Merge "audio: configs: Support compress record and SVA usecase" (71039967) · Commits · e / devices / Fairphone 4 / android_hardware_qcom_audio

configs/sdm670/audio_output_policy.conf

+34 −0

Original line number	Diff line number	Diff line
		@@ -82,3 +82,37 @@ voip_rx {
		app_type 69940
		}
		}

		inputs {
		primary {
		flags AUDIO_INPUT_FLAG_NONE
		formats AUDIO_FORMAT_PCM_16_BIT
		sampling_rates 8000\|11025\|12000\|16000\|22050\|24000\|32000\|44100\|48000
		bit_width 16
		app_type 69938
		}
		record_24bit {
		profile none
		flags AUDIO_INPUT_FLAG_NONE
		formats AUDIO_FORMAT_PCM_24_BIT_PACKED
		sampling_rates 16000\|48000
		bit_width 24
		app_type 69945
		}
		record_fluence1 {
		flags AUDIO_INPUT_FLAG_COMPRESS
		profile record_fluence
		formats AUDIO_FORMAT_PCM_16_BIT
		sampling_rates 48000
		bit_width 16
		app_type 69944
		}
		record_fluence {
		flags AUDIO_INPUT_FLAG_TIMESTAMP
		profile record_fluence
		formats AUDIO_FORMAT_PCM_16_BIT
		sampling_rates 48000
		bit_width 16
		app_type 69944
		}
		}

configs/sdm670/audio_platform_info_intcodec.xml

+7 −0

Original line number	Diff line number	Diff line
		@@ -63,6 +63,11 @@
		<usecase name="USECASE_AUDIO_HFP_SCO_WB" type="in" id="12" />
		<usecase name="USECASE_AUDIO_PLAYBACK_VOIP" type="out" id="16" />
		<usecase name="USECASE_AUDIO_RECORD_VOIP" type="in" id="16" />
		<usecase name="USECASE_AUDIO_RECORD_COMPRESS2" type="in" id="41"/>
		<usecase name="USECASE_AUDIO_RECORD_COMPRESS3" type="in" id="42"/>
		<usecase name="USECASE_AUDIO_RECORD_COMPRESS4" type="in" id="43"/>
		<usecase name="USECASE_AUDIO_RECORD_COMPRESS5" type="in" id="44"/>
		<usecase name="USECASE_AUDIO_RECORD_COMPRESS6" type="in" id="45"/>
		</pcm_ids>
		<config_params>
		<!-- In the below value string, the value indicates default mono -->
		@@ -97,6 +102,7 @@
		<device name="SND_DEVICE_IN_UNPROCESSED_THREE_MIC" acdb_id="145"/>
		<device name="SND_DEVICE_IN_UNPROCESSED_QUAD_MIC" acdb_id="146"/>
		<device name="SND_DEVICE_IN_UNPROCESSED_HEADSET_MIC" acdb_id="147"/>
		<device name="SND_DEVICE_IN_HANDSET_GENERIC_QMIC" acdb_id="157"/>
		</acdb_ids>
		<backend_names>
		<device name="SND_DEVICE_OUT_HEADPHONES" backend="headphones" interface="INT0_MI2S_RX"/>
		@@ -122,6 +128,7 @@
		<device name="SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_1" interface="INT4_MI2S_RX-and-INT0_MI2S_RX"/>
		<device name="SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES_EXTERNAL_2" interface="INT4_MI2S_RX-and-INT0_MI2S_RX"/>
		<device name="SND_DEVICE_OUT_VOICE_HANDSET" interface="INT4_MI2S_RX"/>
		<device name="SND_DEVICE_IN_HANDSET_GENERIC_QMIC" interface="INT3_MI2S_TX"/>
		<device name="SND_DEVICE_OUT_VOICE_SPEAKER" interface="INT4_MI2S_RX"/>
		<device name="SND_DEVICE_OUT_VOICE_SPEAKER_VBAT" interface="INT4_MI2S_RX"/>
		<device name="SND_DEVICE_OUT_VOICE_SPEAKER_2" interface="INT4_MI2S_RX"/>

configs/sdm670/mixer_paths_mtp.xml

+25 −0

Original line number	Diff line number	Diff line
		@@ -373,6 +373,11 @@
		<ctl name="IIR1 INP3 Volume" value="53" />
		<ctl name="IIR1 INP4 Volume" value="53" />
		<ctl name="IIR1 INP1 MUX" value="ZERO" />
		<ctl name="MultiMedia17 Mixer INT3_MI2S_TX" value="0" />
		<ctl name="MultiMedia18 Mixer INT3_MI2S_TX" value="0" />
		<ctl name="MultiMedia19 Mixer INT3_MI2S_TX" value="0" />
		<ctl name="MultiMedia28 Mixer INT3_MI2S_TX" value="0" />
		<ctl name="MultiMedia29 Mixer INT3_MI2S_TX" value="0" />

		<!-- anc related -->

		@@ -1270,6 +1275,26 @@
		<ctl name="MultiMedia8 Mixer INT3_MI2S_TX" value="1" />
		</path>

		<path name="audio-record-compress2">
		<ctl name="MultiMedia17 Mixer INT3_MI2S_TX" value="1" />
		</path>

		<path name="audio-record-compress3">
		<ctl name="MultiMedia18 Mixer INT3_MI2S_TX" value="1" />
		</path>

		<path name="audio-record-compress4">
		<ctl name="MultiMedia19 Mixer INT3_MI2S_TX" value="1" />
		</path>

		<path name="audio-record-compress5">
		<ctl name="MultiMedia28 Mixer INT3_MI2S_TX" value="1" />
		</path>

		<path name="audio-record-compress6">
		<ctl name="MultiMedia29 Mixer INT3_MI2S_TX" value="1" />
		</path>

		<path name="audio-record-compress bt-sco">
		<ctl name="MultiMedia8 Mixer SLIM_7_TX" value="1" />
		</path>

configs/sdm670/sdm670.mk

+7 −0

Original line number	Diff line number	Diff line
		@@ -6,6 +6,7 @@ BOARD_USES_ALSA_AUDIO := true
		ifneq ($(TARGET_USES_AOSP_FOR_AUDIO), true)
		USE_CUSTOM_AUDIO_POLICY := 1
		AUDIO_FEATURE_ENABLED_COMPRESS_CAPTURE := false
		AUDIO_FEATURE_ENABLED_COMPRESS_INPUT := true
		AUDIO_FEATURE_ENABLED_COMPRESS_VOIP := false
		AUDIO_FEATURE_ENABLED_DYNAMIC_ECNS := false
		AUDIO_FEATURE_ENABLED_EXTN_FORMATS := true
		@@ -140,8 +141,14 @@ ro.vendor.audio.sdk.fluencetype=none\
		persist.vendor.audio.fluence.voicecall=true\
		persist.vendor.audio.fluence.voicerec=false\
		persist.vendor.audio.fluence.speaker=true\
		persist.vendor.audio.fluence.audiorec=false\
		persist.vendor.audio.fluence.tmic.enabled=false

		# Mutlirec Apptype
		PRODUCT_PROPERTY_OVERRIDES += \
		vendor.audio.apptype.multirec.enabled=false \
		vendor.audio.record.multiple.enabled=false

		##speaker protection v3 switch and ADSP AFE API version
		PRODUCT_PROPERTY_OVERRIDES += \
		persist.vendor.audio.spv3.enable=true\

configs/sdm670/sound_trigger_platform_info.xml

+2 −0

Original line number	Diff line number	Diff line
		@@ -38,8 +38,10 @@
		<!-- Below backend params must match with port used in mixer path file -->
		<!-- param used to configure backend sample rate, format and channels -->
		<param backend_port_name="SLIM_0_TX" />
		<param backend_port_name="INT3_MI2S_TX" />
		<!-- Param used to match and obtain device backend index -->
		<param backend_dai_name="SLIMBUS_0_TX" />
		<param backend_dai_name="INT3_MI2S_TX" />
		</common_config>
		<acdb_ids>
		<param DEVICE_HANDSET_MIC_APE="100" />