From faab5a32f4d0784d6bde57963267be0453be3546 Mon Sep 17 00:00:00 2001 From: Karl Beldan Date: Thu, 20 Nov 2008 15:39:27 +0100 Subject: ASoC: ssm2602: Fix priv substreams refs Clean up our record of the active streams in shutdown(), fixing subsequent failures of snd_pcm_hw_constraints_complete after closure of a stream. NOTE: - The ssm2602 allows pairs of non-matching PB/REC rates. - This is a fix for less evil: The logic is flawed (e.g. the slave might startup before the master's rate and sample_bits are set). Signed-off-by: Karl Beldan Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound/soc/codecs/ssm2602.c') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 44ef0dacd564..0e522e718dfc 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -292,9 +292,15 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct ssm2602_priv *ssm2602 = codec->private_data; + struct i2c_client *i2c = codec->control_data; u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; int i = get_coeff(ssm2602->sysclk, params_rate(params)); + if (substream == ssm2602->slave_substream) { + dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n"); + return 0; + } + /*no match is found*/ if (i == ARRAY_SIZE(coeff_div)) return -EINVAL; @@ -330,13 +336,19 @@ static int ssm2602_startup(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct ssm2602_priv *ssm2602 = codec->private_data; + struct i2c_client *i2c = codec->control_data; struct snd_pcm_runtime *master_runtime; /* The DAI has shared clocks so if we already have a playback or * capture going then constrain this substream to match it. + * TODO: the ssm2602 allows pairs of non-matching PB/REC rates */ if (ssm2602->master_substream) { master_runtime = ssm2602->master_substream->runtime; + dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", + master_runtime->sample_bits, + master_runtime->rate); + snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, master_runtime->rate, @@ -370,9 +382,15 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; /* deactivate */ if (!codec->active) ssm2602_write(codec, SSM2602_ACTIVE, 0); + + if (ssm2602->master_substream == substream) + ssm2602->master_substream = ssm2602->slave_substream; + + ssm2602->slave_substream = NULL; } static int ssm2602_mute(struct snd_soc_dai *dai, int mute) -- cgit v1.2.3 From 5de27b6cc0a8a1d27158ec9047cb5981745edfc0 Mon Sep 17 00:00:00 2001 From: Karl Beldan Date: Thu, 20 Nov 2008 15:39:31 +0100 Subject: ASoC: ssm2602: Update supported stream formats Signed-off-by: Karl Beldan Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs/ssm2602.c') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 0e522e718dfc..56dc1c9c7c52 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -514,6 +514,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ SNDRV_PCM_RATE_96000) +#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + struct snd_soc_dai ssm2602_dai = { .name = "SSM2602", .playback = { @@ -521,13 +524,13 @@ struct snd_soc_dai ssm2602_dai = { .channels_min = 2, .channels_max = 2, .rates = SSM2602_RATES, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .formats = SSM2602_FORMATS,}, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SSM2602_RATES, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .formats = SSM2602_FORMATS,}, .ops = { .startup = ssm2602_startup, .prepare = ssm2602_pcm_prepare, -- cgit v1.2.3 From dee89c4d94433520e4e3977ae203d4cfbfe385fb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Nov 2008 22:11:38 +0000 Subject: ASoC: Merge snd_soc_ops into snd_soc_dai_ops Liam Girdwood's ASoC v2 work avoids having two different ops structures for DAIs by merging the members of struct snd_soc_ops into struct snd_soc_dai_ops, allowing per DAI configuration for everything. Backport this change. This paves the way for future work allowing any combination of DAIs to be connected rather than having fixed purpose CODEC and CPU DAIs and only allowing CODEC<->CPU interconnections. Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs/ssm2602.c') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 56dc1c9c7c52..0c5884ea1b00 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -285,7 +285,8 @@ static inline int get_coeff(int mclk, int rate) } static int ssm2602_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { u16 srate; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -330,7 +331,8 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ssm2602_startup(struct snd_pcm_substream *substream) +static int ssm2602_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -366,7 +368,8 @@ static int ssm2602_startup(struct snd_pcm_substream *substream) return 0; } -static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream) +static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -377,7 +380,8 @@ static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream) return 0; } -static void ssm2602_shutdown(struct snd_pcm_substream *substream) +static void ssm2602_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -536,8 +540,6 @@ struct snd_soc_dai ssm2602_dai = { .prepare = ssm2602_pcm_prepare, .hw_params = ssm2602_hw_params, .shutdown = ssm2602_shutdown, - }, - .dai_ops = { .digital_mute = ssm2602_mute, .set_sysclk = ssm2602_set_dai_sysclk, .set_fmt = ssm2602_set_dai_fmt, -- cgit v1.2.3 From 968a6025aa9f909d487988efb542217a126023a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 28 Nov 2008 11:49:07 +0000 Subject: ASoC: Rename snd_soc_register_card() to snd_soc_init_card() Currently ASoC card initialisation is completed by a function called snd_soc_register_card(). As part of the work to allow independant registration of cards, codecs and machines in ASoC v2 a new function of the same name has been added so rename the existing function to facilitate the merge of v2. Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs/ssm2602.c') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 0c5884ea1b00..973844973fe1 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -624,7 +624,7 @@ static int ssm2602_init(struct snd_soc_device *socdev) ssm2602_add_controls(codec); ssm2602_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { pr_err("ssm2602: failed to register card\n"); goto card_err; -- cgit v1.2.3 From 64089b84abfe2f26a864ebd968429302dcb071de Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 8 Dec 2008 19:17:58 +0000 Subject: ASoC: Register non-AC97 codec DAIs Currently this is done at module probe time since ASoC ties in codec device probe to the instantiation of the entire ASoC device. Subsequent patches will refactor the codec drivers to handle probing separately. Note that the core does not yet use this information. AC97 is special since the codec is controlled over the AC97 link but we want to give the machine driver a chance to set up the system before trying to instantiate since it may need to do configuration before the AC97 link will operate Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound/soc/codecs/ssm2602.c') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 973844973fe1..77fdcb4b9a1b 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -793,6 +793,18 @@ struct snd_soc_codec_device soc_codec_dev_ssm2602 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602); +static int __devinit ssm2602_modinit(void) +{ + return snd_soc_register_dai(&ssm2602_dai); +} +module_init(ssm2602_modinit); + +static void __exit ssm2602_exit(void) +{ + snd_soc_unregister_dai(&ssm2602_dai); +} +module_exit(ssm2602_exit); + MODULE_DESCRIPTION("ASoC ssm2602 driver"); MODULE_AUTHOR("Cliff Cai"); MODULE_LICENSE("GPL"); -- cgit v1.2.3 From c9b3a40ff2b3dea9914e36965a17c802650bb603 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 10 Dec 2008 07:47:22 +0100 Subject: ALSA: ASoC - Fix wrong section types The module init entries should be __init instead of __devinit. Signed-off-by: Takashi Iwai --- sound/soc/codecs/ssm2602.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs/ssm2602.c') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 77fdcb4b9a1b..2325aefea411 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -793,7 +793,7 @@ struct snd_soc_codec_device soc_codec_dev_ssm2602 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602); -static int __devinit ssm2602_modinit(void) +static int __init ssm2602_modinit(void) { return snd_soc_register_dai(&ssm2602_dai); } -- cgit v1.2.3 From c69134858722977a82f58cae88e7ffdb28e1e858 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 22 Dec 2008 10:57:33 +0200 Subject: ASoC: Fix DSP formats in SSM2602 audio codec Thanks to Troy Kisky for noticing. - DSP_A format has 1-bit data delay which corresponds to SSM6202 submode 2 - DSP_B has 0-bit data delay which corresponds to submode 1 - Currently driver sets them opposite so swap the submode setting Signed-off-by: Jarkko Nikula Cc: Cliff Cai Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs/ssm2602.c') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 2325aefea411..cac373616768 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -454,10 +454,10 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0001; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x0013; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x0003; break; default: return -EINVAL; -- cgit v1.2.3