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5f712b2b73
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
[Note that this is a backported version for 2.6.34.
Upstream commit is fd23b7dee]
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
255 lines
7.6 KiB
C
255 lines
7.6 KiB
C
/*
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* linux/sound/soc-dai.h -- ALSA SoC Layer
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*
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* Copyright: 2005-2008 Wolfson Microelectronics. PLC.
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License version 2 as
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* published by the Free Software Foundation.
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*
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* Digital Audio Interface (DAI) API.
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*/
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#ifndef __LINUX_SND_SOC_DAI_H
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#define __LINUX_SND_SOC_DAI_H
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#include <linux/list.h>
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#include <sound/soc.h>
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struct snd_pcm_substream;
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/*
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* DAI hardware audio formats.
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*
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* Describes the physical PCM data formating and clocking. Add new formats
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* to the end.
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*/
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#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */
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#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */
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#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */
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#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */
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#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */
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#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
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#define SND_SOC_DAIFMT_PDM 6 /* Pulse density modulation */
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/* left and right justified also known as MSB and LSB respectively */
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#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
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#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
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/*
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* DAI Clock gating.
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*
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* DAI bit clocks can be be gated (disabled) when the DAI is not
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* sending or receiving PCM data in a frame. This can be used to save power.
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*/
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#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
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#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
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/*
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* DAI hardware signal inversions.
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*
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* Specifies whether the DAI can also support inverted clocks for the specified
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* format.
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*/
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#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
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#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */
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#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */
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#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert BCLK + FRM */
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/*
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* DAI hardware clock masters.
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*
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* This is wrt the codec, the inverse is true for the interface
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* i.e. if the codec is clk and FRM master then the interface is
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* clk and frame slave.
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*/
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#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */
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#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */
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#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */
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#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */
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#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
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#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
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#define SND_SOC_DAIFMT_INV_MASK 0x0f00
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#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
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/*
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* Master Clock Directions
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*/
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#define SND_SOC_CLOCK_IN 0
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#define SND_SOC_CLOCK_OUT 1
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#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
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SNDRV_PCM_FMTBIT_S16_LE |\
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SNDRV_PCM_FMTBIT_S16_BE |\
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SNDRV_PCM_FMTBIT_S20_3LE |\
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SNDRV_PCM_FMTBIT_S20_3BE |\
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SNDRV_PCM_FMTBIT_S24_3LE |\
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SNDRV_PCM_FMTBIT_S24_3BE |\
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SNDRV_PCM_FMTBIT_S32_LE |\
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SNDRV_PCM_FMTBIT_S32_BE)
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struct snd_soc_dai_ops;
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struct snd_soc_dai;
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struct snd_ac97_bus_ops;
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/* Digital Audio Interface registration */
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int snd_soc_register_dai(struct snd_soc_dai *dai);
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void snd_soc_unregister_dai(struct snd_soc_dai *dai);
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int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count);
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void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count);
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/* Digital Audio Interface clocking API.*/
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int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
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unsigned int freq, int dir);
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int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
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int div_id, int div);
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int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
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int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
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/* Digital Audio interface formatting */
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int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
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int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
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unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
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int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
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unsigned int tx_num, unsigned int *tx_slot,
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unsigned int rx_num, unsigned int *rx_slot);
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int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
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/* Digital Audio Interface mute */
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int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
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/*
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* Digital Audio Interface.
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*
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* Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
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* operations and capabilities. Codec and platform drivers will register this
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* structure for every DAI they have.
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*
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* This structure covers the clocking, formating and ALSA operations for each
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* interface.
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*/
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struct snd_soc_dai_ops {
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/*
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* DAI clocking configuration, all optional.
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* Called by soc_card drivers, normally in their hw_params.
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*/
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int (*set_sysclk)(struct snd_soc_dai *dai,
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int clk_id, unsigned int freq, int dir);
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int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
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unsigned int freq_in, unsigned int freq_out);
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int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
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/*
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* DAI format configuration
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* Called by soc_card drivers, normally in their hw_params.
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*/
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int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
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int (*set_tdm_slot)(struct snd_soc_dai *dai,
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unsigned int tx_mask, unsigned int rx_mask,
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int slots, int slot_width);
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int (*set_channel_map)(struct snd_soc_dai *dai,
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unsigned int tx_num, unsigned int *tx_slot,
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unsigned int rx_num, unsigned int *rx_slot);
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int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
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/*
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* DAI digital mute - optional.
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* Called by soc-core to minimise any pops.
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*/
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int (*digital_mute)(struct snd_soc_dai *dai, int mute);
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/*
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* ALSA PCM audio operations - all optional.
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* Called by soc-core during audio PCM operations.
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*/
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int (*startup)(struct snd_pcm_substream *,
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struct snd_soc_dai *);
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void (*shutdown)(struct snd_pcm_substream *,
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struct snd_soc_dai *);
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int (*hw_params)(struct snd_pcm_substream *,
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struct snd_pcm_hw_params *, struct snd_soc_dai *);
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int (*hw_free)(struct snd_pcm_substream *,
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struct snd_soc_dai *);
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int (*prepare)(struct snd_pcm_substream *,
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struct snd_soc_dai *);
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int (*trigger)(struct snd_pcm_substream *, int,
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struct snd_soc_dai *);
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/*
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* For hardware based FIFO caused delay reporting.
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* Optional.
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*/
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snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
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struct snd_soc_dai *);
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};
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/*
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* Digital Audio Interface runtime data.
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*
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* Holds runtime data for a DAI.
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*/
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struct snd_soc_dai {
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/* DAI description */
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char *name;
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unsigned int id;
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int ac97_control;
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struct device *dev;
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void *ac97_pdata; /* platform_data for the ac97 codec */
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/* DAI callbacks */
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int (*probe)(struct platform_device *pdev,
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struct snd_soc_dai *dai);
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void (*remove)(struct platform_device *pdev,
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struct snd_soc_dai *dai);
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int (*suspend)(struct snd_soc_dai *dai);
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int (*resume)(struct snd_soc_dai *dai);
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/* ops */
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struct snd_soc_dai_ops *ops;
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/* DAI capabilities */
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struct snd_soc_pcm_stream capture;
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struct snd_soc_pcm_stream playback;
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unsigned int symmetric_rates:1;
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/* DAI runtime info */
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struct snd_soc_codec *codec;
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unsigned int active;
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unsigned char pop_wait:1;
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/* DAI private data */
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void *private_data;
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/* parent platform */
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struct snd_soc_platform *platform;
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struct list_head list;
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};
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static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
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const struct snd_pcm_substream *ss)
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{
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return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
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dai->playback.dma_data : dai->capture.dma_data;
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}
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static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
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const struct snd_pcm_substream *ss,
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void *data)
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{
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if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
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dai->playback.dma_data = data;
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else
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dai->capture.dma_data = data;
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}
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#endif
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