May 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ucm: Handle repeat device and modifier enables/disables
by Arun Raghavan 29 May '12

29 May '12

Currently, enabling a device twice can cause it to be added to snd_use_case_mgr_t->active_devices twice, causing the list to become a loop and subsequent uses to result in an infinite loop. This patch makes sure we don't enable/disable a device twice, and avoid doing the same for modifiers. --- src/ucm/main.c | 62 ++++++++++++++++++++++++++++++------------------------- 1 files changed, 34 insertions(+), 28 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 76ca151..79901f8 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -565,6 +565,34 @@ static struct use_case_modifier * return NULL; } +long device_status(snd_use_case_mgr_t *uc_mgr, + const char *device_name) +{ + struct use_case_device *dev; + struct list_head *pos; + + list_for_each(pos, &uc_mgr->active_devices) { + dev = list_entry(pos, struct use_case_device, active_list); + if (strcmp(dev->name, device_name) == 0) + return 1; + } + return 0; +} + +long modifier_status(snd_use_case_mgr_t *uc_mgr, + const char *modifier_name) +{ + struct use_case_modifier *mod; + struct list_head *pos; + + list_for_each(pos, &uc_mgr->active_modifiers) { + mod = list_entry(pos, struct use_case_modifier, active_list); + if (strcmp(mod->name, modifier_name) == 0) + return 1; + } + return 0; +} + /** * \brief Set verb * \param uc_mgr Use case manager @@ -607,6 +635,9 @@ static int set_modifier(snd_use_case_mgr_t *uc_mgr, struct list_head *seq; int err; + if (modifier_status(uc_mgr, modifier->name) == enable) + return 0; + if (enable) { seq = &modifier->enable_list; } else { @@ -638,6 +669,9 @@ static int set_device(snd_use_case_mgr_t *uc_mgr, struct list_head *seq; int err; + if (device_status(uc_mgr, device->name) == enable) + return 0; + if (enable) { seq = &device->enable_list; } else { @@ -1316,34 +1350,6 @@ int snd_use_case_get(snd_use_case_mgr_t *uc_mgr, return err; } -long device_status(snd_use_case_mgr_t *uc_mgr, - const char *device_name) -{ - struct use_case_device *dev; - struct list_head *pos; - - list_for_each(pos, &uc_mgr->active_devices) { - dev = list_entry(pos, struct use_case_device, active_list); - if (strcmp(dev->name, device_name) == 0) - return 1; - } - return 0; -} - -long modifier_status(snd_use_case_mgr_t *uc_mgr, - const char *modifier_name) -{ - struct use_case_modifier *mod; - struct list_head *pos; - - list_for_each(pos, &uc_mgr->active_modifiers) { - mod = list_entry(pos, struct use_case_modifier, active_list); - if (strcmp(mod->name, modifier_name) == 0) - return 1; - } - return 0; -} - /** * \brief Get current - integer -- 1.7.8.6

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[alsa-devel] [PATCH] ASoC: fsi: use dmaengine helper functions
by Kuninori Morimoto 29 May '12

29 May '12

This patch used dmaengine helper functions instead of using hand setting. And reduced local variables Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This is for v3.6 sound/soc/sh/fsi.c | 19 ++++--------------- 1 files changed, 4 insertions(+), 15 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 18c920f..53486ff 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1089,13 +1089,10 @@ static void fsi_dma_do_tasklet(unsigned long data) { struct fsi_stream *io = (struct fsi_stream *)data; struct fsi_priv *fsi = fsi_stream_to_priv(io); - struct dma_chan *chan; struct snd_soc_dai *dai; struct dma_async_tx_descriptor *desc; - struct scatterlist sg; struct snd_pcm_runtime *runtime; enum dma_data_direction dir; - dma_cookie_t cookie; int is_play = fsi_stream_is_play(fsi, io); int len; dma_addr_t buf; @@ -1104,7 +1101,6 @@ static void fsi_dma_do_tasklet(unsigned long data) return; dai = fsi_get_dai(io->substream); - chan = io->chan; runtime = io->substream->runtime; dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; len = samples_to_bytes(runtime, io->period_samples); @@ -1112,14 +1108,8 @@ static void fsi_dma_do_tasklet(unsigned long data) dma_sync_single_for_device(dai->dev, buf, len, dir); - sg_init_table(&sg, 1); - sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf)), - len , offset_in_page(buf)); - sg_dma_address(&sg) = buf; - sg_dma_len(&sg) = len; - - desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); return; @@ -1128,13 +1118,12 @@ static void fsi_dma_do_tasklet(unsigned long data) desc->callback = fsi_dma_complete; desc->callback_param = io; - cookie = desc->tx_submit(desc); - if (cookie < 0) { + if (dmaengine_submit(desc) < 0) { dev_err(dai->dev, "tx_submit() fail\n"); return; } - dma_async_issue_pending(chan); + dma_async_issue_pending(io->chan); /* * FIXME -- 1.7.5.4

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Re: [alsa-devel] [PATCH 3/4] ASOC: mmp: add sspa support
by Mark Brown 29 May '12

29 May '12

On Fri, May 25, 2012 at 03:11:02PM +0800, Zhangfei Gao wrote: > The SSPA is a configurable multi-channel audio serial (TDM) interface. > It's configurable at runtime to support up to 128 channels and the > number of bits per sample: 8, 12, 16, 20, 24 and 32 bits. It also > support stereo format: I2S, left-justified or right-justified. Mostly looks good. A few fairly minor things... > +static int mmp_sspa_startup(struct snd_pcm_substream *substream, > + struct snd_soc_dai *dai) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; > + struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai); > + struct ssp_device *sspa = sspa_priv->sspa; > + int ret = 0; > + > + if (!cpu_dai->active) { > + clk_enable(sysclk); > + clk_enable(sspa->clk); > + } The clock API is refcounted so you shouldn't need to worry about multiple enables, you should just be able to unconditionally enable and disable. If this is needed we probably have a problem we should fix. > + switch (pll_id) { > + case MMP_SYSCLK: > + clk_set_rate(sysclk, freq_out); > + break; > + case MMP_SSPA_CLK: > + clk_set_rate(sspa->clk, freq_out); > + break; You're ignoring the return values here. > + priv->sspa->clk = clk_get(&pdev->dev, NULL); > + if (IS_ERR(priv->sspa->clk)) { > + ret = PTR_ERR(priv->sspa->clk); > + goto err_free_clk; > + } devm_clk_get(). > + res = request_mem_region(res->start, resource_size(res), > + pdev->name); > + if (res == NULL) { > + dev_err(&pdev->dev, "failed to request memory resource\n"); > + ret = -EBUSY; > + goto err_get_res; > + } > + > + priv->sspa->mmio_base = ioremap(res->start, resource_size(res)); > + if (priv->sspa->mmio_base == NULL) { > + dev_err(&pdev->dev, "failed to ioremap() registers\n"); > + ret = -ENODEV; > + goto err_free_mem; > + } devm_request_and_ioremap(). > +err_free_clk: > + devm_kfree(&pdev->dev, priv->dma_params); > +err_priv_dma_params: > + devm_kfree(&pdev->dev, priv->sspa); > +err_priv_sspa: > + devm_kfree(&pdev->dev, priv); The whole point of the devm_ stuff is that you don't need to do stuff like this.

2 1

[alsa-devel] [PATCH 15/15] ASoC: mxs-saif: Use clk_prepare_enable/clk_disable_unprepare
by Fabio Estevam 28 May '12

28 May '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Prepare the clock before enabling it. Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index aba71bf..15ad61a 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -482,7 +482,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: dev_dbg(cpu_dai->dev, "start\n"); - clk_enable(master_saif->clk); + clk_prepare_enable(master_saif->clk); if (!master_saif->mclk_in_use) __raw_writel(BM_SAIF_CTRL_RUN, master_saif->base + SAIF_CTRL + MXS_SET_ADDR); @@ -492,7 +492,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, * itself clk for its internal basic logic to work. */ if (saif != master_saif) { - clk_enable(saif->clk); + clk_prepare_enable(saif->clk); __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_SET_ADDR); } @@ -534,13 +534,13 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, master_saif->base + SAIF_CTRL + MXS_CLR_ADDR); udelay(delay); } - clk_disable(master_saif->clk); + clk_disable_unprepare(master_saif->clk); if (saif != master_saif) { __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_CLR_ADDR); udelay(delay); - clk_disable(saif->clk); + clk_disable_unprepare(saif->clk); } master_saif->ongoing = 0; -- 1.7.1

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Re: [alsa-devel] [PATCH 4/4] ASoC: add mmp brownstone support
by Mark Brown 28 May '12

28 May '12

On Fri, May 25, 2012 at 03:11:03PM +0800, Zhangfei Gao wrote: > +config SND_MMP_SOC_BROWNSTONE > + tristate "SoC Audio support for Marvell Brownstone" > + depends on SND_MMP_SOC && MACH_BROWNSTONE > + select SND_MMP_SOC_SSPA > + select SND_SOC_WM8994 Should depend on MFD_WM8994. > +static void brownstone_ext_control(struct snd_soc_dapm_context *dapm) > +{ This stuff is really unexpected in a modern machine driver, it was used on things like spitz due to the odd wiring but for modern stuff I'd not expect to see it. > + if (brownstone_spk_func == BROWNSTONE_SPK_ON) { > + snd_soc_dapm_enable_pin(dapm, "Ext Left Spk"); > + snd_soc_dapm_enable_pin(dapm, "Ext Right Spk"); > + } else { > + snd_soc_dapm_disable_pin(dapm, "Ext Left Spk"); > + snd_soc_dapm_disable_pin(dapm, "Ext Right Spk"); > + } Just define a single widget for the speakers and use a SND_SOC_DAPM_PIN_SWITCH(). > + /* set up jack connection */ > + switch (brownstone_jack_func) { > + case BROWNSTONE_HP: > + snd_soc_dapm_disable_pin(dapm, "Headset Mic"); > + snd_soc_dapm_enable_pin(dapm, "Main Mic"); > + snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); > + break; This should all be autodetectable, users having to manually select it is *very* unusual. Is there really no accessory detection hardware on the board? > + snd_soc_dapm_enable_pin(dapm, "Ext Left Spk"); > + snd_soc_dapm_enable_pin(dapm, "Ext Right Spk"); > + snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); > + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); > + snd_soc_dapm_enable_pin(dapm, "Main Mic"); Everything is enable dby default. > + /* turn on micbias 1/2 always */ > + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, > + WM8994_MICB1_ENA_MASK | > + WM8994_MICB2_ENA_MASK, > + WM8994_MICB1_ENA | > + WM8994_MICB2_ENA); If you need to do this force enable them with DAPM, this won't work anyway as the widgets will be powered off as soon as DAPM notices they're on. > + snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | > + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); > + > + snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | > + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); Set this in the dai_link. > +static struct platform_driver mmp_driver = { > + .driver = { > + .name = "mmp-audio", Should probably be something like "brownstone-audio".

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Re: [alsa-devel] [PATCH] ASoC: imx-ssi: Use clk_prepare_enable/clk_disable_unprepare
by Mark Brown 28 May '12

28 May '12

On Fri, May 25, 2012 at 05:27:20PM -0300, Fabio Estevam wrote: > Fix the following kernel crash: Applied, thanks.

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[alsa-devel] [PATCH 3/5] ASoC: Ux500: Add platform-driver
by Ola Lilja 28 May '12

28 May '12

Add platform-driver handling all DMA-activities. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- sound/soc/ux500/Kconfig | 7 + sound/soc/ux500/Makefile | 3 + sound/soc/ux500/ux500_pcm.c | 318 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/ux500/ux500_pcm.h | 35 +++++ 4 files changed, 363 insertions(+), 0 deletions(-) create mode 100644 sound/soc/ux500/ux500_pcm.c create mode 100644 sound/soc/ux500/ux500_pcm.h diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 44cf434..1d38515 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -12,3 +12,10 @@ menuconfig SND_SOC_UX500 config SND_SOC_UX500_PLAT_MSP_I2S tristate depends on SND_SOC_UX500 + +config SND_SOC_UX500_PLAT_DMA + tristate "Platform - DB8500 (DMA)" + depends on SND_SOC_UX500 + select SND_SOC_DMAENGINE_PCM + help + Say Y if you want to enable the Ux500 platform-driver. diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index 19974c5..4634bf0 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -2,3 +2,6 @@ snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o + +snd-soc-ux500-plat-dma-objs := ux500_pcm.o +obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c new file mode 100644 index 0000000..66b080e --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.c @@ -0,0 +1,318 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <asm/page.h> + +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/dmaengine.h> +#include <linux/slab.h> + +#include <plat/ste_dma40.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "ux500_msp_i2s.h" +#include "ux500_pcm.h" + +static struct snd_pcm_hardware ux500_pcm_hw_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_PLAYBACK, + .rate_max = UX500_PLATFORM_MAX_RATE_PLAYBACK, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static struct snd_pcm_hardware ux500_pcm_hw_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_CAPTURE, + .rate_max = UX500_PLATFORM_MAX_RATE_CAPTURE, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static void ux500_pcm_dma_hw_free(struct device *dev, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + + if (runtime->dma_area == NULL) + return; + + if (buf != &substream->dma_buffer) { + dma_free_coherent(buf->dev.dev, buf->bytes, buf->area, + buf->addr); + kfree(runtime->dma_buffer_p); + } + + snd_pcm_set_runtime_buffer(substream, NULL); +} + +static int ux500_pcm_open(struct snd_pcm_substream *substream) +{ + int stream_id = substream->pstr->stream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct device *dev = dai->dev; + int ret; + struct ux500_msp_dma_params *dma_params; + u16 per_data_width, mem_data_width; + struct stedma40_chan_cfg *dma_cfg; + + dev_dbg(dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, + snd_pcm_stream_str(substream)); + + dev_dbg(dev, "%s: Set runtime hwparams.\n", __func__); + if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_playback); + else + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_capture); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(dev, "%s: Error: snd_pcm_hw_constraints failed (%d)\n", + __func__, ret); + return ret; + } + + dev_dbg(dev, "%s: Set hw-struct for %s.\n", __func__, + snd_pcm_stream_str(substream)); + runtime->hw = (stream_id == SNDRV_PCM_STREAM_PLAYBACK) ? + ux500_pcm_hw_playback : ux500_pcm_hw_capture; + + mem_data_width = STEDMA40_HALFWORD_WIDTH; + + dma_params = snd_soc_dai_get_dma_data(dai, substream); + switch (dma_params->data_size) { + case 32: + per_data_width = STEDMA40_WORD_WIDTH; + break; + case 16: + per_data_width = STEDMA40_HALFWORD_WIDTH; + break; + case 8: + per_data_width = STEDMA40_BYTE_WIDTH; + break; + default: + per_data_width = STEDMA40_WORD_WIDTH; + dev_warn(rtd->platform->dev, + "%s: Unknown data-size (%d)! Assuming 32 bits.\n", + __func__, dma_params->data_size); + } + + dma_cfg = dma_params->dma_cfg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_cfg->src_info.data_width = mem_data_width; + dma_cfg->dst_info.data_width = per_data_width; + } else { + dma_cfg->src_info.data_width = per_data_width; + dma_cfg->dst_info.data_width = mem_data_width; + } + + + ret = snd_dmaengine_pcm_open(substream, stedma40_filter, dma_cfg); + if (ret) { + dev_dbg(dai->dev, + "%s: ERROR: snd_dmaengine_pcm_open failed (%d)!\n", + __func__, ret); + return ret; + } + + snd_dmaengine_pcm_set_data(substream, dma_cfg); + + return 0; +} + +static int ux500_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + + dev_dbg(dai->dev, "%s: Enter\n", __func__); + + snd_dmaengine_pcm_close(substream); + + return 0; +} + +static int ux500_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0; + int size; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + size = params_buffer_bytes(hw_params); + + if (buf) { + if (buf->bytes >= size) + goto out; + ux500_pcm_dma_hw_free(NULL, substream); + } + + if (substream->dma_buffer.area != NULL && + substream->dma_buffer.bytes >= size) { + buf = &substream->dma_buffer; + } else { + buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL); + if (!buf) + goto nomem; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = NULL; + buf->area = dma_alloc_coherent(NULL, size, &buf->addr, + GFP_KERNEL); + buf->bytes = size; + buf->private_data = NULL; + + if (!buf->area) + goto free; + } + snd_pcm_set_runtime_buffer(substream, buf); + ret = 1; + out: + runtime->dma_bytes = size; + return ret; + + free: + kfree(buf); + nomem: + return -ENOMEM; +} + +static int ux500_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + ux500_pcm_dma_hw_free(NULL, substream); + + return 0; +} + +static int ux500_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter.\n", __func__); + + return dma_mmap_coherent(NULL, vma, runtime->dma_area, + runtime->dma_addr, runtime->dma_bytes); +} + +static struct snd_pcm_ops ux500_pcm_ops = { + .open = ux500_pcm_open, + .close = ux500_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = ux500_pcm_hw_params, + .hw_free = ux500_pcm_hw_free, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = ux500_pcm_mmap +}; + +int ux500_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + + dev_dbg(rtd->platform->dev, "%s: Enter (id = '%s').\n", __func__, + pcm->id); + + pcm->info_flags = 0; + + return 0; +} + +static struct snd_soc_platform_driver ux500_pcm_soc_drv = { + .ops = &ux500_pcm_ops, + .pcm_new = ux500_pcm_new, +}; + +static int __devexit ux500_pcm_drv_probe(struct platform_device *pdev) +{ + int ret; + + ret = snd_soc_register_platform(&pdev->dev, &ux500_pcm_soc_drv); + if (ret < 0) { + dev_err(&pdev->dev, + "%s: ERROR: Failed to register platform '%s' (%d)!\n", + __func__, pdev->name, ret); + return ret; + } + + return 0; +} + +static int __devinit ux500_pcm_drv_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver ux500_pcm_driver = { + .driver = { + .name = "ux500-pcm", + .owner = THIS_MODULE, + }, + + .probe = ux500_pcm_drv_probe, + .remove = __devexit_p(ux500_pcm_drv_remove), +}; +module_platform_driver(ux500_pcm_driver); + +MODULE_LICENSE("GPLv2"); diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h new file mode 100644 index 0000000..77ed44d --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.h @@ -0,0 +1,35 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ +#ifndef UX500_PCM_H +#define UX500_PCM_H + +#include <asm/page.h> + +#include <linux/workqueue.h> + +#define UX500_PLATFORM_MIN_RATE_PLAYBACK 8000 +#define UX500_PLATFORM_MAX_RATE_PLAYBACK 48000 +#define UX500_PLATFORM_MIN_RATE_CAPTURE 8000 +#define UX500_PLATFORM_MAX_RATE_CAPTURE 48000 + +#define UX500_PLATFORM_MIN_CHANNELS 1 +#define UX500_PLATFORM_MAX_CHANNELS 8 + +#define UX500_PLATFORM_PERIODS_BYTES_MIN 128 +#define UX500_PLATFORM_PERIODS_BYTES_MAX (64 * PAGE_SIZE) +#define UX500_PLATFORM_PERIODS_MIN 2 +#define UX500_PLATFORM_PERIODS_MAX 48 +#define UX500_PLATFORM_BUFFER_BYTES_MAX (2048 * PAGE_SIZE) + +#endif -- 1.7.8.3

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[alsa-devel] [PATCH 2/5] ASoC: core: Add widget SND_SOC_DAPM_CLOCK_SUPPLY
by Ola Lilja 28 May '12

28 May '12

Adds a supply-widget variant for connection to the clock-framework. This widget-type corresponds to the variant for regulators. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- include/sound/soc-dapm.h | 10 ++++++++++ sound/soc/soc-dapm.c | 38 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 48 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e3833d9..05559e5 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -229,6 +229,10 @@ struct device; { .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert, \ .event = wevent, .event_flags = wflags} +#define SND_SOC_DAPM_CLOCK_SUPPLY(wname) \ +{ .id = snd_soc_dapm_clock_supply, .name = wname, \ + .reg = SND_SOC_NOPM, .event = dapm_clock_event, \ + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } /* generic widgets */ #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ @@ -245,6 +249,7 @@ struct device; .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } + /* dapm kcontrol types */ #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -327,6 +332,8 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +int dapm_clock_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, @@ -432,6 +439,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ snd_soc_dapm_regulator_supply, /* external regulator */ + snd_soc_dapm_clock_supply, /* external clock */ snd_soc_dapm_aif_in, /* audio interface input */ snd_soc_dapm_aif_out, /* audio interface output */ snd_soc_dapm_siggen, /* signal generator */ @@ -537,6 +545,8 @@ struct snd_soc_dapm_widget { struct list_head dirty; int inputs; int outputs; + + struct clk *clk; }; struct snd_soc_dapm_update { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 90ee77d..3bb7a6f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -35,6 +35,7 @@ #include <linux/debugfs.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> +#include <linux/clk.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -51,6 +52,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, [snd_soc_dapm_supply] = 1, [snd_soc_dapm_regulator_supply] = 1, + [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_micbias] = 2, [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai] = 3, @@ -92,6 +94,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_aif_out] = 10, [snd_soc_dapm_dai] = 10, [snd_soc_dapm_dai_link] = 11, + [snd_soc_dapm_clock_supply] = 12, [snd_soc_dapm_regulator_supply] = 12, [snd_soc_dapm_supply] = 12, [snd_soc_dapm_post] = 13, @@ -391,6 +394,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_vmid: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_dai: @@ -764,6 +768,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: return 0; default: break; @@ -850,6 +855,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: return 0; default: break; @@ -996,6 +1002,24 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_regulator_event); +/* + * Handler for clock supply widget. + */ +int dapm_clock_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (!w->clk) + return -EIO; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + return clk_enable(w->clk); + } else { + clk_disable(w->clk); + return 0; + } +} +EXPORT_SYMBOL_GPL(dapm_clock_event); + static int dapm_widget_power_check(struct snd_soc_dapm_widget *w) { if (w->power_checked) @@ -1487,6 +1511,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, switch (w->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: /* Supplies can't affect their outputs, only their inputs */ break; default: @@ -1587,6 +1612,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_micbias: if (d->target_bias_level < SND_SOC_BIAS_STANDBY) d->target_bias_level = SND_SOC_BIAS_STANDBY; @@ -1941,6 +1967,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: if (w->name) count += sprintf(buf + count, "%s: %s\n", w->name, w->power ? "On":"Off"); @@ -2187,6 +2214,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_post: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_dai: @@ -2873,6 +2901,15 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, return NULL; } break; + case snd_soc_dapm_clock_supply: + w->clk = clk_get(dapm->dev, w->name); + if (IS_ERR(w->clk)) { + ret = PTR_ERR(w->clk); + dev_err(dapm->dev, "Failed to request %s: %d\n", + w->name, ret); + return NULL; + } + break; default: break; } @@ -2924,6 +2961,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: w->power_check = dapm_supply_check_power; break; case snd_soc_dapm_dai: -- 1.7.8.3

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[alsa-devel] [PATCH 1/5] ALSA: pcm: Add debug-print helper function
by Ola Lilja 28 May '12

28 May '12

Adds a function getting the stream-name as a string for a specific stream. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- include/sound/pcm.h | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0d11128..a55d5db 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1073,4 +1073,15 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) const char *snd_pcm_format_name(snd_pcm_format_t format); +/** + * Get a string naming the direction of a stream + */ +static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return "Playback"; + else + return "Capture"; +} + #endif /* __SOUND_PCM_H */ -- 1.7.8.3

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Re: [alsa-devel] Adjust volume may cause audio playback corruption
by cee1 28 May '12

28 May '12

2012/5/12 cee1 <fykcee1(a)gmail.com>: > Hi all, > > We're using PA on our product, and sometimes audio playback may > corrupt after adjusting volume. > > We found a way to (relatively) easily reproduce the problem: > 1. Play sample.mp3 through totem in repeat mode. > 2. In gnome-volume-control, Hardware tap, repeat to shift between "Off > profile" and "Analog Stereo Duplex profile". > > When the problem happens, the corruption continues until I: > * Do one more shift between "Off profile" and "Analog Stereo Duplex profile". > * Close totem (close all PA playback clients). > * Adjust volume. > > When the problem happens, open another PA client doing audio playback, > is also corrupt. > > Some additional information: > * Output of alsa-info.sh on our product: > http://dev.lemote.com/files/upload/software/PA-apc/alsa-info-lemote > * PA daemon log: > http://dev.lemote.com/files/upload/software/PA-apc/pulseaudio.log > * The corrupted sound: > http://dev.lemote.com/files/upload/software/PA-apc/corrupted_sound.ogg > * The sample mp3: http://dev.lemote.com/files/upload/software/PA-apc/sample.mp3 When the problem happens today, I notice: 1) PA complains "alsa-source.c: Overrun!". I doesn't run any recording application then. Curious. 2) Playback becomes good automatically when play sample.mp3 again(note totem is in repeat playing mode). It says sink becomes idle(I didn't touch totem, so totem should keep playing back). Any idea? Following is PA daemon log: I: [alsa-source] ratelimit.c: 1226 events suppressed I: [alsa-source] alsa-source.c: Overrun! I: [alsa-source] alsa-source.c: Increasing minimal latency to 1.00 ms D: [alsa-source] alsa-source.c: latency set to 20.00ms D: [alsa-source] alsa-source.c: hwbuf_unused=62008 D: [alsa-source] alsa-source.c: setting avail_min=442 I: [alsa-source] alsa-source.c: Overrun! I: [alsa-source] alsa-source.c: Increasing minimal latency to 2.00 ms D: [alsa-source] alsa-source.c: latency set to 20.00ms D: [alsa-source] alsa-source.c: hwbuf_unused=62008 D: [alsa-source] alsa-source.c: setting avail_min=442 I: [alsa-source] alsa-source.c: Overrun! I: [alsa-source] alsa-source.c: Increasing minimal latency to 4.00 ms D: [alsa-source] alsa-source.c: latency set to 20.00ms D: [alsa-source] alsa-source.c: hwbuf_unused=62008 D: [alsa-source] alsa-source.c: setting avail_min=442 I: [alsa-source] alsa-source.c: Overrun! I: [alsa-source] alsa-source.c: Increasing minimal latency to 8.00 ms D: [alsa-source] alsa-source.c: latency set to 20.00ms D: [alsa-source] alsa-source.c: hwbuf_unused=62008 D: [alsa-source] alsa-source.c: setting avail_min=442 I: [alsa-source] alsa-source.c: Overrun! I: [alsa-source] alsa-source.c: Increasing minimal latency to 16.00 ms D: [alsa-source] alsa-source.c: latency set to 20.00ms D: [alsa-source] alsa-source.c: hwbuf_unused=62008 D: [alsa-source] alsa-source.c: setting avail_min=442 I: [alsa-source] alsa-source.c: Overrun! I: [alsa-source] alsa-source.c: Increasing minimal latency to 26.00 ms D: [alsa-source] alsa-source.c: latency set to 26.00ms D: [alsa-source] alsa-source.c: hwbuf_unused=60952 D: [alsa-source] alsa-source.c: setting avail_min=706 D: [alsa-sink] protocol-native.c: Underrun on 'Playback Stream', 0 bytes in queue. D: [alsa-sink] protocol-native.c: Requesting rewind due to rewrite. D: [alsa-sink] sink-input.c: Requesting rewind due to corking D: [alsa-sink] alsa-sink.c: Requested to rewind 16208 bytes. D: [alsa-sink] alsa-sink.c: Limited to 15096 bytes. D: [alsa-sink] alsa-sink.c: before: 3774 D: [alsa-sink] alsa-sink.c: after: 3642 D: [alsa-sink] alsa-sink.c: Rewound 14568 bytes. D: [alsa-sink] sink.c: Processing rewind... D: [alsa-sink] sink.c: latency = 4156 D: [alsa-sink] sink-input.c: Have to rewind 14568 bytes on render memblockq. D: [alsa-sink] sink-input.c: Have to rewind 29136 bytes on implementor. D: [alsa-sink] source.c: Processing rewind... D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes idle, timeout in 5 seconds. D: [alsa-sink] protocol-native.c: Requesting rewind due to end of underrun. D: [alsa-sink] protocol-native.c: Requesting rewind due to end of underrun. D: [alsa-sink] protocol-native.c: Requesting rewind due to end of underrun. D: [alsa-sink] protocol-native.c: Requesting rewind due to end of underrun. D: [alsa-sink] protocol-native.c: Requesting rewind due to end of underrun. D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0004, suspending I: [alsa-sink] alsa-sink.c: Device suspended... I: [pulseaudio] alsa-sink.c: Updating rate for device front:0, new rate is 44100 I: [pulseaudio] sink.c: Changed sampling rate successfully D: [alsa-sink] protocol-native.c: Requesting rewind due to end of underrun. D: [alsa-sink] sink-input.c: Requesting rewind due to uncorking D: [pulseaudio] sink.c: Suspend cause of sink alsa_output.pci-0000_00_14.2.analog-stereo is 0x0000, resuming I: [alsa-sink] alsa-sink.c: Trying resume... I: [alsa-sink] alsa-util.c: Trying to disable ALSA period wakeups, using timers only D: [alsa-sink] alsa-util.c: Maximum hw buffer size is 371 ms D: [alsa-sink] alsa-util.c: Set buffer size first (to 16384 samples), period size second (to 8192 samples). I: [alsa-sink] alsa-util.c: ALSA period wakeups disabled D: [alsa-sink] alsa-sink.c: Latency set to 90.00ms D: [alsa-sink] alsa-sink.c: hwbuf_unused=49660 D: [alsa-sink] alsa-sink.c: setting avail_min=15503 I: [alsa-sink] alsa-sink.c: Time scheduling watermark is 20.00ms I: [alsa-sink] alsa-sink.c: Resumed successfully... D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. I: [alsa-sink] alsa-sink.c: Starting playback. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [pulseaudio] core-subscribe.c: Dropped redundant event due to change event. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [pulseaudio] module-suspend-on-idle.c: Sink alsa_output.pci-0000_00_14.2.analog-stereo becomes busy. D: [pulseaudio] module-udev-detect.c: /dev/snd/controlC0 is accessible: yes D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. D: [alsa-sink] alsa-sink.c: Cutting sleep time for the initial iterations by half. > > We've tried PA 1.0-0ubuntu3, 1.1-0ubuntu15 and 2.0, the problem continues. > > The problem can also be reproduced on an AMD turion 64 x2 board with > ubuntu 10.10 and ubuntu 12.04: > * Output of alsa-info: > ubuntu 10.10(32bit) > http://dev.lemote.com/files/upload/software/PA-apc/alsa-info-ubuntu-10.10 > ubuntu 12.04(32bit) > http://dev.lemote.com/files/upload/software/PA-apc/alsa-info-ubuntu-12.04 > > Note: > * The AMD turion 64 x2 board uses the same chipset as our product. > * gnome-volume-control is removed in ubuntu 12.04, thus we compiled > it: http://dev.lemote.com/files/upload/software/PA-apc/gnome-volume-control -- Regards, - cee1

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