There is no need for implementations of DAI set_channel_map() to modify contents of passed arrays with actual channel mapping. Additionally, the caller keeps full ownership of the array.

Constify these pointer arguments so the code will be safer and easier to read (documenting the caller's ownership).

Acked-by: Charles Keepax ckeepax@opensource.cirrus.com Signed-off-by: Krzysztof Kozlowski krzysztof.kozlowski@linaro.org ---

Changes in v2/v3/v4: 1. None --- include/sound/cs35l41.h | 4 ++-- include/sound/soc-dai.h | 8 ++++---- sound/soc/codecs/adau7118.c | 6 ++++-- sound/soc/codecs/cs35l41-lib.c | 4 ++-- sound/soc/codecs/cs35l41.c | 3 ++- sound/soc/codecs/max98504.c | 6 ++++-- sound/soc/codecs/wcd9335.c | 6 ++++-- sound/soc/codecs/wcd934x.c | 6 ++++-- sound/soc/qcom/qdsp6/q6afe-dai.c | 16 ++++++++++------ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 6 ++++-- sound/soc/soc-dai.c | 4 ++-- 11 files changed, 42 insertions(+), 27 deletions(-)

diff --git a/include/sound/cs35l41.h b/include/sound/cs35l41.h index bb70782d15d0..43c6a9ef8d9f 100644 --- a/include/sound/cs35l41.h +++ b/include/sound/cs35l41.h @@ -896,8 +896,8 @@ int cs35l41_test_key_lock(struct device *dev, struct regmap *regmap); int cs35l41_otp_unpack(struct device *dev, struct regmap *regmap); int cs35l41_register_errata_patch(struct device *dev, struct regmap *reg, unsigned int reg_revid); int cs35l41_set_channels(struct device *dev, struct regmap *reg, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot); + unsigned int tx_num, const unsigned int *tx_slot, + unsigned int rx_num, const unsigned int *rx_slot); int cs35l41_gpio_config(struct regmap *regmap, struct cs35l41_hw_cfg *hw_cfg); void cs35l41_configure_cs_dsp(struct device *dev, struct regmap *reg, struct cs_dsp *dsp); int cs35l41_set_cspl_mbox_cmd(struct device *dev, struct regmap *regmap, diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index adcd8719d343..15ef268c9845 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -188,8 +188,8 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);

int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot); + unsigned int tx_num, const unsigned int *tx_slot, + unsigned int rx_num, const unsigned int *rx_slot);

int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);

@@ -305,8 +305,8 @@ struct snd_soc_dai_ops { unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int (*set_channel_map)(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot); + unsigned int tx_num, const unsigned int *tx_slot, + unsigned int rx_num, const unsigned int *rx_slot); int (*get_channel_map)(struct snd_soc_dai *dai, unsigned int *tx_num, unsigned int *tx_slot, unsigned int *rx_num, unsigned int *rx_slot); diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c index a663d37e5776..abc4764697a5 100644 --- a/sound/soc/codecs/adau7118.c +++ b/sound/soc/codecs/adau7118.c @@ -121,8 +121,10 @@ static const struct snd_soc_dapm_widget adau7118_widgets[] = { };

static int adau7118_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) + unsigned int tx_num, + const unsigned int *tx_slot, + unsigned int rx_num, + const unsigned int *rx_slot) { struct adau7118_data *st = snd_soc_component_get_drvdata(dai->component); diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c index e9993a39f7d0..1702f26049d3 100644 --- a/sound/soc/codecs/cs35l41-lib.c +++ b/sound/soc/codecs/cs35l41-lib.c @@ -936,8 +936,8 @@ int cs35l41_register_errata_patch(struct device *dev, struct regmap *reg, unsign EXPORT_SYMBOL_GPL(cs35l41_register_errata_patch);

int cs35l41_set_channels(struct device *dev, struct regmap *reg, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) + unsigned int tx_num, const unsigned int *tx_slot, + unsigned int rx_num, const unsigned int *rx_slot) { unsigned int val, mask; int i; diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index cb25c33cc9b9..1688c2c688f0 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -673,7 +673,8 @@ static const struct snd_soc_dapm_route cs35l41_audio_map[] = { };

static int cs35l41_set_channel_map(struct snd_soc_dai *dai, unsigned int tx_n, - unsigned int *tx_slot, unsigned int rx_n, unsigned int *rx_slot) + const unsigned int *tx_slot, + unsigned int rx_n, const unsigned int *rx_slot) { struct cs35l41_private *cs35l41 = snd_soc_component_get_drvdata(dai->component);

diff --git a/sound/soc/codecs/max98504.c b/sound/soc/codecs/max98504.c index 93412b966b33..6b6a7ece4cec 100644 --- a/sound/soc/codecs/max98504.c +++ b/sound/soc/codecs/max98504.c @@ -220,8 +220,10 @@ static int max98504_set_tdm_slot(struct snd_soc_dai *dai, return 0; } static int max98504_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) + unsigned int tx_num, + const unsigned int *tx_slot, + unsigned int rx_num, + const unsigned int *rx_slot) { struct max98504_priv *max98504 = snd_soc_dai_get_drvdata(dai); struct regmap *map = max98504->regmap; diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index deb15b95992d..42a99978fe5a 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1983,8 +1983,10 @@ static int wcd9335_trigger(struct snd_pcm_substream *substream, int cmd, }

static int wcd9335_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) + unsigned int tx_num, + const unsigned int *tx_slot, + unsigned int rx_num, + const unsigned int *rx_slot) { struct wcd9335_codec *wcd; int i; diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index de870c7819ca..fcad2c9fba55 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -1923,8 +1923,10 @@ static int wcd934x_trigger(struct snd_pcm_substream *substream, int cmd, }

static int wcd934x_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) + unsigned int tx_num, + const unsigned int *tx_slot, + unsigned int rx_num, + const unsigned int *rx_slot) { struct wcd934x_codec *wcd; int i; diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index a9c4f896a7df..7d9628cda875 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -172,8 +172,8 @@ static int q6tdm_set_tdm_slot(struct snd_soc_dai *dai, }

static int q6tdm_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) + unsigned int tx_num, const unsigned int *tx_slot, + unsigned int rx_num, const unsigned int *rx_slot) {

struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); @@ -250,8 +250,10 @@ static int q6tdm_hw_params(struct snd_pcm_substream *substream, }

static int q6dma_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_ch_mask, - unsigned int rx_num, unsigned int *rx_ch_mask) + unsigned int tx_num, + const unsigned int *tx_ch_mask, + unsigned int rx_num, + const unsigned int *rx_ch_mask) {

struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); @@ -407,8 +409,10 @@ static int q6afe_dai_prepare(struct snd_pcm_substream *substream, }

static int q6slim_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) + unsigned int tx_num, + const unsigned int *tx_slot, + unsigned int rx_num, + const unsigned int *rx_slot) { struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); struct q6afe_port_config *pcfg = &dai_data->port_config[dai->id]; diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c index 68a38f63a2db..6bfbb52345e1 100644 --- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c +++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c @@ -25,8 +25,10 @@ struct q6apm_lpass_dai_data { };

static int q6dma_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_ch_mask, - unsigned int rx_num, unsigned int *rx_ch_mask) + unsigned int tx_num, + const unsigned int *tx_ch_mask, + unsigned int rx_num, + const unsigned int *rx_ch_mask) {

struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev); diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index fefe394dce72..03afd5efb24c 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -304,8 +304,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); * configure the relationship between channel number and TDM slot number. */ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) + unsigned int tx_num, const unsigned int *tx_slot, + unsigned int rx_num, const unsigned int *rx_slot) { int ret = -ENOTSUPP;

When constructing packets to DSP, the Audioreach code uses 'struct audioreach_module_config' to configure parameters like number of channels, bitrate, sample rate etc, but uses defaults for the channel mapping.

Rework this code to copy the channel mapping from 'struct audioreach_module_config', instead of using the default. This requires all callers to fill that structure: add missing initialization of channel mapping.

Entire patch makes code more logical and easier to follow: 1. q6apm-dai and q6apm-lpass-dais code which allocates 'struct audioreach_module_config' initializes it fully, so fills both the number of channels and the channel mapping. 2. Audioreach code, which uses 'struct audioreach_module_config' when constructing packets, copies entire contents of passed config, not only pieces of it.

Signed-off-by: Krzysztof Kozlowski krzysztof.kozlowski@linaro.org

---

Changes in v4: 1. Rename audioreach_set_channel_mapping -> audioreach_set_default_channel_mapping

Changes in v2: 1. Extend commit msg, rationale. 2. Rework to new approach, but most of the code stays. 3. Export audioreach_set_channel_mapping() (needed by Q6APM DAIS and LPASS DAIS) 4. Correct channel mapping also in audioreach_mfc_set_media_format(), because MFC DAI is now part of backend. 5. Do not adjust dynamic DAIs (drop audioreach_dai_load()). --- sound/soc/qcom/qdsp6/audioreach.c | 30 +++++++----------------------- sound/soc/qcom/qdsp6/audioreach.h | 1 + sound/soc/qcom/qdsp6/q6apm-dai.c | 3 +++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 5 ++++- 4 files changed, 15 insertions(+), 24 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c index 5291deac0a0b..4ebaaf736fb9 100644 --- a/sound/soc/qcom/qdsp6/audioreach.c +++ b/sound/soc/qcom/qdsp6/audioreach.c @@ -267,7 +267,7 @@ void *audioreach_alloc_apm_cmd_pkt(int pkt_size, uint32_t opcode, uint32_t token } EXPORT_SYMBOL_GPL(audioreach_alloc_apm_cmd_pkt);

-static void audioreach_set_channel_mapping(u8 *ch_map, int num_channels) +void audioreach_set_default_channel_mapping(u8 *ch_map, int num_channels) { if (num_channels == 1) { ch_map[0] = PCM_CHANNEL_FL; @@ -281,6 +281,7 @@ static void audioreach_set_channel_mapping(u8 *ch_map, int num_channels) ch_map[3] = PCM_CHANNEL_RS; } } +EXPORT_SYMBOL_GPL(audioreach_set_default_channel_mapping);

static void apm_populate_container_config(struct apm_container_obj *cfg, struct audioreach_container *cont) @@ -819,7 +820,7 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph, uint32_t num_channels = cfg->num_channels; int payload_size; struct gpr_pkt *pkt; - int rc; + int rc, i; void *p;

payload_size = APM_MFC_CFG_PSIZE(media_format, num_channels) + @@ -842,18 +843,8 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph, media_format->sample_rate = cfg->sample_rate; media_format->bit_width = cfg->bit_width; media_format->num_channels = cfg->num_channels; - - if (num_channels == 1) { - media_format->channel_mapping[0] = PCM_CHANNEL_FL; - } else if (num_channels == 2) { - media_format->channel_mapping[0] = PCM_CHANNEL_FL; - media_format->channel_mapping[1] = PCM_CHANNEL_FR; - } else if (num_channels == 4) { - media_format->channel_mapping[0] = PCM_CHANNEL_FL; - media_format->channel_mapping[1] = PCM_CHANNEL_FR; - media_format->channel_mapping[2] = PCM_CHANNEL_LS; - media_format->channel_mapping[3] = PCM_CHANNEL_RS; - } + for (i = 0; i < num_channels; i++) + media_format->channel_mapping[i] = cfg->channel_map[i];

rc = q6apm_send_cmd_sync(graph->apm, pkt, 0);

@@ -883,9 +874,6 @@ static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr, mp3_cfg->q_factor = mcfg->bit_width - 1; mp3_cfg->endianness = PCM_LITTLE_ENDIAN; mp3_cfg->num_channels = mcfg->num_channels; - - audioreach_set_channel_mapping(mp3_cfg->channel_mapping, - mcfg->num_channels); break; case SND_AUDIOCODEC_AAC: media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED; @@ -1104,9 +1092,7 @@ static int audioreach_pcm_set_media_format(struct q6apm_graph *graph, media_cfg->num_channels = mcfg->num_channels; media_cfg->q_factor = mcfg->bit_width - 1; media_cfg->bits_per_sample = mcfg->bit_width; - - audioreach_set_channel_mapping(media_cfg->channel_mapping, - num_channels); + memcpy(media_cfg->channel_mapping, mcfg->channel_map, mcfg->num_channels);

rc = q6apm_send_cmd_sync(graph->apm, pkt, 0);

@@ -1163,9 +1149,7 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph, cfg->q_factor = mcfg->bit_width - 1; cfg->endianness = PCM_LITTLE_ENDIAN; cfg->num_channels = mcfg->num_channels; - - audioreach_set_channel_mapping(cfg->channel_mapping, - num_channels); + memcpy(cfg->channel_mapping, mcfg->channel_map, mcfg->num_channels); } else { rc = audioreach_set_compr_media_format(header, p, mcfg); if (rc) { diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h index eb9306280988..61a69df4f50f 100644 --- a/sound/soc/qcom/qdsp6/audioreach.h +++ b/sound/soc/qcom/qdsp6/audioreach.h @@ -766,6 +766,7 @@ struct audioreach_module_config { /* Packet Allocation routines */ void *audioreach_alloc_apm_cmd_pkt(int pkt_size, uint32_t opcode, uint32_t token); +void audioreach_set_default_channel_mapping(u8 *ch_map, int num_channels); void *audioreach_alloc_cmd_pkt(int payload_size, uint32_t opcode, uint32_t token, uint32_t src_port, uint32_t dest_port); diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 00bbd291be5c..3361c10d69a4 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -243,6 +243,7 @@ static int q6apm_dai_prepare(struct snd_soc_component *component, cfg.num_channels = runtime->channels; cfg.bit_width = prtd->bits_per_sample; cfg.fmt = SND_AUDIOCODEC_PCM; + audioreach_set_default_channel_mapping(cfg.channel_map, runtime->channels);

if (prtd->state) { /* clear the previous setup if any */ @@ -669,6 +670,8 @@ static int q6apm_dai_compr_set_params(struct snd_soc_component *component, cfg.num_channels = 2; cfg.bit_width = prtd->bits_per_sample; cfg.fmt = codec->id; + audioreach_set_default_channel_mapping(cfg.channel_map, + cfg.num_channels); memcpy(&cfg.codec, codec, sizeof(*codec));

ret = q6apm_graph_media_format_shmem(prtd->graph, &cfg); diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c index a4ad1d0e6abd..ba28ec9dff86 100644 --- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c +++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c @@ -106,6 +106,7 @@ static int q6hdmi_hw_params(struct snd_pcm_substream *substream, cfg->bit_width = params_width(params); cfg->sample_rate = params_rate(params); cfg->num_channels = channels; + audioreach_set_default_channel_mapping(cfg->channel_map, channels);

switch (dai->id) { case DISPLAY_PORT_RX_0: @@ -130,10 +131,12 @@ static int q6dma_hw_params(struct snd_pcm_substream *substream, { struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev); struct audioreach_module_config *cfg = &dai_data->module_config[dai->id]; + int channels = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max;

cfg->bit_width = params_width(params); cfg->sample_rate = params_rate(params); - cfg->num_channels = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max; + cfg->num_channels = channels; + audioreach_set_default_channel_mapping(cfg->channel_map, channels);

return 0; }