[PATCH 5/5] ASoC: sdw_utils: cs_amp: Delete bogus and incorrect capture channel fixup

[Richard Fitzgerald <rf@opensource.cirrus.com>]

Delete the asoc_sdw_cs_spk_feedback_rtd_init(). This is not needed now
that the ASoC bug it was working around has been fixed. And it was broken
anyway because it didn't match the way the core SoundWire code mapped
codec channels to frame bitslots.

This code was added to avoid a problem where multiple codec DP outputs
were mapped to the same SoundWire frame bit slot. This would allow a
user to break the SoundWire bus just by enabling mixer outputs using
ALSA controls.

As no production system has used the capture stream, this workaround
was of little consequence and the problem of conflicting DP mappings
was not investigated.

The ASoC bug that enabled too many channels on each codec has now been
fixed. So this workaround can be completely deleted.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
---
 include/sound/soc_sdw_utils.h        |  2 --
 sound/soc/sdw_utils/soc_sdw_cs_amp.c | 46 ----------------------------
 sound/soc/sdw_utils/soc_sdw_utils.c  |  4 ---
 3 files changed, 52 deletions(-)

diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h
index 3c1d177a2f2d..418466438ea1 100644
--- a/include/sound/soc_sdw_utils.h
+++ b/include/sound/soc_sdw_utils.h
@@ -250,8 +250,6 @@ int asoc_sdw_cs_amp_init(struct snd_soc_card *card,
 			 struct snd_soc_dai_link *dai_links,
 			 struct asoc_sdw_codec_info *info,
 			 bool playback);
-int asoc_sdw_cs_spk_feedback_rtd_init(struct snd_soc_pcm_runtime *rtd,
-				      struct snd_soc_dai *dai);
 int asoc_sdw_cs35l56_volume_limit(struct snd_soc_card *card, const char *name_prefix);
 
 /* MAXIM codec support */
diff --git a/sound/soc/sdw_utils/soc_sdw_cs_amp.c b/sound/soc/sdw_utils/soc_sdw_cs_amp.c
index 325ab7230481..6e21ef8f87e2 100644
--- a/sound/soc/sdw_utils/soc_sdw_cs_amp.c
+++ b/sound/soc/sdw_utils/soc_sdw_cs_amp.c
@@ -14,7 +14,6 @@
 #include <sound/soc-dai.h>
 #include <sound/soc_sdw_utils.h>
 
-#define CS_AMP_CHANNELS_PER_AMP	4
 #define CS35L56_SPK_VOLUME_0DB 400 /* 0dB Max */
 
 int asoc_sdw_cs35l56_volume_limit(struct snd_soc_card *card, const char *name_prefix)
@@ -64,51 +63,6 @@ int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai
 }
 EXPORT_SYMBOL_NS(asoc_sdw_cs_spk_rtd_init, "SND_SOC_SDW_UTILS");
 
-int asoc_sdw_cs_spk_feedback_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai)
-{
-	const struct snd_soc_dai_link *dai_link = rtd->dai_link;
-	const struct snd_soc_dai_link_ch_map *ch_map;
-	const struct snd_soc_dai_link_component *codec_dlc;
-	struct snd_soc_dai *codec_dai;
-	u8 ch_slot[8] = {};
-	unsigned int amps_per_bus, ch_per_amp, mask;
-	int i, ret;
-
-	WARN_ON(dai_link->num_cpus > ARRAY_SIZE(ch_slot));
-
-	/*
-	 * CS35L56 has 4 TX channels. When the capture is aggregated the
-	 * same bus slots will be allocated to all the amps on a bus. Only
-	 * one amp on that bus can be transmitting in each slot so divide
-	 * the available 4 slots between all the amps on a bus.
-	 */
-	amps_per_bus = dai_link->num_codecs / dai_link->num_cpus;
-	if ((amps_per_bus == 0) || (amps_per_bus > CS_AMP_CHANNELS_PER_AMP)) {
-		dev_err(rtd->card->dev, "Illegal num_codecs:%u / num_cpus:%u\n",
-			dai_link->num_codecs, dai_link->num_cpus);
-		return -EINVAL;
-	}
-
-	ch_per_amp = CS_AMP_CHANNELS_PER_AMP / amps_per_bus;
-
-	for_each_rtd_ch_maps(rtd, i, ch_map) {
-		codec_dlc = snd_soc_link_to_codec(rtd->dai_link, i);
-		codec_dai = snd_soc_find_dai(codec_dlc);
-		mask = GENMASK(ch_per_amp - 1, 0) << ch_slot[ch_map->cpu];
-
-		ret = snd_soc_dai_set_tdm_slot(codec_dai, 0, mask, 4, 32);
-		if (ret < 0) {
-			dev_err(rtd->card->dev, "Failed to set TDM slot:%d\n", ret);
-			return ret;
-		}
-
-		ch_slot[ch_map->cpu] += ch_per_amp;
-	}
-
-	return 0;
-}
-EXPORT_SYMBOL_NS(asoc_sdw_cs_spk_feedback_rtd_init, "SND_SOC_SDW_UTILS");
-
 int asoc_sdw_cs_amp_init(struct snd_soc_card *card,
 			 struct snd_soc_dai_link *dai_links,
 			 struct asoc_sdw_codec_info *info,
diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c
index 778ba6531434..0d7182bebcdc 100644
--- a/sound/soc/sdw_utils/soc_sdw_utils.c
+++ b/sound/soc/sdw_utils/soc_sdw_utils.c
@@ -820,7 +820,6 @@ struct asoc_sdw_codec_info codec_info_list[] = {
 				.dai_name = "cs35l56-sdw1c",
 				.dai_type = SOC_SDW_DAI_TYPE_AMP,
 				.dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID},
-				.rtd_init = asoc_sdw_cs_spk_feedback_rtd_init,
 			},
 		},
 		.dai_num = 2,
@@ -849,7 +848,6 @@ struct asoc_sdw_codec_info codec_info_list[] = {
 				.dai_name = "cs35l56-sdw1c",
 				.dai_type = SOC_SDW_DAI_TYPE_AMP,
 				.dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID},
-				.rtd_init = asoc_sdw_cs_spk_feedback_rtd_init,
 			},
 		},
 		.dai_num = 2,
@@ -878,7 +876,6 @@ struct asoc_sdw_codec_info codec_info_list[] = {
 				.dai_name = "cs35l56-sdw1c",
 				.dai_type = SOC_SDW_DAI_TYPE_AMP,
 				.dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID},
-				.rtd_init = asoc_sdw_cs_spk_feedback_rtd_init,
 			},
 		},
 		.dai_num = 2,
@@ -907,7 +904,6 @@ struct asoc_sdw_codec_info codec_info_list[] = {
 				.dai_name = "cs35l56-sdw1c",
 				.dai_type = SOC_SDW_DAI_TYPE_AMP,
 				.dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID},
-				.rtd_init = asoc_sdw_cs_spk_feedback_rtd_init,
 			},
 		},
 		.dai_num = 2,
-- 
2.47.3