame width */ + if (frame_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(frame_width - 16); + else + sscr0 |= SSCR0_DataSize(frame_width); + + if (slots > 1) { + /* enable network mode */ + sscr0 |= SSCR0_MOD; - /* set number of active slots */ - sscr0 |= SSCR0_SlotsPerFrm(slots); + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, tx_mask); + ssp_write_reg(ssp, SSRSA, rx_mask); + } ssp_write_reg(ssp, SSCR0, sscr0); - /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, mask); - ssp_write_reg(ssp, SSRSA, mask); return 0; } @@ -439,8 +452,8 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* reset port settings */ - sscr0 = ssp_read_reg(ssp, SSCR0) & - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + sscr0 = ssp_read_reg(ssp, SSCR0) & (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | + SSCR0_ACS | SSCR0_EDSS | SSCR0_DSS); sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); sspsp = 0; @@ -487,7 +500,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; case SND_SOC_DAIFMT_DSP_B: - sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -537,48 +550,70 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; int chn = params_channels(params); - u32 sscr0; - u32 sspsp; + u32 sscr0, sscr1, sspsp; int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + int frame_width; + + /* check if the user explicitly set a frame_width */ + sscr0 = ssp_read_reg(ssp, SSCR0); + + if (sscr0 & (SSCR0_EDSS | SSCR0_DSS)) + frame_width = (sscr0 & SSCR0_DSS) + + (sscr0 & SSCR0_EDSS ? 17 : 1); + else + frame_width = width * chn; /* generate correct DMA params */ if (cpu_dai->dma_data) kfree(cpu_dai->dma_data); - /* Network mode with one active slot (ttsa == 1) can be used - * to force 16-bit frame width on the wire (for S16_LE), even - * with two channels. Use 16-bit DMA transfers for this case. - */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, - ((chn == 2) && (ttsa != 1)) || (width == 32), + cpu_dai->dma_data = ssp_get_dma_params(ssp, frame_width > 16, substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + if (sscr0 & SSCR0_SSE) return 0; - /* clear selected SSP bits */ - sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - ssp_write_reg(ssp, SSCR0, sscr0); - - /* bit size */ - sscr0 = ssp_read_reg(ssp, SSCR0); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + /* FIXME: What this is for? */ #ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) - sscr0 |= SSCR0_FPCKE; + if (width == 16 && cpu_is_pxa3xx()) + sscr0 |= SSCR0_FPCKE; #endif - sscr0 |= SSCR0_DataSize(16); - break; - case SNDRV_PCM_FORMAT_S24_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - break; - case SNDRV_PCM_FORMAT_S32_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - break; + + if (!(sscr0 & (SSCR0_EDSS | SSCR0_DSS))) { + /* Frame width not set yet, we are not using network mode */ + if (frame_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(frame_width - 16); + else + sscr0 |= SSCR0_DataSize(frame_width); + + if (frame_width > 32) { + /* + * Network mode is needed to support this frame_width + * We assume that the wire is not networked and setup + * a "fake" network mode here. + */ + int slots = frame_width / 32; + + sscr0 |= SSCR0_MOD; + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* + * Set active slots. Only set an active TX slot + * if we are going to use it. + */ + ssp_write_reg(ssp, SSRSA, slots - 1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ssp_write_reg(ssp, SSTSA, slots - 1); + } } + + /* If SSCR0_MOD is set we can't use SSCR1_RWOT */ + if (sscr0 & SSCR0_MOD) { + sscr1 = ssp_read_reg(ssp, SSCR1); + ssp_write_reg(ssp, SSCR1, sscr1 & ~SSCR1_RWOT); + } + ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -625,14 +660,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, break; } - /* When we use a network mode, we always require TDM slots - * - complain loudly and fail if they've not been set up yet. - */ - if ((sscr0 & SSCR0_MOD) && !ttsa) { - dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); - return -EINVAL; - } - dump_registers(ssp); return 0; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1a920c..69becf2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2133,17 +2133,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** * snd_soc_dai_set_tdm_slot - configure DAI TDM. * @dai: DAI - * @mask: DAI specific mask representing used slots. + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. + * @frame_width: Width in bits for each slot. * * Configures a DAI for TDM operation. Both mask and slots are codec and DAI * specific. */ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int frame_width) { if (dai->ops && dai->ops->set_tdm_slot) - return dai->ops->set_tdm_slot(dai, mask, slots); + return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, frame_width); else return -EINVAL; } -- Daniel Ribeiro

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[alsa-devel] X-Fi suspend/resume support patch
by Takashi Iwai 10 Jun '09

10 Jun '09

Hi, can anyone test the patch below for supporting suspend/resume with X-Fi driver? Note that it's just compile tested but not really checked with any real hardware, so highly likely incomplete. Keep backup before trying this :) thanks, Takashi --- diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 80fb2ba..9861418 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1119,18 +1119,67 @@ static int atc_spdif_out_passthru(struct ct_atc *atc, unsigned char state) return err; } -static int ct_atc_destroy(struct ct_atc *atc) +static void ct_daio_clear_connections(struct ct_atc *atc) { struct daio_mgr *daio_mgr; struct dao *dao; struct dai *dai; struct daio *daio; + int i; + + daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO]; + for (i = 0; i < atc->n_daio; i++) { + daio = atc->daios[i]; + if (daio->type < LINEIM) { + dao = container_of(daio, struct dao, daio); + dao->ops->clear_left_input(dao); + dao->ops->clear_right_input(dao); + } else { + dai = container_of(daio, struct dai, daio); + /* some thing to do for dai ... */ + } + daio_mgr->put_daio(daio_mgr, daio); + } +} + +static void ct_sum_clear_connections(struct ct_atc *atc) +{ struct sum_mgr *sum_mgr; + int i; + + sum_mgr = atc->rsc_mgrs[SUM]; + for (i = 0; i < atc->n_pcm; i++) + sum_mgr->put_sum(sum_mgr, atc->pcm[i]); +} + +static void ct_src_clear_connections(struct ct_atc *atc) +{ struct src_mgr *src_mgr; + int i; + + src_mgr = atc->rsc_mgrs[SRC]; + for (i = 0; i < atc->n_src; i++) + src_mgr->put_src(src_mgr, atc->srcs[i]); +} + +static void ct_srcimp_clear_connections(struct ct_atc *atc) +{ struct srcimp_mgr *srcimp_mgr; struct srcimp *srcimp; + int i; + + srcimp_mgr = atc->rsc_mgrs[SRCIMP]; + for (i = 0; i < atc->n_srcimp; i++) { + srcimp = atc->srcimps[i]; + srcimp->ops->unmap(srcimp); + srcimp_mgr->put_srcimp(srcimp_mgr, atc->srcimps[i]); + } +} + +static int ct_atc_destroy(struct ct_atc *atc) +{ struct ct_mixer *mixer; - int i = 0; + int i; if (NULL == atc) return 0; @@ -1157,45 +1206,22 @@ static int ct_atc_destroy(struct ct_atc *atc) } if (NULL != atc->daios) { - daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO]; - for (i = 0; i < atc->n_daio; i++) { - daio = atc->daios[i]; - if (daio->type < LINEIM) { - dao = container_of(daio, struct dao, daio); - dao->ops->clear_left_input(dao); - dao->ops->clear_right_input(dao); - } else { - dai = container_of(daio, struct dai, daio); - /* some thing to do for dai ... */ - } - daio_mgr->put_daio(daio_mgr, daio); - } + ct_daio_clear_connections(atc); kfree(atc->daios); } if (NULL != atc->pcm) { - sum_mgr = atc->rsc_mgrs[SUM]; - for (i = 0; i < atc->n_pcm; i++) - sum_mgr->put_sum(sum_mgr, atc->pcm[i]); - + ct_sum_clear_connections(atc); kfree(atc->pcm); } if (NULL != atc->srcs) { - src_mgr = atc->rsc_mgrs[SRC]; - for (i = 0; i < atc->n_src; i++) - src_mgr->put_src(src_mgr, atc->srcs[i]); - + ct_src_clear_connections(atc); kfree(atc->srcs); } if (NULL != atc->srcimps) { - srcimp_mgr = atc->rsc_mgrs[SRCIMP]; - for (i = 0; i < atc->n_srcimp; i++) { - srcimp = atc->srcimps[i]; - srcimp->ops->unmap(srcimp); - srcimp_mgr->put_srcimp(srcimp_mgr, atc->srcimps[i]); - } + ct_srcimp_clear_connections(atc); kfree(atc->srcimps); } @@ -1314,37 +1340,15 @@ static int __devinit atc_create_hw_devs(struct ct_atc *atc) return 0; } -static int __devinit atc_get_resources(struct ct_atc *atc) +static int ct_daio_init_connections(struct ct_atc *atc) { struct daio_desc da_desc = {0}; struct daio_mgr *daio_mgr; - struct src_desc src_dsc = {0}; - struct src_mgr *src_mgr; - struct srcimp_desc srcimp_dsc = {0}; - struct srcimp_mgr *srcimp_mgr; - struct sum_desc sum_dsc = {0}; - struct sum_mgr *sum_mgr; - int err, i; - - atc->daios = kzalloc(sizeof(void *)*(DAIONUM), GFP_KERNEL); - if (NULL == atc->daios) - return -ENOMEM; - - atc->srcs = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL); - if (NULL == atc->srcs) - return -ENOMEM; - - atc->srcimps = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL); - if (NULL == atc->srcimps) - return -ENOMEM; - - atc->pcm = kzalloc(sizeof(void *)*(2*4), GFP_KERNEL); - if (NULL == atc->pcm) - return -ENOMEM; + int i, err; daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO]; da_desc.msr = atc->msr; - for (i = 0, atc->n_daio = 0; i < DAIONUM-1; i++) { + for (i = 0, atc->n_daio = 0; i < DAIONUM - 1; i++) { da_desc.type = i; err = daio_mgr->get_daio(daio_mgr, &da_desc, (struct daio **)&atc->daios[i]); @@ -1366,12 +1370,20 @@ static int __devinit atc_get_resources(struct ct_atc *atc) return err; } atc->n_daio++; + return 0; +} + +static int ct_src_init_connections(struct ct_atc *atc) +{ + struct src_desc src_dsc = {0}; + struct src_mgr *src_mgr; + int i, err; src_mgr = atc->rsc_mgrs[SRC]; src_dsc.multi = 1; src_dsc.msr = atc->msr; src_dsc.mode = ARCRW; - for (i = 0, atc->n_src = 0; i < (2*2); i++) { + for (i = 0, atc->n_src = 0; i < (2 * 2); i++) { err = src_mgr->get_src(src_mgr, &src_dsc, (struct src **)&atc->srcs[i]); if (err) @@ -1379,10 +1391,18 @@ static int __devinit atc_get_resources(struct ct_atc *atc) atc->n_src++; } + return 0; +} + +static int ct_srcimp_init_connections(struct ct_atc *atc) +{ + struct srcimp_desc srcimp_dsc = {0}; + struct srcimp_mgr *srcimp_mgr; + int i, err; srcimp_mgr = atc->rsc_mgrs[SRCIMP]; srcimp_dsc.msr = 8; /* SRCIMPs for S/PDIFIn SRT */ - for (i = 0, atc->n_srcimp = 0; i < (2*1); i++) { + for (i = 0, atc->n_srcimp = 0; i < (2 * 1); i++) { err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc, (struct srcimp **)&atc->srcimps[i]); if (err) @@ -1391,7 +1411,7 @@ static int __devinit atc_get_resources(struct ct_atc *atc) atc->n_srcimp++; } srcimp_dsc.msr = 8; /* SRCIMPs for LINE/MICIn SRT */ - for (i = 0; i < (2*1); i++) { + for (i = 0; i < (2 * 1); i++) { err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc, (struct srcimp **)&atc->srcimps[2*1+i]); if (err) @@ -1399,10 +1419,18 @@ static int __devinit atc_get_resources(struct ct_atc *atc) atc->n_srcimp++; } + return 0; +} + +static int ct_sum_init_connections(struct ct_atc *atc) +{ + struct sum_desc sum_dsc = {0}; + struct sum_mgr *sum_mgr; + int i, err; sum_mgr = atc->rsc_mgrs[SUM]; sum_dsc.msr = atc->msr; - for (i = 0, atc->n_pcm = 0; i < (2*4); i++) { + for (i = 0, atc->n_pcm = 0; i < (2 * 4); i++) { err = sum_mgr->get_sum(sum_mgr, &sum_dsc, (struct sum **)&atc->pcm[i]); if (err) @@ -1410,6 +1438,41 @@ static int __devinit atc_get_resources(struct ct_atc *atc) atc->n_pcm++; } + return 0; +} + +static int __devinit atc_get_resources(struct ct_atc *atc) +{ + int err; + + atc->daios = kzalloc(sizeof(void *)*(DAIONUM), GFP_KERNEL); + if (NULL == atc->daios) + return -ENOMEM; + + atc->srcs = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL); + if (NULL == atc->srcs) + return -ENOMEM; + + atc->srcimps = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL); + if (NULL == atc->srcimps) + return -ENOMEM; + + atc->pcm = kzalloc(sizeof(void *)*(2*4), GFP_KERNEL); + if (NULL == atc->pcm) + return -ENOMEM; + + err = ct_daio_init_connections(atc); + if (err < 0) + return err; + err = ct_src_init_connections(atc); + if (err < 0) + return err; + err = ct_srcimp_init_connections(atc); + if (err < 0) + return err; + err = ct_sum_init_connections(atc); + if (err < 0) + return err; err = ct_mixer_create(atc, (struct ct_mixer **)&atc->mixer); if (err) { @@ -1420,7 +1483,7 @@ static int __devinit atc_get_resources(struct ct_atc *atc) return 0; } -static void __devinit +static void atc_connect_dai(struct src_mgr *src_mgr, struct dai *dai, struct src **srcs, struct srcimp **srcimps) { @@ -1459,7 +1522,7 @@ atc_connect_dai(struct src_mgr *src_mgr, struct dai *dai, src_mgr->commit_write(src_mgr); /* Synchronously enable SRCs */ } -static void __devinit atc_connect_resources(struct ct_atc *atc) +static void atc_connect_resources(struct ct_atc *atc) { struct dai *dai; struct dao *dao; @@ -1617,3 +1680,51 @@ error1: printk(KERN_ERR "ctxfi: Something wrong!!!\n"); return err; } + +#ifdef CONFIG_PM +int ct_atc_suspend(struct ct_atc *atc) +{ + struct hw *hw = atc->hw; + int i; + + snd_power_change_state(atc->card, SNDRV_CTL_POWER_D3hot); + for (i = FRONT; i <= IEC958; i++) + ct_pcm_suspend(atc->pcm_devs[i]); + ct_mixer_suspend(atc->mixer); + + ct_daio_clear_connections(atc); + ct_sum_clear_connections(atc); + ct_src_clear_connections(atc); + ct_srcimp_clear_connections(atc); + + memset(atc->daios, 0, sizeof(void *) * DAIONUM); + memset(atc->pcm, 0, sizeof(void *) * (2 * 4)); + memset(atc->srcs, 0, sizeof(void *) * (2 *2)); + memset(atc->srcimps, 0, sizeof(void *) * (2*2)); + + hw->card_suspend(hw); + return 0; +} + +int ct_atc_resume(struct ct_atc *atc) +{ + struct hw *hw = atc->hw; + int err; + + err = hw->card_resume(hw); + if (err < 0) { + snd_card_disconnect(atc->card); + return -EIO; + } + + ct_daio_init_connections(atc); + ct_src_init_connections(atc); + ct_srcimp_init_connections(atc); + ct_sum_init_connections(atc); + + atc_connect_resources(atc); + ct_mixer_resume(atc->mixer); + snd_power_change_state(atc->card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index a033472..e8251ff 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -79,6 +79,7 @@ struct ct_atc { unsigned int rsr; /* reference sample rate in Hz */ unsigned int msr; /* master sample rate in rsr */ unsigned int pll_rate; /* current rate of Phase Lock Loop */ + struct snd_pcm *pcm_devs[NUM_CTALSADEVS]; int chip_type; int model; @@ -144,4 +145,9 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, struct ct_atc **ratc); int __devinit ct_atc_create_alsa_devs(struct ct_atc *atc); +#ifdef CONFIG_PM +int ct_atc_suspend(struct ct_atc *atc); +int ct_atc_resume(struct ct_atc *atc); +#endif + #endif /* CTATC_H */ diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h index 4a8e04f..6a3de6a 100644 --- a/sound/pci/ctxfi/cthardware.h +++ b/sound/pci/ctxfi/cthardware.h @@ -60,6 +60,10 @@ struct card_conf { struct hw { int (*card_init)(struct hw *hw, struct card_conf *info); int (*card_stop)(struct hw *hw); +#ifdef CONFIG_PM + int (*card_suspend)(struct hw *hw); + int (*card_resume)(struct hw *hw); +#endif int (*pll_init)(struct hw *hw, unsigned int rsr); int (*is_adc_source_selected)(struct hw *hw, enum ADCSRC source); int (*select_adc_source)(struct hw *hw, enum ADCSRC source); @@ -164,6 +168,8 @@ struct hw { void (*irq_callback)(void *data, unsigned int bit); void *irq_callback_data; + struct card_conf info; + struct pci_dev *pci; /* the pci kernel structure of this card */ int irq; unsigned long io_base; diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index cb69d9d..e1b6d8d 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1976,7 +1976,7 @@ static int hw_card_shutdown(struct hw *hw) return 0; } -static int hw_card_init(struct hw *hw, struct card_conf *info) +static int hw_init(struct hw *hw, struct card_conf *info) { int err; unsigned int gctl; @@ -2119,11 +2119,54 @@ static void hw_write_pci(struct hw *hw, u32 reg, u32 data) &container_of(hw, struct hw20k1, hw)->reg_pci_lock, flags); } +static int hw_card_init(struct hw *hw, struct card_conf *info) +{ + hw->info = *info; + return hw_init(hw, info); +} + +#ifdef CONFIG_PM +static int hw_card_suspend(struct hw *hw) +{ + hw_card_stop(hw); + pci_disable_device(hw->pci); + pci_save_state(hw->pci); + return 0; +} + +static int hw_card_resume(struct hw *hw) +{ + int err; + struct pci_dev *pci = hw->pci; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + if (pci_enable_device(pci) < 0) { + printk(KERN_ERR "ctxfi: pci_enable_device failed, " + "disabling device\n"); + return -EIO; + } + if (hw->model == CTUAA) { + err = uaa_to_xfi(pci); + if (err) + return err; + } + pci_set_master(pci); + + hw_init(hw, &hw->info); + return 0; +} +#endif + static struct hw ct20k1_preset __devinitdata = { .irq = -1, .card_init = hw_card_init, .card_stop = hw_card_stop, +#ifdef CONFIG_PM + .card_suspend = hw_card_suspend, + .card_resume = hw_card_resume, +#endif .pll_init = hw_pll_init, .is_adc_source_selected = hw_is_adc_input_selected, .select_adc_source = hw_adc_input_select, diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 4493a51..5db75ea 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -1845,7 +1845,6 @@ static int hw_card_start(struct hw *hw) { int err = 0; struct pci_dev *pci = hw->pci; - unsigned int gctl; err = pci_enable_device(pci); if (err < 0) @@ -1872,11 +1871,6 @@ static int hw_card_start(struct hw *hw) goto error2; } - /* Switch to 20k2 mode from UAA mode. */ - gctl = hw_read_20kx(hw, GLOBAL_CNTL_GCTL); - set_field(&gctl, GCTL_UAA, 0); - hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl); - /*if ((err = request_irq(pci->irq, ct_atc_interrupt, IRQF_SHARED, atc->chip_details->nm_card, hw))) { goto error3; @@ -1927,7 +1921,7 @@ static int hw_card_shutdown(struct hw *hw) return 0; } -static int hw_card_init(struct hw *hw, struct card_conf *info) +static int hw_init(struct hw *hw, struct card_conf *info) { int err; unsigned int gctl; @@ -1945,6 +1939,11 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) return err; } + /* Switch to 20k2 mode from UAA mode. */ + gctl = hw_read_20kx(hw, GLOBAL_CNTL_GCTL); + set_field(&gctl, GCTL_UAA, 0); + hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl); + /* PLL init */ err = hw_pll_init(hw, info->rsr); if (err < 0) @@ -2006,6 +2005,39 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) return 0; } +static int hw_card_init(struct hw *hw, struct card_conf *info) +{ + hw->info = *info; + return hw_init(hw, info); +} + +#ifdef CONFIG_PM +static int hw_card_suspend(struct hw *hw) +{ + hw_card_stop(hw); + pci_disable_device(hw->pci); + pci_save_state(hw->pci); + return 0; +} + +static int hw_card_resume(struct hw *hw) +{ + struct pci_dev *pci = hw->pci; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + if (pci_enable_device(pci) < 0) { + printk(KERN_ERR "ctxfi: pci_enable_device failed, " + "disabling device\n"); + return -EIO; + } + pci_set_master(pci); + + hw_init(hw, &hw->info); + return 0; +} +#endif + static u32 hw_read_20kx(struct hw *hw, u32 reg) { return readl((void *)(hw->mem_base + reg)); @@ -2021,6 +2053,10 @@ static struct hw ct20k2_preset __devinitdata = { .card_init = hw_card_init, .card_stop = hw_card_stop, +#ifdef CONFIG_PM + .card_suspend = hw_card_suspend, + .card_resume = hw_card_resume, +#endif .pll_init = hw_pll_init, .is_adc_source_selected = hw_is_adc_input_selected, .select_adc_source = hw_adc_input_select, diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index 666722d..4e5547a 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -485,6 +485,57 @@ static int ct_alsa_mix_switch_get(struct snd_kcontrol *kcontrol, return 0; } +static void ct_mixer_switch_update(struct ct_atc *atc, + struct ct_mixer *mixer, + enum CTALSA_MIXER_CTL type, + int state) +{ + /* Do changes in mixer. */ + if (SWH_CAPTURE_START <= type && SWH_CAPTURE_END >= type) { + enum CT_AMIXER_CTL idx = get_amixer_index(type); + if (state) + ct_mixer_recording_select(mixer, idx); + else + ct_mixer_recording_unselect(mixer, idx); + return; + } + + /* Do changes out of mixer. */ + switch (type) { + case MIXER_LINEIN_C_S: + case MIXER_MIC_C_S: + if (state) + do_line_mic_switch(atc, type); + break; + case MIXER_WAVEF_P_S: + atc->line_front_unmute(atc, state); + break; + case MIXER_WAVES_P_S: + atc->line_surround_unmute(atc, state); + break; + case MIXER_WAVEC_P_S: + atc->line_clfe_unmute(atc, state); + break; + case MIXER_WAVER_P_S: + atc->line_rear_unmute(atc, state); + break; + case MIXER_LINEIN_P_S: + atc->line_in_unmute(atc, state); + break; + case MIXER_SPDIFO_P_S: + atc->spdif_out_unmute(atc, state); + break; + case MIXER_SPDIFI_P_S: + atc->spdif_in_unmute(atc, state); + break; + case MIXER_DIGITAL_IO_S: + do_digit_io_switch(atc, state); + break; + default: + break; + } +} + static int ct_alsa_mix_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -498,35 +549,7 @@ static int ct_alsa_mix_switch_put(struct snd_kcontrol *kcontrol, return 0; set_switch_state(mixer, type, state); - /* Do changes in mixer. */ - if ((SWH_CAPTURE_START <= type) && (SWH_CAPTURE_END >= type)) { - if (state) { - ct_mixer_recording_select(mixer, - get_amixer_index(type)); - } else { - ct_mixer_recording_unselect(mixer, - get_amixer_index(type)); - } - } - /* Do changes out of mixer. */ - if (state && (MIXER_LINEIN_C_S == type || MIXER_MIC_C_S == type)) - do_line_mic_switch(atc, type); - else if (MIXER_WAVEF_P_S == type) - atc->line_front_unmute(atc, state); - else if (MIXER_WAVES_P_S == type) - atc->line_surround_unmute(atc, state); - else if (MIXER_WAVEC_P_S == type) - atc->line_clfe_unmute(atc, state); - else if (MIXER_WAVER_P_S == type) - atc->line_rear_unmute(atc, state); - else if (MIXER_LINEIN_P_S == type) - atc->line_in_unmute(atc, state); - else if (MIXER_SPDIFO_P_S == type) - atc->spdif_out_unmute(atc, state); - else if (MIXER_SPDIFI_P_S == type) - atc->spdif_in_unmute(atc, state); - else if (MIXER_DIGITAL_IO_S == type) - do_digit_io_switch(atc, state); + ct_mixer_switch_update(atc, mixer, type, state); return 1; } @@ -823,7 +846,6 @@ error1: static int ct_mixer_get_mem(struct ct_mixer **rmixer) { struct ct_mixer *mixer; - int err; *rmixer = NULL; /* Allocate mem for mixer obj */ @@ -833,24 +855,27 @@ static int ct_mixer_get_mem(struct ct_mixer **rmixer) mixer->amixers = kzalloc(sizeof(void *)*(NUM_CT_AMIXERS*CHN_NUM), GFP_KERNEL); - if (NULL == mixer->amixers) { - err = -ENOMEM; - goto error1; - } + if (!mixer->amixers) + goto error; mixer->sums = kzalloc(sizeof(void *)*(NUM_CT_SUMS*CHN_NUM), GFP_KERNEL); - if (NULL == mixer->sums) { - err = -ENOMEM; - goto error2; - } + if (!mixer->sums) + goto error; + +#ifdef CONFIG_PM + mixer->saved_vol = kmalloc(sizeof(u32) * VOL_MIXER_NUM * CHN_NUM, + GFP_KERNEL); + if (!mixer->saved_vol) + goto error; +#endif *rmixer = mixer; return 0; -error2: +error: kfree(mixer->amixers); -error1: + kfree(mixer->sums); kfree(mixer); - return err; + return -ENOMEM; } static int ct_mixer_topology_build(struct ct_mixer *mixer) @@ -1039,13 +1064,13 @@ mixer_set_input_right(struct ct_mixer *mixer, return 0; } -int ct_mixer_destroy(struct ct_mixer *mixer) +static void ct_mixer_release_resources(struct ct_mixer *mixer) { struct sum_mgr *sum_mgr = (struct sum_mgr *)mixer->atc->rsc_mgrs[SUM]; struct amixer_mgr *amixer_mgr = (struct amixer_mgr *)mixer->atc->rsc_mgrs[AMIXER]; struct amixer *amixer; - int i = 0; + int i; /* Release amixer resources */ for (i = 0; i < (NUM_CT_AMIXERS * CHN_NUM); i++) { @@ -1060,8 +1085,15 @@ int ct_mixer_destroy(struct ct_mixer *mixer) if (NULL != mixer->sums[i]) sum_mgr->put_sum(sum_mgr, (struct sum *)mixer->sums[i]); } +} +int ct_mixer_destroy(struct ct_mixer *mixer) +{ + ct_mixer_release_resources(mixer); /* Release mem assigned to mixer object */ +#ifdef CONFIG_PM + kfree(mixer->saved_vol); +#endif kfree(mixer->sums); kfree(mixer->amixers); kfree(mixer); @@ -1121,3 +1153,49 @@ int ct_alsa_mix_create(struct ct_atc *atc, return 0; } + +#ifdef CONFIG_PM +void ct_mixer_suspend(struct ct_mixer *mixer) +{ + struct ct_atc *atc = mixer->atc; + int i, c, idx; + + idx = 0; + for (i = VOL_MIXER_START; i <= VOL_MIXER_END; i++) { + for (c = 0; c < CHN_NUM; c++) { + struct amixer *amixer = mixer->amixers[idx]; + mixer->saved_vol[idx] = amixer->ops->get_scale(amixer); + idx++; + } + } + + atc->spdif_out_get_status(atc, &mixer->saved_spdif_status); + ct_mixer_release_resources(mixer); +} + +void ct_mixer_resume(struct ct_mixer *mixer) +{ + struct ct_atc *atc = mixer->atc; + int i, c, idx; + + ct_mixer_topology_build(mixer); + + idx = 0; + for (i = VOL_MIXER_START; i <= VOL_MIXER_END; i++) { + for (c = 0; c < CHN_NUM; c++) { + struct amixer *amixer = mixer->amixers[idx]; + amixer->ops->set_scale(amixer, mixer->saved_vol[idx]); + idx++; + } + } + + for (i = SWH_MIXER_START; i <= SWH_MIXER_END; i++) { + if (i == MIXER_DIGITAL_IO_S && + !atc->have_digit_io_switch(atc)) + continue; + ct_mixer_switch_update(atc, mixer, i, + get_switch_state(mixer, i)); + } + atc->spdif_out_set_status(atc, mixer->saved_spdif_status); +} +#endif diff --git a/sound/pci/ctxfi/ctmixer.h b/sound/pci/ctxfi/ctmixer.h index e2d96eb..597a0c9 100644 --- a/sound/pci/ctxfi/ctmixer.h +++ b/sound/pci/ctxfi/ctmixer.h @@ -48,6 +48,10 @@ struct ct_mixer { void **amixers; /* amixer resources for volume control */ void **sums; /* sum resources for signal collection */ unsigned int switch_state; /* A bit-map to indicate state of switches */ +#ifdef CONFIG_PM + u32 *saved_vol; + unsigned int saved_spdif_status; +#endif int (*get_output_ports)(struct ct_mixer *mixer, enum MIXER_PORT_T type, struct rsc **rleft, struct rsc **rright); @@ -64,4 +68,9 @@ int ct_alsa_mix_create(struct ct_atc *atc, int ct_mixer_create(struct ct_atc *atc, struct ct_mixer **rmixer); int ct_mixer_destroy(struct ct_mixer *mixer); +#ifdef CONFIG_PM +void ct_mixer_suspend(struct ct_mixer *mixer); +void ct_mixer_resume(struct ct_mixer *mixer); +#endif + #endif /* CTMIXER_H */ diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 9e5c0c4..6f872ba 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -422,5 +422,31 @@ int ct_alsa_pcm_create(struct ct_atc *atc, snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(atc->pci), 128*1024, 128*1024); + atc->pcm_devs[device] = pcm; + return 0; } + +#ifdef CONFIG_PM +void ct_pcm_suspend(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *s; + int i; + + if (!pcm) + return; + snd_pcm_suspend_all(pcm); + /* clear all PCM resources once */ + for (i = 0; i < 2; i++) { + for (s = pcm->streams[i].substream; s; s = s->next) { + struct ct_atc *atc; + struct ct_atc_pcm *apcm; + if (!s->runtime) + continue; + apcm = s->runtime->private_data; + atc = snd_pcm_substream_chip(s); + atc->pcm_release_resources(atc, apcm); + } + } +} +#endif diff --git a/sound/pci/ctxfi/ctpcm.h b/sound/pci/ctxfi/ctpcm.h index 178da0d..867139b 100644 --- a/sound/pci/ctxfi/ctpcm.h +++ b/sound/pci/ctxfi/ctpcm.h @@ -24,4 +24,8 @@ int ct_alsa_pcm_create(struct ct_atc *atc, enum CTALSADEVS device, const char *device_name); +#ifdef CONFIG_PM +void ct_pcm_suspend(struct snd_pcm *pcm); +#endif + #endif /* CTPCM_H */ diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index 2d3dd89..59a0876 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -121,11 +121,31 @@ static void __devexit ct_card_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } +#ifdef CONFIG_PM +static int ct_card_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct ct_atc *atc = card->private_data; + return ct_atc_suspend(atc); +} + +static int ct_card_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct ct_atc *atc = card->private_data; + return ct_atc_resume(atc); +} +#endif + static struct pci_driver ct_driver = { .name = "SB-XFi", .id_table = ct_pci_dev_ids, .probe = ct_card_probe, .remove = __devexit_p(ct_card_remove), +#ifdef CONFIG_PM + .suspend = ct_card_suspend, + .resume = ct_card_resume, +#endif }; static int __init ct_card_init(void)

1 1

[alsa-devel] ASoC update for 2.6.31
by Mark Brown 10 Jun '09

10 Jun '09

The following changes since commit 74b8f955a73d20b1e22403fd1ef85834fbf38d98: Mark Brown (1): ASoC: Apostrophe patrol are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Mark Brown (1): ASoC: Fix lm4857 control sound/soc/s3c24xx/neo1973_wm8753.c | 16 ++++++++++------ 1 files changed, 10 insertions(+), 6 deletions(-)

2 1

[alsa-devel] Sound streaming application: which device to use?
by Lex Wassenberg 10 Jun '09

10 Jun '09

Hi all, I'm completely new to ALSA, so forgive me if these are newby questions. I'm working on an application which receives sound chunks via an IP connection, and it should use the local sound card to make this sound audible. Up to now it used OSS, but there were some problems with that, so I'm busy with rewriting the application to ALSA. The sound is 16-bit stereo, sampled at 8000 Hz, and it comes in chunks of 1024 frames. I managed to get things more or less working, but there are still some things unclear to me. 1. If I open the soundcard with device "default" or device "plughw" I get good sound, but the device "default" only accepts 1020 frames of every 1024 that I offer it (as can be seen from the return value of snd_pcm_writei() ). The device "plughw" nicely accepts all 1024 frames. Why is this? In both cases I have set the sample rate to 8000 via a call to snd_pcm_hw_params_set_rate_near() and the buffer size to 2048 frames via snd_pcm_hw_params_set_buffer_size_near(). 2. If I use device "default", there seems to be no obligation to set the buffer size nor the period size, I still get good sound at the output. If I use "plughw", I have to set either the buffer size (2048 frames) or the period size (1024 frames), otherwise I get distorted sound. If I set both (the buffer size being 2048 and the period size being 1024) I also get distorted sound. Why? 3. If I set the buffer size twice the period size (as above), and ask the sizes back, ALSA nicely reports the buffer size as being twice the period size. But if I ask the buffer and period TIMES, they are reported as being the same: 128000 us! Huh???? 4. The return value of snd_pcm_hw_params_set_rate_near() is sometimes 8000, sometimes 7999, from the very same program. I would expect ALSA to always return the same value, be it what I request or not, but at least being deterministic. How can this be? 5. There is another application running on the same target system, which will sometimes produce audible alarms. However, if my application is running that other application cannot generate the alarms, it produces a "resource busy" error. It is clear to me that my application is using the sound card exclusively, but how can I get it to allow other applications access to it too? I tried using the device "dmix", but then my sound is again very distorted. Hope that somebody can help me out. Thanks in advance. Lex

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[alsa-devel] Codec 1708b
by Dark Dragon 10 Jun '09

10 Jun '09

I cant seem to get my m3873-vm asus motherboard to play Sorround Sound audio via digital (HDMI) Profile at http://www.alsa-project.org/db/?f=26d1122ea384bc64805e516610c1bab3bea4d75c Works when playing sterio audio but "file now found" errors when trying to play sorround sound audio

1 0

[alsa-devel] Problems with X-Fi driver
by Daniel Bankmann 09 Jun '09

09 Jun '09

Takashi Iwai schrieb: > At Fri, 22 May 2009 11:54:47 +0200, > Daniel Bankmann wrote: >> Takashi Iwai schrieb: >>> At Thu, 21 May 2009 09:53:23 +0200, >>> Daniel Bankmann wrote: >>>> Hi! >>>> >>>> I just tested the newest driver version with Open Suse 11.1 and >>>> pae-kernel. I seems not to work probably, although the latest "official" >>>> driver by Creative worked more or less good. >>>> >>>> The problem is, that always, when hearing some music or any other sound, >>>> the signal gets interrupted every few seconds. At the other side, the >>>> startup jingle for example is accompanied by a knock every few seconds. >>>> >>>> Any idea, or what information do you need to fix this? >>> If you are using pulseaudio, try to upgrade to the one in Factory. >>> PA on 11.1 is really a crappy version, unfortunately. >>> >>> >>> Takashi >>> >> Hi, >> >> are you sure, that this is a problem with PA? > > Depends on your system setup. As default, PA is used on KDE and GNOME, > regardless of h/w multi-stream functions. > >> Upgrading to the newest >> version or the one in Factory seems to be a bit tricky (because auf >> missing dependencies). > > You can upgrade the packages on OBS multimedia:libs repo. > >> Shouldn't the driver just work without PA and ESD installed? I tried >> that out, but the problem still resists. > > It's not the driver to decide whether to use PA or not. It's the > upper part. > > If the problem still occurs without PA (and you make sure that you > don't use it), the issue is likely the hwptr update timing of the > ctxfi driver. The recent PCM core has a more strict check, and > possibly ctxfi might not provide the correct DMA position. > > Anyway, please check the problem occurs with which apps more exactly. > At best, try to reproduce with aplay and speaker-test programs, i.e. > with less complexity. And, please report back to alsa-devel ML. > (Note that your post may appear later after manual approvals if you > aren't subscribed yet). > > > thanks, > > Takashi > Hi, so I tried reproducing this problem. Here's a link to a wav-file which includes tests with aplay and the starting/shutdown -jingle. I hope this can give you a better idea of my problem;) Daniel http://rapidshare.com/files/239574432/output_test.wav

2 4

[alsa-devel] Default enabling of mixer controls
by Aggarwal, Anuj 09 Jun '09

09 Jun '09

Hi, I want to by-default enable the following mixer controls for my platform: Analog Left Capture Route AUXL Analog Right Capture Route AUXR Right now, they both come disabled in the default configuration and I have to enable them explicitly by issuing the amixer cset command. How can I do that automatically? Thanks and Regards, Anuj Aggarwal

3 6

[alsa-devel] Setting gain/volume of PCM capture or playback
by erumusele Odigie 09 Jun '09

09 Jun '09

Hi, I am new to programming with ALSA, and I would like to find out how to change the gain (capture or playback) for my PCM device in my C code. I am aware I am supposed to use the mixer API, possibly functions like: snd_mixer_selem_set_playback_volume() but I don't know how this function can be connected to my PCM structure/device. Please whoever answers this should kindly attach some sample code, or a link to sample code because I can't find any on line. Thanks a lot. Erumusele

2 3

[alsa-devel] Testers wanted: New SB X-Fi driver
by Gernot Kohlhaas 09 Jun '09

09 Jun '09

Hi and thanks for this driver. It works great and even 6- and 8-Channel-Output is supported now. The only two little things I noticed are: 1. Most programs are looking for "surround51" to use for multichannel sound. But this driver only provides the "default"-Device (in it's description it says: aplay -L default:CARD=XFi Creative X-Fi, Front/WaveIn Default Audio Device ) 2. I can hear a gentle grumbling out of my speakers. But nothing serious. -- http://www.gnupp.de/verschluesselung/warum.html Key ID: 0x4C0B2250 Key: http://home.arcor.de/mr_maniac/ANDERES/public.gpg

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[alsa-devel] [PATCH 4/7] Alchemy: DB1200 AC97+I2S audio support.
by Manuel Lauss 08 Jun '09

08 Jun '09

Replaces the sample Alchemy PSC AC97 machine code with a DB1200 machine driver with AC97 and I2S support. AC97/I2S can be selected at boot time by setting switch S6.7. Cc: alsa-devel(a)alsa-project.org Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- arch/mips/alchemy/devboards/db1200/platform.c | 12 ++ sound/soc/au1x/Kconfig | 10 +- sound/soc/au1x/Makefile | 4 +- sound/soc/au1x/db1200.c | 189 +++++++++++++++++++++++++ sound/soc/au1x/sample-ac97.c | 144 ------------------- 5 files changed, 209 insertions(+), 150 deletions(-) create mode 100644 sound/soc/au1x/db1200.c delete mode 100644 sound/soc/au1x/sample-ac97.c diff --git a/arch/mips/alchemy/devboards/db1200/platform.c b/arch/mips/alchemy/devboards/db1200/platform.c index cc3f663..b3142e0 100644 --- a/arch/mips/alchemy/devboards/db1200/platform.c +++ b/arch/mips/alchemy/devboards/db1200/platform.c @@ -428,6 +428,7 @@ static int __init db1200_dev_init(void) ARRAY_SIZE(db1200_i2c_devs)); /* SWITCHES: S6.8 I2C/SPI selector (OFF=I2C ON=SPI) + * S6.7 AC97/I2S selector (OFF=AC97 ON=I2S) */ /* NOTE: GPIO215 controls OTG VBUS supply. In SPI mode however @@ -466,6 +467,17 @@ static int __init db1200_dev_init(void) au_writel(pfc, SYS_PINFUNC); au_sync(); + /* audio is handled in sound/soc/au1x/db1200.c; + * however we need I2C to talk to I2S codec. + */ + if ((bcsr->switches & 3) == BCSR_SWITCHES_DIP_8) { + bcsr->resets |= BCSR_RESETS_PSC1MUX; + printk(KERN_INFO " S6.7 ON : PSC1 mode I2S\n"); + } else { + bcsr->resets &= ~BCSR_RESETS_PSC1MUX; + printk(KERN_INFO " S6.7 OFF: PSC1 mode AC97\n"); + } + return platform_add_devices(db1200_devs, ARRAY_SIZE(db1200_devs)); } device_initcall(db1200_dev_init); diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig index 410a893..4b67140 100644 --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -22,11 +22,13 @@ config SND_SOC_AU1XPSC_AC97 ## ## Boards ## -config SND_SOC_SAMPLE_PSC_AC97 - tristate "Sample Au12x0/Au1550 PSC AC97 sound machine" +config SND_SOC_DB1200 + tristate "DB1200 AC97+I2S audio support" depends on SND_SOC_AU1XPSC select SND_SOC_AU1XPSC_AC97 select SND_SOC_AC97_CODEC + select SND_SOC_AU1XPSC_I2S + select SND_SOC_WM8731 help - This is a sample AC97 sound machine for use in Au12x0/Au1550 - based systems which have audio on PSC1 (e.g. Db1200 demoboard). + Select this option to enable audio (AC97 or I2S) on the + Alchemy/AMD/RMI DB1200 demoboard. diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile index 6c6950b..1687307 100644 --- a/sound/soc/au1x/Makefile +++ b/sound/soc/au1x/Makefile @@ -8,6 +8,6 @@ obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o # Boards -snd-soc-sample-ac97-objs := sample-ac97.o +snd-soc-db1200-objs := db1200.o -obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o +obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c new file mode 100644 index 0000000..a2cb02c --- /dev/null +++ b/sound/soc/au1x/db1200.c @@ -0,0 +1,189 @@ +/* + * DB1200 ASoC audio support code. + * + * (c) 2008 Manuel Lauss <manuel.lauss(a)gmail.com> + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/mach-au1x00/au1000.h> +#include <asm/mach-au1x00/au1xxx.h> +#include <asm/mach-au1x00/au1xxx_psc.h> +#include <asm/mach-au1x00/au1xxx_dbdma.h> + +/* it sucks that the ASoC headers are not under include/ */ +#include "../codecs/ac97.h" +#include "../codecs/wm8731.h" +#include "psc.h" + +/*------------------------- AC97 PART ---------------------------*/ + +static int db1200_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_sync(codec); + return 0; +} + +static struct snd_soc_dai_link db1200_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &au1xpsc_ac97_dai, + .codec_dai = &ac97_dai, + .init = db1200_ac97_init, +}; + +static struct snd_soc_card db1200_ac97_machine = { + .name = "DB1200 AC97 Audio", + .dai_link = &db1200_ac97_dai, + .num_links = 1, + .platform = &au1xpsc_soc_platform, +}; + +static struct snd_soc_device db1200_ac97_devdata = { + .card = &db1200_ac97_machine, + .codec_dev = &soc_codec_dev_ac97, +}; + +/*------------------------- I2S PART ---------------------------*/ + +static int db1200_i2s_startup(struct snd_pcm_substream *substream) +{ + 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; + + /* WM8731 has its own 12MHz crystal */ + snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + + /* codec is bitclock and lrclk master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + goto out; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + goto out; + + ret = 0; +out: + return ret; +} + +static struct snd_soc_ops db1200_i2s_wm8731_ops = { + .startup = db1200_i2s_startup, +}; + +static int db1200_i2s_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_sync(codec); + return 0; +} + +static struct snd_soc_dai_link db1200_i2s_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &au1xpsc_i2s_dai, + .codec_dai = &wm8731_dai, + .init = db1200_i2s_init, + .ops = &db1200_i2s_wm8731_ops, +}; + +static struct snd_soc_card db1200_i2s_machine = { + .name = "DB1200 I2S Audio", + .dai_link = &db1200_i2s_dai, + .num_links = 1, + .platform = &au1xpsc_soc_platform, +}; + +static struct snd_soc_device db1200_i2s_devdata = { + .card = &db1200_i2s_machine, + .codec_dev = &soc_codec_dev_wm8731, +}; + +/*------------------------- COMMON PART ---------------------------*/ + +static struct resource psc1_res[] = { + [0] = { + .start = CPHYSADDR(PSC1_BASE_ADDR), + .end = CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff, + .flags = IORESOURCE_MEM, + }, + [1] = { + .start = AU1200_PSC1_INT, + .end = AU1200_PSC1_INT, + .flags = IORESOURCE_IRQ, + }, + [2] = { + .start = DSCR_CMD0_PSC1_TX, + .end = DSCR_CMD0_PSC1_TX, + .flags = IORESOURCE_DMA, + }, + [3] = { + .start = DSCR_CMD0_PSC1_RX, + .end = DSCR_CMD0_PSC1_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device *db1200_asoc_dev; + +static int __init db1200_audio_load(void) +{ + int ret; + + /* DB1200: PSC clock is supplied externally */ + au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET); + au_sync(); + + ret = -ENOMEM; + db1200_asoc_dev = platform_device_alloc("soc-audio", -1); + if (!db1200_asoc_dev) + goto out; + + db1200_asoc_dev->resource = + kmemdup(psc1_res, sizeof(struct resource) * + ARRAY_SIZE(psc1_res), GFP_KERNEL); + db1200_asoc_dev->num_resources = ARRAY_SIZE(psc1_res); + db1200_asoc_dev->id = 1; + + /* DB1200 board setup set PSC1MUX to preferred audio device */ + if (bcsr->resets & BCSR_RESETS_PSC1MUX) + platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_devdata); + else + platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_devdata); + + db1200_ac97_devdata.dev = &db1200_asoc_dev->dev; + db1200_i2s_devdata.dev = &db1200_asoc_dev->dev; + ret = platform_device_add(db1200_asoc_dev); + + if (ret) { + platform_device_put(db1200_asoc_dev); + db1200_asoc_dev = NULL; + } +out: + return ret; +} + +static void __exit db1200_audio_unload(void) +{ + platform_device_unregister(db1200_asoc_dev); +} + +module_init(db1200_audio_load); +module_exit(db1200_audio_unload); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("DB1200 ASoC audio support"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c deleted file mode 100644 index 27683eb..0000000 --- a/sound/soc/au1x/sample-ac97.c +++ /dev/null @@ -1,144 +0,0 @@ -/* - * Sample Au12x0/Au1550 PSC AC97 sound machine. - * - * Copyright (c) 2007-2008 Manuel Lauss <mano(a)roarinelk.homelinux.net> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms outlined in the file COPYING at the root of this - * source archive. - * - * This is a very generic AC97 sound machine driver for boards which - * have (AC97) audio at PSC1 (e.g. DB1200 demoboards). - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <asm/mach-au1x00/au1000.h> -#include <asm/mach-au1x00/au1xxx_psc.h> -#include <asm/mach-au1x00/au1xxx_dbdma.h> - -#include "../codecs/ac97.h" -#include "psc.h" - -static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec) -{ - snd_soc_dapm_sync(codec); - return 0; -} - -static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = { - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai = &au1xpsc_ac97_dai, /* see psc-ac97.c */ - .codec_dai = &ac97_dai, /* see codecs/ac97.c */ - .init = au1xpsc_sample_ac97_init, - .ops = NULL, -}; - -static struct snd_soc_card au1xpsc_sample_ac97_machine = { - .name = "Au1xxx PSC AC97 Audio", - .dai_link = &au1xpsc_sample_ac97_dai, - .num_links = 1, -}; - -static struct snd_soc_device au1xpsc_sample_ac97_devdata = { - .card = &au1xpsc_sample_ac97_machine, - .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ - .codec_dev = &soc_codec_dev_ac97, -}; - -static struct resource au1xpsc_psc1_res[] = { - [0] = { - .start = CPHYSADDR(PSC1_BASE_ADDR), - .end = CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff, - .flags = IORESOURCE_MEM, - }, - [1] = { -#ifdef CONFIG_SOC_AU1200 - .start = AU1200_PSC1_INT, - .end = AU1200_PSC1_INT, -#elif defined(CONFIG_SOC_AU1550) - .start = AU1550_PSC1_INT, - .end = AU1550_PSC1_INT, -#endif - .flags = IORESOURCE_IRQ, - }, - [2] = { - .start = DSCR_CMD0_PSC1_TX, - .end = DSCR_CMD0_PSC1_TX, - .flags = IORESOURCE_DMA, - }, - [3] = { - .start = DSCR_CMD0_PSC1_RX, - .end = DSCR_CMD0_PSC1_RX, - .flags = IORESOURCE_DMA, - }, -}; - -static struct platform_device *au1xpsc_sample_ac97_dev; - -static int __init au1xpsc_sample_ac97_load(void) -{ - int ret; - -#ifdef CONFIG_SOC_AU1200 - unsigned long io; - - /* modify sys_pinfunc for AC97 on PSC1 */ - io = au_readl(SYS_PINFUNC); - io |= SYS_PINFUNC_P1C; - io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B); - au_writel(io, SYS_PINFUNC); - au_sync(); -#endif - - ret = -ENOMEM; - - /* setup PSC clock source for AC97 part: external clock provided - * by codec. The psc-ac97.c driver depends on this setting! - */ - au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET); - au_sync(); - - au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1); - if (!au1xpsc_sample_ac97_dev) - goto out; - - au1xpsc_sample_ac97_dev->resource = - kmemdup(au1xpsc_psc1_res, sizeof(struct resource) * - ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL); - au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res); - au1xpsc_sample_ac97_dev->id = 1; - - platform_set_drvdata(au1xpsc_sample_ac97_dev, - &au1xpsc_sample_ac97_devdata); - au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev; - ret = platform_device_add(au1xpsc_sample_ac97_dev); - - if (ret) { - platform_device_put(au1xpsc_sample_ac97_dev); - au1xpsc_sample_ac97_dev = NULL; - } - -out: - return ret; -} - -static void __exit au1xpsc_sample_ac97_exit(void) -{ - platform_device_unregister(au1xpsc_sample_ac97_dev); -} - -module_init(au1xpsc_sample_ac97_load); -module_exit(au1xpsc_sample_ac97_exit); - -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); -- 1.6.3.1

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[alsa-devel] PATCH: C-Media 6206 USB Audio
by Dan Allongo 08 Jun '09

08 Jun '09

I've been posting on the alsa-users list and I think I finally have a working patch now that I'd like to submit. The C-Media 6206 device is based off of CM106 and just needed a boot quirk added to set the registers properly for output. At least one other user can confirm that this patch works and that the SPDIF-out is also functioning now with this change. This is diff'd against today's snapshot so it should be good to go. Let me know if there's anything else that I need to do or change with this. Thanks, ---Dan --- alsa-driver/alsa-kernel/usb/usbaudio.c 2009-06-07 06:20:32.000000000 -0400 +++ alsa-driver/alsa-kernel/usb/usbaudio.c 2009-06-07 22:24:44.000000000 -0400 @@ -3291,6 +3291,21 @@ return snd_usb_cm106_write_int_reg(dev, 2, 0x8004); } +static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) +{ + /* + * C-Media CM6206 is based on CM106 with two additional + * registers that are not documented in the data sheet. + * Values here are chosen based on sniffing USB traffic + * under Windows. + */ + return snd_usb_cm106_write_int_reg(dev, 0, 0x200c) + + snd_usb_cm106_write_int_reg(dev, 1, 0x3000) + + snd_usb_cm106_write_int_reg(dev, 2, 0xf800) + + snd_usb_cm106_write_int_reg(dev, 3, 0x143f) + + snd_usb_cm106_write_int_reg(dev, 4, 0x0000) + + snd_usb_cm106_write_int_reg(dev, 5, 0x3000); +} /* * Setup quirks @@ -3577,6 +3592,12 @@ goto __err_val; } + /* C-Media CM6206 / CM106-Like Sound Device */ + if (id == USB_ID(0x0d8c, 0x0102)) { + if (snd_usb_cm6206_boot_quirk(dev) < 0) + goto __err_val; + } + /* * found a config. now register to ALSA */

2 2

[PATCH - CM6206 1/1] modified: sound/usb/usbaudio.c
by Dan Allongo 08 Jun '09

08 Jun '09

Added boot quirk for C-Media CM6206 device in snd_usb_audio_probe. The function snd_usb_cm6206_boot_quirk sets up six internal 16-bit registers in order to initialize the device. Values for the registers came from sniffing USB traffic under Windows since only four of the six are documented in the datasheet for CM106 and some reserved bits were also being set. Signed-off-by: Dan Allongo <gongo2k1(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index a8ef2cb..94e43ee 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3291,6 +3291,24 @@ static int snd_usb_cm106_boot_quirk(struct usb_device *dev) return snd_usb_cm106_write_int_reg(dev, 2, 0x8004); } +/* + * C-Media CM6206 is based on CM106 with two additional + * registers that are not documented in the data sheet. + * Values here are chosen based on sniffing USB traffic + * under Windows. + */ +static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) +{ + int err, reg; + int val[] = {0x200c,0x3000,0xf800,0x143f,0x0000,0x3000}; + + for (reg = 0; reg < 6; reg++) { + if ((err = snd_usb_cm106_write_int_reg(dev, reg, val[reg])) < 0) + return err; + } + + return err; +} /* * Setup quirks @@ -3577,6 +3595,12 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __err_val; } + /* C-Media CM6206 / CM106-Like Sound Device */ + if (id == USB_ID(0x0d8c, 0x0102)) { + if (snd_usb_cm6206_boot_quirk(dev) < 0) + goto __err_val; + } + /* * found a config. now register to ALSA */ -- 1.5.6.5 On Mon, Jun 8, 2009 at 1:41 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Sun, 7 Jun 2009 22:46:51 -0400, > Dan Allongo wrote: > > > > I've been posting on the alsa-users list and I think I finally have a > > working patch now that I'd like to submit. > > The C-Media 6206 device is based off of CM106 and just needed a boot > quirk > > added to set the registers properly for output. > > At least one other user can confirm that this patch works and that the > > SPDIF-out is also functioning now with this change. > > This is diff'd against today's snapshot so it should be good to go. > > Let me know if there's anything else that I need to do or change with > this. > > Thanks for the patch! I'd like certainly to apply it, however... > > > + return snd_usb_cm106_write_int_reg(dev, 0, 0x200c) + > > + snd_usb_cm106_write_int_reg(dev, 1, 0x3000) + > > + snd_usb_cm106_write_int_reg(dev, 2, 0xf800) + > > + snd_usb_cm106_write_int_reg(dev, 3, 0x143f) + > > + snd_usb_cm106_write_int_reg(dev, 4, 0x0000) + > > + snd_usb_cm106_write_int_reg(dev, 5, 0x3000); > > Summing the return values here look weird. > You want to return an error code, right? Then use a simple if > sequence, or write a loop there. > > Also, run $LINUX/scripts/checkpatch.pl to your patch and fix errors / > warnings suggested there. Then please repost the patch with your > sign-off. > > > thanks, > > Takashi >

1 0

[alsa-devel] [PATCH 0/7] DAPM power sequencing changes
by Mark Brown 08 Jun '09

08 Jun '09

I've implemented what I hope are some improvements to the DAPM power sequence application. Currently when it is implementing power changes DAPM will do one register write per widget. Normally this leads to multiple writes to the same register since power control tends to be clustered in a few registers, slowing things down. This can get noticable on slow or heavily contended buses, especially around resume. The overall effect of these changes is that DAPM will try to minimise the number of register writes it does when implementing power sequences. As with any substantial change in the power sequencing there is a risk that there may have been some performance regressions on some systems - my biggest concern is that some thing were being helped by the delays introduced by the repeated register writes and may require special handling to fix. My hope is that this won't happen if anything it should fix some problems by ensuring that both left and right channels of stereo paths get powered up together but without wide testing that's hard to guarantee. Any testing that you can do on your systems would be greatly appreciated. Obviously this is quite an invasive change so I don't intend to submit to ALSA until after the 2.6.31 merge window has closed. The patches can be found here for the time being: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git dapm Mark Brown (7): ASoC: Sort DAPM power sequences while building lists ASoC: Factor out DAPM sequence execution ASoC: Coalesce register writes for DAPM sequences ASoC: Sort specialised mixers and muxes together ASoC: Coalesce power updates for DAPM widgets with events ASoC: Coalesce power updates for PGAs ASoC: Make DAPM power sequence lists local variables include/sound/soc.h | 2 - sound/soc/soc-dapm.c | 302 +++++++++++++++++++++++++++++++++++++------------- 2 files changed, 227 insertions(+), 77 deletions(-)

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[alsa-devel] [PATCH] ARM: DaVinci: ASoc: Minor fixes to probe function
by chaithrika＠ti.com 08 Jun '09

08 Jun '09

From: Chaithrika U S <chaithrika(a)ti.com> Assign proper errors when platform resource claims fail. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to branch 'davinci' of: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git sound/soc/davinci/davinci-i2s.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 500d2f5..2f11970 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -469,7 +469,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea, *res; - int ret = 0; + int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { @@ -511,6 +511,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); + ret = -ENXIO; goto err_free_mem; } dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; @@ -518,6 +519,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); + ret = -ENXIO; goto err_free_mem; } dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; -- 1.5.6

2 1

[alsa-devel] M-Audio Transit: alsamixer shows 100% and -0.5db at the same time !?
by Klaus Schulz 08 Jun '09

08 Jun '09

Hi folks. My M-Audio Transit shows at 100% and 0.5db attentuation in alsamixer at the same time. See below printout. What could be the explanation for this behaviour? THX \Klaus amixer -Dhw:0 Simple mixer control 'PCM',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 254 Mono: Front Left: Playback 254 [100%] [0.50dB] [on] Front Right: Playback 254 [100%] [0.50dB] [on]

3 2

[alsa-devel] [PATCH v4 2/4] ARM: DaVinci: ASoC: Introduce platform driver model for dm644x, dm355
by Chaithrika U S 08 Jun '09

08 Jun '09

Introduce the platform driver model to get platform data for dm355 and dm644x. Register platform driver and acquire the resources in the probe function Since the platform specific code had been moved from machine driver to dm<soc>.c Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-i2s.c | 117 ++++++++++++++++++++++----------------- 1 files changed, 67 insertions(+), 50 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index b1ea52f..500d2f5 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -436,16 +436,40 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params, + .set_fmt = davinci_i2s_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_i2s_dai = { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_i2s_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_i2s_dai); + +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->cpu_dai; + struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; - struct resource *mem, *ioarea; - struct evm_snd_platform_data *pdata; - int ret; + struct resource *mem, *ioarea, *res; + int ret = 0; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { @@ -466,9 +490,7 @@ static int davinci_i2s_probe(struct platform_device *pdev, goto err_release_region; } - cpu_dai->private_data = dev; - - dev->clk = clk_get(&pdev->dev, NULL); + dev->clk = clk_get(&pdev->dev, pdata->clk_name); if (IS_ERR(dev->clk)) { ret = -ENODEV; goto err_free_mem; @@ -476,18 +498,35 @@ static int davinci_i2s_probe(struct platform_device *pdev, clk_enable(dev->clk); dev->base = (void __iomem *)IO_ADDRESS(mem->start); - pdata = pdev->dev.platform_data; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG); dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_free_mem; + } + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_free_mem; + } + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; + + davinci_i2s_dai.private_data = dev; + ret = snd_soc_register_dai(&davinci_i2s_dai); + if (ret != 0) + goto err_free_mem; + return 0; err_free_mem: @@ -498,62 +537,40 @@ err_release_region: return ret; } -static void davinci_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int 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->cpu_dai; - struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = davinci_i2s_dai.private_data; struct resource *mem; + snd_soc_unregister_dai(&davinci_i2s_dai); clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; - kfree(dev); - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(mem->start, (mem->end - mem->start) + 1); -} -#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 - -static struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, - .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params, - .set_fmt = davinci_i2s_set_dai_fmt, -}; + return 0; +} -struct snd_soc_dai davinci_i2s_dai = { - .name = "davinci-i2s", - .id = 0, - .probe = davinci_i2s_probe, - .remove = davinci_i2s_remove, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &davinci_i2s_dai_ops, +static struct platform_driver davinci_mcbsp_driver = { + .probe = davinci_i2s_probe, + .remove = davinci_i2s_remove, + .driver = { + .name = "davinci-asp", + .owner = THIS_MODULE, + }, }; -EXPORT_SYMBOL_GPL(davinci_i2s_dai); static int __init davinci_i2s_init(void) { - return snd_soc_register_dai(&davinci_i2s_dai); + return platform_driver_register(&davinci_mcbsp_driver); } module_init(davinci_i2s_init); static void __exit davinci_i2s_exit(void) { - snd_soc_unregister_dai(&davinci_i2s_dai); + platform_driver_unregister(&davinci_mcbsp_driver); } module_exit(davinci_i2s_exit); -- 1.5.6

3 2

[alsa-devel] [PATCH] ASoC: Allow 32bits registers
by Daniel Ribeiro 08 Jun '09

08 Jun '09

Replace the remaining unsigned shorts with unsigned ints. Tested with pcap2 codec (25 bits registers). Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- include/sound/soc.h | 4 ++-- sound/soc/soc-core.c | 28 ++++++++++++++-------------- sound/soc/soc-dapm.c | 16 ++++++++-------- 3 files changed, 24 insertions(+), 24 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a40bc6f..33b7de1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -212,9 +212,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, /* codec register bit access */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); + unsigned int mask, unsigned int value); int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); + unsigned int mask, unsigned int value); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1cd149b..c9f19d0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1262,10 +1262,10 @@ EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); * Returns 1 for change else 0. */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value) + unsigned int mask, unsigned int value) { int change; - unsigned short old, new; + unsigned int old, new; mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); @@ -1292,10 +1292,10 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits); * Returns 1 for change else 0. */ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value) + unsigned int mask, unsigned int value) { int change; - unsigned short old, new; + unsigned int old, new; mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); @@ -1580,7 +1580,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, bitmask; + unsigned int val, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1609,8 +1609,8 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val; - unsigned short mask, bitmask; + unsigned int val; + unsigned int mask, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1646,7 +1646,7 @@ int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short reg_val, val, mux; + unsigned int reg_val, val, mux; reg_val = snd_soc_read(codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; @@ -1685,8 +1685,8 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val; - unsigned short mask; + unsigned int val; + unsigned int mask; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -1846,7 +1846,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; val = (ucontrol->value.integer.value[0] & mask); if (invert) @@ -1912,7 +1912,7 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, unsigned int reg2 = mc->rreg; unsigned int shift = mc->shift; int max = mc->max; - unsigned int mask = (1<<fls(max))-1; + unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; ucontrol->value.integer.value[0] = @@ -1952,7 +1952,7 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; int err; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; val_mask = mask << shift; val = (ucontrol->value.integer.value[0] & mask); @@ -2044,7 +2044,7 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; int min = mc->min; - unsigned short val; + unsigned int val; val = (ucontrol->value.integer.value[0]+min) & 0xff; val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 735903a..75c372f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -227,7 +227,7 @@ static int dapm_connect_mixer(struct snd_soc_codec *codec, static int dapm_update_bits(struct snd_soc_dapm_widget *widget) { int change, power; - unsigned short old, new; + unsigned int old, new; struct snd_soc_codec *codec = widget->codec; /* check for valid widgets */ @@ -1205,7 +1205,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; int ret; val = (ucontrol->value.integer.value[0] & mask); @@ -1269,7 +1269,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, bitmask; + unsigned int val, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1297,8 +1297,8 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, mux; - unsigned short mask, bitmask; + unsigned int val, mux; + unsigned int mask, bitmask; int ret = 0; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) @@ -1356,7 +1356,7 @@ int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short reg_val, val, mux; + unsigned int reg_val, val, mux; reg_val = snd_soc_read(widget->codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; @@ -1396,8 +1396,8 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, mux; - unsigned short mask; + unsigned int val, mux; + unsigned int mask; int ret = 0; if (ucontrol->value.enumerated.item[0] > e->max - 1) -- tg: (81ee1ba..) asoc/allow-32bits-registers (depends on: master) -- Daniel Ribeiro

2 2

[alsa-devel] ASoC update for 2.6.31
by Mark Brown 08 Jun '09

08 Jun '09

The following changes since commit e3509ff0fb9df53e45cd68488e3b463a80455db7: Daniel Mack (1): ASoC: fix NULL pointer dereference in soc_suspend() are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Mark Brown (1): ASoC: Apostrophe patrol Troy Kisky (1): ASoC: codec tlv320aic23 fix bogus divide by 0 message sound/soc/codecs/tlv320aic23.c | 10 +++++----- sound/soc/soc-dapm.c | 10 +++++----- 2 files changed, 10 insertions(+), 10 deletions(-)

2 1

[alsa-devel] Audio driver for dbAu1250 Development board
by Manjunath A M 08 Jun '09

08 Jun '09

Hi I am trying to make audio up in dbAu1250 Development board, I downloaded linux-2.6.29.4 kernel from the site "http://www.linux-mips.org/pub/linux/mips/kernel/v2.6/ " and applied the patch "asoc-v2.6.29-rc8-2009-03-17.patch"downloaded from "http://opensource.wolfsonmicro.com/~git/asoc/" but while compiling i am getting the bellow errors ********Linux compilation errors************* sound/soc/au1x/exmmb-ac97.c:24:38: error: asm/mach-exm1200/exm1200.h: No such file or directory sound/soc/au1x/exmmb-ac97.c:26:25: error: linux/exm32.h: No such file or directory sound/soc/au1x/exmmb-ac97.c:58: error: subscripted value is neither array nor pointer sound/soc/au1x/exmmb-ac97.c:65: error: variable âsnd_soc_machine_exm97â has initializer but incomplete type sound/soc/au1x/exmmb-ac97.c:66: error: unknown field ânameâ specified in initializer sound/soc/au1x/exmmb-ac97.c:66: warning: excess elements in struct initializer sound/soc/au1x/exmmb-ac97.c:66: warning: (near initialization for âsnd_soc_machine_exm97â) sound/soc/au1x/exmmb-ac97.c:67: error: unknown field âdai_linkâ specified in initializer sound/soc/au1x/exmmb-ac97.c:67: warning: excess elements in struct initializer sound/soc/au1x/exmmb-ac97.c:67: warning: (near initialization for âsnd_soc_machine_exm97â) sound/soc/au1x/exmmb-ac97.c:68: error: unknown field ânum_linksâ specified in initializer sound/soc/au1x/exmmb-ac97.c:68: warning: excess elements in struct initializer sound/soc/au1x/exmmb-ac97.c:68: warning: (near initialization for âsnd_soc_machine_exm97â) sound/soc/au1x/exmmb-ac97.c:73: error: unknown field âmachineâ specified in initializer sound/soc/au1x/exmmb-ac97.c:73: warning: initialization from incompatible pointer type sound/soc/au1x/exmmb-ac97.c:74: error: unknown field âplatformâ specified in initializer sound/soc/au1x/exmmb-ac97.c:74: warning: initialization from incompatible pointer type sound/soc/au1x/exmmb-ac97.c:80: error: âPSC1_BASEâ undeclared here (not in a function) sound/soc/au1x/exmmb-ac97.c: In function âpsc_to_ac97â: sound/soc/au1x/exmmb-ac97.c:107: error: implicit declaration of function âau_readlâ sound/soc/au1x/exmmb-ac97.c:110: error: implicit declaration of function âau_writelâ sound/soc/au1x/exmmb-ac97.c:111: error: implicit declaration of function âau_syncâ sound/soc/au1x/exmmb-ac97.c:115: error: âPSC_CTLâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:115: error: (Each undeclared identifier is reported only once sound/soc/au1x/exmmb-ac97.c:115: error: for each function it appears in.) sound/soc/au1x/exmmb-ac97.c:117: error: âPSC_SELâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:122: error: âPSC_AC97RSTâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:124: error: implicit declaration of function âmsleepâ sound/soc/au1x/exmmb-ac97.c:133: error: âPSC_AC97STATâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:140: error: âPSC_AC97CFGâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c: In function âexm1200_mobo_ac97_initâ: sound/soc/au1x/exmmb-ac97.c:160: error: implicit declaration of function âau_readwâ sound/soc/au1x/exmmb-ac97.c:160: error: âEXM1200_FPGA_BRDCTRLâ undeclared (first use in this function) sound/soc/au1x/exmmb-ac97.c:162: error: implicit declaration of function âau_writewâ make[3]: *** [sound/soc/au1x/exmmb-ac97.o] Error 1 make[2]: *** [sound/soc/au1x] Error 2 **************************************************** before applying the patch the board was working but the audio was not working.. can you suggest me which kernel source you used and how you got the audio working for your board. can you send me the sample kernel image "vmlinux.srec" image for testing.. Thanks Manjunath.AM The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it.

3 6

Re: [alsa-devel] ALSA patch Nvidia 8 channel hdmi audio
by Takashi Iwai 08 Jun '09

08 Jun '09

At Thu, 4 Jun 2009 08:18:23 +0200, Alan Alan wrote: > > 2009/6/4 Takashi Iwai <tiwai(a)suse.de> > > At Thu, 4 Jun 2009 01:28:24 +0200, > Alan Alan wrote: > > > > Hello Takashi ! > > > > I tried to apply the patch that removes restrictions to 16bits 48khz > format. > > Well now we have the following: > > 16bit 48khz works > > 32bit 48khz works > > 32bit 44khz, 96khz, 192khz don't work > > 24bit 44khz, 48khz, 96khz, 192khz don't work > > 16bit 44khz, 96khz, 192khz don't work > > > > I only get > > Sample format not available for playback: > > Setting of hwparams failed: > > As you can find in alsa-info output, it seems that the codec supports > only 48kHz and 88.2kHz, strangely. I don't know whether it's a > (hardware) bug or not. > > The codec surely supports up to 192khz as it works on Windows. > > I think this line is completely wrong as here it says: It is not "wrong". These are values that the codec reports, so it's more or less correct as the h/w information. > rates [0xc0]: 48000 88200 Wei, is it right? Or is anything missing here? > bits [0xf]: 8 16 20 24 > > 8, 20, 24bits are supported, but they all give an error message. 24bit here means 24bit in 4 bytes, thus it's essentially 32bit format. 24bit in 3 bytes format isn't supported on HD-audio. > 32 bits are not supposed to be supported but it works with 48 and 88.2khz. > > How can we change the software side to accept 96 or 192khz spample formats and > 24 bits ? As long as the codec reports so, it's no "software" problem. > I think it is not hardware limited and we just need to change the allowed > formats to be played. If it's just a hardware problem that reports wrong values, you can override the information, simply replacing the bits my patch disabled to the right values. Takashi > > > > > I enclosed to pastebin my alsainfo script output. > > > > http://pastebin.com/m77dcfb3f > > > > Thanks for the help ! > > > > 2009/6/3 Takashi Iwai <tiwai(a)suse.de> > > > > At Wed, 3 Jun 2009 11:34:00 +0200, > > Alan Alan wrote: > > > > > > Hello Takashi ! > > > > > > What should this patch do ? > > > > The patch removes the restriction patch_nvhdmi.c sets. Instead, > > the driver tries to detect the supported format, rate, etc from the > > codec widget information. > > > > > Is it a cure for the 96, 192khz or 24bit sample playback problem ? > > > > Possibly. > > > > > If so, will it resample or natievly pass the stream to the > amplifier ? > > > > It just enables the native support, possibly. > > > > And, don't forget to give back the alsa-info.sh output (run with > > --no-upload option). > > > > Takashi > > > > > > > > Thank you > > > > > > Alan > > > > > > 2009/6/3 Takashi Iwai <tiwai(a)suse.de> > > > > > > At Wed, 3 Jun 2009 10:00:24 +0200, > > > Alan Alan wrote: > > > > > > > > Hello Wei ! > > > > > > > > Thanks for replying. > > > > I tested it with speaker-test and xbmc babylon. Both had > problems > > with > > > > channel mapping. > > > > I used this command: > > > > > > > > speaker-test -Dhdmi -c6 > > > > or > > > > speaker-test -Dhdmi -c8 > > > > > > > > alternatively you could try > > > > > > > > speaker-test -Dhw:0,3 -c6 > > > > > > > > > > > > I had to rund Xorg first and have the command from an xterm, > > becasue > > > from a > > > > normal terminal it would not work. > > > > I think that is normal as a limitation of hdmi. > > > > > > > > What about the support for 24bits 192khz sound ? > > > > > > > > speaker-test -Dhdmi -c6 -r192000 does not work. > > > > > > How about the patch below? > > > > > > Also, could you give alsa-info.sh output? > > > > > > Takashi > > > > > > --- > > > diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/ > > patch_nvhdmi.c > > > index f5792e2..48443de 100644 > > > --- a/sound/pci/hda/patch_nvhdmi.c > > > +++ b/sound/pci/hda/patch_nvhdmi.c > > > @@ -258,9 +258,11 @@ static struct hda_pcm_stream > > > nvhdmi_pcm_digital_playback_8ch = { > > > .channels_min = 2, > > > .channels_max = 8, > > > .nid = Nv_Master_Convert_nid, > > > +#if 0 > > > .rates = SNDRV_PCM_RATE_48000, > > > .maxbps = 16, > > > .formats = SNDRV_PCM_FMTBIT_S16_LE, > > > +#endif > > > .ops = { > > > .open = nvhdmi_dig_playback_pcm_open, > > > .close = nvhdmi_dig_playback_pcm_close_8ch, > > > @@ -273,9 +275,11 @@ static struct hda_pcm_stream > > > nvhdmi_pcm_digital_playback_2ch = { > > > .channels_min = 2, > > > .channels_max = 2, > > > .nid = Nv_Master_Convert_nid, > > > +#if 0 > > > .rates = SNDRV_PCM_RATE_48000, > > > .maxbps = 16, > > > .formats = SNDRV_PCM_FMTBIT_S16_LE, > > > +#endif > > > .ops = { > > > .open = nvhdmi_dig_playback_pcm_open, > > > .close = nvhdmi_dig_playback_pcm_close_2ch, > > > > > > > > > > > >

3 7

[alsa-devel] [PATCH v4 0/4] ARM: DaVinci: ASoC: Audio support for DM646x
by Chaithrika U S 07 Jun '09

07 Jun '09

From: Chaithrika U S <chaithrika(a)ti.com> Add audio support for DM646x Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch set adds audio support for DM646x This updated version has included most of the review comments received for the previous version of the patch set. Patch 1: Registers the platform devices for ASP on dm355, dm644x and dm646x This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 2: Introduces platform driver model for dm644x and dm355 Applies against ALSA mainline - to the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 3: Adds driver support for the two instances of McASP on TI's DM646x Applies against ALSA mainline - to the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 4: Add machine driver support for DM646x Applies against ALSA mainline - to the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git These patches are based on the work done by Steve Chen <schen(a)mvista.com> and Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> on Monstavista 2.6.18 kernel.

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Gernot Kohlhaas 07 Jun '09

07 Jun '09

Hi! I found a little strange thing: Recording from microphone is only possible, when you unmute the "Line-In" playback-device (and set microphone as capture device, of course). Oh, and by the way: Is there any possibility to apply a low pass filter to the subwoofer-channel without using ladspa (xmms and audacious won't play songs with my low pass filter. Mplayer, xine and aplay are working fine). Thanks again for this driver :) -- http://www.gnupp.de/verschluesselung/warum.html Key ID: 0x4C0B2250 Key: http://home.arcor.de/mr_maniac/ANDERES/public.gpg

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[alsa-devel] [RFC/PATCH] asoc: pxa-ssp.c -> enchance the I2S enchancement :)
by Daniel Ribeiro 06 Jun '09

06 Jun '09

Hi, this patch is what i need to use Paul's pxa-ssp.c I2S enchancements on my board. Basically, it doesn't enforce the "fake" network mode, as i dont need it. And i need _real_ network mode, so i prefer to set SSCR0_MOD only when really using network mode. And it uses PXA_SSP_FRM_??FS to actually set SSCR0_DataSize() too. This allows me to use 32bit frames for 16 bit stereo audio. (No, I don't want to fake network mode for this, and SSCR0_MOD doesn't work as expected when PXA is slave). This patch is incremental to the recent patch from Paul Shen. Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index ed0906e..762cb93 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -579,7 +579,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_LEFT_J: - sscr0 |= SSCR0_PSP | SSCR0_MOD; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; /* See hw_params() */ @@ -655,7 +655,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, int dma = 0, chn = params_channels(params); u32 sscr0; u32 sspsp; - int frame_width; + int frame_width = 0; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; @@ -698,7 +698,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); break; } - ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -742,6 +741,15 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, break; } + if (frame_width) { + sscr0 &= ~(SSCR0_EDSS | SSCR0_DataSize(16)); + if (frame_width % 48 == 0) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(8); + else + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(16); + } + ssp_write_reg(ssp, SSCR0, sscr0); + /* When we use a network mode, we always require TDM slots * - complain loudly and fail if they've not been set up yet. */ -- Daniel Ribeiro

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[alsa-devel] Testers wanted: New SB X-Fi driver
by MonsieurWeller 06 Jun '09

06 Jun '09

Hello. Works fine, but the external rack of the "x-fi fatality" didn't seems to work. I plug my headphone on it and i have no sound. Thanks.

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