Merge "configs: msmcobalt: Add mixer controls for native playback" into audio-userspace.lnx.2.1-dev
diff --git a/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml b/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml
new file mode 100755
index 0000000..3c75b8e
--- /dev/null
+++ b/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml
@@ -0,0 +1,115 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<!--- Copyright (c) 2014-2016, The Linux Foundation. All rights reserved. -->
+<!--- -->
+<!--- Redistribution and use in source and binary forms, with or without -->
+<!--- modification, are permitted provided that the following conditions are -->
+<!--- met: -->
+<!--- * Redistributions of source code must retain the above copyright -->
+<!--- notice, this list of conditions and the following disclaimer. -->
+<!--- * Redistributions in binary form must reproduce the above -->
+<!--- copyright notice, this list of conditions and the following -->
+<!--- disclaimer in the documentation and/or other materials provided -->
+<!--- with the distribution. -->
+<!--- * Neither the name of The Linux Foundation nor the names of its -->
+<!--- contributors may be used to endorse or promote products derived -->
+<!--- from this software without specific prior written permission. -->
+<!--- -->
+<!--- THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED -->
+<!--- WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF -->
+<!--- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT -->
+<!--- ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS -->
+<!--- BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR -->
+<!--- CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF -->
+<!--- SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR -->
+<!--- BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, -->
+<!--- WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE -->
+<!--- OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN -->
+<!--- IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. -->
+
+<mixer>
+ <!-- These are the initial mixer settings -->
+ <ctl name="LSM1 MUX" value="None" />
+ <ctl name="LSM2 MUX" value="None" />
+ <ctl name="LSM3 MUX" value="None" />
+ <ctl name="LSM4 MUX" value="None" />
+ <ctl name="LSM5 MUX" value="None" />
+ <ctl name="LSM6 MUX" value="None" />
+ <ctl name="LSM7 MUX" value="None" />
+ <ctl name="LSM8 MUX" value="None" />
+ <ctl name="SLIMBUS_5_TX LSM Function" value="None" />
+ <ctl name="MADONOFF Switch" value="0" />
+ <ctl name="MAD Input" value="DMIC1" />
+ <ctl name="MAD_BROADCAST Switch" value="0" />
+ <ctl name="TX13 INP MUX" value="CDC_DEC_5" />
+ <ctl name="AIF4_MAD Mixer SLIM TX12" value="0" />
+ <ctl name="AIF4_MAD Mixer SLIM TX13" value="0" />
+ <ctl name="CPE AFE MAD Enable" value="0"/>
+ <ctl name="CLK MODE" value="EXTERNAL" />
+ <ctl name="EC BUF MUX INP" value="ZERO" />
+ <ctl name="ADC MUX1" value="DMIC" />
+ <ctl name="DMIC MUX1" value="ZERO" />
+
+ <path name="listen-voice-wakeup-1">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM1 MUX" value="SLIMBUS_5_TX" />
+ </path>
+
+ <path name="listen-voice-wakeup-2">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM2 MUX" value="SLIMBUS_5_TX" />
+ </path>
+ <path name="listen-voice-wakeup-3">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM3 MUX" value="SLIMBUS_5_TX" />
+ </path>
+ <path name="listen-voice-wakeup-4">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM4 MUX" value="SLIMBUS_5_TX" />
+ </path>
+ <path name="listen-voice-wakeup-5">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM5 MUX" value="SLIMBUS_5_TX" />
+ </path>
+ <path name="listen-voice-wakeup-6">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM6 MUX" value="SLIMBUS_5_TX" />
+ </path>
+ <path name="listen-voice-wakeup-7">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM7 MUX" value="SLIMBUS_5_TX" />
+ </path>
+ <path name="listen-voice-wakeup-8">
+ <ctl name="SLIMBUS_5_TX LSM Function" value="AUDIO" />
+ <ctl name="LSM8 MUX" value="SLIMBUS_5_TX" />
+ </path>
+
+ <path name="listen-cpe-handset-mic">
+ <ctl name="MAD Input" "DMIC0" />
+ <ctl name="MAD_SEL MUX" "SPE" />
+ <ctl name="MAD_INP MUX" "MAD" />
+ <ctl name="MAD_CPE1 Switch" 1 />
+ </path>
+
+ <path name="listen-cpe-handset-mic-ecpp">
+ <ctl name="CLK MODE" value="INTERNAL" />
+ <ctl name="EC BUF MUX INP" value="DEC1" />
+ <ctl name="ADC MUX1" value="DMIC" />
+ <ctl name="DMIC MUX1" value="DMIC0" />
+ </path>
+
+ <!-- path name used for low bandwidth FTRT codec interface -->
+ <path name="listen-cpe-handset-mic low-speed-intf">
+ <ctl name="MADONOFF Switch" value="1" />
+ <ctl name="AIF4_MAD Mixer SLIM TX12" value="1" />
+ <ctl name="MAD Input" value="DMIC0" />
+ <ctl name="CPE AFE MAD Enable" value="1"/>
+ </path>
+
+ <path name="listen-ape-handset-mic">
+ <ctl name="MAD_BROADCAST Switch" value="1" />
+ <ctl name="TX13 INP MUX" value="MAD_BRDCST" />
+ <ctl name="AIF4_MAD Mixer SLIM TX13" value="1" />
+ <ctl name="MAD Input" value="DMIC0" />
+ </path>
+
+</mixer>
diff --git a/configs/msmcobalt/sound_trigger_platform_info.xml b/configs/msmcobalt/sound_trigger_platform_info.xml
index b92ea48..1f90bd5 100644
--- a/configs/msmcobalt/sound_trigger_platform_info.xml
+++ b/configs/msmcobalt/sound_trigger_platform_info.xml
@@ -29,8 +29,7 @@
<param version="0x0101" /> <!-- this must be the first param -->
<common_config>
- <param execution_type="CPE" /> <!-- value: "CPE" "APE" -->
- <param max_cpe_sessions="1" />
+ <param max_cpe_sessions="2" />
<param max_ape_sessions="8" />
<param enable_failure_detection="false" />
</common_config>
@@ -41,11 +40,12 @@
<param DEVICE_HANDSET_CPE_ECPP_ACDB_ID="128" />
</acdb_ids>
- <!-- Multiple sound_model_config tags can be listed, each with unique -->
- <!-- vendor_uuid. The below tag represents QTI SVA engine sound model -->
- <!-- configuration. ISV must use their own unique vendor_uuid. -->
+ <!-- Multiple sound_model_config tags can be listed, each with unique -->
+ <!-- vendor_uuid. The below tag represents QTI SVA engine sound model -->
+ <!-- configuration. ISV must use their own unique vendor_uuid. -->
<sound_model_config>
<param vendor_uuid="68ab2d40-e860-11e3-95ef-0002a5d5c51b" />
+ <param execution_type="WDSP" /> <!-- value: "WDSP" "ADSP" "DYNAMIC" -->
<param app_type="2" /> <!-- app type used in ACDB -->
<param library="libsmwrapper.so" />
<param max_cpe_phrases="6" />
@@ -54,7 +54,18 @@
<param max_ape_users="10" />
<param sample_rate="16000" />
- <!-- Module and param ids with which the algorithm is integrated in firmware -->
+ <gcs_uid>
+ <param uid="0x1" />
+ <param did="0x4" />
+ <param load_sound_model_ids="0x00012C0D, 0x0, 0x00012C14" />
+ <param confidence_levels_ids="0x00012C0D, 0x0, 0x00012C28" />
+ <param operation_mode_ids="0x00012C0D, 0x0, 0x00012C28" />
+ <param detection_event_ids="0x00012C0D, 0x0, 0x00012C29" />
+ <param capture_event_ids="0x00020013, 0x0,0x00020015" />
+ </gcs_uid>
+
+ <!-- Module and param ids with which the algorithm is integrated
+ in non-graphite firmware (note these must come after gcs params) -->
<param load_sound_model_ids="0x00012C0D, 0x00012C14" />
<param unload_sound_model_ids="0x00012C0D, 0x00012C15" />
<param confidence_levels_ids="0x00012C0D, 0x00012C07" />
@@ -62,7 +73,8 @@
<!-- format: "ADPCM_packet" or "PCM_packet" !-->
<!-- transfer_mode: "FTRT" or "RT" -->
- <!-- kw_duration is in milli seconds. It is valid only for FTRT transfer mode -->
+ <!-- kw_duration is in milli seconds. It is valid only for FTRT
+ transfer mode -->
<param capture_keyword="PCM_packet, RT, 2000" />
<param client_capture_read_delay="2000" />
</sound_model_config>
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index d186a5f..12a8082 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -401,6 +401,10 @@
#endif
+#ifndef AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH
+#define AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH 0x10000
+#endif
+
#ifndef HDMI_PASSTHROUGH_ENABLED
#define audio_extn_passthru_update_stream_configuration(adev, out) (0)
#define audio_extn_passthru_is_convert_supported(adev, out) (0)
@@ -419,8 +423,6 @@
#define audio_extn_passthru_set_parameters(a, p) (-ENOSYS)
#define audio_extn_passthru_init(a) do {} while(0)
#define audio_extn_passthru_should_standby(o) (1)
-
-#define AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH 0x1000
#else
bool audio_extn_passthru_is_convert_supported(struct audio_device *adev,
struct stream_out *out);
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index e3f1b6c..26c43b4 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -90,9 +90,7 @@
#ifdef INCALL_MUSIC_ENABLED
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_INCALL_MUSIC),
#endif
-#ifdef HDMI_PASSTHROUGH_ENABLED
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH),
-#endif
};
const struct string_to_enum s_format_name_to_enum_table[] = {
@@ -133,6 +131,7 @@
STRING_TO_ENUM(AUDIO_FORMAT_AAC_ADTS_LC),
STRING_TO_ENUM(AUDIO_FORMAT_AAC_ADTS_HE_V1),
STRING_TO_ENUM(AUDIO_FORMAT_AAC_ADTS_HE_V2),
+ STRING_TO_ENUM(AUDIO_FORMAT_DSD),
#endif
};
@@ -515,6 +514,21 @@
__func__, sample_rate);
}
}
+
+ /* Set sampling rate to 176.4 for DSD64
+ * and 352.8Khz for DSD128.
+ * Set Bit Width to 16. output will be 16 bit
+ * post DoP in ASM.
+ */
+ if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH) &&
+ (format == AUDIO_FORMAT_DSD)) {
+ bit_width = 16;
+ if (sample_rate == INPUT_SAMPLING_RATE_DSD64)
+ sample_rate = OUTPUT_SAMPLING_RATE_DSD64;
+ else if (sample_rate == INPUT_SAMPLING_RATE_DSD128)
+ sample_rate = OUTPUT_SAMPLING_RATE_DSD128;
+ }
+
ALOGV("%s: flags: %x, format: %x sample_rate %d",
__func__, flags, format, sample_rate);
list_for_each(node_i, streams_output_cfg_list) {
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index af399a1..70909f5 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -82,6 +82,7 @@
/* ToDo: Check and update a proper value in msec */
#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 50
#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
+#define DSD_VOLUME_MIN_DB (-110)
#define PROXY_OPEN_RETRY_COUNT 100
#define PROXY_OPEN_WAIT_TIME 20
@@ -501,6 +502,7 @@
format == AUDIO_FORMAT_FLAC ||
format == AUDIO_FORMAT_ALAC ||
format == AUDIO_FORMAT_APE ||
+ format == AUDIO_FORMAT_DSD ||
format == AUDIO_FORMAT_VORBIS ||
format == AUDIO_FORMAT_WMA ||
format == AUDIO_FORMAT_WMA_PRO)
@@ -541,6 +543,9 @@
case AUDIO_FORMAT_APE:
id = SND_AUDIOCODEC_APE;
break;
+ case AUDIO_FORMAT_DSD:
+ id = SND_AUDIOCODEC_DSD;
+ break;
case AUDIO_FORMAT_VORBIS:
id = SND_AUDIOCODEC_VORBIS;
break;
@@ -1166,6 +1171,28 @@
return active;
}
+/*
+ * if native DSD playback active
+ */
+bool audio_is_dsd_native_stream_active(struct audio_device *adev)
+{
+ bool active = false;
+ struct listnode *node = NULL;
+ struct audio_usecase *uc = NULL;
+ struct stream_out *curr_out = NULL;
+
+ list_for_each(node, &adev->usecase_list) {
+ uc = node_to_item(node, struct audio_usecase, list);
+ curr_out = (struct stream_out*) uc->stream.out;
+
+ if (curr_out && PCM_PLAYBACK == uc->type &&
+ (DSD_NATIVE_BACKEND == platform_get_backend_index(uc->out_snd_device))) {
+ active = true;
+ ALOGV("%s:DSD playback is active", __func__);
+ }
+ }
+ return active;
+}
static bool force_device_switch(struct audio_usecase *usecase)
{
@@ -2537,6 +2564,14 @@
return latency;
}
+static float AmpToDb(float amplification)
+{
+ if (amplification == 0) {
+ return DSD_VOLUME_MIN_DB;
+ }
+ return 20 * log10(amplification);
+}
+
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
@@ -2555,6 +2590,20 @@
* Mute is 0 and unmute 1
*/
audio_extn_passthru_set_volume(out, (left == 0.0f));
+ } else if (out->format == AUDIO_FORMAT_DSD){
+ char mixer_ctl_name[128] = "DSD Volume";
+ struct audio_device *adev = out->dev;
+ struct mixer_ctl *ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+ volume[0] = (int)(AmpToDb(left));
+ volume[1] = (int)(AmpToDb(right));
+ mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0]));
+ return 0;
} else {
char mixer_ctl_name[128];
struct audio_device *adev = out->dev;
@@ -3666,12 +3715,24 @@
__func__, config->offload_info.version,
config->offload_info.bit_rate);
+ /*Check if DSD audio format is supported in codec
+ *and there is no active native DSD use case
+ */
+
+ if ((config->format == AUDIO_FORMAT_DSD) &&
+ (!platform_check_codec_dsd_support(adev->platform) ||
+ audio_is_dsd_native_stream_active(adev))) {
+ ret = -EINVAL;
+ goto error_open;
+ }
+
/* Disable gapless if any of the following is true
* passthrough playback
* AV playback
* Direct PCM playback
*/
if (audio_extn_passthru_is_passthrough_stream(out) ||
+ (config->format == AUDIO_FORMAT_DSD) ||
config->offload_info.has_video ||
out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
check_and_set_gapless_mode(adev, false);
@@ -3681,6 +3742,10 @@
if (audio_extn_passthru_is_passthrough_stream(out)) {
out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
}
+ if (config->format == AUDIO_FORMAT_DSD) {
+ out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
+ out->compr_config.codec->compr_passthr = PASSTHROUGH_DSD;
+ }
} else if (out->flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) {
ret = voice_extn_check_and_set_incall_music_usecase(adev, out);
if (ret != 0) {
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index ee28157..1b5c6c9 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -410,6 +410,7 @@
bool audio_is_true_native_stream_active(struct audio_device *adev);
+bool audio_is_dsd_native_stream_active(struct audio_device *adev);
int pcm_ioctl(struct pcm *pcm, int request, ...);
int get_snd_card_state(struct audio_device *adev);
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index e0400c6..8894b2f 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -2448,7 +2448,7 @@
return ret;
}
-static int platform_get_backend_index(snd_device_t snd_device)
+int platform_get_backend_index(snd_device_t snd_device)
{
int32_t port = DEFAULT_CODEC_BACKEND;
@@ -4343,8 +4343,8 @@
struct audio_usecase *uc;
uc = node_to_item(node, struct audio_usecase, list);
struct stream_out *out = (struct stream_out*) uc->stream.out;
- unsigned int out_channels = audio_channel_count_from_out_mask(out->channel_mask);
if (uc->type == PCM_PLAYBACK && out && usecase != uc) {
+ unsigned int out_channels = audio_channel_count_from_out_mask(out->channel_mask);
ALOGD("%s:napb: (%d) - (%s)id (%d) sr %d bw "
"(%d) ch (%d) device %s", __func__, i++, use_case_table[uc->id],
@@ -5408,3 +5408,8 @@
}
return 0;
}
+
+bool platform_check_codec_dsd_support(void *platform __unused)
+{
+ return false;
+}
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index dcd351a..6c89d0a 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -198,9 +198,12 @@
SND_DEVICE_MAX = SND_DEVICE_IN_END,
};
-
+#define INPUT_SAMPLING_RATE_DSD64 2822400
+#define INPUT_SAMPLING_RATE_DSD128 5644800
#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
#define OUTPUT_SAMPLING_RATE_44100 44100
+#define OUTPUT_SAMPLING_RATE_DSD64 176400
+#define OUTPUT_SAMPLING_RATE_DSD128 352800
#define MAX_PORT 6
#define ALL_CODEC_BACKEND_PORT 0
#define HEADPHONE_44_1_BACKEND_PORT 5
@@ -208,6 +211,8 @@
enum {
DEFAULT_CODEC_BACKEND,
SLIMBUS_0_RX = DEFAULT_CODEC_BACKEND,
+ DSD_NATIVE_BACKEND,
+ SLIMBUS_2_RX = DSD_NATIVE_BACKEND,
HEADPHONE_44_1_BACKEND,
SLIMBUS_5_RX = HEADPHONE_44_1_BACKEND,
HEADPHONE_BACKEND,
@@ -356,7 +361,8 @@
enum {
LEGACY_PCM = 0,
PASSTHROUGH,
- PASSTHROUGH_CONVERT
+ PASSTHROUGH_CONVERT,
+ PASSTHROUGH_DSD
};
/*
* ID for setting mute and lateny on the device side
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index 2b6a1d7..1b47e7d 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1257,3 +1257,13 @@
}
return 0;
}
+
+bool platform_check_codec_dsd_support(void *platform __unused)
+{
+ return false;
+}
+
+int platform_get_backend_index(snd_device_t snd_device __unused);
+{
+ return 0;
+}
\ No newline at end of file
diff --git a/hal/msm8960/platform.h b/hal/msm8960/platform.h
index e42af8c..07060b6 100644
--- a/hal/msm8960/platform.h
+++ b/hal/msm8960/platform.h
@@ -112,6 +112,12 @@
#define SOUND_CARD 0
#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
+#define INPUT_SAMPLING_RATE_DSD64 2822400
+#define INPUT_SAMPLING_RATE_DSD128 5644800
+#define OUTPUT_SAMPLING_RATE_DSD64 176400
+#define OUTPUT_SAMPLING_RATE_DSD128 352800
+#define DSD_NATIVE_BACKEND 1
+#define PASSTHROUGH_DSD 3
#define ALL_SESSION_VSID 0xFFFFFFFF
#define DEFAULT_MUTE_RAMP_DURATION_MS 20
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 8daa715..9c85d10 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -247,6 +247,7 @@
int metainfo_key;
int source_mic_type;
int max_mic_count;
+ bool is_dsd_supported;
};
static int pcm_device_table[AUDIO_USECASE_MAX][2] = {
@@ -334,6 +335,7 @@
[SND_DEVICE_OUT_SPEAKER_VBAT] = "speaker-vbat",
[SND_DEVICE_OUT_SPEAKER_REVERSE] = "speaker-reverse",
[SND_DEVICE_OUT_HEADPHONES] = "headphones",
+ [SND_DEVICE_OUT_HEADPHONES_DSD] = "headphones-dsd",
[SND_DEVICE_OUT_HEADPHONES_44_1] = "headphones-44.1",
[SND_DEVICE_OUT_LINE] = "line",
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = "speaker-and-headphones",
@@ -451,6 +453,7 @@
[SND_DEVICE_OUT_SPEAKER_REVERSE] = 14,
[SND_DEVICE_OUT_LINE] = 10,
[SND_DEVICE_OUT_HEADPHONES] = 10,
+ [SND_DEVICE_OUT_HEADPHONES_DSD] = 10,
[SND_DEVICE_OUT_HEADPHONES_44_1] = 10,
[SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES] = 10,
[SND_DEVICE_OUT_SPEAKER_AND_LINE] = 10,
@@ -568,6 +571,7 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_VBAT)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_REVERSE)},
{TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES_DSD)},
{TO_NAME_INDEX(SND_DEVICE_OUT_HEADPHONES_44_1)},
{TO_NAME_INDEX(SND_DEVICE_OUT_LINE)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES)},
@@ -1102,11 +1106,13 @@
backend_tag_table[SND_DEVICE_IN_USB_HEADSET_MIC] = strdup("usb-headset-mic");
backend_tag_table[SND_DEVICE_IN_CAPTURE_FM] = strdup("capture-fm");
backend_tag_table[SND_DEVICE_OUT_TRANSMISSION_FM] = strdup("transmission-fm");
+ backend_tag_table[SND_DEVICE_OUT_HEADPHONES_DSD] = strdup("headphones-dsd");
backend_tag_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("voice-speaker-vbat");
backend_tag_table[SND_DEVICE_OUT_BT_A2DP] = strdup("bt-a2dp");
backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = strdup("speaker-and-bt-a2dp");
+ hw_interface_table[SND_DEVICE_OUT_HEADPHONES_DSD] = strdup("SLIMBUS_2_RX");
hw_interface_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("SLIMBUS_5_RX");
hw_interface_table[SND_DEVICE_OUT_HDMI] = strdup("HDMI_RX");
hw_interface_table[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = strdup("SLIMBUS_0_RX-and-HDMI_RX");
@@ -1715,6 +1721,11 @@
my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
strdup("SLIM_0_RX SampleRate");
+ my_data->current_backend_cfg[DSD_NATIVE_BACKEND].bitwidth_mixer_ctl =
+ strdup("SLIM_2_RX Format");
+ my_data->current_backend_cfg[DSD_NATIVE_BACKEND].samplerate_mixer_ctl =
+ strdup("SLIM_2_RX SampleRate");
+
my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].bitwidth_mixer_ctl =
strdup("SLIM_5_RX Format");
my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].samplerate_mixer_ctl =
@@ -1745,6 +1756,12 @@
}
}
+ if(strstr(snd_card_name, "tavil")) {
+ ALOGD("%s:DSD playback is supported", __func__);
+ my_data->is_dsd_supported = true;
+ platform_set_native_support(NATIVE_AUDIO_MODE_MULTIPLE_44_1);
+ }
+
my_data->current_backend_cfg[HEADPHONE_BACKEND].bitwidth_mixer_ctl =
strdup("SLIM_6_RX Format");
my_data->current_backend_cfg[HEADPHONE_BACKEND].samplerate_mixer_ctl =
@@ -2092,7 +2109,8 @@
int platform_set_native_support(int na_mode)
{
- if (NATIVE_AUDIO_MODE_SRC == na_mode || NATIVE_AUDIO_MODE_TRUE_44_1 == na_mode) {
+ if (NATIVE_AUDIO_MODE_SRC == na_mode || NATIVE_AUDIO_MODE_TRUE_44_1 == na_mode
+ || NATIVE_AUDIO_MODE_MULTIPLE_44_1 == na_mode) {
na_props.platform_na_prop_enabled = na_props.ui_na_prop_enabled = true;
na_props.na_mode = na_mode;
ALOGD("%s:napb: native audio playback enabled in (%s) mode v2.0", __func__,
@@ -2107,6 +2125,12 @@
return 0;
}
+bool platform_check_codec_dsd_support(void *platform)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ return my_data->is_dsd_supported;
+}
+
int platform_get_native_support()
{
int ret = NATIVE_AUDIO_MODE_INVALID;
@@ -2159,6 +2183,8 @@
mode = NATIVE_AUDIO_MODE_SRC;
else if (value && !strncmp(value, "true", sizeof("true")))
mode = NATIVE_AUDIO_MODE_TRUE_44_1;
+ else if (value && !strncmp(value, "multiple", sizeof("multiple")))
+ mode = NATIVE_AUDIO_MODE_MULTIPLE_44_1;
else {
mode = NATIVE_AUDIO_MODE_INVALID;
ALOGE("%s:napb:native_audio_mode in platform info xml,invalid mode string",
@@ -2238,7 +2264,7 @@
return ret;
}
-static int platform_get_backend_index(snd_device_t snd_device)
+int platform_get_backend_index(snd_device_t snd_device)
{
int32_t port = DEFAULT_CODEC_BACKEND;
@@ -2247,6 +2273,9 @@
if (strncmp(backend_tag_table[snd_device], "headphones-44.1",
sizeof("headphones-44.1")) == 0)
port = HEADPHONE_44_1_BACKEND;
+ else if (strncmp(backend_tag_table[snd_device], "headphones-dsd",
+ sizeof("headphones-dsd")) == 0)
+ port = DSD_NATIVE_BACKEND;
else if (strncmp(backend_tag_table[snd_device], "headphones",
sizeof("headphones")) == 0)
port = HEADPHONE_BACKEND;
@@ -2764,6 +2793,12 @@
} else if (NATIVE_AUDIO_MODE_SRC == na_mode &&
OUTPUT_SAMPLING_RATE_44100 == sample_rate) {
snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
+ } else if (NATIVE_AUDIO_MODE_MULTIPLE_44_1 == na_mode &&
+ (sample_rate % OUTPUT_SAMPLING_RATE_44100 == 0) &&
+ (out->format != AUDIO_FORMAT_DSD)) {
+ snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
+ } else if (out->format == AUDIO_FORMAT_DSD) {
+ snd_device = SND_DEVICE_OUT_HEADPHONES_DSD;
} else
snd_device = SND_DEVICE_OUT_HEADPHONES;
} else if (devices & AUDIO_DEVICE_OUT_LINE) {
@@ -4070,14 +4105,6 @@
my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
}
- /*
- * Backend sample rate configuration follows:
- * 16 bit playback - 48khz for streams at any valid sample rate
- * 24 bit playback - 48khz for stream sample rate less than 48khz
- * 24 bit playback - 96khz for sample rate range of 48khz to 96khz
- * 24 bit playback - 192khz for sample rate range of 96khz to 192 khz
- * Upper limit is inclusive in the sample rate range.
- */
if (sample_rate !=
my_data->current_backend_cfg[backend_idx].sample_rate) {
char *rate_str = NULL;
@@ -4096,14 +4123,24 @@
rate_str = "KHZ_44P1";
break;
case 64000:
- case 88200:
case 96000:
rate_str = "KHZ_96";
break;
+ case 88200:
+ rate_str = "KHZ_88P2";
+ break;
case 176400:
+ rate_str = "KHZ_176P4";
+ break;
case 192000:
rate_str = "KHZ_192";
break;
+ case 352800:
+ rate_str = "KHZ_352P8";
+ break;
+ case 384000:
+ rate_str = "KHZ_384";
+ break;
default:
rate_str = "KHZ_48";
break;
@@ -4310,8 +4347,8 @@
struct audio_usecase *uc;
uc = node_to_item(node, struct audio_usecase, list);
struct stream_out *out = (struct stream_out*) uc->stream.out;
- unsigned int out_channels = audio_channel_count_from_out_mask(out->channel_mask);
if (uc->type == PCM_PLAYBACK && out && usecase != uc) {
+ unsigned int out_channels = audio_channel_count_from_out_mask(out->channel_mask);
ALOGD("%s:napb: (%d) - (%s)id (%d) sr %d bw "
"(%d) ch (%d) device %s", __func__, i++, use_case_table[uc->id],
@@ -4400,6 +4437,24 @@
channels_updated = true;
}
+ /*
+ * Map native sampling rates to upper limit range
+ * if multiple of native sampling rates are not supported.
+ */
+ if (NATIVE_AUDIO_MODE_MULTIPLE_44_1 != na_mode) {
+ switch (sample_rate) {
+ case 88200:
+ sample_rate = 96000;
+ break;
+ case 176400:
+ sample_rate = 192000;
+ break;
+ case 352800:
+ sample_rate = 192000;
+ break;
+ }
+ }
+
ALOGI("%s:becf: afe: Codec selected backend: %d updated bit width: %d and sample rate: %d",
__func__, backend_idx , bit_width, sample_rate);
@@ -4440,6 +4495,17 @@
/*this is populated by check_codec_backend_cfg hence set default value to false*/
backend_cfg.passthrough_enabled = false;
+ /* Set Backend sampling rate to 176.4 for DSD64 and
+ * 352.8Khz for DSD128.
+ * Set Bit Width to 16
+ */
+ if ((backend_idx == DSD_NATIVE_BACKEND) && (backend_cfg.format == AUDIO_FORMAT_DSD)) {
+ backend_cfg.bit_width = 16;
+ if (backend_cfg.sample_rate == INPUT_SAMPLING_RATE_DSD64)
+ backend_cfg.sample_rate = OUTPUT_SAMPLING_RATE_DSD64;
+ else if (backend_cfg.sample_rate == INPUT_SAMPLING_RATE_DSD128)
+ backend_cfg.sample_rate = OUTPUT_SAMPLING_RATE_DSD128;
+ }
ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
", backend_idx %d usecase = %d device (%s)", __func__, backend_cfg.bit_width,
backend_cfg.sample_rate, backend_cfg.channels, backend_idx, usecase->id,
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index 48bfb2b..9394ef8 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -80,6 +80,7 @@
SND_DEVICE_OUT_SPEAKER_VBAT,
SND_DEVICE_OUT_LINE,
SND_DEVICE_OUT_HEADPHONES,
+ SND_DEVICE_OUT_HEADPHONES_DSD,
SND_DEVICE_OUT_HEADPHONES_44_1,
SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
SND_DEVICE_OUT_SPEAKER_AND_LINE,
@@ -192,13 +193,18 @@
SND_DEVICE_MAX = SND_DEVICE_IN_END,
};
-
+#define INPUT_SAMPLING_RATE_DSD64 2822400
+#define INPUT_SAMPLING_RATE_DSD128 5644800
#define DEFAULT_OUTPUT_SAMPLING_RATE 48000
#define OUTPUT_SAMPLING_RATE_44100 44100
+#define OUTPUT_SAMPLING_RATE_DSD64 176400
+#define OUTPUT_SAMPLING_RATE_DSD128 352800
#define MAX_CODEC_TX_BACKENDS 1
enum {
DEFAULT_CODEC_BACKEND,
SLIMBUS_0_RX = DEFAULT_CODEC_BACKEND,
+ DSD_NATIVE_BACKEND,
+ SLIMBUS_2_RX = DSD_NATIVE_BACKEND,
HEADPHONE_44_1_BACKEND,
SLIMBUS_5_RX = HEADPHONE_44_1_BACKEND,
HEADPHONE_BACKEND,
@@ -447,7 +453,8 @@
enum {
LEGACY_PCM = 0,
PASSTHROUGH,
- PASSTHROUGH_CONVERT
+ PASSTHROUGH_CONVERT,
+ PASSTHROUGH_DSD
};
/*
* ID for setting mute and lateny on the device side
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 0bb73f3..60e46f1 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -27,6 +27,7 @@
enum {
NATIVE_AUDIO_MODE_SRC = 1,
NATIVE_AUDIO_MODE_TRUE_44_1,
+ NATIVE_AUDIO_MODE_MULTIPLE_44_1,
NATIVE_AUDIO_MODE_INVALID
};
@@ -151,4 +152,6 @@
bool enable,
char * str);
bool platform_supports_true_32bit();
+bool platform_check_codec_dsd_support(void *platform);
+int platform_get_backend_index(snd_device_t snd_device);
#endif // AUDIO_PLATFORM_API_H