May 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH][ASoC V2] Additional fixes for the Freescale MPC8610 drivers
by Timur Tabi 23 May '08

23 May '08

More miscellaneous fixes for the Freescale MPC8610 ASoC V2 drivers. CS4270 driver: Replace printk() with dev_xxx(). Use snd_soc_add_new_controls(). Report an error if a CS4270 is not found at the given I2C address. Additional initialization of the 'codec' structure. DMA driver: when an OSS application requests a sync, ASoC calls fsl_dma_hw_params() after calling fsl_dma_prepare(). This caused the DMA controller to be reprogrammed, which made it stop working. So the bulk of DMA programming was moved to fsl_dma_open(). Initialized some more fields in dma_private, so now the ISR won't panic. SSI driver: The SSI is now enabled only when it starts playback or capture, to prevent premature communication with the DMA controller. Added spinlocks to prevent the SSI from being enabled inadvertently by another stream. Fabric driver: Turned the driver into a platform driver. Added calls to snd_soc_card_config_codec() and snd_soc_card_init_codec(). Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 54 +++++++++-------- sound/soc/fsl/fsl_dma.c | 139 ++++++++++++++++++++++------------------- sound/soc/fsl/fsl_ssi.c | 79 +++++++++++++++++------- sound/soc/fsl/fsl_ssi.h | 1 + sound/soc/fsl/mpc8610_hpcd.c | 130 ++++++++++++++++++++++++--------------- 5 files changed, 240 insertions(+), 163 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 13b9d56..6bb4bc7 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -223,7 +223,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, rates &= ~SNDRV_PCM_RATE_KNOT; if (!rates) { - printk(KERN_ERR "cs4270: could not find a valid sample rate\n"); + dev_err(codec_dai->dev, "could not find a valid sample rate\n"); return -EINVAL; } @@ -263,7 +263,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, cs4270->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; break; default: - printk(KERN_ERR "cs4270: invalid DAI format\n"); + dev_err(codec_dai->dev, "invalid DAI format\n"); ret = -EINVAL; } @@ -315,7 +315,7 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, struct i2c_client *client = codec->control_data; if (i2c_smbus_write_byte_data(client, reg, value)) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "I2C write failed\n"); return -EIO; } @@ -359,7 +359,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, if (i == NUM_MCLK_RATIOS) { /* We did not find a matching ratio */ - printk(KERN_ERR "cs4270: could not find matching ratio\n"); + dev_err(codec->dev, "could not find matching ratio\n"); return -EINVAL; } @@ -369,7 +369,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | CS4270_PWRCTL_PDN); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "I2C write failed\n"); return ret; } @@ -381,7 +381,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, ret = cs4270_i2c_write(codec, CS4270_MODE, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "I2C write failed\n"); return ret; } @@ -398,13 +398,13 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg |= CS4270_FORMAT_DAC_LJ | CS4270_FORMAT_ADC_LJ; break; default: - printk(KERN_ERR "cs4270: unknown format\n"); + dev_err(codec->dev, "unknown format\n"); return -EINVAL; } ret = cs4270_i2c_write(codec, CS4270_FORMAT, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "I2C write failed\n"); return ret; } @@ -415,7 +415,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg &= ~CS4270_MUTE_AUTO; ret = cs4270_i2c_write(codec, CS4270_MUTE, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "I2C write failed\n"); return ret; } @@ -423,7 +423,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, ret = cs4270_i2c_write(codec, CS4270_PWRCTL, 0); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "I2C write failed\n"); return ret; } @@ -469,23 +469,16 @@ static int cs4270_codec_init(struct snd_soc_codec *codec, struct snd_soc_card *soc_card) { int ret; - unsigned int i; /* Add the non-DAPM controls */ - for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { - struct snd_kcontrol *kctrl; - - kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); + ret = snd_soc_add_new_controls(soc_card, cs4270_snd_controls, codec, + ARRAY_SIZE(cs4270_snd_controls)); - ret = snd_ctl_add(soc_card->card, kctrl); - if (ret < 0) { - dev_err(soc_card->card->dev, "could not add control\n"); - return ret; - } - } + if (ret < 0) + dev_err(soc_card->card->dev, "could not add control\n"); - return 0; + return ret; } /* @@ -513,9 +506,14 @@ static int cs4270_i2c_probe(struct i2c_client *client, return ret; } /* The top four bits of the chip ID should be 1100. */ - if ((ret & 0xF0) != 0xC0) - /* The device at this address is not a CS4270 codec */ + if ((ret & 0xF0) != 0xC0) { + /* This is a new-style I2C driver, which means we're supposed + * to be given the actual I2C address that the chip is on. If + * this device is not a CS4270, then that's a bug. + */ + dev_err(&client->dev, "device is not a CS4270\n"); return -ENODEV; + } dev_info(&client->dev, "found device at address %X\n", client->addr); dev_info(&client->dev, "hardware revision %X\n", ret & 0xF); @@ -567,10 +565,13 @@ static int cs4270_i2c_probe(struct i2c_client *client, codec->control_data = client; codec->init = cs4270_codec_init; + codec->reg_cache = cs4270->reg_cache; + codec->codec_read = cs4270_read_reg_cache; + codec->codec_write = cs4270_i2c_write; ret = snd_soc_register_codec(codec, &client->dev); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to register card\n"); + dev_err(&client->dev, "failed to register card\n"); goto error; } registered = 1; @@ -583,6 +584,7 @@ static int cs4270_i2c_probe(struct i2c_client *client, cs4270->dai = snd_soc_register_codec_dai(&cs4270->dai_new, &client->dev); if (!cs4270->dai) { + dev_err(&client->dev, "failed to register DAI\n"); ret = -EINVAL; goto error; } @@ -650,7 +652,7 @@ static int __init cs4270_init(void) ret = i2c_add_driver(&cs4270_i2c_driver); if (ret) - printk(KERN_ERR "cs4270: failed to register I2C driver\n"); + pr_err("cs4270: failed to register I2C driver\n"); return ret; } diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 4b48f8a..5a266fc 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -232,10 +232,8 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) ret = IRQ_HANDLED; if (sr & CCSR_DMA_SR_EOSI) { - struct snd_pcm_substream *substream = dma_private->substream; - /* Tell ALSA we completed a period. */ - snd_pcm_period_elapsed(substream); + snd_pcm_period_elapsed(dma_private->substream); /* * Update our link descriptors to point to the next period. We @@ -332,9 +330,13 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) struct fsl_dma_info *dma_info = cpu_dai->dma_data; struct fsl_dma_private *dma_private; + struct ccsr_dma_channel __iomem *dma_channel; dma_addr_t ld_buf_phys; + u64 temp_link; /* Pointer to next link descriptor */ + u32 mr; unsigned int channel; int ret = 0; + unsigned int i; /* * Reject any DMA buffer whose size is not a multiple of the period @@ -363,6 +365,10 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) else dma_private->dma_info = &dma_info[1]; + dma_private->substream = substream; + dma_private->ld_buf_phys = ld_buf_phys; + dma_private->dma_buf_phys = substream->dma_buffer.addr; + ret = request_irq(dma_private->dma_info->irq, fsl_dma_isr, 0, "DMA", dma_private); if (ret) { @@ -379,6 +385,66 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); runtime->private_data = dma_private; + /* Program the fixed DMA controller parameters */ + + dma_channel = dma_private->dma_info->channel; + + temp_link = dma_private->ld_buf_phys + + sizeof(struct fsl_dma_link_descriptor); + + for (i = 0; i < NUM_DMA_LINKS; i++) { + struct fsl_dma_link_descriptor *link = &dma_private->link[i]; + + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->next = cpu_to_be64(temp_link); + + temp_link += sizeof(struct fsl_dma_link_descriptor); + } + /* The last link descriptor points to the first */ + dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); + + /* Tell the DMA controller where the first link descriptor is */ + out_be32(&dma_channel->clndar, + CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys)); + out_be32(&dma_channel->eclndar, + CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys)); + + /* The manual says the BCR must be clear before enabling EMP */ + out_be32(&dma_channel->bcr, 0); + + /* + * Program the mode register for interrupts, external master control, + * and source/destination hold. Also clear the Channel Abort bit. + */ + mr = in_be32(&dma_channel->mr) & + ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); + + /* + * We want External Master Start and External Master Pause enabled, + * because the SSI is controlling the DMA controller. We want the DMA + * controller to be set up in advance, and then we signal only the SSI + * to start transfering. + * + * We want End-Of-Segment Interrupts enabled, because this will generate + * an interrupt at the end of each segment (each link descriptor + * represents one segment). Each DMA segment is the same thing as an + * ALSA period, so this is how we get an interrupt at the end of every + * period. + * + * We want Error Interrupt enabled, so that we can get an error if + * the DMA controller is mis-programmed somehow. + */ + mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | + CCSR_DMA_MR_EMS_EN; + + /* For playback, we want the destination address to be held. For + capture, set the source address to be held. */ + mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE; + + out_be32(&dma_channel->mr, mr); + return 0; } @@ -452,12 +518,8 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; - struct ccsr_dma_channel __iomem *dma_channel = - dma_private->dma_info->channel; dma_addr_t temp_addr; /* Pointer to next period */ - u64 temp_link; /* Pointer to next link descriptor */ - uint32_t mr; /* Temporary variable for MR register */ unsigned int i; @@ -475,8 +537,6 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, dma_private->dma_buf_next = dma_private->dma_buf_phys; /* - * Initialize each link descriptor. - * * The actual address in STX0 (destination for playback, source for * capture) is based on the sample size, but we don't know the sample * size in this function, so we'll have to adjust that later. See @@ -492,16 +552,11 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, * buffer itself. */ temp_addr = substream->dma_buffer.addr; - temp_link = dma_private->ld_buf_phys + - sizeof(struct fsl_dma_link_descriptor); for (i = 0; i < NUM_DMA_LINKS; i++) { struct fsl_dma_link_descriptor *link = &dma_private->link[i]; link->count = cpu_to_be32(period_size); - link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->next = cpu_to_be64(temp_link); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) link->source_addr = cpu_to_be32(temp_addr); @@ -509,51 +564,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, link->dest_addr = cpu_to_be32(temp_addr); temp_addr += period_size; - temp_link += sizeof(struct fsl_dma_link_descriptor); } - /* The last link descriptor points to the first */ - dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); - - /* Tell the DMA controller where the first link descriptor is */ - out_be32(&dma_channel->clndar, - CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys)); - out_be32(&dma_channel->eclndar, - CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys)); - - /* The manual says the BCR must be clear before enabling EMP */ - out_be32(&dma_channel->bcr, 0); - - /* - * Program the mode register for interrupts, external master control, - * and source/destination hold. Also clear the Channel Abort bit. - */ - mr = in_be32(&dma_channel->mr) & - ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); - - /* - * We want External Master Start and External Master Pause enabled, - * because the SSI is controlling the DMA controller. We want the DMA - * controller to be set up in advance, and then we signal only the SSI - * to start transfering. - * - * We want End-Of-Segment Interrupts enabled, because this will generate - * an interrupt at the end of each segment (each link descriptor - * represents one segment). Each DMA segment is the same thing as an - * ALSA period, so this is how we get an interrupt at the end of every - * period. - * - * We want Error Interrupt enabled, so that we can get an error if - * the DMA controller is mis-programmed somehow. - */ - mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | - CCSR_DMA_MR_EMS_EN; - - /* For playback, we want the destination address to be held. For - capture, set the source address to be held. */ - mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE; - - out_be32(&dma_channel->mr, mr); return 0; } @@ -793,8 +804,6 @@ static struct snd_soc_platform_new fsl_dma_platform = { .pcm_free = fsl_dma_free_dma_buffers, }; -/* - */ static int fsl_dma_probe(struct platform_device *pdev) { struct snd_soc_platform *platform; @@ -806,15 +815,16 @@ static int fsl_dma_probe(struct platform_device *pdev) return -ENOMEM; } - platform_set_drvdata(pdev, platform); - ret = snd_soc_register_platform(platform, &pdev->dev); if (ret < 0) { dev_err(&pdev->dev, "could not register platform\n"); snd_soc_free_platform(platform); + return ret; } - return ret; + platform_set_drvdata(pdev, platform); + + return 0; } static int fsl_dma_remove(struct platform_device *pdev) @@ -852,7 +862,8 @@ static __init int fsl_dma_init(void) pdev = platform_device_register_simple(fsl_platform_id, 0, NULL, 0); if (!pdev) { - pr_err("fsl-dma: could not register platform\n"); + pr_err("fsl-dma: could not register device\n"); + platform_driver_unregister(&fsl_dma_driver); return ret; } diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9263fd5..0d531e9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -245,11 +245,13 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, /* * Program the SSI into I2S Slave Non-Network Synchronous mode. - * Also enable the transmit and receive FIFO. + * Also enable the transmit and receive FIFO. Make sure TE and + * RE are disabled. * * FIXME: Little-endian samples require a different shift dir */ - clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK, + clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK | + CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE, CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE | CCSR_SSI_SCR_SYN); @@ -284,6 +286,8 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, out_be32(&ssi->sfcsr, CCSR_SSI_SFCSR_TFWM0(6) | CCSR_SSI_SFCSR_RFWM0(2)); + spin_unlock(&ssi_info->lock); + /* * We keep the SSI disabled because if we enable it, then the * DMA controller will start. It's not supposed to start until @@ -296,10 +300,9 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ssi_info->playback++; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ssi_info->capture++; + ssi_info->playback = 1; + else + ssi_info->capture = 1; cpu_dai->dma_data = ssi_info->dma_info; @@ -326,20 +329,33 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream, struct fsl_ssi_info *ssi_info = cpu_dai->private_data; struct ccsr_ssi __iomem *ssi = ssi_info->ssi; uint32_t wl; + u32 scr; + u32 sxccr; wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); - /* FIXME: We should read and write the registers manually so as to - minimize the amount of time the SSI is disabled. */ + spin_lock(&ssi_info->lock); - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + scr = in_be32(&ssi->scr); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); - else - clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); + /* If the SSI is currently enabled, then we want to keep it disabled for + * as short as time as possible, so we pre-calculate all the values for + * the appropriate registers, and then program them rapidly. + */ - setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sxccr = (in_be32(&ssi->stccr) & ~CCSR_SSI_SxCCR_WL_MASK) | wl; + out_be32(&ssi->scr, scr & ~CCSR_SSI_SCR_SSIEN); + out_be32(&ssi->stccr, sxccr); + out_be32(&ssi->scr, scr); + } else { + sxccr = (in_be32(&ssi->srccr) & ~CCSR_SSI_SxCCR_WL_MASK) | wl; + out_be32(&ssi->scr, scr & ~CCSR_SSI_SCR_SSIEN); + out_be32(&ssi->srccr, sxccr); + out_be32(&ssi->scr, scr); + } + + spin_unlock(&ssi_info->lock); return 0; } @@ -364,10 +380,14 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - setbits32(&ssi->scr, CCSR_SSI_SCR_TE); - } else { - setbits32(&ssi->scr, CCSR_SSI_SCR_RE); + spin_lock(&ssi_info->lock); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); + else { + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); /* * I think we need this delay to allow time for the SSI @@ -376,6 +396,8 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, */ mdelay(1); } + + spin_unlock(&ssi_info->lock); break; case SNDRV_PCM_TRIGGER_STOP: @@ -406,10 +428,9 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, struct fsl_ssi_info *ssi_info = cpu_dai->private_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ssi_info->playback--; - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ssi_info->capture--; + ssi_info->playback = 0; + else + ssi_info->capture = 0; /* * If this is the last active substream, disable the SSI and release @@ -427,7 +448,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, /** * fsl_ssi_set_sysclk: set the clock frequency and direction * - * This function is called by the soc_card driver to tell us what the clock + * This function is called by the fabric driver to tell us what the clock * frequency and direction are. * * Currently, we only support operating as a clock slave (SND_SOC_CLOCK_IN), @@ -523,6 +544,16 @@ static struct snd_soc_dai_ops ops = { .set_fmt = fsl_ssi_set_fmt, }; +/** + * find_dma_node: find the DMA node to use in a legacy device tree + * + * This function returns a device node matching a given DMA channel on a + * given DMA controller. + * + * The original device tree for the MPC8610 HPCD did not include phandles + * from the SSI nodes to the DMA nodes. To support these device trees, we + * declare that SSIx will use DMA channels 0 and 1 of DMA controller DMAx. + */ static struct device_node *find_dma_node(unsigned int controller, unsigned int channel) { @@ -698,7 +729,7 @@ static int fsl_ssi_probe(struct of_device *ofdev, ssi_info->dai_format = SND_SOC_DAIFMT_LEFT_J; ssi_info->codec_clk_direction = SND_SOC_CLOCK_IN; ssi_info->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "rj-master") == 0) { + } else if (strcasecmp(sprop, "rj-slave") == 0) { ssi_info->dai_format = SND_SOC_DAIFMT_RIGHT_J; ssi_info->codec_clk_direction = SND_SOC_CLOCK_OUT; ssi_info->cpu_clk_direction = SND_SOC_CLOCK_IN; diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 05fd84d..9a22b04 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -219,6 +219,7 @@ struct fsl_ssi_info { struct device *dev; unsigned int playback; unsigned int capture; + spinlock_t lock; struct snd_soc_dai *dai; struct device_attribute dev_attr; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index a857e57..10c7fd7 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -52,7 +52,24 @@ static int mpc8610_hpcd_audio_init(struct snd_soc_card *soc_card) struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; struct snd_soc_dai *cpu_dai = snd_soc_card_get_dai(soc_card, "fsl,mpc8610-ssi"); + struct snd_soc_codec *codec = + snd_soc_card_get_codec(soc_card, "cirrus,cs4270"); struct fsl_ssi_info *ssi_info = cpu_dai->private_data; + int ret; + + /* This is stupid. snd_soc_card_config_codec() initializes the + * codec->control_data structure, which is supposed to be a pointer to + * the i2c_client structure. But I already assigned that variable in + * the codec driver. After all, the codec driver is the one that + * supposed to get probed by the I2C bus. So in order to avoid + * overwriting codec->control_data, I pass it as the 4th parameter. + */ + snd_soc_card_config_codec(codec, NULL, NULL, codec->control_data); + ret = snd_soc_card_init_codec(codec, soc_card); + if (ret < 0) { + dev_err(soc_card->dev, "could not initialize codec\n"); + return ret; + } /* Program the signal routing between the SSI and the DMA */ guts_set_dmacr(soc_card_data->guts, @@ -94,17 +111,37 @@ static int mpc8610_hpcd_audio_init(struct snd_soc_card *soc_card) * This function takes board-specific information, like clock frequencies * and serial data formats, and passes that information to the codec and * transport drivers. + * + * The clock information is stored in the ssi_info data structure because + * it's gathered by the SSI driver from the device tree. This isn't really + * the best way to handle it. In fact, this problem is a core dillema with + * ASoC V2: whether the fabric driver should hand the information to the + * platform/codec drivers, or whether these drivers should get the + * information themselves. With device trees, the necessary data may be + * scattered in various device nodes, so it might be impossible to obtain + * that information in a generic manner. + * + * What makes this problem even more thorny is that some of these drivers + * are OF drivers and the rest aren't. The CS4270 can't be an OF driver, + * yet the information it needs is in an OF device node. So perhaps the + * best approach is to have generic codec OF->platform code in the fabric + * driver (normally this code is in fsl_soc.c). + * + * Even more frustrating is the fact that the CS4270 driver is an I2C + * driver, so it's probed via the I2C bus. We need to update the powerpc + * platform to initialize client->dev->archdata.of_node to point to the + * device node. Then the CS4270 driver can get its own clock rate. */ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct mpc8610_hpcd_data *soc_card_data = rtd->soc_card->private_data; + struct fsl_ssi_info *ssi_info = cpu_dai->private_data; int ret = 0; /* Tell the CPU driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(cpu_dai, soc_card_data->dai_format); + ret = snd_soc_dai_set_fmt(cpu_dai, ssi_info->dai_format); if (ret < 0) { dev_err(substream->pcm->card->dev, "could not set CPU driver audio format\n"); @@ -112,7 +149,7 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) } /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(codec_dai, soc_card_data->dai_format); + ret = snd_soc_dai_set_fmt(codec_dai, ssi_info->dai_format); if (ret < 0) { dev_err(substream->pcm->card->dev, "could not set codec driver audio format\n"); @@ -123,8 +160,8 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) * Tell the CPU driver what the clock frequency is, and whether it's a * slave or master. */ - ret = snd_soc_dai_set_sysclk(cpu_dai, 0, soc_card_data->clk_frequency, - soc_card_data->cpu_clk_direction); + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, ssi_info->clk_frequency, + ssi_info->cpu_clk_direction); if (ret < 0) { dev_err(substream->pcm->card->dev, "could not set CPU driver clock parameters\n"); @@ -135,8 +172,8 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) * Tell the codec driver what the MCLK frequency is, and whether it's * a slave or master. */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, soc_card_data->clk_frequency, - soc_card_data->codec_clk_direction); + ret = snd_soc_dai_set_sysclk(codec_dai, 0, ssi_info->clk_frequency, + ssi_info->codec_clk_direction); if (ret < 0) { dev_err(substream->pcm->card->dev, "could not set codec driver clock params\n"); @@ -207,39 +244,38 @@ static struct snd_soc_pcm_config mpc8610_pcm_config = { * respect to audio hardware connections. Therefore, we create a new ASoC * device for each new SSI node that has a codec attached. */ -static int mpc8610_hpcd_probe(struct of_device *ofdev, - const struct of_device_id *match) +static int mpc8610_hpcd_probe(struct platform_device *pdev) { struct snd_soc_card *soc_card = NULL; - struct device_node *guts_np = NULL; struct mpc8610_hpcd_data *soc_card_data; + struct device_node *guts_np = NULL; int ret = -ENODEV; soc_card_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); if (!soc_card_data) { - dev_err(&ofdev->dev, "could not allocate card structure\n"); + dev_err(&pdev->dev, "could not allocate card structure\n"); return -ENOMEM; } /* Map the global utilities registers. */ guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); if (!guts_np) { - dev_err(&ofdev->dev, "could not obtain address of GUTS\n"); + dev_err(&pdev->dev, "could not obtain address of GUTS\n"); ret = -EINVAL; goto error; } soc_card_data->guts = of_iomap(guts_np, 0); of_node_put(guts_np); if (!soc_card_data->guts) { - dev_err(&ofdev->dev, "could not map GUTS\n"); + dev_err(&pdev->dev, "could not map GUTS\n"); ret = -EINVAL; goto error; } - soc_card = snd_soc_card_create("MPC8610", &ofdev->dev, + soc_card = snd_soc_card_create("MPC8610", &pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (!soc_card) { - dev_err(&ofdev->dev, "could not create card\n"); + dev_err(&pdev->dev, "could not create card\n"); goto error; } @@ -247,19 +283,19 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, soc_card->init = mpc8610_hpcd_audio_init; soc_card->exit = mpc8610_hpcd_audio_exit; soc_card->private_data = soc_card_data; - soc_card->dev = &ofdev->dev; + soc_card->dev = &pdev->dev; ret = snd_soc_card_create_pcms(soc_card, &mpc8610_pcm_config, 1); if (ret) { - dev_err(&ofdev->dev, "could not create PCMs\n"); + dev_err(&pdev->dev, "could not create PCMs\n"); goto error; } - /* every has been added at this point */ - dev_set_drvdata(&ofdev->dev, soc_card); + platform_set_drvdata(pdev, soc_card); + ret = snd_soc_card_register(soc_card); if (ret) { - dev_err(&ofdev->dev, "could not register card\n"); + dev_err(&pdev->dev, "could not register card\n"); goto error; } @@ -277,14 +313,9 @@ error: return ret; } -/** - * mpc8610_hpcd_remove: remove the OF device - * - * This function is called when the OF device is removed. - */ -static int mpc8610_hpcd_remove(struct of_device *ofdev) +static int mpc8610_hpcd_remove(struct platform_device *pdev) { - struct snd_soc_card *soc_card = dev_get_drvdata(&ofdev->dev); + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; if (soc_card_data->guts) @@ -294,27 +325,20 @@ static int mpc8610_hpcd_remove(struct of_device *ofdev) snd_soc_card_free(soc_card); - dev_set_drvdata(&ofdev->dev, NULL); - return 0; } -static struct of_device_id mpc8610_hpcd_match[] = { - { - .compatible = "fsl,mpc8610-ssi", +static struct platform_driver mpc8610_hpcd_driver = { + .driver = { + .name = "mpc8610_hpcd", + .owner = THIS_MODULE, }, - {} -}; -MODULE_DEVICE_TABLE(of, mpc8610_hpcd_match); - -static struct of_platform_driver mpc8610_hpcd_of_driver = { - .owner = THIS_MODULE, - .name = "mpc8610_hpcd", - .match_table = mpc8610_hpcd_match, .probe = mpc8610_hpcd_probe, - .remove = mpc8610_hpcd_remove, + .remove = __devexit_p(mpc8610_hpcd_remove), }; +static struct platform_device *pdev; + /** * mpc8610_hpcd_init: fabric driver initialization. * @@ -324,15 +348,22 @@ static int __init mpc8610_hpcd_init(void) { int ret; - printk(KERN_INFO "Freescale MPC8610 HPCD ALSA SoC fabric driver\n"); + printk(KERN_INFO "Freescale MPC8610 HPCD ASoC fabric driver\n"); - ret = of_register_platform_driver(&mpc8610_hpcd_of_driver); + ret = platform_driver_register(&mpc8610_hpcd_driver); + if (ret < 0) { + pr_err("mpc8610-hpcd: could not register platform\n"); + return ret; + } - if (ret) - printk(KERN_ERR - "mpc8610-hpcd: failed to register platform driver\n"); + pdev = platform_device_register_simple("mpc8610_hpcd", 0, NULL, 0); + if (!pdev) { + pr_err("mpc8610-hpcd: could not register device\n"); + platform_driver_unregister(&mpc8610_hpcd_driver); + return ret; + } - return ret; + return 0; } /** @@ -342,12 +373,13 @@ static int __init mpc8610_hpcd_init(void) */ static void __exit mpc8610_hpcd_exit(void) { - of_unregister_platform_driver(&mpc8610_hpcd_of_driver); + platform_device_unregister(pdev); + platform_driver_unregister(&mpc8610_hpcd_driver); } module_init(mpc8610_hpcd_init); module_exit(mpc8610_hpcd_exit); MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); -MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC fabric driver"); +MODULE_DESCRIPTION("Freescale MPC8610 HPCD ASoC fabric driver"); MODULE_LICENSE("GPL"); -- 1.5.5

2 1

[alsa-devel] Problem with intel-hda integrated audio
by Matteo Sasso 23 May '08

23 May '08

Hello, I'm having a problem with a soundcard under ubuntu. I tried nearly every workaround I could find on the Internet but I have not been successful so far. The computer in question is a Toshiba Satellite Pro U200 series. The soundcard seems to be correctly identified by ALSA but it seems muted, nothing I tried seems to work. I followed the instructions given at http://www.mjmwired.net/kernel/Documentation/sound/alsa/ALSA-Configuration.… I determined that my card uses a AD1981 codec, so I added the following to /etc/modprobe.d/alsa-base: options snd-hda-intel model=toshiba I tried many other combinations (3stack, basic, laptop), with and without "position_fix=1". I also tried using underscores instead of hypens ("snd_hda_intel"). I run update-initramfs -u after every try. lspci -vn has to say the following: 00:1b.0 0403: 8086:27d8 (rev 02) Sybsystem: 1179:0001 Flags: bus master, fast devsel, latency 0, IRQ 23 Memory at 50080000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 I'll keep tabs on this mailing list, so feel free to ask for more info. Any help is appreciated. Thanks.

2 2

[alsa-devel] Question about "control names"
by Juergen Beisert 23 May '08

23 May '08

Hi, I'm playing with the control names of my "snd_kcontrol_new" controls. When I'm using "PCM Playback Switch" and "PCM Playback Volume" alsamixer shows me a slider with a switch at its bottom. That's what I expected. But when I'm using a second set ("line in" to "line out" bypass), with "Line Bypass Capture Switch" and "Line Bypass Capture Volume" control names, its confusing (BTW: "Line Bypass Capture Volume" is of type SNDRV_CTL_ELEM_TYPE_ENUMERATED). The "Line Bypass Capture Volume" control is shown on the [Playback] page with the name "Line Byp" and the default value. I can walk through the enumerated values. But the "Line Bypass Capture Switch" is shown on the [Capture] page in alsamixer as: L R CAPTUR <Line Byp> What does it mean? Problem of alsamixer or the name of this set? Juergen -- Dipl.-Ing. Juergen Beisert | http://www.pengutronix.de Pengutronix - Linux Solutions for Science and Industry Handelsregister: Amtsgericht Hildesheim, HRA 2686 Vertretung Sued/Muenchen, Germany Phone: +49-8766-939 228 | Fax: +49-5121-206917-9

1 0

[alsa-devel] softvol plug-in dependancies?
by Simon Dunn 23 May '08

23 May '08

Hi Guys The softvol plug-in, are there any pre-requisites or dependancies to using this? I've tried various entries in the /etc/asound.conf configuration file all with the same result. I can see the control in alsamixer, once I've run the speaker-test through it, but cannot adjust any settings. Trying to run 'alsactl store 0' results in this error : . alsactl: control.c:2437: snd_ctl_elem_value_get_integer: Assertion `idx < sizeof (obj->value.integer.value) / sizeof(obj->value.integer.value[0])' failed. Aborted . Curiously using amixer the device is listed but its properties are displayed like this . Simple mixer control 'PCM Vol',0 Capabilities: volume Playback channels: Front Left - Front Right - Rear Left - Rear Right - Front C enter - Woofer - Side Left - Side Right - Rear Center - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? Capture channels: Front Left - Front Right - Rear Left - Rear Right - Front Ce nter - Woofer - Side Left - Side Right - Rear Center - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? - ? Limits: 0 - 0 Front Left: 0 [0%] Front Right: 0 [0%] Rear Left: 0 [0%] Rear Right: 0 [0%] Front Center: 0 [0%] Woofer: 0 [0%] Side Left: 0 [0%] Side Right: 0 [0%] Rear Center: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] ?: 0 [0%] Any clues as to what's going on here would be much appreciated before I have to start dissecting the library. TIA for any hints Simon -- Simon Dunn, Software Engineer MPC Data Limited e-mail: sdunn(a)mpc-data.co.uk web: www.mpc-data.co.uk tel: +44 (0) 1225 710600 fax: +44 (0) 1225 710601 ddi: +44 (0) 1225 710631 MPC Data Limited is a company registered in England and Wales with company number 05507446 Registered Address: County Gate, County Way, Trowbridge, Wiltshire, BA14 7FJ VAT no: 850625238 The information in this email and in the attached documents is confidential and may be legally privileged. Any unauthorized review, copying, disclosure or distribution is prohibited and may be unlawful. It is intended solely for the addressee. Access to this email by anyone else is unauthorized. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. When addressed to our clients any opinions or advice contained in this email is subject to the terms and conditions expressed in the governing contract.

2 3

[alsa-devel] HG -> GIT migration
by Jaroslav Kysela 23 May '08

23 May '08

Hi all, I finished migration from HG to GIT. All repositories are converted and we have one "new" repository - alsa-kernel.git which is full Linux 2.6 GIT tree. The old HG alsa-kernel repository was renamed to alsa-kmirror - just to track our changes in continuous timeline for releases and packaging. HG access to all repositories is closed now. More information can be obtained here: http://www.alsa-project.org/main/index.php/GIT_Server Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

10 35

Re: [alsa-devel] [Alsa-user] Pc Speaker
by Rene Herman 23 May '08

23 May '08

On 13-05-08 12:44, Armin ranjbar wrote: >> Device Drivers -> >> Input device support -> >> Miscelaneous devices >> >> With current mainline, it's: >> >> General setup -> >> Configure standard kernel features (for small systems) >> Enable PC-Speaker support > > Thank you very much for your reply , > > but i have to say that i have this options activated , still no pc > speaker . Rechecked. Yes, you need this option, you need the ALSA one disabled (not modular, disabled) and then the PC speaker driver option appears in its input menu where you want to enable it. Paths as given in last reply. Yes, that would seem to be amazingly clumsily done. Perhaps there was a reason (adding alsa-devel). Rene.

3 6

[alsa-devel] [PATCH] Patch for alsa-info.sh to check if pastebin.ca is down, and handle with grace.
by Travis Place 22 May '08

22 May '08

Changelog: This patch will allow alsa-info.sh to check if the pastebin site (www.pastebin.ca) is down, before trying to upload to it. It will fail gracefully, with a message letting the user know what happened. The script will also keep the ALSA information it collected (same way it does with --no-upload). Patch is attached. Signed-off-by: Travis Place <wishie at wishie dot net> -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

1 0

[alsa-devel] Moving sound/* to drivers/ ?
by Takashi Iwai 22 May '08

22 May '08

While we are at the issue ALSA HG <-> GIT transition... One thing that annoys me sometimes is that the sound driver codes are on a different subdirectory than other normal drivers, namely under /sound. I don't remember exactly why it came there, but this often makes people forget to change the necessary code for the sound subsystem, or slip from the statistics (I'm not sure whether Linus counted sound/* with recent his posts). If I understand correctly, with git, we can move the files in relatively little costs. So, what about moving sound/* back to drivers/sound/* or drivers/media/sound/*? Of course, the primary question is whether it's really worth. The obvious drawback is that patches won't be applicable after the move. So, if we do it, doing at the last seems practical. But, this can be a bit problem with keeping such a change on linux-next until the next merge... And, another question is to where. drivers/sound or drivers/media/sound, or whatever. Any comments appreciated. thanks, Takashi

9 20

Re: [alsa-devel] [pulseaudio-discuss] Asynchronous notification in ALSA - I/O Plugin - Pulseaudio
by Lennart Poettering 22 May '08

22 May '08

On Thu, 22.05.08 14:11, Colin Guthrie (gmane(a)colin.guthr.ie) wrote: > > That freeze is an unfixable race condition that Flash has when > > shutting down audio. > > > > See the explanation down on http://pulseaudio.org/ticket/267 > > I thought the whole point was that by adding async code to ioplug, the > libflashsupport stuff wasn't needed and flash could just use it's native > alsa support + alsa's pulse plugin.... or am I missing the point here? The destruction dead lock issue is present both in libflashsupport and the native ALSA backend. It's just that it is almost never triggered if a hw device is used. Adobe released the sources of libflashsupport where you can actually check this. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH] Add async support to ioplug
by Takashi Iwai 22 May '08

22 May '08

Hi, the patch below adds a framework to support async on ioplug. Now an external ioplug plugin can implement appropriate async methods if possible. The async callback is just a copy of snd_pcm_async(). When sig >= 0, activate the async, and with a negative value, deactivate async. The test patch to alsa-plugins will follow. Takashi --- diff -r 8b22022a861d include/local.h --- a/include/local.h Fri May 09 16:02:02 2008 +0200 +++ b/include/local.h Tue May 20 09:02:35 2008 +0200 @@ -149,7 +149,13 @@ void *private_data; struct list_head glist; struct list_head hlist; + int no_signal; }; + +int _snd_async_add_handler(snd_async_handler_t **handler, int fd, + snd_async_callback_t callback, + void *private_data, int no_signal); +void _snd_async_call_handlers(int fd); typedef enum _snd_set_mode { SND_CHANGE, diff -r 8b22022a861d include/pcm_ioplug.h --- a/include/pcm_ioplug.h Fri May 09 16:02:02 2008 +0200 +++ b/include/pcm_ioplug.h Tue May 20 09:02:35 2008 +0200 @@ -66,7 +66,7 @@ */ #define SND_PCM_IOPLUG_VERSION_MAJOR 1 /**< Protocol major version */ #define SND_PCM_IOPLUG_VERSION_MINOR 0 /**< Protocol minor version */ -#define SND_PCM_IOPLUG_VERSION_TINY 1 /**< Protocol tiny version */ +#define SND_PCM_IOPLUG_VERSION_TINY 2 /**< Protocol tiny version */ /** * IO-plugin protocol version */ @@ -114,6 +114,10 @@ unsigned int rate; /**< rate; filled after hw_params is called */ snd_pcm_uframes_t period_size; /**< period size; filled after hw_params is called */ snd_pcm_uframes_t buffer_size; /**< buffer size; filled after hw_params is called */ + /** + * fields added by version 1.0.2 + */ + unsigned int no_signal_async; /**< async support without signal */ }; /** Callback table of ioplug */ @@ -189,6 +193,10 @@ * get the delay for the running PCM; optional */ int (*delay)(snd_pcm_ioplug_t *io, snd_pcm_sframes_t *delayp); + /** + * set up async handler; optional; added in version 1.0.2 + */ + int (*async)(snd_pcm_ioplug_t *io, int sig, pid_t pid); }; @@ -212,6 +220,9 @@ /* change PCM status */ int snd_pcm_ioplug_set_state(snd_pcm_ioplug_t *ioplug, snd_pcm_state_t state); +/* call async handlers */ +void snd_pcm_ioplug_call_async_handlers(snd_pcm_ioplug_t *ioplug); + /** \} */ #endif /* __ALSA_PCM_IOPLUG_H */ diff -r 8b22022a861d src/async.c --- a/src/async.c Fri May 09 16:02:02 2008 +0200 +++ b/src/async.c Tue May 20 09:02:35 2008 +0200 @@ -48,19 +48,48 @@ #endif static LIST_HEAD(snd_async_handlers); +static int sig_count; -static void snd_async_handler(int signo ATTRIBUTE_UNUSED, siginfo_t *siginfo, void *context ATTRIBUTE_UNUSED) +#ifndef DOC_HIDDEN +void _snd_async_call_handlers(int fd) { - int fd; struct list_head *i; - //assert(siginfo->si_code == SI_SIGIO); - fd = siginfo->si_fd; list_for_each(i, &snd_async_handlers) { snd_async_handler_t *h = list_entry(i, snd_async_handler_t, glist); if (h->fd == fd && h->callback) h->callback(h); } } +#endif + +static void snd_async_handler(int signo ATTRIBUTE_UNUSED, siginfo_t *siginfo, void *context ATTRIBUTE_UNUSED) +{ + int fd; + //assert(siginfo->si_code == SI_SIGIO); + fd = siginfo->si_fd; + _snd_async_call_handlers(fd); +} + +#ifndef DOC_HIDDEN +int _snd_async_add_handler(snd_async_handler_t **handler, int fd, + snd_async_callback_t callback, + void *private_data, int no_signal) +{ + snd_async_handler_t *h; + assert(handler); + h = malloc(sizeof(*h)); + if (!h) + return -ENOMEM; + h->fd = fd; + h->callback = callback; + h->private_data = private_data; + list_add_tail(&h->glist, &snd_async_handlers); + INIT_LIST_HEAD(&h->hlist); + h->no_signal = no_signal; + *handler = h; + return 0; +} +#endif /** * \brief Registers an async handler. @@ -94,20 +123,13 @@ int snd_async_add_handler(snd_async_handler_t **handler, int fd, snd_async_callback_t callback, void *private_data) { - snd_async_handler_t *h; - int was_empty; - assert(handler); - h = malloc(sizeof(*h)); - if (!h) - return -ENOMEM; - h->fd = fd; - h->callback = callback; - h->private_data = private_data; - was_empty = list_empty(&snd_async_handlers); - list_add_tail(&h->glist, &snd_async_handlers); - INIT_LIST_HEAD(&h->hlist); - *handler = h; - if (was_empty) { + int err; + + err = _snd_async_add_handler(handler, fd, callback, private_data, 0); + if (err < 0) + return err; + + if (!sig_count) { int err; struct sigaction act; memset(&act, 0, sizeof(act)); @@ -116,10 +138,15 @@ sigemptyset(&act.sa_mask); err = sigaction(snd_async_signo, &act, NULL); if (err < 0) { + err = -errno; SYSERR("sigaction"); - return -errno; + list_del(&(*handler)->glist); + free(*handler); + *handler = NULL; + return err; } } + sig_count++; return 0; } @@ -133,15 +160,18 @@ int err = 0; assert(handler); list_del(&handler->glist); - if (list_empty(&snd_async_handlers)) { - struct sigaction act; - memset(&act, 0, sizeof(act)); - act.sa_flags = 0; - act.sa_handler = SIG_DFL; - err = sigaction(snd_async_signo, &act, NULL); - if (err < 0) { - SYSERR("sigaction"); - return -errno; + if (!handler->no_signal) { + sig_count--; + if (!sig_count) { + struct sigaction act; + memset(&act, 0, sizeof(act)); + act.sa_flags = 0; + act.sa_handler = SIG_DFL; + err = sigaction(snd_async_signo, &act, NULL); + if (err < 0) { + SYSERR("sigaction"); + /* return -errno; */ + } } } if (handler->type == SND_ASYNC_HANDLER_GENERIC) diff -r 8b22022a861d src/pcm/pcm.c --- a/src/pcm/pcm.c Fri May 09 16:02:02 2008 +0200 +++ b/src/pcm/pcm.c Tue May 20 09:02:36 2008 +0200 @@ -1984,8 +1984,10 @@ int err; int was_empty; snd_async_handler_t *h; - err = snd_async_add_handler(&h, _snd_pcm_async_descriptor(pcm), - callback, private_data); + + err = _snd_async_add_handler(&h, _snd_pcm_async_descriptor(pcm), + callback, private_data, + pcm->no_signal_async); if (err < 0) return err; h->type = SND_ASYNC_HANDLER_PCM; diff -r 8b22022a861d src/pcm/pcm_ioplug.c --- a/src/pcm/pcm_ioplug.c Fri May 09 16:02:02 2008 +0200 +++ b/src/pcm/pcm_ioplug.c Tue May 20 09:02:36 2008 +0200 @@ -698,11 +698,15 @@ return 0; } -static int snd_pcm_ioplug_async(snd_pcm_t *pcm ATTRIBUTE_UNUSED, - int sig ATTRIBUTE_UNUSED, - pid_t pid ATTRIBUTE_UNUSED) +static int snd_pcm_ioplug_async(snd_pcm_t *pcm, int sig, pid_t pid) { - return -ENOSYS; + ioplug_priv_t *io = pcm->private_data; + + if (io->data->version >= 0x010002 && + io->data->callback->async) + return io->data->callback->async(io->data, sig, pid); + else + return -ENOSYS; } static int snd_pcm_ioplug_munmap(snd_pcm_t *pcm ATTRIBUTE_UNUSED) @@ -1037,6 +1041,9 @@ ioplug->pcm->poll_events = ioplug->poll_events; ioplug->pcm->monotonic = (ioplug->flags & SND_PCM_IOPLUG_FLAG_MONOTONIC) != 0; ioplug->pcm->mmap_rw = ioplug->mmap_rw; + if (ioplug->version >= 0x010002) + ioplug->pcm->no_signal_async = ioplug->no_signal_async; + return 0; } @@ -1070,3 +1077,16 @@ ioplug->state = state; return 0; } + +/** + * \brief Call the associated async handlers + * \param ioplug the ioplug handle + * + * Call the async handlers associated with the given ioplug. + * This function is usually called from the plugin's async handling + * routine appropriately. + */ +void snd_pcm_ioplug_call_async_handlers(snd_pcm_ioplug_t *ioplug) +{ + _snd_async_call_handlers(ioplug->poll_fd); +} diff -r 8b22022a861d src/pcm/pcm_local.h --- a/src/pcm/pcm_local.h Fri May 09 16:02:02 2008 +0200 +++ b/src/pcm/pcm_local.h Tue May 20 09:02:36 2008 +0200 @@ -220,6 +220,7 @@ * use the mmaped buffer of the slave */ unsigned int donot_close: 1; /* don't close this PCM */ + unsigned int no_signal_async :1; /* async without signal */ snd_pcm_channel_info_t *mmap_channels; snd_pcm_channel_area_t *running_areas; snd_pcm_channel_area_t *stopped_areas;

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