Merge "hal: add USB TTY support"
diff --git a/configs/msm8909/mixer_paths.xml b/configs/msm8909/mixer_paths.xml
index 13da80e..0f5b333 100644
--- a/configs/msm8909/mixer_paths.xml
+++ b/configs/msm8909/mixer_paths.xml
@@ -78,7 +78,7 @@
<ctl name="EAR_S" value="ZERO" />
<ctl name="HPHL" value="ZERO" />
<ctl name="HPHR" value="ZERO" />
- <ctl name="SPK DAC Switch" value="0" />
+ <ctl name="SPK" value="ZERO" />
<ctl name="EAR PA Gain" value="POS_1P5_DB" />
<ctl name="MI2S_RX Channels" value="One" />
<ctl name="MI2S_TX Channels" value="One" />
diff --git a/configs/msm8909/mixer_paths_msm8909_pm8916.xml b/configs/msm8909/mixer_paths_msm8909_pm8916.xml
index 5dbeaa7..559a5bf 100644
--- a/configs/msm8909/mixer_paths_msm8909_pm8916.xml
+++ b/configs/msm8909/mixer_paths_msm8909_pm8916.xml
@@ -78,7 +78,7 @@
<ctl name="EAR_S" value="ZERO" />
<ctl name="HPHL" value="ZERO" />
<ctl name="HPHR" value="ZERO" />
- <ctl name="SPK DAC Switch" value="0" />
+ <ctl name="SPK" value="ZERO" />
<ctl name="Speaker Boost" value="ENABLE" />
<ctl name="MICBIAS CAPLESS Switch" value="0" />
<ctl name="EAR PA Boost" value="ENABLE" />
@@ -604,7 +604,7 @@
<path name="speaker">
<ctl name="RX3 MIX1 INP1" value="RX1" />
- <ctl name="SPK DAC Switch" value="1" />
+ <ctl name="SPK" value="Switch" />
</path>
<path name="speaker-mic">
diff --git a/configs/msm8909/mixer_paths_qrd_skuh.xml b/configs/msm8909/mixer_paths_qrd_skuh.xml
index 067d316..d3b232c 100644
--- a/configs/msm8909/mixer_paths_qrd_skuh.xml
+++ b/configs/msm8909/mixer_paths_qrd_skuh.xml
@@ -80,7 +80,7 @@
<ctl name="EAR_S Switch" value="0" />
<ctl name="HPHL" value="ZERO" />
<ctl name="HPHR" value="ZERO" />
- <ctl name="SPK DAC Switch" value="0" />
+ <ctl name="SPK" value="ZERO" />
<ctl name="Speaker Boost" value="DISABLE" />
<ctl name="EAR PA Boost" value="DISABLE" />
<ctl name="EAR PA Gain" value="POS_6_DB" />
@@ -589,7 +589,7 @@
<path name="speaker">
<ctl name="RX3 MIX1 INP1" value="RX1" />
- <ctl name="SPK DAC Switch" value="1" />
+ <ctl name="SPK" value="Switch" />
<ctl name="Speaker Boost" value="ENABLE" />
</path>
diff --git a/configs/msm8909/mixer_paths_qrd_skui.xml b/configs/msm8909/mixer_paths_qrd_skui.xml
index 067d316..d3b232c 100644
--- a/configs/msm8909/mixer_paths_qrd_skui.xml
+++ b/configs/msm8909/mixer_paths_qrd_skui.xml
@@ -80,7 +80,7 @@
<ctl name="EAR_S Switch" value="0" />
<ctl name="HPHL" value="ZERO" />
<ctl name="HPHR" value="ZERO" />
- <ctl name="SPK DAC Switch" value="0" />
+ <ctl name="SPK" value="ZERO" />
<ctl name="Speaker Boost" value="DISABLE" />
<ctl name="EAR PA Boost" value="DISABLE" />
<ctl name="EAR PA Gain" value="POS_6_DB" />
@@ -589,7 +589,7 @@
<path name="speaker">
<ctl name="RX3 MIX1 INP1" value="RX1" />
- <ctl name="SPK DAC Switch" value="1" />
+ <ctl name="SPK" value="Switch" />
<ctl name="Speaker Boost" value="ENABLE" />
</path>
diff --git a/configs/msm8909/mixer_paths_qrd_skut.xml b/configs/msm8909/mixer_paths_qrd_skut.xml
index 45c4581..60c79b7 100644
--- a/configs/msm8909/mixer_paths_qrd_skut.xml
+++ b/configs/msm8909/mixer_paths_qrd_skut.xml
@@ -80,7 +80,7 @@
<ctl name="EAR_S" value="ZERO" />
<ctl name="HPHL" value="ZERO" />
<ctl name="HPHR" value="ZERO" />
- <ctl name="SPK DAC Switch" value="0" />
+ <ctl name="SPK" value="ZERO" />
<ctl name="EAR PA Gain" value="POS_1P5_DB" />
<ctl name="MI2S_RX Channels" value="One" />
<ctl name="MI2S_TX Channels" value="One" />
@@ -603,7 +603,7 @@
<path name="speaker">
<ctl name="RX3 MIX1 INP1" value="RX1" />
- <ctl name="SPK DAC Switch" value="1" />
+ <ctl name="SPK" value="Switch" />
</path>
<path name="speaker-mic">
diff --git a/configs/msm8909/mixer_paths_skua.xml b/configs/msm8909/mixer_paths_skua.xml
index 0ed2211..33efc0b 100644
--- a/configs/msm8909/mixer_paths_skua.xml
+++ b/configs/msm8909/mixer_paths_skua.xml
@@ -80,7 +80,7 @@
<ctl name="EAR_S" value="ZERO" />
<ctl name="HPHL" value="ZERO" />
<ctl name="HPHR" value="ZERO" />
- <ctl name="SPK DAC Switch" value="0" />
+ <ctl name="SPK" value="ZERO" />
<ctl name="EAR PA Gain" value="POS_1P5_DB" />
<ctl name="MI2S_RX Channels" value="One" />
<ctl name="MI2S_TX Channels" value="One" />
@@ -603,7 +603,7 @@
<path name="speaker">
<ctl name="RX3 MIX1 INP1" value="RX1" />
- <ctl name="SPK DAC Switch" value="1" />
+ <ctl name="SPK" value="Switch" />
</path>
<path name="speaker-mic">
diff --git a/configs/msm8909/mixer_paths_skuc.xml b/configs/msm8909/mixer_paths_skuc.xml
index e35788b..1bdb050 100644
--- a/configs/msm8909/mixer_paths_skuc.xml
+++ b/configs/msm8909/mixer_paths_skuc.xml
@@ -80,7 +80,7 @@
<ctl name="EAR_S" value="ZERO" />
<ctl name="HPHL" value="ZERO" />
<ctl name="HPHR" value="ZERO" />
- <ctl name="SPK DAC Switch" value="0" />
+ <ctl name="SPK" value="ZERO" />
<ctl name="EAR PA Gain" value="POS_1P5_DB" />
<ctl name="MI2S_RX Channels" value="One" />
<ctl name="MI2S_TX Channels" value="One" />
@@ -603,7 +603,7 @@
<path name="speaker">
<ctl name="RX3 MIX1 INP1" value="RX1" />
- <ctl name="SPK DAC Switch" value="1" />
+ <ctl name="SPK" value="Switch" />
</path>
<path name="speaker-mic">
diff --git a/configs/msm8909/mixer_paths_skue.xml b/configs/msm8909/mixer_paths_skue.xml
index 86c47ae..e35ddef 100644
--- a/configs/msm8909/mixer_paths_skue.xml
+++ b/configs/msm8909/mixer_paths_skue.xml
@@ -80,7 +80,7 @@
<ctl name="EAR_S" value="ZERO" />
<ctl name="HPHL" value="ZERO" />
<ctl name="HPHR" value="ZERO" />
- <ctl name="SPK DAC Switch" value="0" />
+ <ctl name="SPK" value="ZERO" />
<ctl name="MICBIAS CAPLESS Switch" value="0" />
<ctl name="EAR PA Gain" value="POS_1P5_DB" />
<ctl name="MI2S_RX Channels" value="One" />
@@ -604,7 +604,7 @@
<path name="speaker">
<ctl name="RX3 MIX1 INP1" value="RX1" />
- <ctl name="SPK DAC Switch" value="1" />
+ <ctl name="SPK" value="Switch" />
</path>
<path name="speaker-mic">
diff --git a/configs/msm8909/msm8909.mk b/configs/msm8909/msm8909.mk
index cfd71ef..3405db7 100755
--- a/configs/msm8909/msm8909.mk
+++ b/configs/msm8909/msm8909.mk
@@ -32,6 +32,7 @@
AUDIO_FEATURE_ENABLED_MULTI_VOICE_SESSIONS := true
AUDIO_FEATURE_ENABLED_KPI_OPTIMIZE := true
AUDIO_FEATURE_ENABLED_ACDB_LICENSE := true
+AUDIO_FEATURE_ENABLED_DYNAMIC_LOG := true
MM_AUDIO_ENABLED_FTM := true
TARGET_USES_QCOM_MM_AUDIO := true
@@ -47,11 +48,10 @@
device/qcom/common/media/audio_policy.conf:system/etc/audio_policy.conf
else
PRODUCT_COPY_FILES += \
- hardware/qcom/audio/configs/msm8909/audio_policy.conf:system/etc/audio_policy.conf
+ hardware/qcom/audio/configs/msm8909/audio_policy.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy.conf
endif
PRODUCT_COPY_FILES += \
- hardware/qcom/audio/configs/msm8909/audio_policy.conf:system/etc/audio_policy.conf \
- hardware/qcom/audio/configs/msm8909/audio_effects.conf:system/vendor/etc/audio_effects.conf \
+ hardware/qcom/audio/configs/msm8909/audio_effects.conf:$(TARGET_COPY_OUT_VENDOR)/etc/audio_effects.conf \
hardware/qcom/audio/configs/msm8909/mixer_paths_qrd_skuh.xml:system/etc/mixer_paths_qrd_skuh.xml \
hardware/qcom/audio/configs/msm8909/mixer_paths_qrd_skui.xml:system/etc/mixer_paths_qrd_skui.xml \
hardware/qcom/audio/configs/msm8909/mixer_paths.xml:system/etc/mixer_paths.xml \
diff --git a/configs/msm8937/msm8937.mk b/configs/msm8937/msm8937.mk
index d2aab65..4b26d6c 100644
--- a/configs/msm8937/msm8937.mk
+++ b/configs/msm8937/msm8937.mk
@@ -55,6 +55,7 @@
TARGET_USES_QCOM_MM_AUDIO := true
AUDIO_FEATURE_ENABLED_SOURCE_TRACKING := true
BOARD_SUPPORTS_QAHW := true
+AUDIO_FEATURE_ENABLED_DYNAMIC_LOG := true
##AUDIO_FEATURE_FLAGS
#Audio Specific device overlays
diff --git a/configs/msm8953/msm8953.mk b/configs/msm8953/msm8953.mk
index cd1b62e..1adc471 100644
--- a/configs/msm8953/msm8953.mk
+++ b/configs/msm8953/msm8953.mk
@@ -55,6 +55,7 @@
TARGET_USES_QCOM_MM_AUDIO := true
AUDIO_FEATURE_ENABLED_SOURCE_TRACKING := true
BOARD_SUPPORTS_QAHW := true
+AUDIO_FEATURE_ENABLED_DYNAMIC_LOG := true
##AUDIO_FEATURE_FLAGS
#Audio Specific device overlays
diff --git a/configs/msm8996/msm8996.mk b/configs/msm8996/msm8996.mk
index 7591168..7f8d6ec 100644
--- a/configs/msm8996/msm8996.mk
+++ b/configs/msm8996/msm8996.mk
@@ -54,6 +54,7 @@
AUDIO_FEATURE_ENABLED_SOURCE_TRACKING := true
AUDIO_FEATURE_ENABLED_GEF_SUPPORT := true
BOARD_SUPPORTS_QAHW := true
+AUDIO_FEATURE_ENABLED_DYNAMIC_LOG := true
##AUDIO_FEATURE_FLAGS
#Audio Specific device overlays
diff --git a/configs/msm8998/mixer_paths_tavil.xml b/configs/msm8998/mixer_paths_tavil.xml
index 47f6fd1..27ef9b3 100644
--- a/configs/msm8998/mixer_paths_tavil.xml
+++ b/configs/msm8998/mixer_paths_tavil.xml
@@ -2465,6 +2465,31 @@
<path name="unprocessed-handset-mic" />
</path>
+ <!-- USB TTY start -->
+
+ <!-- full: both end tty -->
+ <path name="voice-tty-full-usb">
+ <ctl name="TTY Mode" value="FULL" />
+ <path name="usb-headphones" />
+ </path>
+
+ <path name="voice-tty-full-usb-mic">
+ <path name="usb-headset-mic" />
+ </path>
+
+ <!-- vco, in: handset mic use existing, out: tty -->
+ <path name="voice-tty-vco-usb">
+ <ctl name="TTY Mode" value="VCO" />
+ <path name="usb-headphones" />
+ </path>
+
+ <!-- hco, in: tty, out: speaker, use existing handset -->
+ <path name="voice-tty-hco-usb-mic">
+ <path name="voice-tty-full-usb-mic" />
+ </path>
+
+ <!-- USB TTY end -->
+
<!-- Added for ADSP testfwk -->
<path name="ADSP testfwk">
<ctl name="SLIMBUS_DL_HL Switch" value="1" />
diff --git a/configs/msm8998/msm8998.mk b/configs/msm8998/msm8998.mk
index 90dfc0f..509d798 100644
--- a/configs/msm8998/msm8998.mk
+++ b/configs/msm8998/msm8998.mk
@@ -29,7 +29,7 @@
AUDIO_FEATURE_ENABLED_HW_ACCELERATED_EFFECTS := false
AUDIO_FEATURE_ENABLED_AUDIOSPHERE := true
AUDIO_FEATURE_ENABLED_USB_TUNNEL_AUDIO := true
-AUDIO_FEATURE_ENABLED_SPLIT_A2DP := false
+AUDIO_FEATURE_ENABLED_SPLIT_A2DP := true
AUDIO_FEATURE_ENABLED_3D_AUDIO := false
AUDIO_FEATURE_ENABLED_VOICE_PRINT := false
USE_LEGACY_AUDIO_DAEMON := false
@@ -62,6 +62,7 @@
AUDIO_FEATURE_ENABLED_GEF_SUPPORT := true
BOARD_SUPPORTS_QAHW := true
AUDIO_FEATURE_ENABLED_RAS := true
+AUDIO_FEATURE_ENABLED_DYNAMIC_LOG := true
##AUDIO_FEATURE_FLAGS
#Audio Specific device overlays
diff --git a/configs/msm8998/sound_trigger_mixer_paths_wcd9340.xml b/configs/msm8998/sound_trigger_mixer_paths_wcd9340.xml
index 2e75a8c..64350fc 100644
--- a/configs/msm8998/sound_trigger_mixer_paths_wcd9340.xml
+++ b/configs/msm8998/sound_trigger_mixer_paths_wcd9340.xml
@@ -240,13 +240,13 @@
<ctl name="DMIC MUX5" value="DMIC1" />
<ctl name="CDC_IF TX6 MUX" value="DEC6" />
<ctl name="ADC MUX6" value="DMIC" />
- <ctl name="DMIC MUX6" value="DMIC0" />
+ <ctl name="DMIC MUX6" value="DMIC5" />
<ctl name="CDC_IF TX7 MUX" value="DEC7" />
<ctl name="ADC MUX7" value="DMIC" />
<ctl name="DMIC MUX7" value="DMIC2" />
<ctl name="CDC_IF TX8 MUX" value="DEC8" />
<ctl name="ADC MUX8" value="DMIC" />
- <ctl name="DMIC MUX8" value="DMIC5" />
+ <ctl name="DMIC MUX8" value="DMIC0" />
</path>
<path name="echo-reference">
diff --git a/configs/sdm660/audio_platform_info.xml b/configs/sdm660/audio_platform_info.xml
index dd0d974..00d64c3 100644
--- a/configs/sdm660/audio_platform_info.xml
+++ b/configs/sdm660/audio_platform_info.xml
@@ -121,12 +121,12 @@
<device name="SND_DEVICE_OUT_ANC_FB_HEADSET" interface="INT0_MI2S_RX"/>
<device name="SND_DEVICE_OUT_VOICE_ANC_FB_HEADSET" interface="INT0_MI2S_RX"/>
<device name="SND_DEVICE_OUT_ANC_HANDSET" interface="INT4_MI2S_RX"/>
- <device name="SND_DEVICE_OUT_SPEAKER_PROTECTED" interface="INT5_MI2S_TX"/>
- <device name="SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED" interface="INT5_MI2S_TX"/>
- <device name="SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED" interface="INT5_MI2S_TX"/>
- <device name="SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT" interface="INT5_MI2S_TX"/>
- <device name="SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT" interface="INT5_MI2S_TX"/>
- <device name="SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT" interface="INT5_MI2S_TX"/>
+ <device name="SND_DEVICE_OUT_SPEAKER_PROTECTED" interface="INT4_MI2S_RX"/>
+ <device name="SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED" interface="INT4_MI2S_RX"/>
+ <device name="SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED" interface="INT4_MI2S_RX"/>
+ <device name="SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT" interface="INT4_MI2S_RX"/>
+ <device name="SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT" interface="INT4_MI2S_RX"/>
+ <device name="SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT" interface="INT4_MI2S_RX"/>
<device name="SND_DEVICE_OUT_SPEAKER_WSA" interface="INT4_MI2S_RX"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_WSA" interface="INT4_MI2S_RX"/>
<device name="SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA" interface="INT4_MI2S_RX"/>
diff --git a/configs/sdm660/mixer_paths_skush.xml b/configs/sdm660/mixer_paths_skush.xml
index 546a9c4..df864b8 100644
--- a/configs/sdm660/mixer_paths_skush.xml
+++ b/configs/sdm660/mixer_paths_skush.xml
@@ -297,7 +297,7 @@
<ctl name="HPHL Volume" value="9" />
<ctl name="HPHR Volume" value="9" />
<ctl name="EAR PA Gain" value="POS_1P5_DB" />
- <ctl name="EAR PA Boost" value="ENABLE" />
+ <ctl name="EAR PA Boost" value="DISABLE" />
<ctl name="RX1 Digital Volume" value="84" />
<ctl name="RX2 Digital Volume" value="84" />
@@ -1790,7 +1790,7 @@
<path name="handset">
<ctl name="INT0_MI2S_RX Channels" value="One" />
- <ctl name="EAR PA Boost" value="ENABLE" />
+ <ctl name="EAR PA Boost" value="DISABLE" />
<ctl name="RX1 MIX1 INP1" value="RX1" />
<ctl name="RDAC2 MUX" value="RX1" />
<ctl name="EAR_S" value="Switch" />
diff --git a/configs/sdm660/sdm660.mk b/configs/sdm660/sdm660.mk
index c0bbd86..a0ae641 100644
--- a/configs/sdm660/sdm660.mk
+++ b/configs/sdm660/sdm660.mk
@@ -29,7 +29,7 @@
AUDIO_FEATURE_ENABLED_HW_ACCELERATED_EFFECTS := false
AUDIO_FEATURE_ENABLED_AUDIOSPHERE := true
AUDIO_FEATURE_ENABLED_USB_TUNNEL_AUDIO := true
-AUDIO_FEATURE_ENABLED_SPLIT_A2DP := false
+AUDIO_FEATURE_ENABLED_SPLIT_A2DP := true
AUDIO_FEATURE_ENABLED_3D_AUDIO := false
AUDIO_FEATURE_ENABLED_VOICE_PRINT := false
USE_LEGACY_AUDIO_DAEMON := false
diff --git a/configs/sdm660/sound_trigger_mixer_paths_wcd9335.xml b/configs/sdm660/sound_trigger_mixer_paths_wcd9335.xml
index 0b381cf..691b2e3 100644
--- a/configs/sdm660/sound_trigger_mixer_paths_wcd9335.xml
+++ b/configs/sdm660/sound_trigger_mixer_paths_wcd9335.xml
@@ -103,7 +103,7 @@
<ctl name="MADONOFF Switch" value="1" />
<ctl name="TX13 INP MUX" value="CPE_TX_PP" />
<ctl name="AIF4_MAD Mixer SLIM TX13" value="1" />
- <ctl name="MAD Input" value="DMIC0" />
+ <ctl name="MAD Input" value="DMIC2" />
<ctl name="CPE AFE MAD Enable" value="1"/>
</path>
@@ -111,14 +111,14 @@
<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" />
+ <ctl name="DMIC MUX1" value="DMIC2" />
</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="MAD Input" value="DMIC2" />
<ctl name="CPE AFE MAD Enable" value="1"/>
</path>
@@ -126,7 +126,7 @@
<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" />
+ <ctl name="MAD Input" value="DMIC2" />
</path>
</mixer>
diff --git a/configs/sdm660/sound_trigger_mixer_paths_wcd9340.xml b/configs/sdm660/sound_trigger_mixer_paths_wcd9340.xml
index 545f46b..f328bd6 100644
--- a/configs/sdm660/sound_trigger_mixer_paths_wcd9340.xml
+++ b/configs/sdm660/sound_trigger_mixer_paths_wcd9340.xml
@@ -171,7 +171,7 @@
</path>
<path name="listen-cpe-handset-mic">
- <ctl name="MAD Input" value="DMIC0" />
+ <ctl name="MAD Input" value="DMIC2" />
<ctl name="MAD_SEL MUX" value="SPE" />
<ctl name="MAD_INP MUX" value="MAD" />
<ctl name="MAD_CPE1 Switch" value="1" />
@@ -181,19 +181,19 @@
<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" />
+ <ctl name="DMIC MUX1" value="DMIC2" />
</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="MAD Input" value="DMIC2" />
<ctl name="CPE AFE MAD Enable" value="1"/>
</path>
<path name="listen-ape-handset-mic">
- <ctl name="MAD Input" value="DMIC0" />
+ <ctl name="MAD Input" value="DMIC2" />
<ctl name="MAD_SEL MUX" value="MSM" />
<ctl name="MAD_INP MUX" value="MAD" />
<ctl name="MAD_BROADCAST Switch" value="1" />
diff --git a/configs/sdm845/mixer_paths_tavil.xml b/configs/sdm845/mixer_paths_tavil.xml
index fbe3976..18a9073 100644
--- a/configs/sdm845/mixer_paths_tavil.xml
+++ b/configs/sdm845/mixer_paths_tavil.xml
@@ -2234,6 +2234,31 @@
<path name="unprocessed-handset-mic" />
</path>
+ <!-- USB TTY start -->
+
+ <!-- full: both end tty -->
+ <path name="voice-tty-full-usb">
+ <ctl name="TTY Mode" value="FULL" />
+ <path name="usb-headphones" />
+ </path>
+
+ <path name="voice-tty-full-usb-mic">
+ <path name="usb-headset-mic" />
+ </path>
+
+ <!-- vco, in: handset mic use existing, out: tty -->
+ <path name="voice-tty-vco-usb">
+ <ctl name="TTY Mode" value="VCO" />
+ <path name="usb-headphones" />
+ </path>
+
+ <!-- hco, in: tty, out: speaker, use existing handset -->
+ <path name="voice-tty-hco-usb-mic">
+ <path name="voice-tty-full-usb-mic" />
+ </path>
+
+ <!-- USB TTY end -->
+
<!-- Added for ADSP testfwk -->
<path name="ADSP testfwk">
<ctl name="SLIMBUS_DL_HL Switch" value="1" />
diff --git a/configs/sdm845/sdm845.mk b/configs/sdm845/sdm845.mk
index d69a6fd..257115a 100644
--- a/configs/sdm845/sdm845.mk
+++ b/configs/sdm845/sdm845.mk
@@ -29,7 +29,7 @@
AUDIO_FEATURE_ENABLED_HW_ACCELERATED_EFFECTS := false
AUDIO_FEATURE_ENABLED_AUDIOSPHERE := true
AUDIO_FEATURE_ENABLED_USB_TUNNEL_AUDIO := true
-AUDIO_FEATURE_ENABLED_SPLIT_A2DP := false
+AUDIO_FEATURE_ENABLED_SPLIT_A2DP := true
AUDIO_FEATURE_ENABLED_3D_AUDIO := false
DOLBY_ENABLE := false
endif
diff --git a/hal/Android.mk b/hal/Android.mk
index 9a8d27c..908619d 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -54,7 +54,8 @@
audio_hw.c \
voice.c \
platform_info.c \
- $(AUDIO_PLATFORM)/platform.c
+ $(AUDIO_PLATFORM)/platform.c \
+ acdb.c
LOCAL_SRC_FILES += audio_extn/audio_extn.c \
audio_extn/utils.c
@@ -351,6 +352,12 @@
LOCAL_SRC_FILES += audio_extn/adsp_hdlr.c
endif
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_DYNAMIC_LOG)), true)
+ LOCAL_CFLAGS += -DDYNAMIC_LOG_ENABLED
+ LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/audio-log-utils
+ LOCAL_SHARED_LIBRARIES += libaudio_log_utils
+endif
+
LOCAL_CFLAGS += -Wall -Werror
LOCAL_COPY_HEADERS_TO := mm-audio
diff --git a/hal/Makefile.am b/hal/Makefile.am
index cbce291..0edf504 100644
--- a/hal/Makefile.am
+++ b/hal/Makefile.am
@@ -11,7 +11,8 @@
platform_info.c \
${TARGET_PLATFORM}/platform.c \
audio_extn/audio_extn.c \
- audio_extn/utils.c
+ audio_extn/utils.c \
+ acdb.c
if HDMI_EDID
AM_CFLAGS += -DHDMI_EDID
diff --git a/hal/acdb.c b/hal/acdb.c
new file mode 100644
index 0000000..cbb96bd
--- /dev/null
+++ b/hal/acdb.c
@@ -0,0 +1,185 @@
+/*
+ * Copyright (c) 2013-2017, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audio_hw_acdb"
+//#define LOG_NDEBUG 0
+#define LOG_NDDEBUG 0
+
+#include <stdlib.h>
+#include <dlfcn.h>
+#include <cutils/log.h>
+#include <cutils/list.h>
+#include "acdb.h"
+#include "platform_api.h"
+
+int acdb_init(int snd_card_num)
+{
+
+ int result = -1;
+ char *cvd_version = NULL;
+
+ char *snd_card_name = NULL;
+ struct mixer *mixer = NULL;
+ struct acdb_platform_data *my_data = NULL;
+
+ if(snd_card_num < 0) {
+ ALOGE("invalid sound card number");
+ return result;
+ }
+
+ mixer = mixer_open(snd_card_num);
+ if (!mixer) {
+ ALOGE("%s: Unable to open the mixer card: %d", __func__,
+ snd_card_num);
+ goto cleanup;
+ }
+
+ my_data = calloc(1, sizeof(struct acdb_platform_data));
+ if (!my_data) {
+ ALOGE("failed to allocate acdb platform data");
+ goto cleanup;
+ }
+
+ list_init(&my_data->acdb_meta_key_list);
+
+ /* Extract META KEY LIST INFO */
+ platform_info_init(PLATFORM_INFO_XML_PATH, my_data, ACDB_EXTN);
+
+ my_data->acdb_handle = dlopen(LIB_ACDB_LOADER, RTLD_NOW);
+ if (my_data->acdb_handle == NULL) {
+ ALOGE("%s: DLOPEN failed for %s", __func__, LIB_ACDB_LOADER);
+ goto cleanup;
+ }
+
+ ALOGV("%s: DLOPEN successful for %s", __func__, LIB_ACDB_LOADER);
+
+ my_data->acdb_init_v3 = (acdb_init_v3_t)dlsym(my_data->acdb_handle,
+ "acdb_loader_init_v3");
+ if (my_data->acdb_init_v3 == NULL)
+ ALOGE("%s: dlsym error %s for acdb_loader_init_v3", __func__, dlerror());
+
+ my_data->acdb_init_v2 = (acdb_init_v2_t)dlsym(my_data->acdb_handle,
+ "acdb_loader_init_v2");
+ if (my_data->acdb_init_v2 == NULL)
+ ALOGE("%s: dlsym error %s for acdb_loader_init_v2", __func__, dlerror());
+
+ my_data->acdb_init = (acdb_init_t)dlsym(my_data->acdb_handle,
+ "acdb_loader_init_ACDB");
+ if (my_data->acdb_init == NULL && my_data->acdb_init_v2 == NULL
+ && my_data->acdb_init_v3 == NULL) {
+ ALOGE("%s: dlsym error %s for acdb_loader_init_ACDB", __func__, dlerror());
+ goto cleanup;
+ }
+
+ /* Get CVD version */
+ cvd_version = calloc(1, MAX_CVD_VERSION_STRING_SIZE);
+ if (!cvd_version) {
+ ALOGE("%s: Failed to allocate cvd version", __func__);
+ goto cleanup;
+ } else {
+ struct mixer_ctl *ctl = NULL;
+ int count = 0;
+
+ ctl = mixer_get_ctl_by_name(mixer, CVD_VERSION_MIXER_CTL);
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s", __func__, CVD_VERSION_MIXER_CTL);
+ goto cleanup;
+ }
+ mixer_ctl_update(ctl);
+
+ count = mixer_ctl_get_num_values(ctl);
+ if (count > MAX_CVD_VERSION_STRING_SIZE)
+ count = MAX_CVD_VERSION_STRING_SIZE;
+
+ result = mixer_ctl_get_array(ctl, cvd_version, count);
+ if (result != 0) {
+ ALOGE("%s: ERROR! mixer_ctl_get_array() failed to get CVD Version", __func__);
+ goto cleanup;
+ }
+ }
+
+ /* Get Sound card name */
+ snd_card_name = strdup(mixer_get_name(mixer));
+ if (!snd_card_name) {
+ ALOGE("failed to allocate memory for snd_card_name");
+ result = -1;
+ goto cleanup;
+ }
+
+ int key = 0;
+ struct listnode *node = NULL;
+ struct meta_key_list *key_info = NULL;
+
+ if (my_data->acdb_init_v3) {
+ result = my_data->acdb_init_v3(snd_card_name, cvd_version,
+ &my_data->acdb_meta_key_list);
+ } else if (my_data->acdb_init_v2) {
+ node = list_head(&my_data->acdb_meta_key_list);
+ key_info = node_to_item(node, struct meta_key_list, list);
+ key = key_info->cal_info.nKey;
+ result = my_data->acdb_init_v2(snd_card_name, cvd_version, key);
+ } else {
+ result = my_data->acdb_init();
+ }
+
+cleanup:
+ if (NULL != my_data) {
+ if (my_data->acdb_handle)
+ dlclose(my_data->acdb_handle);
+
+ struct listnode *node;
+ struct meta_key_list *key_info;
+ list_for_each(node, &my_data->acdb_meta_key_list) {
+ key_info = node_to_item(node, struct meta_key_list, list);
+ free(key_info);
+ }
+ free(my_data);
+ }
+
+ if (mixer)
+ mixer_close(mixer);
+
+ if (cvd_version)
+ free(cvd_version);
+
+ if (snd_card_name)
+ free(snd_card_name);
+
+ return result;
+}
+
+int acdb_set_metainfo_key(void *platform, char *name, int key) {
+
+ struct meta_key_list *key_info = (struct meta_key_list *)
+ calloc(1, sizeof(struct meta_key_list));
+ struct acdb_platform_data *pdata = (struct acdb_platform_data *)platform;
+ if (!key_info) {
+ ALOGE("%s: Could not allocate memory for key %d", __func__, key);
+ return -ENOMEM;
+ }
+
+ key_info->cal_info.nKey = key;
+ strlcpy(key_info->name, name, sizeof(key_info->name));
+ list_add_tail(&pdata->acdb_meta_key_list, &key_info->list);
+
+ ALOGD("%s: successfully added module %s and key %d to the list", __func__,
+ key_info->name, key_info->cal_info.nKey);
+
+ return 0;
+}
diff --git a/hal/acdb.h b/hal/acdb.h
new file mode 100644
index 0000000..44a51ed
--- /dev/null
+++ b/hal/acdb.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (c) 2013-2017, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ACDB_H
+#define ACDB_H
+
+#include <linux/msm_audio_calibration.h>
+
+#define MAX_CVD_VERSION_STRING_SIZE 100
+#define LIB_ACDB_LOADER "libacdbloader.so"
+#define CVD_VERSION_MIXER_CTL "CVD Version"
+#define ACDB_METAINFO_KEY_MODULE_NAME_LEN 100
+
+#ifdef LINUX_ENABLED
+#define PLATFORM_INFO_XML_PATH "/etc/audio_platform_info.xml"
+#else
+#define PLATFORM_INFO_XML_PATH "/system/etc/audio_platform_info.xml"
+#endif
+
+/* Audio calibration related functions */
+typedef void (*acdb_deallocate_t)();
+typedef int (*acdb_init_t)();
+typedef int (*acdb_init_v2_t)(const char *, char *, int);
+typedef int (*acdb_init_v3_t)(const char *, char *, struct listnode *);
+typedef void (*acdb_send_audio_cal_t)(int, int, int , int);
+typedef void (*acdb_send_audio_cal_v3_t)(int, int, int, int, int);
+typedef void (*acdb_send_voice_cal_t)(int, int);
+typedef int (*acdb_reload_vocvoltable_t)(int);
+typedef int (*acdb_get_default_app_type_t)(void);
+typedef int (*acdb_loader_get_calibration_t)(char *attr, int size, void *data);
+typedef int (*acdb_set_audio_cal_t) (void *, void *, uint32_t);
+typedef int (*acdb_get_audio_cal_t) (void *, void *, uint32_t*);
+typedef int (*acdb_send_common_top_t) (void);
+typedef int (*acdb_set_codec_data_t) (void *, char *);
+typedef int (*acdb_reload_t) (char *, char *, char *, int);
+typedef int (*acdb_reload_v2_t) (char *, char *, char *, struct listnode *);
+typedef int (*acdb_send_gain_dep_cal_t)(int, int, int, int, int);
+
+struct meta_key_list {
+ struct listnode list;
+ struct audio_cal_info_metainfo cal_info;
+ char name[ACDB_METAINFO_KEY_MODULE_NAME_LEN];
+};
+
+struct acdb_platform_data {
+ /* Audio calibration related functions */
+ void *acdb_handle;
+ acdb_init_t acdb_init;
+ acdb_init_v2_t acdb_init_v2;
+ acdb_init_v3_t acdb_init_v3;
+ struct listnode acdb_meta_key_list;
+};
+
+int acdb_init(int);
+
+int acdb_set_metainfo_key(void *platform, char *name, int key);
+#endif //ACDB_H
diff --git a/hal/audio_extn/a2dp.c b/hal/audio_extn/a2dp.c
index fba7e6c..1ffa968 100644
--- a/hal/audio_extn/a2dp.c
+++ b/hal/audio_extn/a2dp.c
@@ -41,6 +41,12 @@
#include <hardware/hardware.h>
#include <cutils/properties.h>
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_A2DP
+#include <log_utils.h>
+#endif
+
#ifdef SPLIT_A2DP_ENABLED
#define AUDIO_PARAMETER_A2DP_STARTED "A2dpStarted"
#define BT_IPC_LIB_NAME "libbthost_if.so"
@@ -69,7 +75,6 @@
#define MIXER_ENC_FMT_APTXHD "APTXHD"
#define MIXER_ENC_FMT_NONE "NONE"
-
typedef int (*audio_stream_open_t)(void);
typedef int (*audio_stream_close_t)(void);
typedef int (*audio_start_stream_t)(void);
@@ -172,6 +177,46 @@
uint32_t custom_size;
};
+/* TODO: Define the following structures only for O using PLATFORM_VERSION */
+/* Information about BT SBC encoder configuration
+ * This data is used between audio HAL module and
+ * BT IPC library to configure DSP encoder
+ */
+typedef struct {
+ uint32_t subband; /* 4, 8 */
+ uint32_t blk_len; /* 4, 8, 12, 16 */
+ uint16_t sampling_rate; /*44.1khz,48khz*/
+ uint8_t channels; /*0(Mono),1(Dual_mono),2(Stereo),3(JS)*/
+ uint8_t alloc; /*0(Loudness),1(SNR)*/
+ uint8_t min_bitpool; /* 2 */
+ uint8_t max_bitpool; /*53(44.1khz),51 (48khz) */
+ uint32_t bitrate; /* 320kbps to 512kbps */
+} audio_sbc_encoder_config;
+
+
+/* Information about BT APTX encoder configuration
+ * This data is used between audio HAL module and
+ * BT IPC library to configure DSP encoder
+ */
+typedef struct {
+ uint16_t sampling_rate;
+ uint8_t channels;
+ uint32_t bitrate;
+} audio_aptx_encoder_config;
+
+
+/* Information about BT AAC encoder configuration
+ * This data is used between audio HAL module and
+ * BT IPC library to configure DSP encoder
+ */
+typedef struct {
+ uint32_t enc_mode; /* LC, SBR, PS */
+ uint16_t format_flag; /* RAW, ADTS */
+ uint16_t channels; /* 1-Mono, 2-Stereo */
+ uint32_t sampling_rate;
+ uint32_t bitrate;
+} audio_aac_encoder_config;
+
/*********** END of DSP configurable structures ********************/
/* API to identify DSP encoder captabilities */
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index 4573ecc..3c9330c 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -55,6 +55,12 @@
#include "sound/compress_params.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_AUDIO_EXTN
+#include <log_utils.h>
+#endif
+
#define MAX_SLEEP_RETRY 100
#define WIFI_INIT_WAIT_SLEEP 50
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index bdb039f..e8210ac 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -224,7 +224,7 @@
#else
void audio_extn_a2dp_init(void *adev);
int audio_extn_a2dp_start_playback();
-void audio_extn_a2dp_stop_playback();
+int audio_extn_a2dp_stop_playback();
void audio_extn_a2dp_set_parameters(struct str_parms *parms);
bool audio_extn_a2dp_is_force_device_switch();
void audio_extn_a2dp_set_handoff_mode(bool is_on);
@@ -587,6 +587,8 @@
void audio_extn_utils_update_stream_app_type_cfg_for_usecase(
struct audio_device *adev,
struct audio_usecase *usecase);
+int audio_extn_utils_get_snd_card_num();
+
#ifdef DS2_DOLBY_DAP_ENABLED
#define LIB_DS2_DAP_HAL "vendor/lib/libhwdaphal.so"
#define SET_HW_INFO_FUNC "dap_hal_set_hw_info"
diff --git a/hal/audio_extn/bt_hal.c b/hal/audio_extn/bt_hal.c
index 21baa9c..6441bef 100644
--- a/hal/audio_extn/bt_hal.c
+++ b/hal/audio_extn/bt_hal.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2016, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2016-2017, 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
@@ -41,6 +41,12 @@
#include <../../../../system/bt/audio_a2dp_hw/bthost_ipc.h>
#include <dlfcn.h>
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_BT_HAL
+#include <log_utils.h>
+#endif
+
#define DEFAULT_BUF_SIZE 6144
#define UNUSED(x) (void)(x)
diff --git a/hal/audio_extn/compress_capture.c b/hal/audio_extn/compress_capture.c
index 47e6a9d..2d43446 100644
--- a/hal/audio_extn/compress_capture.c
+++ b/hal/audio_extn/compress_capture.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013 - 2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013 - 2014, 2017, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -35,6 +35,12 @@
#include "sound/compress_params.h"
#include "sound/compress_offload.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_COMPR_CAP
+#include <log_utils.h>
+#endif
+
#ifdef COMPRESS_CAPTURE_ENABLED
#define COMPRESS_IN_CONFIG_CHANNELS 1
diff --git a/hal/audio_extn/compress_in.c b/hal/audio_extn/compress_in.c
index 6b1f6e4..156e3bc 100644
--- a/hal/audio_extn/compress_in.c
+++ b/hal/audio_extn/compress_in.c
@@ -1,5 +1,5 @@
/*
-* Copyright (c) 2016, The Linux Foundation. All rights reserved.
+* Copyright (c) 2016-2017, 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
@@ -51,6 +51,11 @@
#include "audio_defs.h"
#include "sound/compress_params.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_COMPR_IN
+#include <log_utils.h>
+#endif
/* default timestamp metadata definition if not defined in kernel*/
#ifndef COMPRESSED_TIMESTAMP_FLAG
#define COMPRESSED_TIMESTAMP_FLAG 0
diff --git a/hal/audio_extn/dev_arbi.c b/hal/audio_extn/dev_arbi.c
index 69d8568..9c5382a 100644
--- a/hal/audio_extn/dev_arbi.c
+++ b/hal/audio_extn/dev_arbi.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2014, 2016 The Linux Foundation. All rights reserved.
+ * Copyright (c) 2014, 2016-2017 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
@@ -43,6 +43,12 @@
#include <cutils/properties.h>
#include "audio_extn.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_DEV_ARBI
+#include <log_utils.h>
+#endif
+
#ifdef DEV_ARBI_ENABLED
typedef int (init_fn_t)();
diff --git a/hal/audio_extn/dolby.c b/hal/audio_extn/dolby.c
index fee0543..a0f17be 100644
--- a/hal/audio_extn/dolby.c
+++ b/hal/audio_extn/dolby.c
@@ -34,6 +34,12 @@
#include "sound/compress_params.h"
#include "sound/devdep_params.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_DOLBY
+#include <log_utils.h>
+#endif
+
#ifdef DS1_DOLBY_DDP_ENABLED
#define AUDIO_PARAMETER_DDP_DEV "ddp_device"
diff --git a/hal/audio_extn/dts_eagle.c b/hal/audio_extn/dts_eagle.c
index 71bfea6..b8de2ca 100644
--- a/hal/audio_extn/dts_eagle.c
+++ b/hal/audio_extn/dts_eagle.c
@@ -33,6 +33,12 @@
#include "platform.h"
#include "platform_api.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_DTS_EAGLE
+#include <log_utils.h>
+#endif
+
#ifdef DTS_EAGLE
#define AUDIO_PARAMETER_KEY_DTS_EAGLE "DTS_EAGLE"
diff --git a/hal/audio_extn/fm.c b/hal/audio_extn/fm.c
index a28d52f..5da494d 100644
--- a/hal/audio_extn/fm.c
+++ b/hal/audio_extn/fm.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2016, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2017, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -31,6 +31,12 @@
#include <stdlib.h>
#include <cutils/str_parms.h>
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_FM
+#include <log_utils.h>
+#endif
+
#ifdef FM_POWER_OPT
#define AUDIO_PARAMETER_KEY_HANDLE_FM "handle_fm"
#define AUDIO_PARAMETER_KEY_FM_VOLUME "fm_volume"
diff --git a/hal/audio_extn/gef.c b/hal/audio_extn/gef.c
index d5e090a..19f9dfb 100644
--- a/hal/audio_extn/gef.c
+++ b/hal/audio_extn/gef.c
@@ -47,6 +47,12 @@
#include "audio_extn.h"
#include "audio_hw.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_GEF
+#include <log_utils.h>
+#endif
+
#ifdef AUDIO_GENERIC_EFFECT_FRAMEWORK_ENABLED
#if LINUX_ENABLED
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index 3c1d0ef..685078b 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -39,6 +39,12 @@
#include <stdlib.h>
#include <cutils/str_parms.h>
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_HFP
+#include <log_utils.h>
+#endif
+
#ifdef HFP_ENABLED
#define AUDIO_PARAMETER_HFP_ENABLE "hfp_enable"
#define AUDIO_PARAMETER_HFP_SET_SAMPLING_RATE "hfp_set_sampling_rate"
diff --git a/hal/audio_extn/keep_alive.c b/hal/audio_extn/keep_alive.c
index bcc12d4..87cb122 100644
--- a/hal/audio_extn/keep_alive.c
+++ b/hal/audio_extn/keep_alive.c
@@ -29,6 +29,7 @@
#define LOG_TAG "keep_alive"
/*#define LOG_NDEBUG 0*/
+
#include <stdlib.h>
#include <cutils/log.h>
#include "audio_hw.h"
@@ -36,6 +37,12 @@
#include "platform_api.h"
#include <platform.h>
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_KEEP_ALIVE
+#include <log_utils.h>
+#endif
+
#define SILENCE_INTERVAL 2 /*In secs*/
typedef enum {
diff --git a/hal/audio_extn/listen.c b/hal/audio_extn/listen.c
index 4cb2d2d..b98a429 100644
--- a/hal/audio_extn/listen.c
+++ b/hal/audio_extn/listen.c
@@ -1,4 +1,4 @@
-/* Copyright (c) 2013-2014, The Linux Foundation. All rights reserved.
+/* Copyright (c) 2013-2014, 2017, 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
@@ -41,6 +41,11 @@
#include "platform.h"
#include "platform_api.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_AUDIO_LISTEN
+#include <log_utils.h>
+#endif
#ifdef AUDIO_LISTEN_ENABLED
diff --git a/hal/audio_extn/passthru.c b/hal/audio_extn/passthru.c
index dd4d4d4..24208ab 100644
--- a/hal/audio_extn/passthru.c
+++ b/hal/audio_extn/passthru.c
@@ -40,6 +40,11 @@
#include <cutils/properties.h>
#include "sound/compress_params.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_PASSTH
+#include <log_utils.h>
+#endif
static const audio_format_t audio_passthru_formats[] = {
AUDIO_FORMAT_AC3,
diff --git a/hal/audio_extn/pm.c b/hal/audio_extn/pm.c
index 69e19cb..65aa1fe 100644
--- a/hal/audio_extn/pm.c
+++ b/hal/audio_extn/pm.c
@@ -34,6 +34,12 @@
#include <cutils/log.h>
#include <cutils/str_parms.h>
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_PM
+#include <log_utils.h>
+#endif
+
/* Device state*/
#define AUDIO_PARAMETER_KEY_DEV_SHUTDOWN "dev_shutdown"
diff --git a/hal/audio_extn/qaf.c b/hal/audio_extn/qaf.c
index caf64ee..d0a9a95 100644
--- a/hal/audio_extn/qaf.c
+++ b/hal/audio_extn/qaf.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2017, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2016-2017, 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
@@ -36,6 +36,7 @@
#define ALOGVV(a...) do { } while(0)
#endif
+#define DEBUG_MSG_VV(arg,...) ALOGVV("%s: %d: " arg, __func__, __LINE__, ##__VA_ARGS__)
#define DEBUG_MSG(arg,...) ALOGV("%s: %d: " arg, __func__, __LINE__, ##__VA_ARGS__)
#define ERROR_MSG(arg,...) ALOGE("%s: %d: " arg, __func__, __LINE__, ##__VA_ARGS__)
@@ -58,6 +59,7 @@
#define QAF_MODULE_PCM_INPUT_BUFFER_LATENCY 32
#define MS12_PCM_OUT_FRAGMENT_SIZE 1536 //samples
+#define MS12_PCM_IN_FRAGMENT_SIZE 1536 //samples
#define DD_FRAME_SIZE 1536
#define DDP_FRAME_SIZE DD_FRAME_SIZE
@@ -115,6 +117,12 @@
#include <qti_audio.h>
#include "sound/compress_params.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_QAF
+#include <log_utils.h>
+#endif
+
//TODO: Need to remove this.
#define QAF_OUTPUT_SAMPLING_RATE 48000
@@ -147,9 +155,9 @@
typedef enum {
QAF_IN_MAIN = 0, /* Single PID Main/Primary or Dual-PID stream */
- QAF_IN_ASSOC, /* Associated/Secondary stream */
- QAF_IN_PCM, /* PCM stream. */
-
+ QAF_IN_ASSOC, /* Associated/Secondary stream */
+ QAF_IN_PCM, /* PCM stream. */
+ QAF_IN_MAIN_2, /* Single PID Main2 stream */
MAX_QAF_MODULE_IN
} mm_module_input_type;
@@ -305,6 +313,16 @@
}
}
+static bool is_main_active(struct qaf_module* qaf_mod)
+{
+ return (qaf_mod->stream_in[QAF_IN_MAIN] || qaf_mod->stream_in[QAF_IN_MAIN_2]);
+}
+
+static bool is_dual_main_active(struct qaf_module* qaf_mod)
+{
+ return (qaf_mod->stream_in[QAF_IN_MAIN] && qaf_mod->stream_in[QAF_IN_MAIN_2]);
+}
+
/* Gets the pcm output buffer size(in samples) for the mm module. */
static uint32_t get_pcm_output_buffer_size_samples(struct qaf_module *qaf_mod)
{
@@ -491,7 +509,7 @@
/* Sends a command to output stream offload thread. */
static int qaf_send_offload_cmd_l(struct stream_out* out, int command)
{
- DEBUG_MSG("command is %d", command);
+ DEBUG_MSG_VV("command is %d", command);
struct offload_cmd *cmd = (struct offload_cmd *)calloc(1, sizeof(struct offload_cmd));
@@ -637,7 +655,6 @@
{
int ret = -EINVAL;
struct qaf_module *qaf_mod = NULL;
- DEBUG_MSG("bytes = %d [%p]", bytes, out->qaf_stream_handle);
qaf_mod = get_qaf_module_for_input_stream(out);
if ((!qaf_mod) || (!qaf_mod->qaf_audio_stream_write)) {
@@ -657,7 +674,7 @@
struct audio_device *adev = out->dev;
ssize_t ret = 0;
- DEBUG_MSG("bytes = %d, usecase[%d] and flags[%x] for handle[%p]",
+ DEBUG_MSG_VV("bytes = %d, usecase[%d] and flags[%x] for handle[%p]",
(int)bytes, out->usecase, out->flags, out);
lock_output_stream(out);
@@ -691,7 +708,7 @@
}
ret = qaf_module_write_input_buffer(out, buffer, bytes);
- DEBUG_MSG("ret [%d]", (int)ret);
+ DEBUG_MSG_VV("ret [%d]", (int)ret);
if (ret >= 0) {
bytes = ret;
@@ -704,7 +721,7 @@
if (ret < 0) {
if (ret == -EAGAIN) {
- DEBUG_MSG("No space available in mm module, post msg to cb thread");
+ DEBUG_MSG_VV("No space available in mm module, post msg to cb thread");
ret = qaf_send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
bytes = 0;
} else if (ret == -ENOMEM || ret == -EPERM) {
@@ -719,11 +736,10 @@
return bytes;
}
-/* Gets PCM offload buffer size for QAF module output. */
-static uint32_t qaf_get_pcm_offload_buffer_size(struct qaf_module *qaf_mod,
- audio_offload_info_t* info)
+/* Gets PCM offload buffer size for a given config. */
+static uint32_t qaf_get_pcm_offload_buffer_size(audio_offload_info_t* info,
+ uint32_t samples_per_frame)
{
- uint32_t samples_per_frame = get_pcm_output_buffer_size_samples(qaf_mod);
uint32_t fragment_size = 0;
fragment_size = (samples_per_frame * (info->bit_width >> 3) * popcount(info->channel_mask));
@@ -739,11 +755,22 @@
fragment_size = ALIGN(fragment_size,
((info->bit_width >> 3) * popcount(info->channel_mask) * 32));
- ALOGI("Qaf PCM offload Fragment size to %d bytes", fragment_size);
+ ALOGI("Qaf PCM offload Fragment size is %d bytes", fragment_size);
return fragment_size;
}
+static uint32_t qaf_get_pcm_offload_input_buffer_size(info)
+{
+ return qaf_get_pcm_offload_buffer_size(info, MS12_PCM_IN_FRAGMENT_SIZE);
+}
+
+static uint32_t qaf_get_pcm_offload_output_buffer_size(struct qaf_module *qaf_mod,
+ audio_offload_info_t* info)
+{
+ return qaf_get_pcm_offload_buffer_size(info, get_pcm_output_buffer_size_samples(qaf_mod));
+}
+
/* Gets buffer latency in samples. */
static int get_buffer_latency(struct stream_out *out, uint32_t buffer_size, uint32_t *latency)
{
@@ -1047,7 +1074,7 @@
struct stream_out *out = (struct stream_out *)stream;
uint32_t latency = 0;
struct qaf_module *qaf_mod = NULL;
- DEBUG_MSG("Output Stream %p", out);
+ DEBUG_MSG_VV("Output Stream %p", out);
qaf_mod = get_qaf_module_for_input_stream(out);
if (!qaf_mod) {
@@ -1091,7 +1118,7 @@
}
}
- DEBUG_MSG("Latency %d", latency);
+ DEBUG_MSG_VV("Latency %d", latency);
return latency;
}
@@ -1139,12 +1166,10 @@
struct audio_stream_out *bt_stream = NULL;
int format;
- DEBUG_MSG("Device 0x%X, Event = 0x%X", device, event_id);
+ DEBUG_MSG_VV("Device 0x%X, Event = 0x%X, Bytes to write %d", device, event_id, size);
pthread_mutex_lock(&p_qaf->lock);
if (event_id == AUDIO_DATA_EVENT) {
- DEBUG_MSG("Device id 0x%X, bytes to write %d", device, size);
-
if (p_qaf->passthrough_out != NULL) {
//If QAF passthrough is active then all the module output will be dropped.
pthread_mutex_unlock(&p_qaf->lock);
@@ -1284,17 +1309,27 @@
qaf_mod->stream_out[QAF_OUT_OFFLOAD_MCH]->compr_config.fragments =
COMPRESS_OFFLOAD_NUM_FRAGMENTS;
qaf_mod->stream_out[QAF_OUT_OFFLOAD_MCH]->compr_config.fragment_size =
- qaf_get_pcm_offload_buffer_size(qaf_mod, &config.offload_info);
+ qaf_get_pcm_offload_output_buffer_size(qaf_mod, &config.offload_info);
p_qaf->mch_pcm_hdmi_enabled = true;
- if (qaf_mod->stream_in[QAF_IN_MAIN]
- && qaf_mod->stream_in[QAF_IN_MAIN]->client_callback != NULL) {
+ if ((qaf_mod->stream_in[QAF_IN_MAIN]
+ && qaf_mod->stream_in[QAF_IN_MAIN]->client_callback != NULL) ||
+ (qaf_mod->stream_in[QAF_IN_MAIN_2]
+ && qaf_mod->stream_in[QAF_IN_MAIN_2]->client_callback != NULL)) {
- qaf_mod->stream_out[QAF_OUT_OFFLOAD_MCH]->stream.set_callback(
+ if (qaf_mod->stream_in[QAF_IN_MAIN]) {
+ qaf_mod->stream_out[QAF_OUT_OFFLOAD_MCH]->stream.set_callback(
(struct audio_stream_out *)qaf_mod->stream_out[QAF_OUT_OFFLOAD_MCH],
qaf_mod->stream_in[QAF_IN_MAIN]->client_callback,
qaf_mod->stream_in[QAF_IN_MAIN]->client_cookie);
+ }
+ if (qaf_mod->stream_in[QAF_IN_MAIN_2]) {
+ qaf_mod->stream_out[QAF_OUT_OFFLOAD_MCH]->stream.set_callback(
+ (struct audio_stream_out *)qaf_mod->stream_out[QAF_OUT_OFFLOAD_MCH],
+ qaf_mod->stream_in[QAF_IN_MAIN_2]->client_callback,
+ qaf_mod->stream_in[QAF_IN_MAIN_2]->client_cookie);
+ }
} else if (qaf_mod->stream_in[QAF_IN_PCM]
&& qaf_mod->stream_in[QAF_IN_PCM]->client_callback != NULL) {
@@ -1307,6 +1342,8 @@
int index = -1;
if (qaf_mod->adsp_hdlr_config[QAF_IN_MAIN].adsp_hdlr_config_valid)
index = (int) QAF_IN_MAIN;
+ else if (qaf_mod->adsp_hdlr_config[QAF_IN_MAIN_2].adsp_hdlr_config_valid)
+ index = (int) QAF_IN_MAIN_2;
else if (qaf_mod->adsp_hdlr_config[QAF_IN_PCM].adsp_hdlr_config_valid)
index = (int) QAF_IN_PCM;
@@ -1389,13 +1426,23 @@
return;
}
- if (qaf_mod->stream_in[QAF_IN_MAIN]
- && qaf_mod->stream_in[QAF_IN_MAIN]->client_callback != NULL) {
+ if ((qaf_mod->stream_in[QAF_IN_MAIN]
+ && qaf_mod->stream_in[QAF_IN_MAIN]->client_callback != NULL) ||
+ (qaf_mod->stream_in[QAF_IN_MAIN_2]
+ && qaf_mod->stream_in[QAF_IN_MAIN_2]->client_callback != NULL)) {
- qaf_mod->stream_out[QAF_OUT_OFFLOAD]->stream.set_callback(
+ if (qaf_mod->stream_in[QAF_IN_MAIN]) {
+ qaf_mod->stream_out[QAF_OUT_OFFLOAD]->stream.set_callback(
(struct audio_stream_out *)qaf_mod->stream_out[QAF_OUT_OFFLOAD],
qaf_mod->stream_in[QAF_IN_MAIN]->client_callback,
qaf_mod->stream_in[QAF_IN_MAIN]->client_cookie);
+ }
+ if (qaf_mod->stream_in[QAF_IN_MAIN_2]) {
+ qaf_mod->stream_out[QAF_OUT_OFFLOAD]->stream.set_callback(
+ (struct audio_stream_out *)qaf_mod->stream_out[QAF_OUT_OFFLOAD],
+ qaf_mod->stream_in[QAF_IN_MAIN_2]->client_callback,
+ qaf_mod->stream_in[QAF_IN_MAIN_2]->client_cookie);
+ }
} else if (qaf_mod->stream_in[QAF_IN_PCM]
&& qaf_mod->stream_in[QAF_IN_PCM]->client_callback != NULL) {
@@ -1408,7 +1455,7 @@
qaf_mod->stream_out[QAF_OUT_OFFLOAD]->compr_config.fragments =
COMPRESS_OFFLOAD_NUM_FRAGMENTS;
qaf_mod->stream_out[QAF_OUT_OFFLOAD]->compr_config.fragment_size =
- qaf_get_pcm_offload_buffer_size(qaf_mod, &config.offload_info);
+ qaf_get_pcm_offload_output_buffer_size(qaf_mod, &config.offload_info);
if (qaf_mod->is_vol_set) {
DEBUG_MSG("Setting Volume Left[%f], Right[%f]", qaf_mod->vol_left, qaf_mod->vol_right);
@@ -1421,6 +1468,8 @@
int index = -1;
if (qaf_mod->adsp_hdlr_config[QAF_IN_MAIN].adsp_hdlr_config_valid)
index = (int) QAF_IN_MAIN;
+ else if (qaf_mod->adsp_hdlr_config[QAF_IN_MAIN_2].adsp_hdlr_config_valid)
+ index = (int) QAF_IN_MAIN_2;
else if (qaf_mod->adsp_hdlr_config[QAF_IN_PCM].adsp_hdlr_config_valid)
index = (int) QAF_IN_PCM;
if (index >= 0) {
@@ -1449,16 +1498,18 @@
size);
}
}
-
- DEBUG_MSG("Bytes written = %d", ret);
+ DEBUG_MSG_VV("Bytes written = %d", ret);
}
else if (event_id == AUDIO_EOS_MAIN_DD_DDP_EVENT
+ || event_id == AUDIO_EOS_MAIN_2_DD_DDP_EVENT
|| event_id == AUDIO_EOS_MAIN_AAC_EVENT
|| event_id == AUDIO_EOS_MAIN_AC4_EVENT
|| event_id == AUDIO_EOS_ASSOC_DD_DDP_EVENT) {
struct stream_out *out = qaf_mod->stream_in[QAF_IN_MAIN];
+ struct stream_out *out_main2 = qaf_mod->stream_in[QAF_IN_MAIN_2];
struct stream_out *out_assoc = qaf_mod->stream_in[QAF_IN_ASSOC];
bool *main_drain_received = &qaf_mod->drain_received[QAF_IN_MAIN];
+ bool *main2_drain_received = &qaf_mod->drain_received[QAF_IN_MAIN_2];
bool *assoc_drain_received = &qaf_mod->drain_received[QAF_IN_ASSOC];
/**
@@ -1474,6 +1525,15 @@
*assoc_drain_received = false;
unlock_output_stream(out_assoc);
DEBUG_MSG("sent associated DRAIN_READY");
+ } else if (event_id == AUDIO_EOS_MAIN_2_DD_DDP_EVENT
+ && (out_main2 != NULL)
+ && (*main2_drain_received)) {
+
+ lock_output_stream(out_main2);
+ out_main2->client_callback(STREAM_CBK_EVENT_DRAIN_READY, NULL, out_main2->client_cookie);
+ *main2_drain_received = false;
+ unlock_output_stream(out_main2);
+ DEBUG_MSG("sent main2 DRAIN_READY");
} else if ((out != NULL) && (*main_drain_received)) {
lock_output_stream(out);
out->client_callback(STREAM_CBK_EVENT_DRAIN_READY, NULL, out->client_cookie);
@@ -1718,44 +1778,61 @@
devices,
AUDIO_STREAM_SYSTEM_TONE);
qaf_mod->stream_in[QAF_IN_PCM] = out;
- } else {
- if (!qaf_mod->stream_in[QAF_IN_MAIN]) {
- if ((!(flags & AUDIO_OUTPUT_FLAG_MAIN)) && (flags & AUDIO_OUTPUT_FLAG_ASSOCIATED)) {
- ERROR_MSG("Error main input is not active.");
- return -EINVAL;
- }
-
- status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle,
- &out->qaf_stream_handle,
- input_config,
- devices,
- AUDIO_STREAM_MAIN);
- if (status == 0) {
- DEBUG_MSG("Open stream for Input with Main stream contents with flag [%x] and stream handle [%p]",
- flags, out->qaf_stream_handle);
- qaf_mod->stream_in[QAF_IN_MAIN] = out;
- }
+ } else if ((flags & AUDIO_OUTPUT_FLAG_MAIN) && (flags & AUDIO_OUTPUT_FLAG_ASSOCIATED)) {
+ if (is_main_active(qaf_mod) || is_dual_main_active(qaf_mod)) {
+ ERROR_MSG("Dual Main or Main already active. So, Cannot open main and associated stream");
+ return -EINVAL;
} else {
- if (flags & AUDIO_OUTPUT_FLAG_MAIN) {
- ERROR_MSG("Error main input is already active");
- return -EINVAL;
- } else if ((flags & AUDIO_OUTPUT_FLAG_ASSOCIATED)
- && (!qaf_mod->stream_in[QAF_IN_ASSOC])) {
- status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle,
- &out->qaf_stream_handle,
- input_config,
- devices,
- AUDIO_STREAM_ASSOCIATED);
- if (status == 0) {
- DEBUG_MSG("Open stream for Input with only Associated flag [%x] stream handle [%p]",
- flags, out->qaf_stream_handle);
- qaf_mod->stream_in[QAF_IN_ASSOC] = out;
+ status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle, &out->qaf_stream_handle, input_config, devices, /*flags*/AUDIO_STREAM_MAIN);
+ if (status == 0) {
+ DEBUG_MSG("Open stream for Input with both Main and Associated stream contents with flag(%x) and stream_handle(%p)", flags, out->qaf_stream_handle);
+ qaf_mod->stream_in[QAF_IN_MAIN] = out;
+ } else {
+ ERROR_MSG("Stream Open FAILED !!!");
+ }
+ }
+ } else if ((flags & AUDIO_OUTPUT_FLAG_MAIN) || ((!(flags & AUDIO_OUTPUT_FLAG_MAIN)) && (!(flags & AUDIO_OUTPUT_FLAG_ASSOCIATED)))) {
+ /* Assume Main if no flag is set */
+ if (is_dual_main_active(qaf_mod)) {
+ ERROR_MSG("Dual Main already active. So, Cannot open main stream");
+ return -EINVAL;
+ } else if (is_main_active(qaf_mod) && qaf_mod->stream_in[QAF_IN_ASSOC]) {
+ ERROR_MSG("Main and Associated already active. So, Cannot open main stream");
+ return -EINVAL;
+ } else if (is_main_active(qaf_mod) && (mmtype != MS12)) {
+ ERROR_MSG("Main already active and Not an MS12 format. So, Cannot open another main stream");
+ return -EINVAL;
+ } else {
+ status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle, &out->qaf_stream_handle, input_config, devices, /*flags*/AUDIO_STREAM_MAIN);
+ if (status == 0) {
+ DEBUG_MSG("Open stream for Input with only Main flag(%x) stream_handle(%p)", flags, out->qaf_stream_handle);
+ if(qaf_mod->stream_in[QAF_IN_MAIN]) {
+ qaf_mod->stream_in[QAF_IN_MAIN_2] = out;
+ } else {
+ qaf_mod->stream_in[QAF_IN_MAIN] = out;
}
} else {
- ERROR_MSG("Invalid flag or associated is already active");
- status = -EINVAL;
+ ERROR_MSG("Stream Open FAILED !!!");
}
}
+ } else if ((flags & AUDIO_OUTPUT_FLAG_ASSOCIATED)) {
+ if (is_dual_main_active(qaf_mod)) {
+ ERROR_MSG("Dual Main already active. So, Cannot open associated stream");
+ return -EINVAL;
+ } else if (!is_main_active(qaf_mod)) {
+ ERROR_MSG("Main not active. So, Cannot open associated stream");
+ return -EINVAL;
+ } else if (qaf_mod->stream_in[QAF_IN_ASSOC]) {
+ ERROR_MSG("Associated already active. So, Cannot open associated stream");
+ return -EINVAL;
+ }
+ status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle, &out->qaf_stream_handle, input_config, devices, /*flags*/AUDIO_STREAM_ASSOCIATED);
+ if (status == 0) {
+ DEBUG_MSG("Open stream for Input with only Associated flag(%x) stream handle(%p)", flags, out->qaf_stream_handle);
+ qaf_mod->stream_in[QAF_IN_ASSOC] = out;
+ } else {
+ ERROR_MSG("Stream Open FAILED !!!");
+ }
}
if (status != 0) {
@@ -1849,11 +1926,11 @@
stream_callback_event_t event;
bool send_callback = false;
- DEBUG_MSG("List Empty %d (1:TRUE, 0:FALSE)", list_empty(&out->qaf_offload_cmd_list));
+ DEBUG_MSG_VV("List Empty %d (1:TRUE, 0:FALSE)", list_empty(&out->qaf_offload_cmd_list));
if (list_empty(&out->qaf_offload_cmd_list)) {
- DEBUG_MSG("SLEEPING");
+ DEBUG_MSG_VV("SLEEPING");
pthread_cond_wait(&out->qaf_offload_cond, &out->lock);
- DEBUG_MSG("RUNNING");
+ DEBUG_MSG_VV("RUNNING");
continue;
}
@@ -1871,7 +1948,7 @@
send_callback = false;
switch (cmd->cmd) {
case OFFLOAD_CMD_WAIT_FOR_BUFFER: {
- DEBUG_MSG("wait for buffer availability");
+ DEBUG_MSG_VV("wait for buffer availability");
while (1) {
kvpairs = qaf_mod->qaf_audio_stream_get_param(out->qaf_stream_handle,
@@ -1881,12 +1958,12 @@
ret = str_parms_get_int(parms, "buf_available", &value);
if (ret >= 0) {
if (value >= (int)out->compr_config.fragment_size) {
- DEBUG_MSG("buffer available");
+ DEBUG_MSG_VV("buffer available");
str_parms_destroy(parms);
parms = NULL;
break;
} else {
- DEBUG_MSG("sleep");
+ DEBUG_MSG_VV("sleep");
str_parms_destroy(parms);
parms = NULL;
usleep(10000);
@@ -1978,11 +2055,7 @@
/* Setting new device information to the mm module input streams.
* This is needed if QAF module output streams are not created yet.
*/
- if (qaf_mod->stream_in[QAF_IN_MAIN] == out || qaf_mod->stream_in[QAF_IN_ASSOC] == out) {
- qaf_mod->stream_in[QAF_IN_MAIN]->devices = val;
- } else {
- out->devices = val;
- }
+ out->devices = val;
if (val == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) {
//If device is BT then open the BT stream if not already opened.
@@ -2214,7 +2287,10 @@
out->config.period_count = DEEP_BUFFER_OUTPUT_PERIOD_COUNT;
out->config.start_threshold = QAF_DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4;
out->config.avail_min = QAF_DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4;
+ } else if(out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
+ out->compr_config.fragment_size = qaf_get_pcm_offload_input_buffer_size(&(config->offload_info));
}
+
*stream_out = &out->stream;
if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
qaf_create_offload_callback_thread(out);
diff --git a/hal/audio_extn/sndmonitor.c b/hal/audio_extn/sndmonitor.c
index 89a6670..b560c9d 100644
--- a/hal/audio_extn/sndmonitor.c
+++ b/hal/audio_extn/sndmonitor.c
@@ -1,5 +1,5 @@
/*
-* Copyright (c) 2016, The Linux Foundation. All rights reserved.
+* Copyright (c) 2016-2017, 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
@@ -58,6 +58,12 @@
#include "audio_hw.h"
#include "audio_extn.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_SND_MONITOR
+#include <log_utils.h>
+#endif
+
//#define MONITOR_DEVICE_EVENTS
#define CPE_MAGIC_NUM 0x2000
#define MAX_CPE_SLEEP_RETRY 2
diff --git a/hal/audio_extn/soundtrigger.c b/hal/audio_extn/soundtrigger.c
index cecc843..94a8a2b 100644
--- a/hal/audio_extn/soundtrigger.c
+++ b/hal/audio_extn/soundtrigger.c
@@ -41,6 +41,12 @@
#include "platform_api.h"
#include "sound_trigger_prop_intf.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_SND_TRIGGER
+#include <log_utils.h>
+#endif
+
#define XSTR(x) STR(x)
#define STR(x) #x
#define MAX_LIBRARY_PATH 100
diff --git a/hal/audio_extn/source_track.c b/hal/audio_extn/source_track.c
index 5bced66..e5e6c06 100644
--- a/hal/audio_extn/source_track.c
+++ b/hal/audio_extn/source_track.c
@@ -41,6 +41,12 @@
#include <stdlib.h>
#include <cutils/str_parms.h>
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_SRC_TRACK
+#include <log_utils.h>
+#endif
+
#ifdef SOURCE_TRACKING_ENABLED
/* Audio Paramater Key to identify the list of start angles.
* Starting angle (in degrees) defines the boundary starting angle for each sector.
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index 52bf3a6..710fd31 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013 - 2016, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013 - 2017, 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
@@ -47,6 +47,12 @@
#include "audio_extn.h"
#include <linux/msm_audio_calibration.h>
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_SPKR_PROT
+#include <log_utils.h>
+#endif
+
#ifdef SPKR_PROT_ENABLED
/*Range of spkr temparatures -30C to 80C*/
diff --git a/hal/audio_extn/ssr.c b/hal/audio_extn/ssr.c
index f64a861..7467579 100644
--- a/hal/audio_extn/ssr.c
+++ b/hal/audio_extn/ssr.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2016, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2017, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -38,6 +38,12 @@
#include "platform_api.h"
#include "surround_rec_interface.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_SSR
+#include <log_utils.h>
+#endif
+
#ifdef SSR_ENABLED
#define COEFF_ARRAY_SIZE 4
#define FILT_SIZE ((512+1)* 6) /* # ((FFT bins)/2+1)*numOutputs */
diff --git a/hal/audio_extn/usb.c b/hal/audio_extn/usb.c
index 456382e..5c397a7 100644
--- a/hal/audio_extn/usb.c
+++ b/hal/audio_extn/usb.c
@@ -36,6 +36,12 @@
#include <ctype.h>
#include <math.h>
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_USB
+#include <log_utils.h>
+#endif
+
#ifdef USB_HEADSET_ENABLED
#define USB_BUFF_SIZE 2048
#define CHANNEL_NUMBER_STR "Channels: "
diff --git a/hal/audio_extn/utils.c b/hal/audio_extn/utils.c
index 27bbae8..335cdbc 100644
--- a/hal/audio_extn/utils.c
+++ b/hal/audio_extn/utils.c
@@ -39,6 +39,13 @@
#include <sound/compress_params.h>
#include <sound/compress_offload.h>
#include <tinycompress/tinycompress.h>
+
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_UTILS
+#include <log_utils.h>
+#endif
+
#ifdef AUDIO_EXTERNAL_HDMI_ENABLED
#ifdef HDMI_PASSTHROUGH_ENABLED
#include "audio_parsers.h"
@@ -111,6 +118,7 @@
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_INCALL_MUSIC),
#endif
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_TIMESTAMP),
STRING_TO_ENUM(AUDIO_INPUT_FLAG_NONE),
STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST),
STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD),
@@ -1945,3 +1953,66 @@
return 0;
}
#endif
+
+#define MAX_SND_CARD 8
+#define RETRY_US 500000
+#define RETRY_NUMBER 10
+
+int audio_extn_utils_get_snd_card_num()
+{
+
+ void *hw_info = NULL;
+ struct mixer *mixer = NULL;
+ int retry_num = 0;
+ int snd_card_num = 0;
+ char* snd_card_name = NULL;
+
+ while (snd_card_num < MAX_SND_CARD) {
+ mixer = mixer_open(snd_card_num);
+
+ while (!mixer && retry_num < RETRY_NUMBER) {
+ usleep(RETRY_US);
+ mixer = mixer_open(snd_card_num);
+ retry_num++;
+ }
+
+ if (!mixer) {
+ ALOGE("%s: Unable to open the mixer card: %d", __func__,
+ snd_card_num);
+ retry_num = 0;
+ snd_card_num++;
+ continue;
+ }
+
+ snd_card_name = strdup(mixer_get_name(mixer));
+ if (!snd_card_name) {
+ ALOGE("failed to allocate memory for snd_card_name\n");
+ mixer_close(mixer);
+ return -1;
+ }
+ ALOGD("%s: snd_card_name: %s", __func__, snd_card_name);
+
+ hw_info = hw_info_init(snd_card_name);
+ if (hw_info) {
+ ALOGD("%s: Opened sound card:%d", __func__, snd_card_num);
+ break;
+ }
+ ALOGE("%s: Failed to init hardware info", __func__);
+ retry_num = 0;
+ snd_card_num++;
+ free(snd_card_name);
+ mixer_close(mixer);
+ }
+
+ mixer_close(mixer);
+ hw_info_deinit(hw_info);
+ if (snd_card_name)
+ free(snd_card_name);
+
+ if (snd_card_num >= MAX_SND_CARD) {
+ ALOGE("%s: Unable to find correct sound card, aborting.", __func__);
+ return -1;
+ }
+
+ return snd_card_num;
+}
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index de8b388..1013332 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -76,6 +76,12 @@
#include "sound/compress_params.h"
#include "sound/asound.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_AUDIO_HW
+#include <log_utils.h>
+#endif
+
#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
/*DIRECT PCM has same buffer sizes as DEEP Buffer*/
#define DIRECT_PCM_NUM_FRAGMENTS 2
@@ -1248,7 +1254,7 @@
/* Update voc calibration before enabling VoIP route */
if (usecase->type == VOIP_CALL)
status = platform_switch_voice_call_device_post(adev->platform,
- usecase->out_snd_device,
+ platform_get_output_snd_device(adev->platform, uc_info->stream.out),
usecase->in_snd_device);
enable_audio_route(adev, usecase);
}
@@ -1535,7 +1541,9 @@
} else if (voice_extn_compress_voip_is_active(adev)) {
bool out_snd_device_backend_match = true;
voip_usecase = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
- if (usecase->stream.out != NULL) {
+ if ((voip_usecase != NULL) &&
+ (usecase->type == PCM_PLAYBACK) &&
+ (usecase->stream.out != NULL)) {
out_snd_device_backend_match = platform_check_backends_match(
voip_usecase->out_snd_device,
platform_get_output_snd_device(
@@ -2579,9 +2587,12 @@
{
struct stream_out *out = (struct stream_out *)stream;
- if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
- return out->compr_config.fragment_size;
- else if(out->usecase == USECASE_COMPRESS_VOIP_CALL)
+ if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ if (out->flags & AUDIO_OUTPUT_FLAG_TIMESTAMP)
+ return out->compr_config.fragment_size - sizeof(struct snd_codec_metadata);
+ else
+ return out->compr_config.fragment_size;
+ } else if(out->usecase == USECASE_COMPRESS_VOIP_CALL)
return voice_extn_compress_voip_out_get_buffer_size(out);
else if (is_offload_usecase(out->usecase) &&
out->flags == AUDIO_OUTPUT_FLAG_DIRECT)
@@ -4097,6 +4108,11 @@
*/
if (!audio_extn_passthru_is_passthrough_stream(out))
out->bit_width = AUDIO_OUTPUT_BIT_WIDTH;
+
+ if (out->flags & AUDIO_OUTPUT_FLAG_TIMESTAMP)
+ out->compr_config.codec->flags |= COMPRESSED_TIMESTAMP_FLAG;
+ ALOGVV("%s : out->compr_config.codec->flags -> (%#x) ", __func__, out->compr_config.codec->flags);
+
/*TODO: Do we need to change it for passthrough */
out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;
@@ -4159,6 +4175,9 @@
out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
}
+ if (out->flags & AUDIO_OUTPUT_FLAG_TIMESTAMP) {
+ out->compr_config.fragment_size += sizeof(struct snd_codec_metadata);
+ }
if (config->offload_info.format == AUDIO_FORMAT_FLAC)
out->compr_config.codec->options.flac_dec.sample_size = AUDIO_OUTPUT_BIT_WIDTH;
@@ -5173,6 +5192,10 @@
pthread_mutex_init(&adev->lock, (const pthread_mutexattr_t *) NULL);
+#ifdef DYNAMIC_LOG_ENABLED
+ register_for_dynamic_logging("hal");
+#endif
+
adev->device.common.tag = HARDWARE_DEVICE_TAG;
adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
adev->device.common.module = (struct hw_module_t *)module;
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index ff9149f..9f10efa 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -270,6 +270,7 @@
struct audio_out_start_delay_param delay_param; /*start delay*/
audio_offload_info_t info;
+ qahwi_stream_out_t qahwi_out;
};
struct stream_in {
diff --git a/hal/audio_hw_extn_api.c b/hal/audio_hw_extn_api.c
index a1bd04d..63f8d0d 100644
--- a/hal/audio_hw_extn_api.c
+++ b/hal/audio_hw_extn_api.c
@@ -31,6 +31,7 @@
/*#define LOG_NDEBUG 0*/
#define LOG_NDDEBUG 0
+#include <inttypes.h>
#include <errno.h>
#include <cutils/log.h>
@@ -40,6 +41,12 @@
#include "audio_extn.h"
#include "audio_hw_extn_api.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_AUDIO_HW_EXTN_API
+#include <log_utils.h>
+#endif
+
/* default timestamp metadata definition if not defined in kernel*/
#ifndef COMPRESSED_TIMESTAMP_FLAG
#define COMPRESSED_TIMESTAMP_FLAG 0
@@ -286,6 +293,117 @@
return ret;
}
+ssize_t qahwi_out_write_v2(struct audio_stream_out *stream, const void* buffer,
+ size_t bytes, int64_t* timestamp)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ struct snd_codec_metadata *mdata = NULL;
+ size_t mdata_size = 0, bytes_written = 0;
+ char *buf = NULL;
+ ssize_t ret = 0;
+
+ if (!out->qahwi_out.is_inititalized) {
+ ALOGE("%s: invalid state!", __func__);
+ return -EINVAL;
+ }
+ if (COMPRESSED_TIMESTAMP_FLAG &&
+ (out->flags & AUDIO_OUTPUT_FLAG_TIMESTAMP)) {
+
+ mdata_size = sizeof(struct snd_codec_metadata);
+ buf = (char *) out->qahwi_out.obuf;
+ if (timestamp) {
+ mdata = (struct snd_codec_metadata *) buf;
+ mdata->length = bytes;
+ mdata->offset = mdata_size;
+ mdata->timestamp = *timestamp;
+ }
+ memcpy(buf + mdata_size, buffer, bytes);
+ ret = out->qahwi_out.base.write(&out->stream, (void *)buf, out->qahwi_out.buf_size);
+ if (ret <= 0) {
+ ALOGE("%s: error! write returned %zd", __func__, ret);
+ } else {
+ bytes_written = bytes;
+ }
+ ALOGV("%s: flag 0x%x, bytes %zd, read %zd, ret %zd timestamp 0x%"PRIx64"",
+ __func__, out->flags, bytes, bytes_written, ret, *timestamp);
+ } else {
+ bytes_written = out->qahwi_out.base.write(&out->stream, buffer, bytes);
+ ALOGV("%s: flag 0x%x, bytes %zd, read %zd, ret %zd",
+ __func__, out->flags, bytes, bytes_written, ret);
+ }
+ return bytes_written;
+}
+
+static void qahwi_close_output_stream(struct audio_hw_device *dev,
+ struct audio_stream_out *stream_out)
+{
+ struct audio_device *adev = (struct audio_device *) dev;
+ struct stream_out *out = (struct stream_out *)stream_out;
+
+ ALOGV("%s", __func__);
+ if (!adev->qahwi_dev.is_inititalized || !out->qahwi_out.is_inititalized) {
+ ALOGE("%s: invalid state!", __func__);
+ return;
+ }
+ if (out->qahwi_out.obuf)
+ free(out->qahwi_out.obuf);
+ out->qahwi_out.buf_size = 0;
+ adev->qahwi_dev.base.close_output_stream(dev, stream_out);
+}
+
+static int qahwi_open_output_stream(struct audio_hw_device *dev,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ audio_output_flags_t flags,
+ struct audio_config *config,
+ struct audio_stream_out **stream_out,
+ const char *address)
+{
+ struct audio_device *adev = (struct audio_device *) dev;
+ struct stream_out *out = NULL;
+ size_t buf_size = 0, mdata_size = 0;
+ int ret = 0;
+
+ ALOGV("%s: dev_init %d, flags 0x%x", __func__,
+ adev->qahwi_dev.is_inititalized, flags);
+ if (!adev->qahwi_dev.is_inititalized) {
+ ALOGE("%s: invalid state!", __func__);
+ return -EINVAL;
+ }
+
+ ret = adev->qahwi_dev.base.open_output_stream(dev, handle, devices, flags,
+ config, stream_out, address);
+ if (ret)
+ return ret;
+
+ out = (struct stream_out *)*stream_out;
+ // keep adev fptrs before overriding
+ out->qahwi_out.base = out->stream;
+
+ out->qahwi_out.is_inititalized = true;
+
+ if (COMPRESSED_TIMESTAMP_FLAG &&
+ (flags & AUDIO_OUTPUT_FLAG_TIMESTAMP)) {
+ // set write to NULL as this is not supported in timestamp mode
+ out->stream.write = NULL;
+
+ mdata_size = sizeof(struct snd_codec_metadata);
+ buf_size = out->qahwi_out.base.common.get_buffer_size(&out->stream.common);
+ buf_size += mdata_size;
+ out->qahwi_out.buf_size = buf_size;
+ out->qahwi_out.obuf = malloc(buf_size);
+ if (!out->qahwi_out.obuf) {
+ ALOGE("%s: allocation failed for timestamp metadata!", __func__);
+ qahwi_close_output_stream(dev, &out->stream);
+ *stream_out = NULL;
+ ret = -ENOMEM;
+ }
+ ALOGD("%s: obuf %p, buff_size %zd",
+ __func__, out->qahwi_out.obuf, buf_size);
+ }
+ return ret;
+}
+
void qahwi_init(hw_device_t *device)
{
struct audio_device *adev = (struct audio_device *) device;
@@ -299,6 +417,9 @@
adev->device.open_input_stream = qahwi_open_input_stream;
adev->device.close_input_stream = qahwi_close_input_stream;
+ adev->device.open_output_stream = qahwi_open_output_stream;
+ adev->device.close_output_stream = qahwi_close_output_stream;
+
adev->qahwi_dev.is_inititalized = true;
}
void qahwi_deinit(hw_device_t *device)
diff --git a/hal/audio_hw_extn_api.h b/hal/audio_hw_extn_api.h
index e5fa9ec..4123461 100644
--- a/hal/audio_hw_extn_api.h
+++ b/hal/audio_hw_extn_api.h
@@ -33,6 +33,7 @@
#ifdef AUDIO_HW_EXTN_API_ENABLED
#include <hardware/audio.h>
typedef struct qahwi_stream_in qahwi_stream_in_t;
+typedef struct qahwi_stream_out qahwi_stream_out_t;
typedef struct qahwi_device qahwi_device_t;
struct qahwi_stream_in {
@@ -41,6 +42,13 @@
void *ibuf;
};
+struct qahwi_stream_out {
+ struct audio_stream_out base;
+ bool is_inititalized;
+ size_t buf_size;
+ void *obuf;
+};
+
struct qahwi_device {
struct audio_hw_device base;
bool is_inititalized;
@@ -50,6 +58,7 @@
void qahwi_deinit(hw_device_t *device);
#else
typedef void *qahwi_stream_in_t;
+typedef void *qahwi_stream_out_t;
typedef void *qahwi_device_t;
#define qahwi_init(device) (0)
diff --git a/hal/edid.c b/hal/edid.c
index e889530..f7259c7 100644
--- a/hal/edid.c
+++ b/hal/edid.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2014, 2016, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2014, 2016-2017, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2014 The Android Open Source Project
@@ -33,6 +33,12 @@
#include "platform_api.h"
#include "edid.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_EDID
+#include <log_utils.h>
+#endif
+
static const char * edid_format_to_str(unsigned char format)
{
char * format_str = "??";
@@ -798,4 +804,4 @@
ALOGV("%s: returns [%d] for highest supported sr",
__func__, highest_sr);
return highest_sr;
-}
\ No newline at end of file
+}
diff --git a/hal/msm8916/hw_info.c b/hal/msm8916/hw_info.c
index 652afab..a384827 100644
--- a/hal/msm8916/hw_info.c
+++ b/hal/msm8916/hw_info.c
@@ -39,6 +39,11 @@
#include "platform.h"
#include "platform_api.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_HW_INFO
+#include <log_utils.h>
+#endif
struct hardware_info {
char name[HW_INFO_ARRAY_MAX_SIZE];
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 2ab9408..22215e3 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -32,13 +32,19 @@
#include <platform_api.h>
#include "platform.h"
#include "audio_extn.h"
+#include "acdb.h"
#include "voice_extn.h"
#include "edid.h"
#include "sound/compress_params.h"
#include "sound/msmcal-hwdep.h"
#include <dirent.h>
#include <linux/msm_audio.h>
-#include "linux/msm_audio_calibration.h"
+
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_PLATFORM
+#include <log_utils.h>
+#endif
#define SOUND_TRIGGER_DEVICE_HANDSET_MONO_LOW_POWER_ACDB_ID (100)
#define MAX_MIXER_XML_PATH 100
@@ -132,11 +138,6 @@
#define DEFAULT_APP_TYPE_RX_PATH 0x11130
#define DEFAULT_APP_TYPE_TX_PATH 0x11132
-/* Retry for delay in FW loading*/
-#define RETRY_NUMBER 20
-#define RETRY_US 500000
-#define MAX_SND_CARD 8
-
#define SAMPLE_RATE_8KHZ 8000
#define SAMPLE_RATE_16KHZ 16000
@@ -178,6 +179,11 @@
static char *default_rx_backend = NULL;
+#ifdef DYNAMIC_LOG_ENABLED
+extern void log_utils_init(void);
+extern void log_utils_deinit(void);
+#endif
+
char dsp_only_decoders_mime[][MAX_MIME_TYPE_LENGTH] = {
"audio/x-ms-wma" /* wma*/ ,
"audio/x-ms-wma-lossless" /* wma lossless */ ,
@@ -217,24 +223,7 @@
CAL_MODE_RTAC = 0x4
};
-/* Audio calibration related functions */
-typedef void (*acdb_deallocate_t)();
-typedef int (*acdb_init_t)(const char *, char *, int);
-typedef int (*acdb_init_v3_t)(const char *, char *, struct listnode *);
-typedef void (*acdb_send_audio_cal_t)(int, int, int , int);
-typedef void (*acdb_send_audio_cal_v3_t)(int, int, int, int, int);
-typedef void (*acdb_send_voice_cal_t)(int, int);
-typedef int (*acdb_reload_vocvoltable_t)(int);
-typedef int (*acdb_get_default_app_type_t)(void);
-typedef int (*acdb_loader_get_calibration_t)(char *attr, int size, void *data);
acdb_loader_get_calibration_t acdb_loader_get_calibration;
-typedef int (*acdb_set_audio_cal_t) (void *, void *, uint32_t);
-typedef int (*acdb_get_audio_cal_t) (void *, void *, uint32_t*);
-typedef int (*acdb_send_common_top_t) (void);
-typedef int (*acdb_set_codec_data_t) (void *, char *);
-typedef int (*acdb_reload_t) (char *, char *, char *, int);
-typedef int (*acdb_send_gain_dep_cal_t)(int, int, int, int, int);
-typedef int (*acdb_reload_v2_t) (char *, char *, char *, struct listnode *);
typedef struct codec_backend_cfg {
uint32_t sample_rate;
@@ -248,12 +237,6 @@
static native_audio_prop na_props = {0, 0, NATIVE_AUDIO_MODE_INVALID};
static bool supports_true_32_bit = false;
-struct meta_key_list {
- struct listnode list;
- struct audio_cal_info_metainfo cal_info;
- char name[ACDB_METAINFO_KEY_MODULE_NAME_LEN];
-};
-
static int max_be_dai_names = 0;
static const struct be_dai_name_struct *be_dai_name_table;
@@ -2061,7 +2044,7 @@
{
char value[PROPERTY_VALUE_MAX];
struct platform_data *my_data = NULL;
- int retry_num = 0, snd_card_num = 0;
+ int snd_card_num = 0;
const char *snd_card_name;
char mixer_xml_path[MAX_MIXER_XML_PATH],ffspEnable[PROPERTY_VALUE_MAX];
const char *mixer_ctl_name = "Set HPX ActiveBe";
@@ -2070,6 +2053,25 @@
int wsaCount =0;
bool is_wsa_combo_supported = false;
+ snd_card_num = audio_extn_utils_get_snd_card_num();
+ if(snd_card_num < 0) {
+ ALOGE("%s: Unable to find correct sound card", __func__);
+ return NULL;
+ }
+
+ adev->snd_card = snd_card_num;
+ ALOGD("%s: Opened sound card:%d", __func__, snd_card_num);
+
+ adev->mixer = mixer_open(snd_card_num);
+ if (!adev->mixer) {
+ ALOGE("%s: Unable to open the mixer card: %d", __func__,
+ snd_card_num);
+ return NULL;
+ }
+
+ snd_card_name = mixer_get_name(adev->mixer);
+ ALOGD("%s: snd_card_name: %s", __func__, snd_card_name);
+
my_data = calloc(1, sizeof(struct platform_data));
if (!my_data) {
@@ -2077,62 +2079,31 @@
return NULL;
}
- while (snd_card_num < MAX_SND_CARD) {
- adev->mixer = mixer_open(snd_card_num);
-
- while (!adev->mixer && retry_num < RETRY_NUMBER) {
- usleep(RETRY_US);
- adev->mixer = mixer_open(snd_card_num);
- retry_num++;
- }
-
- if (!adev->mixer) {
- ALOGE("%s: Unable to open the mixer card: %d", __func__,
- snd_card_num);
- retry_num = 0;
- snd_card_num++;
- continue;
- }
-
- snd_card_name = mixer_get_name(adev->mixer);
- ALOGD("%s: snd_card_name: %s", __func__, snd_card_name);
-
- my_data->hw_info = hw_info_init(snd_card_name);
- if (!my_data->hw_info) {
- ALOGE("%s: Failed to init hardware info", __func__);
- } else {
- query_platform(snd_card_name, mixer_xml_path);
- ALOGD("%s: mixer path file is %s", __func__,
- mixer_xml_path);
- if (audio_extn_read_xml(adev, snd_card_num, mixer_xml_path,
- MIXER_XML_PATH_AUXPCM) == -ENOSYS) {
- adev->audio_route = audio_route_init(snd_card_num,
- mixer_xml_path);
- }
- if (!adev->audio_route) {
- ALOGE("%s: Failed to init audio route controls, aborting.",
- __func__);
- free(my_data);
- mixer_close(adev->mixer);
- return NULL;
- }
- adev->snd_card = snd_card_num;
- update_codec_type(snd_card_name);
- update_interface(snd_card_name);
- ALOGD("%s: Opened sound card:%d", __func__, snd_card_num);
- break;
- }
- retry_num = 0;
- snd_card_num++;
- mixer_close(adev->mixer);
- }
-
- if (snd_card_num >= MAX_SND_CARD) {
- ALOGE("%s: Unable to find correct sound card, aborting.", __func__);
+ my_data->hw_info = hw_info_init(snd_card_name);
+ if (!my_data->hw_info) {
+ ALOGE("%s: Failed to init hardware info", __func__);
free(my_data);
return NULL;
}
+ query_platform(snd_card_name, mixer_xml_path);
+ ALOGD("%s: mixer path file is %s", __func__,
+ mixer_xml_path);
+ if (audio_extn_read_xml(adev, snd_card_num, mixer_xml_path,
+ MIXER_XML_PATH_AUXPCM) == -ENOSYS) {
+ adev->audio_route = audio_route_init(snd_card_num,
+ mixer_xml_path);
+ }
+ if (!adev->audio_route) {
+ ALOGE("%s: Failed to init audio route controls, aborting.",
+ __func__);
+ free(my_data);
+ mixer_close(adev->mixer);
+ return NULL;
+ }
+ update_codec_type(snd_card_name);
+ update_interface(snd_card_name);
+
my_data->adev = adev;
my_data->fluence_in_spkr_mode = false;
my_data->fluence_in_voice_call = false;
@@ -2239,12 +2210,12 @@
/* Initialize ACDB and PCM ID's */
if (is_external_codec)
- platform_info_init(PLATFORM_INFO_XML_PATH_EXTCODEC, my_data);
+ platform_info_init(PLATFORM_INFO_XML_PATH_EXTCODEC, my_data, PLATFORM);
else if (!strncmp(snd_card_name, "sdm660-snd-card-skush",
sizeof("sdm660-snd-card-skush")))
- platform_info_init(PLATFORM_INFO_XML_PATH_SKUSH, my_data);
+ platform_info_init(PLATFORM_INFO_XML_PATH_SKUSH, my_data, PLATFORM);
else
- platform_info_init(PLATFORM_INFO_XML_PATH, my_data);
+ platform_info_init(PLATFORM_INFO_XML_PATH, my_data, PLATFORM);
my_data->voice_feature_set = VOICE_FEATURE_SET_DEFAULT;
my_data->acdb_handle = dlopen(LIB_ACDB_LOADER, RTLD_NOW);
@@ -2347,10 +2318,20 @@
goto acdb_init_fail;
}
- platform_acdb_init(my_data);
+ int result = acdb_init(adev->snd_card);
+ if (!result) {
+ my_data->is_acdb_initialized = true;
+ ALOGD("ACDB initialized");
+ audio_hwdep_send_cal(my_data);
+ } else {
+ my_data->is_acdb_initialized = false;
+ ALOGD("ACDB initialization failed");
+ }
}
audio_extn_pm_vote();
-
+#ifdef DYNAMIC_LOG_ENABLED
+ log_utils_init();
+#endif
/* Configure active back end for HPX*/
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (ctl) {
@@ -2586,6 +2567,9 @@
/* deinit usb */
audio_extn_usb_deinit();
audio_extn_dap_hal_deinit();
+#ifdef DYNAMIC_LOG_ENABLED
+ log_utils_deinit();
+#endif
}
static int platform_is_acdb_initialized(void *platform)
@@ -6339,9 +6323,18 @@
audio_offload_info_t* info)
{
uint32_t fragment_size = MIN_COMPRESS_PASSTHROUGH_FRAGMENT_SIZE;
+ char value[PROPERTY_VALUE_MAX] = {0};
+
+ if (((info->format == AUDIO_FORMAT_DOLBY_TRUEHD) ||
+ (info->format == AUDIO_FORMAT_IEC61937)) &&
+ property_get("audio.truehd.buffer.size.kb", value, "") &&
+ atoi(value)) {
+ fragment_size = atoi(value) * 1024;
+ goto done;
+ }
if (!info->has_video)
fragment_size = MIN_COMPRESS_PASSTHROUGH_FRAGMENT_SIZE;
-
+done:
return fragment_size;
}
diff --git a/hal/msm8974/hw_info.c b/hal/msm8974/hw_info.c
index dd74877..1187f4b 100644
--- a/hal/msm8974/hw_info.c
+++ b/hal/msm8974/hw_info.c
@@ -39,6 +39,11 @@
#include "platform.h"
#include "platform_api.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_HW_INFO
+#include <log_utils.h>
+#endif
struct hardware_info {
char name[HW_INFO_ARRAY_MAX_SIZE];
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 77baa93..47cae0c 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -38,11 +38,17 @@
#include <platform_api.h>
#include "platform.h"
#include "audio_extn.h"
+#include "acdb.h"
#include "voice_extn.h"
#include "edid.h"
#include "sound/compress_params.h"
#include "sound/msmcal-hwdep.h"
-#include <linux/msm_audio_calibration.h>
+
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_PLATFORM
+#include <log_utils.h>
+#endif
#define SOUND_TRIGGER_DEVICE_HANDSET_MONO_LOW_POWER_ACDB_ID (100)
#define MIXER_FILE_DELIMITER "_"
@@ -103,11 +109,6 @@
#define DEFAULT_APP_TYPE_RX_PATH 0x11130
#define DEFAULT_APP_TYPE_TX_PATH 0x11132
-/* Retry for delay in FW loading*/
-#define RETRY_NUMBER 10
-#define RETRY_US 500000
-#define MAX_SND_CARD 8
-
#define SAMPLE_RATE_8KHZ 8000
#define SAMPLE_RATE_16KHZ 16000
@@ -139,6 +140,11 @@
#define MAX_CAL_NAME 20
#define MAX_MIME_TYPE_LENGTH 30
+#ifdef DYNAMIC_LOG_ENABLED
+extern void log_utils_init(void);
+extern void log_utils_deinit(void);
+#endif
+
char cal_name_info[WCD9XXX_MAX_CAL][MAX_CAL_NAME] = {
[WCD9XXX_ANC_CAL] = "anc_cal",
[WCD9XXX_MBHC_CAL] = "mbhc_cal",
@@ -187,23 +193,7 @@
CAL_MODE_RTAC = 0x4
};
-/* Audio calibration related functions */
-typedef void (*acdb_deallocate_t)();
-typedef int (*acdb_init_t)(const char *, char *, int);
-typedef int (*acdb_init_v3_t)(const char *, char *, struct listnode *);
-typedef void (*acdb_send_audio_cal_t)(int, int, int , int);
-typedef void (*acdb_send_audio_cal_v3_t)(int, int, int, int, int);
-typedef void (*acdb_send_voice_cal_t)(int, int);
-typedef int (*acdb_reload_vocvoltable_t)(int);
-typedef int (*acdb_get_default_app_type_t)(void);
-typedef int (*acdb_loader_get_calibration_t)(char *attr, int size, void *data);
acdb_loader_get_calibration_t acdb_loader_get_calibration;
-typedef int (*acdb_set_audio_cal_t) (void *, void *, uint32_t);
-typedef int (*acdb_get_audio_cal_t) (void *, void *, uint32_t*);
-typedef int (*acdb_send_common_top_t) (void);
-typedef int (*acdb_set_codec_data_t) (void *, char *);
-typedef int (*acdb_reload_t) (char *, char *, char *, int);
-typedef int (*acdb_reload_v2_t) (char *, char *, char *, struct listnode *);
typedef struct codec_backend_cfg {
uint32_t sample_rate;
@@ -216,13 +206,6 @@
static native_audio_prop na_props = {0, 0, NATIVE_AUDIO_MODE_INVALID};
static bool supports_true_32_bit = false;
-typedef int (*acdb_send_gain_dep_cal_t)(int, int, int, int, int);
-
-struct meta_key_list {
- struct listnode list;
- struct audio_cal_info_metainfo cal_info;
- char name[ACDB_METAINFO_KEY_MODULE_NAME_LEN];
-};
static int max_be_dai_names = 0;
static const struct be_dai_name_struct *be_dai_name_table;
@@ -1764,125 +1747,109 @@
char baseband[PROPERTY_VALUE_MAX];
char value[PROPERTY_VALUE_MAX];
struct platform_data *my_data = NULL;
- int retry_num = 0, snd_card_num = 0;
char *snd_card_name = NULL, *snd_card_name_t = NULL;
char *snd_internal_name = NULL;
char *tmp = NULL;
char mixer_xml_file[MIXER_PATH_MAX_LENGTH]= {0};
int idx;
- my_data = calloc(1, sizeof(struct platform_data));
+ adev->snd_card = audio_extn_utils_get_snd_card_num();
+ if (adev->snd_card < 0) {
+ ALOGE("%s: Unable to find correct sound card", __func__);
+ return NULL;
+ }
+ ALOGD("%s: Opened sound card:%d", __func__, adev->snd_card);
- if (!my_data) {
- ALOGE("failed to allocate platform data");
+ adev->mixer = mixer_open(adev->snd_card);
+ if (!adev->mixer) {
+ ALOGE("%s: Unable to open the mixer card: %d", __func__,
+ adev->snd_card);
return NULL;
}
- while (snd_card_num < MAX_SND_CARD) {
- adev->mixer = mixer_open(snd_card_num);
-
- while (!adev->mixer && retry_num < RETRY_NUMBER) {
- usleep(RETRY_US);
- adev->mixer = mixer_open(snd_card_num);
- retry_num++;
- }
-
- if (!adev->mixer) {
- ALOGE("%s: Unable to open the mixer card: %d", __func__,
- snd_card_num);
- retry_num = 0;
- snd_card_num++;
- continue;
- }
-
- snd_card_name = strdup(mixer_get_name(adev->mixer));
- if (!snd_card_name) {
- ALOGE("failed to allocate memory for snd_card_name\n");
- free(my_data);
- mixer_close(adev->mixer);
- return NULL;
- }
- ALOGD("%s: snd_card_name: %s", __func__, snd_card_name);
-
- my_data->hw_info = hw_info_init(snd_card_name);
- if (!my_data->hw_info) {
- ALOGE("%s: Failed to init hardware info", __func__);
- } else {
- if (platform_is_i2s_ext_modem(snd_card_name, my_data)) {
- ALOGD("%s: Call MIXER_XML_PATH_I2S", __func__);
-
- adev->audio_route = audio_route_init(snd_card_num,
- MIXER_XML_PATH_I2S);
- } else {
- /* Get the codec internal name from the sound card name
- * and form the mixer paths file name dynamically. This
- * is generic way of picking any codec name based mixer
- * files in future with no code change. This code
- * assumes mixer files are formed with format as
- * mixer_paths_internalcodecname.xml
-
- * If this dynamically read mixer files fails to open then it
- * falls back to default mixer file i.e mixer_paths.xml. This is
- * done to preserve backward compatibility but not mandatory as
- * long as the mixer files are named as per above assumption.
- */
- snd_card_name_t = strdup(snd_card_name);
- snd_internal_name = strtok_r(snd_card_name_t, "-", &tmp);
-
- if (snd_internal_name != NULL)
- snd_internal_name = strtok_r(NULL, "-", &tmp);
-
- if (snd_internal_name != NULL) {
- strlcpy(mixer_xml_file, MIXER_XML_BASE_STRING,
- MIXER_PATH_MAX_LENGTH);
- strlcat(mixer_xml_file, MIXER_FILE_DELIMITER,
- MIXER_PATH_MAX_LENGTH);
- strlcat(mixer_xml_file, snd_internal_name,
- MIXER_PATH_MAX_LENGTH);
- strlcat(mixer_xml_file, MIXER_FILE_EXT,
- MIXER_PATH_MAX_LENGTH);
- } else {
- strlcpy(mixer_xml_file, MIXER_XML_DEFAULT_PATH,
- MIXER_PATH_MAX_LENGTH);
- }
-
- if (F_OK == access(mixer_xml_file, 0)) {
- ALOGD("%s: Loading mixer file: %s", __func__, mixer_xml_file);
- if (audio_extn_read_xml(adev, snd_card_num, mixer_xml_file,
- MIXER_XML_PATH_AUXPCM) == -ENOSYS)
- adev->audio_route = audio_route_init(snd_card_num,
- mixer_xml_file);
- } else {
- ALOGD("%s: Loading default mixer file", __func__);
- if(audio_extn_read_xml(adev, snd_card_num, MIXER_XML_DEFAULT_PATH,
- MIXER_XML_PATH_AUXPCM) == -ENOSYS)
- adev->audio_route = audio_route_init(snd_card_num,
- MIXER_XML_DEFAULT_PATH);
- }
- }
- if (!adev->audio_route) {
- ALOGE("%s: Failed to init audio route controls, aborting.",
- __func__);
- if (my_data)
- free(my_data);
- if (snd_card_name)
- free(snd_card_name);
- if (snd_card_name_t)
- free(snd_card_name_t);
- mixer_close(adev->mixer);
- return NULL;
- }
- adev->snd_card = snd_card_num;
- ALOGD("%s: Opened sound card:%d", __func__, snd_card_num);
- break;
- }
- retry_num = 0;
- snd_card_num++;
+ snd_card_name = strdup(mixer_get_name(adev->mixer));
+ if (!snd_card_name) {
+ ALOGE("failed to allocate memory for snd_card_name\n");
mixer_close(adev->mixer);
+ return NULL;
}
- if (snd_card_num >= MAX_SND_CARD) {
- ALOGE("%s: Unable to find correct sound card, aborting.", __func__);
+ my_data = calloc(1, sizeof(struct platform_data));
+ if (!my_data) {
+ ALOGE("failed to allocate platform data");
+ if (snd_card_name)
+ free(snd_card_name);
+ mixer_close(adev->mixer);
+ return NULL;
+ }
+
+ my_data->hw_info = hw_info_init(snd_card_name);
+ if (!my_data->hw_info) {
+ ALOGE("failed to init hw_info");
+ mixer_close(adev->mixer);
+ if (my_data)
+ free(my_data);
+
+ if (snd_card_name)
+ free(snd_card_name);
+ return NULL;
+ }
+
+ if (platform_is_i2s_ext_modem(snd_card_name, my_data)) {
+ ALOGD("%s: Call MIXER_XML_PATH_I2S", __func__);
+
+ adev->audio_route = audio_route_init(adev->snd_card,
+ MIXER_XML_PATH_I2S);
+ } else {
+ /* Get the codec internal name from the sound card name
+ * and form the mixer paths file name dynamically. This
+ * is generic way of picking any codec name based mixer
+ * files in future with no code change. This code
+ * assumes mixer files are formed with format as
+ * mixer_paths_internalcodecname.xml
+
+ * If this dynamically read mixer files fails to open then it
+ * falls back to default mixer file i.e mixer_paths.xml. This is
+ * done to preserve backward compatibility but not mandatory as
+ * long as the mixer files are named as per above assumption.
+ */
+ snd_card_name_t = strdup(snd_card_name);
+ snd_internal_name = strtok_r(snd_card_name_t, "-", &tmp);
+
+ if (snd_internal_name != NULL) {
+ snd_internal_name = strtok_r(NULL, "-", &tmp);
+ }
+ if (snd_internal_name != NULL) {
+ strlcpy(mixer_xml_file, MIXER_XML_BASE_STRING,
+ MIXER_PATH_MAX_LENGTH);
+ strlcat(mixer_xml_file, MIXER_FILE_DELIMITER,
+ MIXER_PATH_MAX_LENGTH);
+ strlcat(mixer_xml_file, snd_internal_name,
+ MIXER_PATH_MAX_LENGTH);
+ strlcat(mixer_xml_file, MIXER_FILE_EXT,
+ MIXER_PATH_MAX_LENGTH);
+ } else {
+ strlcpy(mixer_xml_file, MIXER_XML_DEFAULT_PATH,
+ MIXER_PATH_MAX_LENGTH);
+ }
+
+ if (F_OK == access(mixer_xml_file, 0)) {
+ ALOGD("%s: Loading mixer file: %s", __func__, mixer_xml_file);
+ if (audio_extn_read_xml(adev, adev->snd_card, mixer_xml_file,
+ MIXER_XML_PATH_AUXPCM) == -ENOSYS)
+ adev->audio_route = audio_route_init(adev->snd_card,
+ mixer_xml_file);
+ } else {
+ ALOGD("%s: Loading default mixer file", __func__);
+ if (audio_extn_read_xml(adev, adev->snd_card, MIXER_XML_DEFAULT_PATH,
+ MIXER_XML_PATH_AUXPCM) == -ENOSYS)
+ adev->audio_route = audio_route_init(adev->snd_card,
+ MIXER_XML_DEFAULT_PATH);
+ }
+ }
+ if (!adev->audio_route) {
+ ALOGE("%s: Failed to init audio route controls, aborting.",
+ __func__);
if (my_data)
free(my_data);
if (snd_card_name)
@@ -1966,9 +1933,9 @@
/* Initialize ACDB ID's */
if (my_data->is_i2s_ext_modem)
- platform_info_init(PLATFORM_INFO_XML_PATH_I2S, my_data);
+ platform_info_init(PLATFORM_INFO_XML_PATH_I2S, my_data, PLATFORM);
else
- platform_info_init(PLATFORM_INFO_XML_PATH, my_data);
+ platform_info_init(PLATFORM_INFO_XML_PATH, my_data, PLATFORM);
my_data->voice_feature_set = VOICE_FEATURE_SET_DEFAULT;
my_data->acdb_handle = dlopen(LIB_ACDB_LOADER, RTLD_NOW);
@@ -2071,12 +2038,23 @@
ALOGE("%s: dlsym error %s for acdb_loader_reload_acdb_files", __func__, dlerror());
goto acdb_init_fail;
}
- platform_acdb_init(my_data);
+
+ int result = acdb_init(adev->snd_card);
+ if (!result) {
+ my_data->is_acdb_initialized = true;
+ ALOGD("ACDB initialized");
+ audio_hwdep_send_cal(my_data);
+ } else {
+ my_data->is_acdb_initialized = false;
+ ALOGD("ACDB initialization failed");
+ }
}
/* init keep-alive for compress passthru */
audio_extn_keep_alive_init(adev);
-
+#ifdef DYNAMIC_LOG_ENABLED
+ log_utils_init();
+#endif
acdb_init_fail:
@@ -2277,6 +2255,9 @@
/* deinit usb */
audio_extn_usb_deinit();
audio_extn_dap_hal_deinit();
+#ifdef DYNAMIC_LOG_ENABLED
+ log_utils_deinit();
+#endif
}
static int platform_is_acdb_initialized(void *platform)
@@ -6054,9 +6035,18 @@
audio_offload_info_t* info)
{
uint32_t fragment_size = MIN_COMPRESS_PASSTHROUGH_FRAGMENT_SIZE;
+ char value[PROPERTY_VALUE_MAX] = {0};
+
+ if (((info->format == AUDIO_FORMAT_DOLBY_TRUEHD) ||
+ (info->format == AUDIO_FORMAT_IEC61937)) &&
+ property_get("audio.truehd.buffer.size.kb", value, "") &&
+ atoi(value)) {
+ fragment_size = atoi(value) * 1024;
+ goto done;
+ }
if (!info->has_video)
fragment_size = MIN_COMPRESS_PASSTHROUGH_FRAGMENT_SIZE;
-
+done:
return fragment_size;
}
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 269aedc..1b6c1f1 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -31,7 +31,11 @@
#define SAMPLE_RATE_11025 11025
#define sample_rate_multiple(sr, base) ((sr % base)== 0?true:false)
#define MAX_VOLUME_CAL_STEPS 15
-#define ACDB_METAINFO_KEY_MODULE_NAME_LEN 100
+
+typedef enum {
+ PLATFORM,
+ ACDB_EXTN,
+} caller_t;
struct amp_db_and_gain_table {
float amp;
@@ -142,7 +146,7 @@
int platform_get_snd_device_backend_index(snd_device_t device);
/* From platform_info.c */
-int platform_info_init(const char *filename, void *);
+int platform_info_init(const char *filename, void *, caller_t);
void platform_snd_card_update(void *platform, int snd_scard_state);
diff --git a/hal/platform_info.c b/hal/platform_info.c
index 6b64261..597d1f7 100644
--- a/hal/platform_info.c
+++ b/hal/platform_info.c
@@ -36,10 +36,17 @@
#include <cutils/log.h>
#include <cutils/str_parms.h>
#include <audio_hw.h>
+#include "acdb.h"
#include "platform_api.h"
#include <platform.h>
#include <math.h>
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_PLATFORM_INFO
+#include <log_utils.h>
+#endif
+
#define BUF_SIZE 1024
typedef enum {
@@ -81,6 +88,7 @@
static section_t section;
struct platform_info {
+ caller_t caller;
void *platform;
struct str_parms *kvpairs;
};
@@ -369,9 +377,21 @@
}
int key = atoi((char *)attr[3]);
- if (platform_set_acdb_metainfo_key(my_data.platform, (char*)attr[1], key) < 0) {
- ALOGE("%s: key %d was not set!", __func__, key);
- goto done;
+ switch(my_data.caller) {
+ case ACDB_EXTN:
+ if(acdb_set_metainfo_key(my_data.platform, (char*)attr[1], key) < 0) {
+ ALOGE("%s: key %d was not set!", __func__, key);
+ goto done;
+ }
+ break;
+ case PLATFORM:
+ if(platform_set_acdb_metainfo_key(my_data.platform, (char*)attr[1], key) < 0) {
+ ALOGE("%s: key %d was not set!", __func__, key);
+ goto done;
+ }
+ break;
+ default:
+ ALOGE("%s: unknown caller!", __func__);
}
done:
@@ -381,58 +401,73 @@
static void start_tag(void *userdata __unused, const XML_Char *tag_name,
const XML_Char **attr)
{
- if (strcmp(tag_name, "bit_width_configs") == 0) {
- section = BITWIDTH;
- } else if (strcmp(tag_name, "acdb_ids") == 0) {
- section = ACDB;
- } else if (strcmp(tag_name, "pcm_ids") == 0) {
- section = PCM_ID;
- } else if (strcmp(tag_name, "backend_names") == 0) {
- section = BACKEND_NAME;
- } else if (strcmp(tag_name, "config_params") == 0) {
- section = CONFIG_PARAMS;
- } else if (strcmp(tag_name, "interface_names") == 0) {
- section = INTERFACE_NAME;
- } else if (strcmp(tag_name, "gain_db_to_level_mapping") == 0) {
- section = GAIN_LEVEL_MAPPING;
- } else if(strcmp(tag_name, "acdb_metainfo_key") == 0) {
- section = ACDB_METAINFO_KEY;
- } else if (strcmp(tag_name, "device") == 0) {
- if ((section != ACDB) && (section != BACKEND_NAME) && (section != BITWIDTH) &&
- (section != INTERFACE_NAME)) {
- ALOGE("device tag only supported for acdb/backend names/bitwitdh/interface names");
- return;
- }
+ if (my_data.caller == ACDB_EXTN) {
+ if(strcmp(tag_name, "acdb_metainfo_key") == 0) {
+ section = ACDB_METAINFO_KEY;
+ } else if (strcmp(tag_name, "param") == 0) {
+ if ((section != CONFIG_PARAMS) && (section != ACDB_METAINFO_KEY)) {
+ ALOGE("param tag only supported with CONFIG_PARAMS section");
+ return;
+ }
- /* call into process function for the current section */
- section_process_fn fn = section_table[section];
- fn(attr);
- } else if (strcmp(tag_name, "gain_level_map") == 0) {
- if (section != GAIN_LEVEL_MAPPING) {
- ALOGE("usecase tag only supported with GAIN_LEVEL_MAPPING section");
- return;
+ section_process_fn fn = section_table[section];
+ fn(attr);
}
+ } else if(my_data.caller == PLATFORM) {
+ if (strcmp(tag_name, "bit_width_configs") == 0) {
+ section = BITWIDTH;
+ } else if (strcmp(tag_name, "acdb_ids") == 0) {
+ section = ACDB;
+ } else if (strcmp(tag_name, "pcm_ids") == 0) {
+ section = PCM_ID;
+ } else if (strcmp(tag_name, "backend_names") == 0) {
+ section = BACKEND_NAME;
+ } else if (strcmp(tag_name, "config_params") == 0) {
+ section = CONFIG_PARAMS;
+ } else if (strcmp(tag_name, "interface_names") == 0) {
+ section = INTERFACE_NAME;
+ } else if (strcmp(tag_name, "gain_db_to_level_mapping") == 0) {
+ section = GAIN_LEVEL_MAPPING;
+ } else if(strcmp(tag_name, "acdb_metainfo_key") == 0) {
+ section = ACDB_METAINFO_KEY;
+ } else if (strcmp(tag_name, "device") == 0) {
+ if ((section != ACDB) && (section != BACKEND_NAME) && (section != BITWIDTH) &&
+ (section != INTERFACE_NAME)) {
+ ALOGE("device tag only supported for acdb/backend names/bitwitdh/interface names");
+ return;
+ }
- section_process_fn fn = section_table[GAIN_LEVEL_MAPPING];
- fn(attr);
- } else if (strcmp(tag_name, "usecase") == 0) {
- if (section != PCM_ID) {
- ALOGE("usecase tag only supported with PCM_ID section");
- return;
+ /* call into process function for the current section */
+ section_process_fn fn = section_table[section];
+ fn(attr);
+ } else if (strcmp(tag_name, "gain_level_map") == 0) {
+ if (section != GAIN_LEVEL_MAPPING) {
+ ALOGE("usecase tag only supported with GAIN_LEVEL_MAPPING section");
+ return;
+ }
+
+ section_process_fn fn = section_table[GAIN_LEVEL_MAPPING];
+ fn(attr);
+ } else if (strcmp(tag_name, "usecase") == 0) {
+ if (section != PCM_ID) {
+ ALOGE("usecase tag only supported with PCM_ID section");
+ return;
+ }
+
+ section_process_fn fn = section_table[PCM_ID];
+ fn(attr);
+ } else if (strcmp(tag_name, "param") == 0) {
+ if ((section != CONFIG_PARAMS) && (section != ACDB_METAINFO_KEY)) {
+ ALOGE("param tag only supported with CONFIG_PARAMS section");
+ return;
+ }
+
+ section_process_fn fn = section_table[section];
+ fn(attr);
}
-
- section_process_fn fn = section_table[PCM_ID];
- fn(attr);
- } else if (strcmp(tag_name, "param") == 0) {
- if ((section != CONFIG_PARAMS) && (section != ACDB_METAINFO_KEY)) {
- ALOGE("param tag only supported with CONFIG_PARAMS section");
- return;
- }
-
- section_process_fn fn = section_table[section];
- fn(attr);
+ } else {
+ ALOGE("%s: unknown caller!", __func__);
}
-
return;
}
@@ -448,7 +483,9 @@
section = ROOT;
} else if (strcmp(tag_name, "config_params") == 0) {
section = ROOT;
- platform_set_parameters(my_data.platform, my_data.kvpairs);
+ if (my_data.caller == PLATFORM) {
+ platform_set_parameters(my_data.platform, my_data.kvpairs);
+ }
} else if (strcmp(tag_name, "interface_names") == 0) {
section = ROOT;
} else if (strcmp(tag_name, "gain_db_to_level_mapping") == 0) {
@@ -458,7 +495,7 @@
}
}
-int platform_info_init(const char *filename, void *platform)
+int platform_info_init(const char *filename, void *platform, caller_t caller_type)
{
XML_Parser parser;
FILE *file;
@@ -483,6 +520,7 @@
goto err_close_file;
}
+ my_data.caller = caller_type;
my_data.platform = platform;
my_data.kvpairs = str_parms_create();
diff --git a/hal/voice.c b/hal/voice.c
index 852c3e6..5a3ff33 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2016, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2017, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -34,6 +34,12 @@
#include "platform_api.h"
#include "audio_extn.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_VOICE
+#include <log_utils.h>
+#endif
+
struct pcm_config pcm_config_voice_call = {
.channels = 1,
.rate = 8000,
diff --git a/hal/voice_extn/compress_voip.c b/hal/voice_extn/compress_voip.c
index f23ff5b..6448b38 100644
--- a/hal/voice_extn/compress_voip.c
+++ b/hal/voice_extn/compress_voip.c
@@ -36,6 +36,12 @@
#include "platform.h"
#include "voice_extn.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_COMPR_VOIP
+#include <log_utils.h>
+#endif
+
#define COMPRESS_VOIP_IO_BUF_SIZE_NB 320
#define COMPRESS_VOIP_IO_BUF_SIZE_WB 640
#define COMPRESS_VOIP_IO_BUF_SIZE_SWB 1280
@@ -288,7 +294,7 @@
ALOGV("%s: unexpected because out_stream_count=%d, in_stream_count=%d",
__func__, voip_data.out_stream_count, voip_data.in_stream_count);
uc_info = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
- if (uc_info)
+ if (uc_info && !voip_data.out_stream_count)
uc_info->stream.out = adev->primary_output;
ret = -EINVAL;
}
diff --git a/hal/voice_extn/voice_extn.c b/hal/voice_extn/voice_extn.c
index 3cd3e78..8bc782d 100644
--- a/hal/voice_extn/voice_extn.c
+++ b/hal/voice_extn/voice_extn.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013-2016, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2013-2017, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2013 The Android Open Source Project
@@ -35,6 +35,12 @@
#include "platform_api.h"
#include "voice_extn.h"
+#ifdef DYNAMIC_LOG_ENABLED
+#include <log_xml_parser.h>
+#define LOG_MASK HAL_MOD_FILE_VOICE_EXTN
+#include <log_utils.h>
+#endif
+
#define AUDIO_PARAMETER_KEY_VSID "vsid"
#define AUDIO_PARAMETER_KEY_CALL_STATE "call_state"
#define AUDIO_PARAMETER_KEY_AUDIO_MODE "audio_mode"
diff --git a/qahw_api/src/qahw.c b/qahw_api/src/qahw.c
index c5cd636..df69df5 100644
--- a/qahw_api/src/qahw.c
+++ b/qahw_api/src/qahw.c
@@ -47,6 +47,10 @@
*/
#define QAHW_MODULE_API_VERSION_CURRENT QAHW_MODULE_API_VERSION_0_0
+
+typedef uint64_t (*qahwi_out_write_v2_t)(audio_stream_out_t *out, const void* buffer,
+ size_t bytes, int64_t* timestamp);
+
typedef int (*qahwi_get_param_data_t) (const audio_hw_device_t *,
qahw_param_id, qahw_param_payload *);
@@ -90,6 +94,7 @@
pthread_mutex_t lock;
qahwi_out_set_param_data_t qahwi_out_get_param_data;
qahwi_out_get_param_data_t qahwi_out_set_param_data;
+ qahwi_out_write_v2_t qahwi_out_write_v2;
} qahw_stream_out_t;
typedef struct {
@@ -535,10 +540,13 @@
}
/*TBD:: validate other meta data parameters */
-
pthread_mutex_lock(&qahw_stream_out->lock);
out = qahw_stream_out->stream;
- if (out->write) {
+ if (qahw_stream_out->qahwi_out_write_v2) {
+ rc = qahw_stream_out->qahwi_out_write_v2(out, out_buf->buffer,
+ out_buf->bytes, out_buf->timestamp);
+ out_buf->offset = 0;
+ } else if (out->write) {
rc = out->write(out, out_buf->buffer, out_buf->bytes);
} else {
rc = -ENOSYS;
@@ -1468,6 +1476,19 @@
}
}
+ /* dlsym qahwi_out_write_v2 */
+ if (!rc) {
+ const char *error;
+
+ /* clear any existing errors */
+ dlerror();
+ qahw_stream_out->qahwi_out_write_v2 = (qahwi_out_write_v2_t)dlsym(qahw_module->module->dso, "qahwi_out_write_v2");
+ if ((error = dlerror()) != NULL) {
+ ALOGI("%s: dlsym error %s for qahwi_out_write_v2", __func__, error);
+ qahw_stream_out->qahwi_out_write_v2 = NULL;
+ }
+ }
+
exit:
pthread_mutex_unlock(&qahw_module->lock);
return rc;
diff --git a/qahw_api/test/qahw_multi_record_test.c b/qahw_api/test/qahw_multi_record_test.c
index c9f8b03..f0720f2 100644
--- a/qahw_api/test/qahw_multi_record_test.c
+++ b/qahw_api/test/qahw_multi_record_test.c
@@ -89,6 +89,40 @@
static pthread_mutex_t sourcetrack_lock;
struct qahw_sound_focus_param sound_focus_data;
+static bool request_wake_lock(bool wakelock_acquired, bool enable)
+{
+ int system_ret;
+
+ if (enable) {
+ if (!wakelock_acquired) {
+ system_ret = system("echo audio_services > /sys/power/wake_lock");
+ if (system_ret < 0) {
+ fprintf(stderr, "%s.Failed to acquire audio_service lock\n", __func__);
+ fprintf(log_file, "%s.Failed to acquire audio_service lock\n", __func__);
+ } else {
+ wakelock_acquired = true;
+ fprintf(log_file, "%s.Success to acquire audio_service lock\n", __func__);
+ }
+ } else
+ fprintf(log_file, "%s.Lock is already acquired\n", __func__);
+ }
+
+ if (!enable) {
+ if (wakelock_acquired) {
+ system_ret = system("echo audio_services > /sys/power/wake_unlock");
+ if (system_ret < 0) {
+ fprintf(stderr, "%s.Failed to release audio_service lock\n", __func__);
+ fprintf(log_file, "%s.Failed to release audio_service lock\n", __func__);
+ } else {
+ wakelock_acquired = false;
+ fprintf(log_file, "%s.Success to release audio_service lock\n", __func__);
+ }
+ } else
+ fprintf(log_file, "%s.No Lock is acquired to release\n", __func__);
+ }
+ return wakelock_acquired;
+}
+
void stop_signal_handler(int signal __unused)
{
stop_record = true;
@@ -295,9 +329,12 @@
strlcat(param, params->profile, sizeof(param));
qahw_in_set_parameters(in_handle, param);
- fprintf(log_file, "\n Please speak into the microphone for %lf seconds, handle(%d)\n", params->record_length, params->handle);
+ /* Caution: Below ADL log shouldnt be altered without notifying automation APT since it used for
+ * automation testing
+ */
+ fprintf(log_file, "\n ADL: Please speak into the microphone for %lf seconds, handle(%d)\n", params->record_length, params->handle);
if (log_file != stdout)
- fprintf(stdout, "\n Please speak into the microphone for %lf seconds, handle(%d)\n", params->record_length, params->handle);
+ fprintf(stdout, "\n ADL: Please speak into the microphone for %lf seconds, handle(%d)\n", params->record_length, params->handle);
snprintf(file_name + name_len, sizeof(file_name) - name_len, "%d.wav", (0x99A - params->handle));
FILE *fd = fopen(file_name,"w");
@@ -433,14 +470,17 @@
fprintf(stdout, "could not close input stream %d, handle(%d)\n",rc, params->handle);
}
- /* Print instructions to access the file. */
- fprintf(log_file, "\n\n The audio recording has been saved to %s. Please use adb pull to get "
+ /* Print instructions to access the file.
+ * Caution: Below ADL log shouldnt be altered without notifying automation APT since it used for
+ * automation testing
+ */
+ fprintf(log_file, "\n\n ADL: The audio recording has been saved to %s. Please use adb pull to get "
"the file and play it using audacity. The audio data has the "
"following characteristics:\n Sample rate: %i\n Format: %d\n "
"Num channels: %i, handle(%d)\n\n",
file_name, params->config.sample_rate, params->config.format, params->channels, params->handle);
if (log_file != stdout)
- fprintf(stdout, "\n\n The audio recording has been saved to %s. Please use adb pull to get "
+ fprintf(stdout, "\n\n ADL: The audio recording has been saved to %s. Please use adb pull to get "
"the file and play it using audacity. The audio data has the "
"following characteristics:\n Sample rate: %i\n Format: %d\n "
"Num channels: %i, handle(%d)\n\n",
@@ -547,6 +587,7 @@
bool interactive_mode = false, source_tracking = false;
struct listnode param_list;
char log_filename[256] = "stdout";
+ bool wakelock_acquired = false;
log_file = stdout;
list_init(¶m_list);
@@ -624,6 +665,7 @@
}
}
+ wakelock_acquired = request_wake_lock(wakelock_acquired, true);
qahw_mod_handle = qahw_load_module(mod_name);
if(qahw_mod_handle == NULL) {
fprintf(log_file, " qahw_load_module failed");
@@ -857,10 +899,14 @@
fprintf(log_file, "could not unload hal %d \n",ret);
}
- fprintf(log_file, "\n Done with hal record test \n");
+ /* Caution: Below ADL log shouldnt be altered without notifying automation APT since it used
+ * for automation testing
+ */
+ fprintf(log_file, "\n ADL: Done with hal record test \n");
if (log_file != stdout) {
- fprintf(stdout, "\n Done with hal record test \n");
+ fprintf(stdout, "\n ADL: Done with hal record test \n");
fclose(log_file);
}
+ wakelock_acquired = request_wake_lock(wakelock_acquired, false);
return 0;
}
diff --git a/qahw_api/test/qahw_playback_test.c b/qahw_api/test/qahw_playback_test.c
index 9fe713d..c28f41e 100644
--- a/qahw_api/test/qahw_playback_test.c
+++ b/qahw_api/test/qahw_playback_test.c
@@ -58,6 +58,10 @@
#define KVPAIRS_MAX 100
#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[1]))
+#define FORMAT_DESCRIPTOR_SIZE 12
+#define SUBCHUNK1_SIZE(x) ((8) + (x))
+#define SUBCHUNK2_SIZE 8
+
static int get_wav_header_length (FILE* file_stream);
static void init_streams(void);
@@ -238,6 +242,40 @@
"music_offload_wma_encode_option2=%d;" \
"music_offload_wma_format_tag=%d;"
+static bool request_wake_lock(bool wakelock_acquired, bool enable)
+{
+ int system_ret;
+
+ if (enable) {
+ if (!wakelock_acquired) {
+ system_ret = system("echo audio_services > /sys/power/wake_lock");
+ if (system_ret < 0) {
+ fprintf(stderr, "%s.Failed to acquire audio_service lock\n", __func__);
+ fprintf(log_file, "%s.Failed to acquire audio_service lock\n", __func__);
+ } else {
+ wakelock_acquired = true;
+ fprintf(log_file, "%s.Success to acquire audio_service lock\n", __func__);
+ }
+ } else
+ fprintf(log_file, "%s.Lock is already acquired\n", __func__);
+ }
+
+ if (!enable) {
+ if (wakelock_acquired) {
+ system_ret = system("echo audio_services > /sys/power/wake_unlock");
+ if (system_ret < 0) {
+ fprintf(stderr, "%s.Failed to release audio_service lock\n", __func__);
+ fprintf(log_file, "%s.Failed to release audio_service lock\n", __func__);
+ } else {
+ wakelock_acquired = false;
+ fprintf(log_file, "%s.Success to release audio_service lock\n", __func__);
+ }
+ } else
+ fprintf(log_file, "%s.No Lock is acquired to release\n", __func__);
+ }
+ return wakelock_acquired;
+}
+
void stop_signal_handler(int signal __unused)
{
stop_playback = true;
@@ -374,7 +412,7 @@
qahw_in_buffer_t in_buf;
char *buffer;
int rc = 0;
- int bytes_to_read, bytes_written = 0;
+ int bytes_to_read, bytes_written = 0, bytes_wrote = 0;
FILE *fp = NULL;
qahw_stream_handle_t* in_handle = nullptr;
@@ -416,7 +454,13 @@
while (!(params->acp.thread_exit)) {
rc = qahw_in_read(in_handle, &in_buf);
if (rc > 0) {
- bytes_written += fwrite((char *)(in_buf.buffer), sizeof(char), (int)in_buf.bytes, fp);
+ bytes_wrote = fwrite((char *)(in_buf.buffer), sizeof(char), (int)in_buf.bytes, fp);
+ bytes_written += bytes_wrote;
+ if(bytes_wrote < in_buf.bytes) {
+ stop_playback = true;
+ fprintf(log_file, "Error in fwrite due to no memory(%d)=%s\n",ferror(fp), strerror(ferror(fp)));
+ break;
+ }
}
}
params->hdr.data_sz = bytes_written;
@@ -508,8 +552,6 @@
pthread_t drift_query_thread;
struct drift_data drift_params;
- if (params->output_device & AUDIO_DEVICE_OUT_ALL_A2DP)
- params->output_device = AUDIO_DEVICE_OUT_PROXY;
rc = qahw_open_output_stream(params->qahw_out_hal_handle,
params->handle,
params->output_device,
@@ -680,9 +722,12 @@
qahw_out_drain(params->out_handle, QAHW_DRAIN_ALL);
pthread_cond_wait(¶ms->drain_cond, ¶ms->drain_lock);
fprintf(log_file, "stream %d: out of compress drain\n", params->stream_index);
- fprintf(log_file, "stream %d: playback completed successfully\n", params->stream_index);
pthread_mutex_unlock(¶ms->drain_lock);
}
+ /* Caution: Below ADL log shouldnt be altered without notifying automation APT since
+ * it used for automation testing
+ */
+ fprintf(log_file, "ADL: stream %d: playback completed successfully\n", params->stream_index);
}
exit = true;
continue;
@@ -694,6 +739,10 @@
fprintf(log_file, "stream %d: writing to hal %zd bytes, offset %d, write length %zd\n",
params->stream_index, bytes_remaining, offset, write_length);
bytes_written = write_to_hal(params->out_handle, data_ptr+offset, bytes_remaining, params);
+ if (bytes_written == -1) {
+ fprintf(stderr, "proxy_write failed in usb hal");
+ break;
+ }
bytes_remaining -= bytes_written;
latency = qahw_out_get_latency(params->out_handle);
@@ -974,7 +1023,7 @@
}
stream_info->config.sample_rate = stream_info->config.offload_info.sample_rate;
stream_info->config.format = stream_info->config.offload_info.format;
- stream_info->config.channel_mask = stream_info->config.offload_info.channel_mask = audio_channel_in_mask_from_count(stream_info->channels);
+ stream_info->config.channel_mask = stream_info->config.offload_info.channel_mask = audio_channel_out_mask_from_count(stream_info->channels);
return;
}
@@ -1071,7 +1120,7 @@
event_payload.num_events = 1;
event_payload.event_id = 0x13236;
- event_payload.module_id = 0x10EEC;
+ event_payload.module_id = 0x10940;
event_payload.config_mask = 1;
payload.adsp_event_params.payload_length = sizeof(event_payload);
@@ -1145,6 +1194,9 @@
return -1;
parms = str_parms_create_str(kvpairs);
+ if (parms == NULL)
+ return -1;
+
if (str_parms_get_str(parms, key, value, KVPAIRS_MAX) < 0)
return -1;
@@ -1169,7 +1221,7 @@
/*
* for now we assume usb hal/pcm device announces suport for one format ONLY
*/
- for (i = 0; i < sizeof(format_table); i++) {
+ for (i = 0; i < (sizeof(format_table)/sizeof(format_table[0])); i++) {
if(!strncmp(format_table[i].string, value, sizeof(value))) {
match = true;
break;
@@ -1307,8 +1359,8 @@
param_string = qahw_out_get_parameters(stream->out_handle, QAHW_PARAMETER_STREAM_SUP_CHANNELS);
if ((ch = get_channels(param_string)) <= 0) {
- fprintf(log_file, "Unable to extract channels =(%d) string(%s)\n", ch, param_string);
- fprintf(stderr, "Unable to extract channels =(%d) string(%s)\n", ch, param_string);
+ fprintf(log_file, "Unable to extract channels =(%d) string(%s)\n", ch, param_string == NULL ? "null":param_string);
+ fprintf(stderr, "Unable to extract channels =(%d) string(%s)\n", ch, param_string == NULL ? "null":param_string);
return -1;
}
stream->config.channel_mask = audio_channel_in_mask_from_count(ch);
@@ -1403,10 +1455,13 @@
printf(" hal_play_test -f /data/MateRani.mp3 -t 2 -d 2 -v 0.01 -r 44100 -c 2 \n");
printf(" -> plays MP3 stream(-t = 2) on speaker device(-d = 2)\n");
printf(" -> 2 channels and 44100 sample rate\n\n");
- printf(" hal_play_test -f /data/v1-CBR-32kHz-stereo-40kbps.mp3 -t 2 -d 128 -v 0.01 -r 32000 -c 2 -D /data/proxy_dump.wav\n");
- printf(" -> plays MP3 stream(-t = 2) on BT device(-d = 128)\n");
+ printf(" hal_play_test -f /data/v1-CBR-32kHz-stereo-40kbps.mp3 -t 2 -d 33554432 -v 0.01 -r 32000 -c 2 -D /data/proxy_dump.wav\n");
+ printf(" -> plays MP3 stream(-t = 2) on BT device in non-split path (-d = 33554432)\n");
printf(" -> 2 channels and 32000 sample rate\n");
printf(" -> dumps pcm data to file at /data/proxy_dump.wav\n\n");
+ printf(" hal_play_test -f /data/v1-CBR-32kHz-stereo-40kbps.mp3 -t 2 -d 128 -v 0.01 -r 32000 -c 2 \n");
+ printf(" -> plays MP3 stream(-t = 2) on BT device in split path (-d = 128)\n");
+ printf(" -> 2 channels and 32000 sample rate\n");
printf(" hal_play_test -f /data/AACLC-71-48000Hz-384000bps.aac -t 4 -d 2 -v 0.05 -r 48000 -c 2 -a 1 \n");
printf(" -> plays AAC-ADTS stream(-t = 4) on speaker device(-d = 2)\n");
printf(" -> AAC format type is LC(-a = 1)\n");
@@ -1476,20 +1531,7 @@
fprintf(log_file, "This is not a valid wav file \n");
fprintf(stderr, "This is not a valid wav file \n");
} else {
- switch (subchunk_size) {
- case 16:
- fprintf(log_file, "44-byte wav header \n");
- wav_header_len = 44;
- break;
- case 18:
- fprintf(log_file, "46-byte wav header \n");
- wav_header_len = 46;
- break;
- default:
- fprintf(log_file, "Header contains extra data and is larger than 46 bytes: subchunk_size=%d \n", subchunk_size);
- wav_header_len = subchunk_size;
- break;
- }
+ wav_header_len = FORMAT_DESCRIPTOR_SIZE + SUBCHUNK1_SIZE(subchunk_size) + SUBCHUNK2_SIZE;
}
return wav_header_len;
}
@@ -1578,6 +1620,7 @@
int j = 0;
kpi_mode = false;
event_trigger = false;
+ bool wakelock_acquired = false;
log_file = stdout;
proxy_params.acp.file_name = "/data/pcm_dump.wav";
@@ -1743,8 +1786,12 @@
}
}
+ wakelock_acquired = request_wake_lock(wakelock_acquired, true);
num_of_streams = i+1;
- fprintf(log_file, "Starting audio hal tests for streams : %d\n", num_of_streams);
+ /* Caution: Below ADL log shouldnt be altered without notifying automation APT since it used
+ * for automation testing
+ */
+ fprintf(log_file, "ADL: Starting audio hal tests for streams : %d\n", num_of_streams);
if (kpi_mode == true && num_of_streams > 1) {
fprintf(log_file, "kpi-mode is not supported for multi-playback usecase\n");
@@ -1756,7 +1803,7 @@
goto exit;
}
- if (num_of_streams > 1 && stream_param[num_of_streams-1].output_device & AUDIO_DEVICE_OUT_ALL_A2DP) {
+ if (num_of_streams > 1 && stream_param[num_of_streams-1].output_device & AUDIO_DEVICE_OUT_PROXY) {
fprintf(log_file, "Proxy thread is not supported for multi-playback usecase\n");
fprintf(stderr, "Proxy thread is not supported for multi-playback usecase\n");
goto exit;
@@ -1827,13 +1874,14 @@
} else if (kpi_mode == true)
stream->config.format = stream->config.offload_info.format = AUDIO_FORMAT_PCM_16_BIT;
- if (stream->output_device & AUDIO_DEVICE_OUT_ALL_A2DP)
+ if (stream->output_device & AUDIO_DEVICE_OUT_PROXY)
fprintf(log_file, "Saving pcm data to file: %s\n", proxy_params.acp.file_name);
/* Set device connection state for HDMI */
- if (stream->output_device == AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ if ((stream->output_device == AUDIO_DEVICE_OUT_AUX_DIGITAL) ||
+ (stream->output_device == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP)) {
char param[100] = {0};
- snprintf(param, sizeof(param), "%s=%d", "connect", AUDIO_DEVICE_OUT_AUX_DIGITAL);
+ snprintf(param, sizeof(param), "%s=%d", "connect", stream->output_device);
qahw_set_parameters(stream->qahw_out_hal_handle, param);
}
@@ -1892,16 +1940,17 @@
* reset device connection state for HDMI and close the file streams
*/
for (i = 0; i < num_of_streams; i++) {
- if (stream_param[i].output_device == AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ if ((stream_param[i].output_device == AUDIO_DEVICE_OUT_AUX_DIGITAL) ||
+ (stream_param[i].output_device == AUDIO_DEVICE_OUT_BLUETOOTH_A2DP)) {
char param[100] = {0};
- snprintf(param, sizeof(param), "%s=%d", "disconnect", AUDIO_DEVICE_OUT_AUX_DIGITAL);
+ snprintf(param, sizeof(param), "%s=%d", "disconnect", stream_param[i].output_device);
qahw_set_parameters(stream_param[i].qahw_out_hal_handle, param);
}
if (stream_param[i].file_stream != nullptr)
fclose(stream_param[i].file_stream);
else if (AUDIO_DEVICE_NONE != stream_param[i].input_device) {
- if (stream->in_handle) {
+ if (stream != NULL && stream->in_handle) {
rc = qahw_close_input_stream(stream->in_handle);
if (rc) {
fprintf(log_file, "input stream could not be closed\n");
@@ -1917,6 +1966,10 @@
if ((log_file != stdout) && (log_file != nullptr))
fclose(log_file);
- fprintf(log_file, "\nBYE BYE\n");
+ wakelock_acquired = request_wake_lock(wakelock_acquired, false);
+ /* Caution: Below ADL log shouldnt be altered without notifying automation APT since it used
+ * for automation testing
+ */
+ fprintf(log_file, "\nADL: BYE BYE\n");
return 0;
}