May 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ALSA: hda - allow to mask a buggy stream DMA by setting it as opened
by mengdong.lin＠intel.com 13 May '14

13 May '14

From: Mengdong Lin <mengdong.lin(a)intel.com> The function azx_init_stream() can accept a bit mask parameter. If the bit for a stream DMA is set, that stream device will be masked by setting it as opened and so it will never be assigned to any PCM stream. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 248b90a..378fed7 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -2004,7 +2004,7 @@ EXPORT_SYMBOL_GPL(azx_mixer_create); /* initialize SD streams */ -int azx_init_stream(struct azx *chip) +int azx_init_stream(struct azx *chip, unsigned int mask) { int i; @@ -2022,6 +2022,9 @@ int azx_init_stream(struct azx *chip) /* stream tag: must be non-zero and unique */ azx_dev->index = i; azx_dev->stream_tag = i + 1; + /* mask a buggy stream DMA by setting it as opened */ + if (mask & (1 << i)) + azx_dev->opened = 1; } return 0; diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index baf0e77..2fd8e6d 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -48,6 +48,6 @@ int azx_codec_create(struct azx *chip, const char *model, int *power_save_to); int azx_codec_configure(struct azx *chip); int azx_mixer_create(struct azx *chip); -int azx_init_stream(struct azx *chip); +int azx_init_stream(struct azx *chip, unsigned int mask); #endif /* __SOUND_HDA_CONTROLLER_H */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d6bca62..377c3a5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1364,7 +1364,7 @@ static int azx_first_init(struct azx *chip) return err; /* initialize streams */ - azx_init_stream(chip); + azx_init_stream(chip, 0); /* initialize chip */ azx_init_pci(chip); -- 1.8.1.2

1 0

[alsa-devel] [RFC PATCH 0/2] Add support for audio cards with multiple endpoints on the one interface
by Eduard Gilmutdinov 13 May '14

13 May '14

Devices of company Focusrite (Novation) contain multiple endpoints on a single interface. In particular (1235:0010) "Novation Saffire 6 USB" on one interface have endpoints for playback and capture. Functions "set_format" and "pcm_close" are generated when new stream is creating and when it is closing respectively. They contain a function call "usb_set_interface" which resets interface and endpoints on it. Thus it is imposible to capture and playback a sound at the same time. To avoid this, suggest to monitor the status of the interface and endpoint on it. Status of endpoints can be monitored by the field "enabled" of the structure "usb_host_endpoint". This approach should not affect for the audio cards with one endpoint on the interface. Please correct me if I'm wrong.

1 2

[alsa-devel] [PATCH v4 0/5] ASoC: Move IO and kcontrols to the component level
by Lars-Peter Clausen 13 May '14

13 May '14

Changes since v3: * Removed the omap-hdmi compile fix that slipped in by accident. v3 cover letter: The only change in v3 is the renaming of the component IO mutex from 'mutex' to 'io_mutex'. v2 cover letter: This series contains the remaining patches from the previous series that have not been applied yet. The series has been rebased ontop of asoc/for-next and depends on topic/sta350, topic/core, topic/dapm, topic/cache and topic/component. Some minor updates were made to the patches and three new patches: * One new user of snd_kcontrol_chip() showed up, the patch converts it to snd_soc_kcontrol_codec() * Change return type of snd_soc_write() from unsigned int to int * Remove ASoC level tracing Original cover letter: This series is the first step towards full componentisation of the ASoC core. It moves both the IO abstraction layers within ASoC as well as the standard set of kcontrols to the component level. This for example means we can get rid of constructs like if (w->codec) snd_soc_read(....) else if(w->platform) snd_soc_platform_read(...) Moving the kcontrols to the component level means we can use the same implementation also for other non-CODEC components. E.g. there seems to be an increasing amount of CPU components that have basic signal processing and things like volume controls etc. whose register layout is similar to those used in CODECs. Currently each CPU component driver re-implements these controls by hand. The first two patches introduce two new helper functions which hide the actual implementation on how the CODEC or platform struct that register a control can be obtained from the control. This means that when the actual implementation is changed only the two helper functions need to be updated and not every single driver. The patches that follow that are just cleanups removing unused IO stuff and move all IO functions to soc-io.c. The next step is to make platforms also components. And then finally first the IO abstraction layers in ASoC are unified at the component level and then on top of that the kcontrol helpers are moved to the component level. The series depends on quite a few topic branches related to changes to the core and cleanups for individual drivers. It is probably best to place it on top of asoc-v3.15-2. The patch that moves the kcontrols to the component level also has a runtime dependency on the not yet applied patches that move the ams-delta and mfld_machine controls to the card level. Lars-Peter Clausen (5): ASoC: Move IO abstraction to the component level ASoC: Move standard kcontrol helpers to the component level ASoC: Remove snd_soc_update_bits_locked() ASoC: dapm: Rename soc_widget_update_bits_locked() to soc_widget_update_bits() ASoC: Remove ASoC level IO tracing include/sound/soc-dapm.h | 1 + include/sound/soc.h | 51 +++++-- include/trace/events/asoc.h | 92 ------------- sound/soc/soc-core.c | 285 +++++++++++++++++++++++++------------- sound/soc/soc-dapm.c | 91 ++---------- sound/soc/soc-io.c | 327 ++++++++++++++++++++++++++------------------ 6 files changed, 447 insertions(+), 400 deletions(-) -- 1.8.0

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[alsa-devel] [PATCH] alsa-plugins: fix polling and recovering in alsa-jack plugin
by Sergey 13 May '14

13 May '14

Hello. This patch fixes polling in alsa-to-jack plugin. It makes poll()+_poll_revents() return correct values whenever there's something to read/write. Additionally jack_deactivate() code makes it survive snd_pcm_recover(). It works for the code example I posted before, aplay, arecord, mozilla, etc. Comments and suggestions are welcome. --- diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 7a8b24d..a39fea5 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -42,6 +42,7 @@ typedef struct { unsigned int num_ports; unsigned int hw_ptr; unsigned int sample_bits; + snd_pcm_uframes_t min_avail; unsigned int channels; snd_pcm_channel_area_t *areas; @@ -50,6 +51,41 @@ typedef struct { jack_client_t *client; } snd_pcm_jack_t; +static inline int poll_blocked(snd_pcm_ioplug_t *io) +{ + static char buf[65536]; + snd_pcm_sframes_t avail; + snd_pcm_jack_t *jack = io->private_data; + + if (io->state == SND_PCM_STATE_RUNNING) + { + avail = snd_pcm_avail_update(io->pcm); + if (avail >= 0 && avail < jack->min_avail) + { + read(io->poll_fd, &buf, sizeof buf); + return 1; + } + } + + return 0; +} + +static inline int poll_unblocked(snd_pcm_ioplug_t *io) +{ + static char buf[1]; + snd_pcm_sframes_t avail; + snd_pcm_jack_t *jack = io->private_data; + + avail = snd_pcm_avail_update(io->pcm); + if (avail < 0 || avail >= jack->min_avail) + { + write(jack->fd, &buf, 1); + return 1; + } + + return 0; +} + static void snd_pcm_jack_free(snd_pcm_jack_t *jack) { if (jack) { @@ -81,14 +117,10 @@ static int snd_pcm_jack_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { - static char buf[1]; - assert(pfds && nfds == 1 && revents); - read(pfds[0].fd, buf, 1); - *revents = pfds[0].revents & ~(POLLIN | POLLOUT); - if (pfds[0].revents & POLLIN) + if (pfds[0].revents & POLLIN && !poll_blocked(io)) *revents |= (io->stream == SND_PCM_STREAM_PLAYBACK) ? POLLOUT : POLLIN; return 0; } @@ -105,7 +137,6 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) snd_pcm_jack_t *jack = io->private_data; const snd_pcm_channel_area_t *areas; snd_pcm_uframes_t xfer = 0; - static char buf[1]; unsigned int channel; for (channel = 0; channel < io->channels; channel++) { @@ -145,7 +176,7 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) xfer += frames; } - write(jack->fd, buf, 1); /* for polling */ + poll_unblocked(io); /* unblock socket for polling if needed */ return 0; } @@ -154,9 +185,25 @@ static int snd_pcm_jack_prepare(snd_pcm_ioplug_t *io) { snd_pcm_jack_t *jack = io->private_data; unsigned int i; + snd_pcm_sw_params_t *swparams; + int err; jack->hw_ptr = 0; + jack->min_avail = io->period_size; + snd_pcm_sw_params_alloca(&swparams); + err = snd_pcm_sw_params_current(io->pcm, swparams); + if (err == 0) { + snd_pcm_sw_params_get_avail_min(swparams, &jack->min_avail); + } + + poll_unblocked(io); /* unblock socket for initial polling if needed */ + + if (jack->activated) { + jack_deactivate(jack->client); + jack->activated = 0; + } + if (jack->ports) return 0; --

2 4

[alsa-devel] [PATCH RESEND 0/4] ASoC: fsl_esai: Fix some issues
by Nicolin Chen 13 May '14

13 May '14

Sorry for this resend. I just found my previous patches have not been delivered successfully to the alsa maillist. This series of patches fix the following problems: PATCH-1 for an incorrect ratio range check. PATCH-2 for an incorrect bypass settings of FSYS clock source. PATCH-3 for duplicated calculations and configurations of dividers PATCH-4 for mismatched init flow of configuration comparing to RM. Nicolin Chen (4): ASoC: fsl_esai: Fix incorrect condition within ratio range check for FP ASoC: fsl_esai: Only bypass sck_div for EXTAL source ASoC: fsl_esai: Bypass divider settings if clock requirement is not changed ASoC: fsl_esai: Set PCRC and PRRC registers at the end of hw_params() sound/soc/fsl/fsl_esai.c | 45 ++++++++++++++++++++++++++++++++++----------- 1 file changed, 34 insertions(+), 11 deletions(-) -- 1.8.4

2 5

[alsa-devel] [PATCH 2/2] ASoC: fsl_spdif: Improve coding style
by Nicolin Chen 13 May '14

13 May '14

1) Apply better indentations 2) Drop braces for single statement. 3) Use simpler ternary to reduce code. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 72 +++++++++++++++++++++++------------------------ sound/soc/fsl/fsl_spdif.h | 2 ++ 2 files changed, 37 insertions(+), 37 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 9ea2dd4..9d35c5c 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -32,10 +32,13 @@ #define FSL_SPDIF_TXFIFO_WML 0x8 #define FSL_SPDIF_RXFIFO_WML 0x8 -#define INTR_FOR_PLAYBACK (INT_TXFIFO_RESYNC) -#define INTR_FOR_CAPTURE (INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL | INT_URX_OV|\ - INT_QRX_FUL | INT_QRX_OV | INT_UQ_SYNC | INT_UQ_ERR |\ - INT_RXFIFO_RESYNC | INT_LOSS_LOCK | INT_DPLL_LOCKED) +#define INTR_FOR_PLAYBACK (INT_TXFIFO_RESYNC) +#define INTR_FOR_CAPTURE (INT_SYM_ERR | INT_BIT_ERR | INT_URX_FUL |\ + INT_URX_OV | INT_QRX_FUL | INT_QRX_OV |\ + INT_UQ_SYNC | INT_UQ_ERR | INT_RXFIFO_RESYNC |\ + INT_LOSS_LOCK | INT_DPLL_LOCKED) + +#define SIE_INTR_FOR(tx) (tx ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE) /* Index list for the values that has if (DPLL Locked) condition */ static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb }; @@ -137,10 +140,9 @@ static void spdif_irq_sym_error(struct fsl_spdif_priv *spdif_priv) dev_dbg(&pdev->dev, "isr: receiver found illegal symbol\n"); - if (!spdif_priv->dpll_locked) { - /* DPLL unlocked seems no audio stream */ + /* Clear illegal symbol if DPLL unlocked since no audio stream */ + if (!spdif_priv->dpll_locked) regmap_update_bits(regmap, REG_SPDIF_SIE, INT_SYM_ERR, 0); - } } /* U/Q Channel receive register full */ @@ -335,8 +337,8 @@ static void spdif_write_channel_status(struct fsl_spdif_priv *spdif_priv) u32 ch_status; ch_status = (bitrev8(ctrl->ch_status[0]) << 16) | - (bitrev8(ctrl->ch_status[1]) << 8) | - bitrev8(ctrl->ch_status[2]); + (bitrev8(ctrl->ch_status[1]) << 8) | + bitrev8(ctrl->ch_status[2]); regmap_write(regmap, REG_SPDIF_STCSCH, ch_status); dev_dbg(&pdev->dev, "STCSCH: 0x%06x\n", ch_status); @@ -433,13 +435,12 @@ clk_set_bypass: spdif_set_cstatus(ctrl, IEC958_AES3_CON_FS, csfs); /* select clock source and divisor */ - stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) | STC_TXCLK_DF(txclk_df); - mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK | STC_TXCLK_DF_MASK; + stc = STC_TXCLK_ALL_EN | STC_TXCLK_SRC_SET(clk) | + STC_TXCLK_DF(txclk_df) | STC_SYSCLK_DF(sysclk_df); + mask = STC_TXCLK_ALL_EN_MASK | STC_TXCLK_SRC_MASK | + STC_TXCLK_DF_MASK | STC_SYSCLK_DF_MASK; regmap_update_bits(regmap, REG_SPDIF_STC, mask, stc); - regmap_update_bits(regmap, REG_SPDIF_STC, - STC_SYSCLK_DF_MASK, STC_SYSCLK_DF(sysclk_df)); - dev_dbg(&pdev->dev, "set sample rate to %dHz for %dHz playback\n", spdif_priv->txrate[rate], sample_rate); @@ -553,7 +554,7 @@ static int fsl_spdif_hw_params(struct snd_pcm_substream *substream, return ret; } spdif_set_cstatus(ctrl, IEC958_AES3_CON_CLOCK, - IEC958_AES3_CON_CLOCK_1000PPM); + IEC958_AES3_CON_CLOCK_1000PPM); spdif_write_channel_status(spdif_priv); } else { /* Setup rx clock source */ @@ -569,9 +570,9 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct regmap *regmap = spdif_priv->regmap; - int is_playack = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - u32 intr = is_playack ? INTR_FOR_PLAYBACK : INTR_FOR_CAPTURE; - u32 dmaen = is_playack ? SCR_DMA_TX_EN : SCR_DMA_RX_EN;; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 intr = SIE_INTR_FOR(tx); + u32 dmaen = SCR_DMA_xX_EN(tx); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -662,9 +663,8 @@ static int fsl_spdif_capture_get(struct snd_kcontrol *kcontrol, u32 cstatus, val; regmap_read(regmap, REG_SPDIF_SIS, &val); - if (!(val & INT_CNEW)) { + if (!(val & INT_CNEW)) return -EAGAIN; - } regmap_read(regmap, REG_SPDIF_SRCSH, &cstatus); ucontrol->value.iec958.status[0] = (cstatus >> 16) & 0xFF; @@ -693,15 +693,14 @@ static int fsl_spdif_subcode_get(struct snd_kcontrol *kcontrol, struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; unsigned long flags; - int ret = 0; + int ret = -EAGAIN; spin_lock_irqsave(&ctrl->ctl_lock, flags); if (ctrl->ready_buf) { int idx = (ctrl->ready_buf - 1) * SPDIF_UBITS_SIZE; memcpy(&ucontrol->value.iec958.subcode[0], &ctrl->subcode[idx], SPDIF_UBITS_SIZE); - } else { - ret = -EAGAIN; + ret = 0; } spin_unlock_irqrestore(&ctrl->ctl_lock, flags); @@ -726,15 +725,14 @@ static int fsl_spdif_qget(struct snd_kcontrol *kcontrol, struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; unsigned long flags; - int ret = 0; + int ret = -EAGAIN; spin_lock_irqsave(&ctrl->ctl_lock, flags); if (ctrl->ready_buf) { int idx = (ctrl->ready_buf - 1) * SPDIF_QSUB_SIZE; memcpy(&ucontrol->value.bytes.data[0], &ctrl->qsub[idx], SPDIF_QSUB_SIZE); - } else { - ret = -EAGAIN; + ret = 0; } spin_unlock_irqrestore(&ctrl->ctl_lock, flags); @@ -799,10 +797,10 @@ static int spdif_get_rxclk_rate(struct fsl_spdif_priv *spdif_priv, regmap_read(regmap, REG_SPDIF_SRPC, &phaseconf); clksrc = (phaseconf >> SRPC_CLKSRC_SEL_OFFSET) & 0xf; - if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED)) { - /* Get bus clock from system */ + + /* Get bus clock from system */ + if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED)) busclk_freq = clk_get_rate(spdif_priv->sysclk); - } /* FreqMeas_CLK = (BUS_CLK * FreqMeas) / 2 ^ 10 / GAINSEL / 128 */ tmpval64 = (u64) busclk_freq * freqmeas; @@ -826,12 +824,12 @@ static int fsl_spdif_rxrate_get(struct snd_kcontrol *kcontrol, { struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); - int rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL); + int rate = 0; if (spdif_priv->dpll_locked) - ucontrol->value.integer.value[0] = rate; - else - ucontrol->value.integer.value[0] = 0; + rate = spdif_get_rxclk_rate(spdif_priv, SPDIF_DEFAULT_GAINSEL); + + ucontrol->value.integer.value[0] = rate; return 0; } @@ -1238,12 +1236,12 @@ static int fsl_spdif_probe(struct platform_device *pdev) spin_lock_init(&ctrl->ctl_lock); /* Init tx channel status default value */ - ctrl->ch_status[0] = - IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_5015; + ctrl->ch_status[0] = IEC958_AES0_CON_NOT_COPYRIGHT | + IEC958_AES0_CON_EMPHASIS_5015; ctrl->ch_status[1] = IEC958_AES1_CON_DIGDIGCONV_ID; ctrl->ch_status[2] = 0x00; - ctrl->ch_status[3] = - IEC958_AES3_CON_FS_44100 | IEC958_AES3_CON_CLOCK_1000PPM; + ctrl->ch_status[3] = IEC958_AES3_CON_FS_44100 | + IEC958_AES3_CON_CLOCK_1000PPM; spdif_priv->dpll_locked = false; diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index 16fde4b..b13e30e 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -93,6 +93,8 @@ #define SCR_USRC_SEL_RECV (0x1 << SCR_USRC_SEL_OFFSET) #define SCR_USRC_SEL_CHIP (0x3 << SCR_USRC_SEL_OFFSET) +#define SCR_DMA_xX_EN(tx) (tx ? SCR_DMA_TX_EN : SCR_DMA_RX_EN) + /* SPDIF CDText control */ #define SRCD_CD_USER_OFFSET 1 #define SRCD_CD_USER (1 << SRCD_CD_USER_OFFSET) -- 1.8.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: fsl_spdif: Fix incorrect usage of regmap_read()
by Nicolin Chen 13 May '14

13 May '14

We should not copy the return value into this val since it's supposed to get the value of the register not the success result of regmap_read(). Thus fix it. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index b912d45..9ea2dd4 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -762,7 +762,7 @@ static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol, struct regmap *regmap = spdif_priv->regmap; u32 val; - val = regmap_read(regmap, REG_SPDIF_SIS, &val); + regmap_read(regmap, REG_SPDIF_SIS, &val); ucontrol->value.integer.value[0] = (val & INT_VAL_NOGOOD) != 0; regmap_write(regmap, REG_SPDIF_SIC, INT_VAL_NOGOOD); -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: rt5640: add default case for unexpected ID
by bardliao＠realtek.com 12 May '14

12 May '14

From: Bard Liao <bardliao(a)realtek.com> We may read an unexpected value when detemining which codec is attached. In that case, either a unsupported codec is attached or something wrong with I2C. The driver will not work properly on both cases. So we return an error for that. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b0967df..9e0d48f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2019,6 +2019,10 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5639_specific_dapm_routes, ARRAY_SIZE(rt5639_specific_dapm_routes)); break; + default: + dev_err(codec->dev, + "The driver is for RT5639 RT5640 or RT5642 only\n"); + return -ENODEV; } return 0; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 0/5] ASoC: Trivial bits and cleanups
by Lars-Peter Clausen 12 May '14

12 May '14

Hi, This series contains a few cleanup patches that came up while working on the componentization, but all also make sense on their own. The biggest change is admittedly the removal of the suspend/resume code from the atmel-pcm-pdc driver and may not seem trivial at first. But this code has been fully broken for over 4 years, was always partially broken and was never really needed in the first place. And right now it is blocking other cleanups. Everything else is just removal of unused fields and the consolidation of duplicated fields. Depends on: asoc/topic/component - Lars Lars-Peter Clausen (5): ASoC: Remove card's DAI list ASoC: Remove unused 'list' field form card ASoC: Remove unused num_dai field from CODEC ASoC: atmel-pcm-pdc: Remove broken suspend/resume code ASoC: Remove runtime field from DAI include/sound/soc-dai.h | 2 -- include/sound/soc.h | 4 --- sound/soc/atmel/atmel-pcm-pdc.c | 63 ----------------------------------------- sound/soc/soc-compress.c | 1 - sound/soc/soc-core.c | 7 ----- sound/soc/soc-pcm.c | 1 - 6 files changed, 78 deletions(-) -- 1.8.0

2 10

[alsa-devel] [PATCH v2] ASoC: dapm: Fix SUSPEND -> OFF bias sequence
by Lars-Peter Clausen 12 May '14

12 May '14

Currently when the DAPM context bias level is SUSPEND and the target bias level is OFF dapm_pre_sequence_async() will first transition to PREPARE and dapm_post_sequence_async() will then transition back from PREPARE to STANDBY and then to OFF. This patch makes sure that dapm_pre_sequence_async() only transitions to PREPARE when either going to ON or away from ON. This avoids the extra unnecessary transitions. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Given that SUSPEND -> OFF transitions are rather rare and the code will still work correctly (but inefficent) without this patch, the patch probably does not need to go into stable. Changes since v1: * Change indention --- sound/soc/soc-dapm.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5d54dfc..bfc46a9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1549,8 +1549,11 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) "ASoC: Failed to turn on bias: %d\n", ret); } - /* Prepare for a STADDBY->ON or ON->STANDBY transition */ - if (d->bias_level != d->target_bias_level) { + /* Prepare for a transition to ON or away from ON */ + if ((d->target_bias_level == SND_SOC_BIAS_ON && + d->bias_level != SND_SOC_BIAS_ON) || + (d->target_bias_level != SND_SOC_BIAS_ON && + d->bias_level == SND_SOC_BIAS_ON)) { ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE); if (ret != 0) dev_err(d->dev, -- 1.8.0

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