From: Connor McAdams <conmanx360@gmail.com>
To: unlisted-recipients:; (no To-header on input)
Cc: conmanx360@gmail.com, Jaroslav Kysela <perex@perex.cz>,
Takashi Iwai <tiwai@suse.com>,
alsa-devel@alsa-project.org, linux-kernel@vger.kernel.org
Subject: [PATCH 5/5] ALSA: hda/ca0132 - Remove now unnecessary DSP setup functions.
Date: Tue, 8 Dec 2020 16:25:45 -0500 [thread overview]
Message-ID: <20201208212546.428392-6-conmanx360@gmail.com> (raw)
In-Reply-To: <20201208212546.428392-1-conmanx360@gmail.com>
Now that the DSP's audio configuration is understood, remove previous
hacky methods of trying to properly configure it.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
---
sound/pci/hda/patch_ca0132.c | 105 -----------------------------------
1 file changed, 105 deletions(-)
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 12fee1146fc2..b7d36c9b28b5 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -2436,13 +2436,6 @@ static int dspio_set_uint_param(struct hda_codec *codec, int mod_id,
sizeof(unsigned int));
}
-static int dspio_set_uint_param_no_source(struct hda_codec *codec, int mod_id,
- int req, const unsigned int data)
-{
- return dspio_set_param(codec, mod_id, 0x00, req, &data,
- sizeof(unsigned int));
-}
-
/*
* Allocate a DSP DMA channel via an SCP message
*/
@@ -7789,24 +7782,6 @@ static void ca0132_alt_init_speaker_tuning(struct hda_codec *codec)
SPEAKER_TUNING_FRONT_LEFT_DELAY + i, values[i]);
}
-/*
- * Creates a dummy stream to bind the output to. This seems to have to be done
- * after changing the main outputs source and destination streams.
- */
-static void ca0132_alt_create_dummy_stream(struct hda_codec *codec)
-{
- struct ca0132_spec *spec = codec->spec;
- unsigned int stream_format;
-
- stream_format = snd_hdac_calc_stream_format(48000, 2,
- SNDRV_PCM_FORMAT_S32_LE, 32, 0);
-
- snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
- 0, stream_format);
-
- snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
-}
-
/*
* Initialize mic for non-chromebook ca0132 implementations.
*/
@@ -7848,9 +7823,6 @@ static void sbz_connect_streams(struct hda_codec *codec)
codec_dbg(codec, "Connect Streams entered, mutex locked and loaded.\n");
- chipio_set_stream_channels(codec, 0x0C, 6);
- chipio_set_stream_control(codec, 0x0C, 1);
-
/* This value is 0x43 for 96khz, and 0x83 for 192khz. */
chipio_write_no_mutex(codec, 0x18a020, 0x00000043);
@@ -7898,9 +7870,6 @@ static void sbz_chipio_startup_data(struct hda_codec *codec)
break;
}
- chipio_set_stream_channels(codec, 0x0c, 6);
- chipio_set_stream_control(codec, 0x0c, 1);
-
if (dsp_out_remap_data)
chipio_remap_stream(codec, dsp_out_remap_data);
@@ -7908,57 +7877,6 @@ static void sbz_chipio_startup_data(struct hda_codec *codec)
mutex_unlock(&spec->chipio_mutex);
}
-/*
- * Custom DSP SCP commands where the src value is 0x00 instead of 0x20. This is
- * done after the DSP is loaded.
- */
-static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec)
-{
- struct ca0132_spec *spec = codec->spec;
- unsigned int tmp, i;
-
- /*
- * Gotta run these twice, or else mic works inconsistently. Not clear
- * why this is, but multiple tests have confirmed it.
- */
- for (i = 0; i < 2; i++) {
- switch (ca0132_quirk(spec)) {
- case QUIRK_SBZ:
- case QUIRK_AE5:
- case QUIRK_AE7:
- tmp = 0x00000003;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
- tmp = 0x00000000;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
- tmp = 0x00000001;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
- tmp = 0x00000004;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
- tmp = 0x00000005;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
- tmp = 0x00000000;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
- break;
- case QUIRK_R3D:
- case QUIRK_R3DI:
- tmp = 0x00000000;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
- tmp = 0x00000001;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
- tmp = 0x00000004;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
- tmp = 0x00000005;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
- tmp = 0x00000000;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
- break;
- default:
- break;
- }
- msleep(100);
- }
-}
-
static void ca0132_alt_dsp_initial_mic_setup(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
@@ -8076,9 +7994,6 @@ static void ae5_post_dsp_stream_setup(struct hda_codec *codec)
chipio_set_conn_rate_no_mutex(codec, 0x70, SR_96_000);
- chipio_set_stream_channels(codec, 0x0C, 6);
- chipio_set_stream_control(codec, 0x0C, 1);
-
chipio_set_stream_source_dest(codec, 0x5, 0x43, 0x0);
chipio_set_stream_source_dest(codec, 0x18, 0x9, 0xd0);
@@ -8136,9 +8051,6 @@ static void ae7_post_dsp_setup_ports(struct hda_codec *codec)
mutex_lock(&spec->chipio_mutex);
- chipio_set_stream_channels(codec, 0x0c, 6);
- chipio_set_stream_control(codec, 0x0c, 1);
-
/* Seems to share the same port remapping as the SBZ. */
chipio_remap_stream(codec, &stream_remap_data[1]);
@@ -8164,8 +8076,6 @@ static void ae7_post_dsp_asi_stream_setup(struct hda_codec *codec)
ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00);
chipio_set_conn_rate_no_mutex(codec, 0x70, SR_96_000);
- chipio_set_stream_channels(codec, 0x0c, 6);
- chipio_set_stream_control(codec, 0x0c, 1);
chipio_set_stream_source_dest(codec, 0x05, 0x43, 0x00);
chipio_set_stream_source_dest(codec, 0x18, 0x09, 0xd0);
@@ -8372,7 +8282,6 @@ static void r3d_setup_defaults(struct hda_codec *codec)
if (spec->dsp_state != DSP_DOWNLOADED)
return;
- ca0132_alt_dsp_scp_startup(codec);
ca0132_alt_init_analog_mics(codec);
ca0132_alt_start_dsp_audio_streams(codec);
@@ -8423,15 +8332,11 @@ static void sbz_setup_defaults(struct hda_codec *codec)
if (spec->dsp_state != DSP_DOWNLOADED)
return;
- ca0132_alt_dsp_scp_startup(codec);
ca0132_alt_init_analog_mics(codec);
ca0132_alt_start_dsp_audio_streams(codec);
sbz_connect_streams(codec);
sbz_chipio_startup_data(codec);
- chipio_set_stream_control(codec, 0x03, 1);
- chipio_set_stream_control(codec, 0x04, 1);
-
/*
* Sets internal input loopback to off, used to have a switch to
* enable input loopback, but turned out to be way too buggy.
@@ -8466,8 +8371,6 @@ static void sbz_setup_defaults(struct hda_codec *codec)
}
ca0132_alt_init_speaker_tuning(codec);
-
- ca0132_alt_create_dummy_stream(codec);
}
/*
@@ -8483,11 +8386,8 @@ static void ae5_setup_defaults(struct hda_codec *codec)
if (spec->dsp_state != DSP_DOWNLOADED)
return;
- ca0132_alt_dsp_scp_startup(codec);
ca0132_alt_init_analog_mics(codec);
ca0132_alt_start_dsp_audio_streams(codec);
- chipio_set_stream_control(codec, 0x03, 1);
- chipio_set_stream_control(codec, 0x04, 1);
/* New, unknown SCP req's */
tmp = FLOAT_ZERO;
@@ -8536,8 +8436,6 @@ static void ae5_setup_defaults(struct hda_codec *codec)
}
ca0132_alt_init_speaker_tuning(codec);
-
- ca0132_alt_create_dummy_stream(codec);
}
/*
@@ -8553,7 +8451,6 @@ static void ae7_setup_defaults(struct hda_codec *codec)
if (spec->dsp_state != DSP_DOWNLOADED)
return;
- ca0132_alt_dsp_scp_startup(codec);
ca0132_alt_init_analog_mics(codec);
ca0132_alt_start_dsp_audio_streams(codec);
ae7_post_dsp_setup_ports(codec);
@@ -8622,8 +8519,6 @@ static void ae7_setup_defaults(struct hda_codec *codec)
}
ca0132_alt_init_speaker_tuning(codec);
-
- ca0132_alt_create_dummy_stream(codec);
}
/*
--
2.25.1
WARNING: multiple messages have this Message-ID (diff)
From: Connor McAdams <conmanx360@gmail.com>
Cc: linux-kernel@vger.kernel.org, conmanx360@gmail.com,
Takashi Iwai <tiwai@suse.com>,
alsa-devel@alsa-project.org
Subject: [PATCH 5/5] ALSA: hda/ca0132 - Remove now unnecessary DSP setup functions.
Date: Tue, 8 Dec 2020 16:25:45 -0500 [thread overview]
Message-ID: <20201208212546.428392-6-conmanx360@gmail.com> (raw)
In-Reply-To: <20201208212546.428392-1-conmanx360@gmail.com>
Now that the DSP's audio configuration is understood, remove previous
hacky methods of trying to properly configure it.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
---
sound/pci/hda/patch_ca0132.c | 105 -----------------------------------
1 file changed, 105 deletions(-)
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 12fee1146fc2..b7d36c9b28b5 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -2436,13 +2436,6 @@ static int dspio_set_uint_param(struct hda_codec *codec, int mod_id,
sizeof(unsigned int));
}
-static int dspio_set_uint_param_no_source(struct hda_codec *codec, int mod_id,
- int req, const unsigned int data)
-{
- return dspio_set_param(codec, mod_id, 0x00, req, &data,
- sizeof(unsigned int));
-}
-
/*
* Allocate a DSP DMA channel via an SCP message
*/
@@ -7789,24 +7782,6 @@ static void ca0132_alt_init_speaker_tuning(struct hda_codec *codec)
SPEAKER_TUNING_FRONT_LEFT_DELAY + i, values[i]);
}
-/*
- * Creates a dummy stream to bind the output to. This seems to have to be done
- * after changing the main outputs source and destination streams.
- */
-static void ca0132_alt_create_dummy_stream(struct hda_codec *codec)
-{
- struct ca0132_spec *spec = codec->spec;
- unsigned int stream_format;
-
- stream_format = snd_hdac_calc_stream_format(48000, 2,
- SNDRV_PCM_FORMAT_S32_LE, 32, 0);
-
- snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
- 0, stream_format);
-
- snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
-}
-
/*
* Initialize mic for non-chromebook ca0132 implementations.
*/
@@ -7848,9 +7823,6 @@ static void sbz_connect_streams(struct hda_codec *codec)
codec_dbg(codec, "Connect Streams entered, mutex locked and loaded.\n");
- chipio_set_stream_channels(codec, 0x0C, 6);
- chipio_set_stream_control(codec, 0x0C, 1);
-
/* This value is 0x43 for 96khz, and 0x83 for 192khz. */
chipio_write_no_mutex(codec, 0x18a020, 0x00000043);
@@ -7898,9 +7870,6 @@ static void sbz_chipio_startup_data(struct hda_codec *codec)
break;
}
- chipio_set_stream_channels(codec, 0x0c, 6);
- chipio_set_stream_control(codec, 0x0c, 1);
-
if (dsp_out_remap_data)
chipio_remap_stream(codec, dsp_out_remap_data);
@@ -7908,57 +7877,6 @@ static void sbz_chipio_startup_data(struct hda_codec *codec)
mutex_unlock(&spec->chipio_mutex);
}
-/*
- * Custom DSP SCP commands where the src value is 0x00 instead of 0x20. This is
- * done after the DSP is loaded.
- */
-static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec)
-{
- struct ca0132_spec *spec = codec->spec;
- unsigned int tmp, i;
-
- /*
- * Gotta run these twice, or else mic works inconsistently. Not clear
- * why this is, but multiple tests have confirmed it.
- */
- for (i = 0; i < 2; i++) {
- switch (ca0132_quirk(spec)) {
- case QUIRK_SBZ:
- case QUIRK_AE5:
- case QUIRK_AE7:
- tmp = 0x00000003;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
- tmp = 0x00000000;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
- tmp = 0x00000001;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
- tmp = 0x00000004;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
- tmp = 0x00000005;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
- tmp = 0x00000000;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
- break;
- case QUIRK_R3D:
- case QUIRK_R3DI:
- tmp = 0x00000000;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
- tmp = 0x00000001;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
- tmp = 0x00000004;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
- tmp = 0x00000005;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
- tmp = 0x00000000;
- dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
- break;
- default:
- break;
- }
- msleep(100);
- }
-}
-
static void ca0132_alt_dsp_initial_mic_setup(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
@@ -8076,9 +7994,6 @@ static void ae5_post_dsp_stream_setup(struct hda_codec *codec)
chipio_set_conn_rate_no_mutex(codec, 0x70, SR_96_000);
- chipio_set_stream_channels(codec, 0x0C, 6);
- chipio_set_stream_control(codec, 0x0C, 1);
-
chipio_set_stream_source_dest(codec, 0x5, 0x43, 0x0);
chipio_set_stream_source_dest(codec, 0x18, 0x9, 0xd0);
@@ -8136,9 +8051,6 @@ static void ae7_post_dsp_setup_ports(struct hda_codec *codec)
mutex_lock(&spec->chipio_mutex);
- chipio_set_stream_channels(codec, 0x0c, 6);
- chipio_set_stream_control(codec, 0x0c, 1);
-
/* Seems to share the same port remapping as the SBZ. */
chipio_remap_stream(codec, &stream_remap_data[1]);
@@ -8164,8 +8076,6 @@ static void ae7_post_dsp_asi_stream_setup(struct hda_codec *codec)
ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00);
chipio_set_conn_rate_no_mutex(codec, 0x70, SR_96_000);
- chipio_set_stream_channels(codec, 0x0c, 6);
- chipio_set_stream_control(codec, 0x0c, 1);
chipio_set_stream_source_dest(codec, 0x05, 0x43, 0x00);
chipio_set_stream_source_dest(codec, 0x18, 0x09, 0xd0);
@@ -8372,7 +8282,6 @@ static void r3d_setup_defaults(struct hda_codec *codec)
if (spec->dsp_state != DSP_DOWNLOADED)
return;
- ca0132_alt_dsp_scp_startup(codec);
ca0132_alt_init_analog_mics(codec);
ca0132_alt_start_dsp_audio_streams(codec);
@@ -8423,15 +8332,11 @@ static void sbz_setup_defaults(struct hda_codec *codec)
if (spec->dsp_state != DSP_DOWNLOADED)
return;
- ca0132_alt_dsp_scp_startup(codec);
ca0132_alt_init_analog_mics(codec);
ca0132_alt_start_dsp_audio_streams(codec);
sbz_connect_streams(codec);
sbz_chipio_startup_data(codec);
- chipio_set_stream_control(codec, 0x03, 1);
- chipio_set_stream_control(codec, 0x04, 1);
-
/*
* Sets internal input loopback to off, used to have a switch to
* enable input loopback, but turned out to be way too buggy.
@@ -8466,8 +8371,6 @@ static void sbz_setup_defaults(struct hda_codec *codec)
}
ca0132_alt_init_speaker_tuning(codec);
-
- ca0132_alt_create_dummy_stream(codec);
}
/*
@@ -8483,11 +8386,8 @@ static void ae5_setup_defaults(struct hda_codec *codec)
if (spec->dsp_state != DSP_DOWNLOADED)
return;
- ca0132_alt_dsp_scp_startup(codec);
ca0132_alt_init_analog_mics(codec);
ca0132_alt_start_dsp_audio_streams(codec);
- chipio_set_stream_control(codec, 0x03, 1);
- chipio_set_stream_control(codec, 0x04, 1);
/* New, unknown SCP req's */
tmp = FLOAT_ZERO;
@@ -8536,8 +8436,6 @@ static void ae5_setup_defaults(struct hda_codec *codec)
}
ca0132_alt_init_speaker_tuning(codec);
-
- ca0132_alt_create_dummy_stream(codec);
}
/*
@@ -8553,7 +8451,6 @@ static void ae7_setup_defaults(struct hda_codec *codec)
if (spec->dsp_state != DSP_DOWNLOADED)
return;
- ca0132_alt_dsp_scp_startup(codec);
ca0132_alt_init_analog_mics(codec);
ca0132_alt_start_dsp_audio_streams(codec);
ae7_post_dsp_setup_ports(codec);
@@ -8622,8 +8519,6 @@ static void ae7_setup_defaults(struct hda_codec *codec)
}
ca0132_alt_init_speaker_tuning(codec);
-
- ca0132_alt_create_dummy_stream(codec);
}
/*
--
2.25.1
next prev parent reply other threads:[~2020-12-08 21:27 UTC|newest]
Thread overview: 14+ messages / expand[flat|nested] mbox.gz Atom feed top
2020-12-08 21:25 [PATCH 0/5] ALSA: hda/ca0132 - Fix no-audio issues and add documentation Connor McAdams
2020-12-08 21:25 ` Connor McAdams
2020-12-08 21:25 ` [PATCH 1/5] ALSA: hda/ca0132 - Reset codec upon initialization Connor McAdams
2020-12-08 21:25 ` Connor McAdams
2020-12-08 21:25 ` [PATCH 2/5] ALSA: hda/ca0132 - Add stream port remapping function Connor McAdams
2020-12-08 21:25 ` Connor McAdams
2020-12-08 21:25 ` [PATCH 3/5] ALSA: hda/ca0132 - Add 8051 exram helper functions Connor McAdams
2020-12-08 21:25 ` Connor McAdams
2020-12-09 8:33 ` Takashi Iwai
2020-12-09 8:33 ` Takashi Iwai
2020-12-08 21:25 ` [PATCH 4/5] ALSA: hda/ca0132 - Ensure DSP is properly setup post-firmware download Connor McAdams
2020-12-08 21:25 ` Connor McAdams
2020-12-08 21:25 ` Connor McAdams [this message]
2020-12-08 21:25 ` [PATCH 5/5] ALSA: hda/ca0132 - Remove now unnecessary DSP setup functions Connor McAdams
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