Merge "configs: msm8953: set default IIR volume to 53 for internal codec" into audio-userspace.lnx.2.1-dev
diff --git a/configs/msm8937/audio_policy_configuration.xml b/configs/msm8937/audio_policy_configuration.xml
index 2443d13..44abe28 100644
--- a/configs/msm8937/audio_policy_configuration.xml
+++ b/configs/msm8937/audio_policy_configuration.xml
@@ -81,13 +81,13 @@
<mixPort name="direct_pcm" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
</mixPort>
<mixPort name="compressed_offload" role="source"
diff --git a/configs/msm8953/audio_policy_configuration.xml b/configs/msm8953/audio_policy_configuration.xml
index 2443d13..44abe28 100644
--- a/configs/msm8953/audio_policy_configuration.xml
+++ b/configs/msm8953/audio_policy_configuration.xml
@@ -81,13 +81,13 @@
<mixPort name="direct_pcm" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
</mixPort>
<mixPort name="compressed_offload" role="source"
diff --git a/configs/msm8996/audio_policy_configuration.xml b/configs/msm8996/audio_policy_configuration.xml
index 6f36be5..e8d4cd0 100644
--- a/configs/msm8996/audio_policy_configuration.xml
+++ b/configs/msm8996/audio_policy_configuration.xml
@@ -81,13 +81,13 @@
<mixPort name="direct_pcm" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
</mixPort>
<mixPort name="compressed_offload" role="source"
diff --git a/configs/msmcobalt/audio_policy_configuration.xml b/configs/msmcobalt/audio_policy_configuration.xml
index 2503bd2..66b7d17 100644
--- a/configs/msmcobalt/audio_policy_configuration.xml
+++ b/configs/msmcobalt/audio_policy_configuration.xml
@@ -86,13 +86,13 @@
<mixPort name="direct_pcm" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000,352800"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
</mixPort>
<mixPort name="compressed_offload" role="source"
diff --git a/configs/msmcobalt/mixer_paths_tasha.xml b/configs/msmcobalt/mixer_paths_tasha.xml
index 038408c..efd275d 100644
--- a/configs/msmcobalt/mixer_paths_tasha.xml
+++ b/configs/msmcobalt/mixer_paths_tasha.xml
@@ -205,12 +205,6 @@
<ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
<ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
<ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
- <ctl name="USB_AUDIO_RX Channels" value="One" />
- <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
- <ctl name="USB_AUDIO_RX Format" value="S16_LE" />
- <ctl name="USB_AUDIO_TX Channels" value="One" />
- <ctl name="USB_AUDIO_TX SampleRate" value="KHZ_48" />
- <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
<ctl name="MultiMedia6 Mixer SLIM_0_TX" value="0" />
<ctl name="IIR0 INP0 MUX" value="ZERO" />
<ctl name="IIR0 INP1 MUX" value="ZERO" />
@@ -399,7 +393,7 @@
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia5" value="0" />
<!-- usb headset end -->
<!-- fm -->
- <ctl name="SLIMBUS_8 LOOPBACK Volume" value="1" />
+ <ctl name="SLIMBUS_8 LOOPBACK Volume" value="0" />
<ctl name="SLIMBUS_0_RX Port Mixer SLIM_8_TX" value="0" />
<ctl name="SLIMBUS_DL_HL Switch" value="0" />
<ctl name="SLIMBUS_6_RX Port Mixer SLIM_8_TX" value="0" />
diff --git a/configs/msmcobalt/mixer_paths_tavil.xml b/configs/msmcobalt/mixer_paths_tavil.xml
index 4baaa52..34543f5 100644
--- a/configs/msmcobalt/mixer_paths_tavil.xml
+++ b/configs/msmcobalt/mixer_paths_tavil.xml
@@ -166,12 +166,6 @@
<ctl name="MultiMedia1 Mixer USB_AUDIO_TX" value="0" />
<ctl name="MultiMedia5 Mixer USB_AUDIO_TX" value="0" />
<ctl name="MultiMedia8 Mixer USB_AUDIO_TX" value="0" />
- <ctl name="USB_AUDIO_RX Channels" value="One" />
- <ctl name="USB_AUDIO_RX SampleRate" value="KHZ_48" />
- <ctl name="USB_AUDIO_RX Format" value="S16_LE" />
- <ctl name="USB_AUDIO_TX Channels" value="One" />
- <ctl name="USB_AUDIO_TX SampleRate" value="KHZ_48" />
- <ctl name="USB_AUDIO_TX Format" value="S16_LE" />
<ctl name="MultiMedia6 Mixer SLIM_0_TX" value="0" />
<ctl name="SLIM_2_RX Format" value="UNPACKED" />
<ctl name="SLIM_2_RX SampleRate" value="KHZ_48" />
@@ -222,7 +216,7 @@
<ctl name="AFE_PCM_RX Audio Mixer MultiMedia5" value="0" />
<!-- usb headset end -->
<!-- fm -->
- <ctl name="SLIMBUS_8 LOOPBACK Volume" value="1" />
+ <ctl name="SLIMBUS_8 LOOPBACK Volume" value="0" />
<ctl name="SLIMBUS_0_RX Port Mixer SLIM_8_TX" value="0" />
<ctl name="SLIMBUS_DL_HL Switch" value="0" />
<ctl name="SLIMBUS_6_RX Port Mixer SLIM_8_TX" value="0" />
diff --git a/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml b/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml
index d12b62f..be77fee 100755
--- a/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml
+++ b/configs/msmcobalt/sound_trigger_mixer_paths_wcd9340.xml
@@ -106,10 +106,12 @@
</path>
<path name="listen-ape-handset-mic">
- <ctl name="MAD_BROADCAST Switch" value="1" />
- <ctl name="TX13 INP MUX" value="MAD_BRDCST" />
- <ctl name="AIF4_MAD Mixer SLIM TX13" value="1" />
<ctl name="MAD Input" value="DMIC0" />
+ <ctl name="MAD_SEL MUX" value="MSM" />
+ <ctl name="MAD_INP MUX" value="MAD" />
+ <ctl name="MAD_BROADCAST Switch" value="1" />
+ <ctl name="CDC_IF TX13 MUX" value="MAD_BRDCST" />
+ <ctl name="AIF4_MAD Mixer SLIM TX13" value="1" />
</path>
</mixer>
diff --git a/configs/msmfalcon/audio_policy_configuration.xml b/configs/msmfalcon/audio_policy_configuration.xml
index ea4b140..b1ea1b9 100644
--- a/configs/msmfalcon/audio_policy_configuration.xml
+++ b/configs/msmfalcon/audio_policy_configuration.xml
@@ -81,13 +81,13 @@
<mixPort name="direct_pcm" role="source"
flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_DIRECT_PCM">
<profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_8_24_BIT"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
<profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
- samplingRates="8000,11025,16000,22050,32000,44100,48000,64000,88200,96000,128000,176400,192000"
+ samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000,64000,88200,96000,128000,176400,192000"
channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_2POINT1,AUDIO_CHANNEL_OUT_QUAD,AUDIO_CHANNEL_OUT_PENTA,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_6POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
</mixPort>
<mixPort name="compressed_offload" role="source"
diff --git a/hal/Android.mk b/hal/Android.mk
index f457c7b..daf7397 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -248,6 +248,12 @@
LOCAL_SRC_FILES += audio_extn/a2dp.c
endif
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_QAF)),true)
+ LOCAL_CFLAGS += -DQAF_EXTN_ENABLED
+ LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/qaf/
+ LOCAL_SRC_FILES += audio_extn/qaf.c
+endif
+
LOCAL_SHARED_LIBRARIES := \
liblog \
libcutils \
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index ccf3d64..b57bb81 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -459,9 +459,11 @@
#ifndef HFP_ENABLED
#define audio_extn_hfp_is_active(adev) (0)
#define audio_extn_hfp_get_usecase() (-1)
+#define hfp_set_mic_mute(dev, state) (0)
#else
bool audio_extn_hfp_is_active(struct audio_device *adev);
audio_usecase_t audio_extn_hfp_get_usecase();
+int hfp_set_mic_mute(struct audio_device *dev, bool state);
#endif
#ifndef DEV_ARBI_ENABLED
@@ -581,6 +583,29 @@
void audio_utils_set_hdmi_channel_status(struct stream_out *out, char * buffer, size_t bytes);
#endif
+#ifdef QAF_EXTN_ENABLED
+bool audio_extn_qaf_is_enabled();
+void audio_extn_qaf_deinit();
+void audio_extn_qaf_close_output_stream(struct audio_hw_device *dev __unused,
+ struct audio_stream_out *stream);
+int audio_extn_qaf_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 __unused);
+int audio_extn_qaf_init(struct audio_device *adev);
+int audio_extn_qaf_set_parameters(struct audio_device *adev, struct str_parms *parms);
+#else
+#define audio_extn_qaf_is_enabled() (0)
+#define audio_extn_qaf_deinit() (0)
+#define audio_extn_qaf_close_output_stream adev_close_output_stream
+#define audio_extn_qaf_open_output_stream adev_open_output_stream
+#define audio_extn_qaf_init(adev) (0)
+#define audio_extn_qaf_set_parameters(adev, parms) (0)
+#endif
+
#ifndef KEEP_ALIVE_ENABLED
#define audio_extn_keep_alive_init(a) do {} while(0)
#define audio_extn_keep_alive_start() do {} while(0)
diff --git a/hal/audio_extn/dev_arbi.c b/hal/audio_extn/dev_arbi.c
index d7ab5ff..69d8568 100644
--- a/hal/audio_extn/dev_arbi.c
+++ b/hal/audio_extn/dev_arbi.c
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2014, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2014, 2016 The Linux Foundation. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are
@@ -131,6 +131,7 @@
{SND_DEVICE_OUT_VOICE_HANDSET, AUDIO_DEVICE_OUT_EARPIECE},
{SND_DEVICE_OUT_SPEAKER, AUDIO_DEVICE_OUT_SPEAKER},
{SND_DEVICE_OUT_VOICE_SPEAKER, AUDIO_DEVICE_OUT_SPEAKER},
+ {SND_DEVICE_OUT_VOICE_SPEAKER_2, AUDIO_DEVICE_OUT_SPEAKER},
{SND_DEVICE_OUT_HEADPHONES, AUDIO_DEVICE_OUT_WIRED_HEADPHONE},
{SND_DEVICE_OUT_VOICE_HEADPHONES, AUDIO_DEVICE_OUT_WIRED_HEADPHONE},
{SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
diff --git a/hal/audio_extn/hfp.c b/hal/audio_extn/hfp.c
index 5a45b80..243d48d 100644
--- a/hal/audio_extn/hfp.c
+++ b/hal/audio_extn/hfp.c
@@ -302,6 +302,26 @@
return false;
}
+int hfp_set_mic_mute(struct audio_device *adev, bool state)
+{
+ struct mixer_ctl *ctl;
+ const char *mixer_ctl_name = "HFP TX Mute";
+ uint32_t set_values[ ] = {0};
+
+ ALOGI("%s: enter, state=%d", __func__, state);
+
+ set_values[0] = state;
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+ mixer_ctl_set_array(ctl, set_values, ARRAY_SIZE(set_values));
+ ALOGV("%s: exit", __func__);
+ return 0;
+}
+
audio_usecase_t audio_extn_hfp_get_usecase()
{
return hfpmod.ucid;
diff --git a/hal/audio_extn/qaf.c b/hal/audio_extn/qaf.c
new file mode 100644
index 0000000..d631275
--- /dev/null
+++ b/hal/audio_extn/qaf.c
@@ -0,0 +1,1129 @@
+/*
+ * Copyright (c) 2016, The Linux Foundation. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are
+ * met:
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above
+ * copyright notice, this list of conditions and the following
+ * disclaimer in the documentation and/or other materials provided
+ * with the distribution.
+ * * Neither the name of The Linux Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED "AS IS" AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR
+ * BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE
+ * OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN
+ * IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#define LOG_TAG "audio_hw_qaf"
+/*#define LOG_NDEBUG 0*/
+/*#define VERY_VERY_VERBOSE_LOGGING*/
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
+#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
+#define QAF_DEFAULT_COMPR_AUDIO_HANDLE 1001
+#define QAF_DEFAULT_COMPR_PASSTHROUGH_HANDLE 1002
+
+#include <stdlib.h>
+#include <pthread.h>
+#include <errno.h>
+#include <dlfcn.h>
+#include <sys/resource.h>
+#include <sys/prctl.h>
+#include <cutils/properties.h>
+#include <cutils/str_parms.h>
+#include <cutils/log.h>
+#include <cutils/atomic.h>
+#include "audio_utils/primitives.h"
+#include "audio_hw.h"
+#include "platform_api.h"
+#include <platform.h>
+#include <system/thread_defs.h>
+#include <cutils/sched_policy.h>
+#include "audio_extn.h"
+#include <qti_audio.h>
+#include "sound/compress_params.h"
+
+#define QAF_OUTPUT_SAMPLING_RATE 48000
+#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 50
+#define QAF_PLAYBACK_LATENCY 30
+
+#define QAF_LATENCY (COMPRESS_OFFLOAD_PLAYBACK_LATENCY + QAF_PLAYBACK_LATENCY)
+
+#ifdef QAF_DUMP_ENABLED
+FILE *fp_output_writer_hdmi = NULL;
+#endif
+
+typedef enum {
+AUDIO_OUTPUT_FLAG_MAIN = 0x4000, // Flag for Main Input Stream
+AUDIO_OUTPUT_FLAG_ASSOCIATED = 0x8000, // Flag for Assocated Input Stream
+} qaf_audio_output_flags_t;
+
+struct qaf {
+ struct audio_device *adev;
+ audio_session_handle_t session_handle;
+ void *qaf_lib;
+ int (*qaf_audio_session_open)(audio_session_handle_t* session_handle, void *p_data, void* license_data);
+ int (*qaf_audio_session_close)(audio_session_handle_t session_handle);
+ int (*qaf_audio_stream_open)(audio_session_handle_t session_handle, audio_stream_handle_t* stream_handle,
+ audio_stream_config_t input_config, audio_devices_t devices, stream_type_t flags);
+ int (*qaf_audio_stream_close)(audio_stream_handle_t stream_handle);
+ int (*qaf_audio_stream_set_param)(audio_stream_handle_t stream_handle, const char* kv_pairs);
+ int (*qaf_audio_session_set_param)(audio_session_handle_t handle, const char* kv_pairs);
+ char* (*qaf_audio_stream_get_param)(audio_stream_handle_t stream_handle, const char* key);
+ char* (*qaf_audio_session_get_param)(audio_session_handle_t handle, const char* key);
+ int (*qaf_audio_stream_start)(audio_stream_handle_t handle);
+ int (*qaf_audio_stream_stop)(audio_stream_handle_t stream_handle);
+ int (*qaf_audio_stream_pause)(audio_stream_handle_t stream_handle);
+ int (*qaf_audio_stream_flush)(audio_stream_handle_t stream_handle);
+ int (*qaf_audio_stream_write)(audio_stream_handle_t stream_handle, const void* buf, int size);
+ void (*qaf_register_event_callback)(audio_session_handle_t session_handle, void *priv_data,
+ notify_event_callback_t event_callback, audio_event_id_t event_id);
+ pthread_mutex_t lock;
+ struct stream_out *stream_drain_main;
+ struct stream_out *qaf_compr_offload_out;
+ struct stream_out *qaf_compr_passthrough_out;
+ int passthrough_enabled;
+ int hdmi_sink_channels;
+ bool multi_ch_out_enabled;
+ bool main_output_active;
+ bool assoc_output_active;
+};
+
+static struct qaf *qaf_mod = NULL;
+
+static void lock_output_stream(struct stream_out *out)
+{
+ pthread_mutex_lock(&out->pre_lock);
+ pthread_mutex_lock(&out->lock);
+ pthread_mutex_unlock(&out->pre_lock);
+}
+
+static int qaf_send_offload_cmd_l(struct stream_out* out, int command)
+{
+ struct offload_cmd *cmd = (struct offload_cmd *)calloc(1, sizeof(struct offload_cmd));
+
+ if (!cmd) {
+ ALOGE("failed to allocate mem for command 0x%x", command);
+ return -ENOMEM;
+ }
+
+ ALOGV("%s %d", __func__, command);
+
+ cmd->cmd = command;
+ list_add_tail(&out->qaf_offload_cmd_list, &cmd->node);
+ pthread_cond_signal(&out->qaf_offload_cond);
+ return 0;
+}
+
+static int audio_extn_qaf_stream_stop(struct stream_out *out)
+{
+ ALOGV("%s: %d start", __func__, __LINE__);
+ if (!qaf_mod->qaf_audio_stream_stop)
+ return -EINVAL;
+
+ return qaf_mod->qaf_audio_stream_stop(out->qaf_stream_handle);
+}
+
+static int qaf_out_standby(struct audio_stream *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = 0;
+
+ ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
+ stream, out->usecase, use_case_table[out->usecase]);
+
+ lock_output_stream(out);
+ if (!out->standby) {
+ out->standby = true;
+ status = audio_extn_qaf_stream_stop(out);
+ }
+ pthread_mutex_unlock(&out->lock);
+ out->written = 0;
+ return status;
+}
+
+static int qaf_stream_set_param(struct stream_out *out, const char *kv_pair)
+{
+ ALOGV("%s %d kvpair: %s", __func__, __LINE__, kv_pair);
+ if (!qaf_mod->qaf_audio_stream_set_param)
+ return -EINVAL;
+
+ return qaf_mod->qaf_audio_stream_set_param(out->qaf_stream_handle, kv_pair);
+}
+
+static int qaf_out_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int ret = 0;
+
+ ALOGV("%s: enter: usecase(%d: %s) kvpairs: %s",
+ __func__, out->usecase, use_case_table[out->usecase], kvpairs);
+ lock_output_stream(out);
+ ret = qaf_stream_set_param(out, kvpairs);
+ pthread_mutex_unlock(&out->lock);
+ if ((NULL != qaf_mod->qaf_compr_offload_out)) {
+ qaf_mod->qaf_compr_offload_out->stream.common.set_parameters((struct audio_stream *) qaf_mod->qaf_compr_offload_out, kvpairs);
+ }
+ return ret;
+}
+
+static int qaf_write_input_buffer(struct stream_out *out, const void *buffer, int bytes)
+{
+ int ret = 0;
+ ALOGVV("%s bytes = %d [%p]", __func__, bytes, out->qaf_stream_handle);
+ if (!qaf_mod->qaf_audio_stream_write)
+ return -EINVAL;
+
+ if (out->qaf_stream_handle)
+ ret = qaf_mod->qaf_audio_stream_write(out->qaf_stream_handle, buffer, bytes);
+ return ret;
+}
+
+static int qaf_out_set_volume(struct audio_stream_out *stream __unused, float left,
+ float right)
+{
+ if (qaf_mod->qaf_compr_offload_out != NULL) {
+ return qaf_mod->qaf_compr_offload_out->stream.set_volume(
+ (struct audio_stream_out *)qaf_mod->qaf_compr_offload_out, left, right);
+ }
+ return -ENOSYS;
+}
+
+static int qaf_stream_start(struct stream_out *out)
+{
+ if (!qaf_mod->qaf_audio_stream_start)
+ return -EINVAL;
+
+ return qaf_mod->qaf_audio_stream_start(out->qaf_stream_handle);
+}
+
+static int qaf_start_output_stream(struct stream_out *out)
+{
+ int ret = 0;
+ struct audio_device *adev = out->dev;
+ int snd_card_status = get_snd_card_state(adev);
+
+ if ((out->usecase < 0) || (out->usecase >= AUDIO_USECASE_MAX)) {
+ ret = -EINVAL;
+ usleep(50000);
+ return ret;
+ }
+
+ ALOGD("%s: enter: stream(%p)usecase(%d: %s) devices(%#x)",
+ __func__, &out->stream, out->usecase, use_case_table[out->usecase],
+ out->devices);
+
+ if (SND_CARD_STATE_OFFLINE == snd_card_status) {
+ ALOGE("%s: sound card is not active/SSR returning error", __func__);
+ ret = -EIO;
+ usleep(50000);
+ return ret;
+ }
+
+ return qaf_stream_start(out);
+}
+
+static ssize_t qaf_out_write(struct audio_stream_out *stream, const void *buffer,
+ size_t bytes)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ struct audio_device *adev = out->dev;
+ ssize_t ret = 0;
+
+ ALOGV("qaf_out_write bytes = %d, usecase[%d] and flags[%x] for handle[%p]",(int)bytes, out->usecase, out->flags, out);
+ lock_output_stream(out);
+
+ if (out->standby) {
+ out->standby = false;
+ pthread_mutex_lock(&adev->lock);
+ ret = qaf_start_output_stream(out);
+ pthread_mutex_unlock(&adev->lock);
+ /* ToDo: If use case is compress offload should return 0 */
+ if (ret != 0) {
+ out->standby = true;
+ goto exit;
+ }
+ }
+
+ if (adev->is_channel_status_set == false && (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)){
+ audio_utils_set_hdmi_channel_status(out, (char *)buffer, bytes);
+ adev->is_channel_status_set = true;
+ }
+
+ ret = qaf_write_input_buffer(out, buffer, bytes);
+ ALOGV("%s, ret [%d] ", __func__, (int)ret);
+ if (ret < 0) {
+ goto exit;
+ }
+ out->written += bytes / ((popcount(out->channel_mask) * sizeof(short)));
+
+exit:
+
+ pthread_mutex_unlock(&out->lock);
+
+ if (ret < 0) {
+ if (ret == -EAGAIN) {
+ ALOGV("No space available in ms12 driver, post msg to cb thread");
+ lock_output_stream(out);
+ ret = qaf_send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
+ pthread_mutex_unlock(&out->lock);
+ bytes = 0;
+ }
+ if(ret == -ENOMEM || ret == -EPERM){
+ if (out->pcm)
+ ALOGE("%s: error %d, %s", __func__, (int)ret, pcm_get_error(out->pcm));
+ qaf_out_standby(&out->stream.common);
+ usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) /
+ out->stream.common.get_sample_rate(&out->stream.common));
+ }
+ }
+ return bytes;
+}
+
+static int qaf_get_timestamp(struct stream_out *out, uint64_t *frames, struct timespec *timestamp)
+{
+ int ret = 0;
+ struct str_parms *parms;
+ int value = 0;
+ int signed_frames = 0;
+ const char* kvpairs = NULL;
+
+ ALOGV("%s out->format %d", __func__, out->format);
+ if(out->format & AUDIO_FORMAT_PCM_16_BIT) {
+ *frames = out->written;
+ signed_frames = out->written - (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL);
+ // It would be unusual for this value to be negative, but check just in case ...
+ if (signed_frames >= 0) {
+ *frames = signed_frames;
+ }
+ clock_gettime(CLOCK_MONOTONIC, timestamp);
+ } else if (qaf_mod->qaf_audio_stream_get_param) {
+ kvpairs = qaf_mod->qaf_audio_stream_get_param(out->qaf_stream_handle, "position");
+ if (kvpairs) {
+ parms = str_parms_create_str(kvpairs);
+ ret = str_parms_get_int(parms, "position", &value);
+ if (ret >= 0) {
+ *frames = value;
+ signed_frames = value - (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL);
+ // It would be unusual for this value to be negative, but check just in case ...
+ if (signed_frames >= 0) {
+ *frames = signed_frames;
+ }
+ clock_gettime(CLOCK_MONOTONIC, timestamp);
+ }
+ str_parms_destroy(parms);
+ }
+ } else {
+ ret = -EINVAL;
+ }
+ return ret;
+}
+
+static int qaf_out_get_presentation_position(const struct audio_stream_out *stream,
+ uint64_t *frames, struct timespec *timestamp)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int ret = -1;
+ lock_output_stream(out);
+ ret = qaf_get_timestamp(out, frames, timestamp);
+ pthread_mutex_unlock(&out->lock);
+
+ return ret;
+}
+
+static int qaf_stream_pause(struct stream_out *out)
+{
+ ALOGV("%s: %d start", __func__, __LINE__);
+ if (!qaf_mod->qaf_audio_stream_pause)
+ return -EINVAL;
+
+ return qaf_mod->qaf_audio_stream_pause(out->qaf_stream_handle);
+}
+
+static int qaf_out_pause(struct audio_stream_out* stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = -ENOSYS;
+ ALOGE("%s", __func__);
+ lock_output_stream(out);
+ status = qaf_stream_pause(out);
+ pthread_mutex_unlock(&out->lock);
+ return status;
+}
+
+static int qaf_out_resume(struct audio_stream_out* stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = -ENOSYS;
+ ALOGD("%s", __func__);
+ lock_output_stream(out);
+ status = qaf_stream_start(out);
+ pthread_mutex_unlock(&out->lock);
+ ALOGD("%s Exit", __func__);
+ return status;
+}
+
+static int qaf_out_drain(struct audio_stream_out* stream, audio_drain_type_t type __unused )
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = 0;
+ ALOGV("%s stream_handle = %p , format = %x", __func__, out->qaf_stream_handle, out->format);
+ lock_output_stream(out);
+ if (out->offload_callback && out->qaf_stream_handle) {
+ /* Stream stop will trigger EOS and on EOS_EVENT received
+ from callback DRAIN_READY command is sent */
+ status = audio_extn_qaf_stream_stop(out);
+ if (out->format != AUDIO_FORMAT_PCM_16_BIT)
+ qaf_mod->stream_drain_main = out;
+ }
+ pthread_mutex_unlock(&out->lock);
+ return status;
+}
+
+static int audio_extn_qaf_stream_flush(struct stream_out *out)
+{
+ ALOGV("%s: %d exit", __func__, __LINE__);
+ if (!qaf_mod->qaf_audio_stream_flush)
+ return -EINVAL;
+
+ return qaf_mod->qaf_audio_stream_flush(out->qaf_stream_handle);
+}
+
+static int qaf_out_flush(struct audio_stream_out* stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ ALOGV("%s", __func__);
+ int status = -ENOSYS;
+ lock_output_stream(out);
+ status = audio_extn_qaf_stream_flush(out);
+ pthread_mutex_unlock(&out->lock);
+ ALOGV("%s Exit", __func__);
+ return status;
+}
+
+static uint32_t qaf_out_get_latency(const struct audio_stream_out *stream __unused)
+{
+ uint32_t latency = 0;
+
+ latency = QAF_LATENCY;
+ ALOGV("%s: Latency %d", __func__, latency);
+ return latency;
+}
+
+static void notify_event_callback(audio_session_handle_t session_handle __unused, void *prv_data, void *buf, audio_event_id_t event_id, int size, int device)
+{
+
+/*
+ For SPKR:
+ 1. Open pcm device if device_id passed to it SPKR and write the data to pcm device
+
+ For HDMI
+ 1.Open compress device for HDMI(PCM or AC3) based on current_hdmi_output_format
+ 2.create offload_callback thread to receive async events
+ 3.Write the data to compress device. If not all the data is consumed by the driver,
+ add a command to offload_callback thread.
+*/
+ int ret;
+ audio_output_flags_t flags;
+ struct qaf* qaf_module = (struct qaf* ) prv_data;
+ ALOGV("%s device 0x%X, %d in event = %d", __func__, device, __LINE__, event_id);
+
+ if (event_id == AUDIO_DATA_EVENT) {
+ ALOGVV("Device id %x %s %d, bytes to written %d", device, __func__,__LINE__, size);
+
+ pthread_mutex_lock(&qaf_module->lock);
+ if ((device == (AUDIO_DEVICE_OUT_AUX_DIGITAL | AUDIO_COMPRESSED_OUT_DD)) ||
+ (device == (AUDIO_DEVICE_OUT_AUX_DIGITAL | AUDIO_COMPRESSED_OUT_DDP))) {
+
+ if (NULL == qaf_mod->qaf_compr_passthrough_out) {
+ struct audio_config config;
+ audio_devices_t devices;
+
+ if (qaf_mod->qaf_compr_offload_out) {
+ adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev,
+ (struct audio_stream_out *) (qaf_mod->qaf_compr_offload_out));
+ qaf_mod->qaf_compr_offload_out = NULL;
+ }
+
+ config.sample_rate = config.offload_info.sample_rate = QAF_OUTPUT_SAMPLING_RATE;
+ config.offload_info.version = AUDIO_INFO_INITIALIZER.version;
+ config.offload_info.size = AUDIO_INFO_INITIALIZER.size;
+
+ if (device == (AUDIO_DEVICE_OUT_AUX_DIGITAL | AUDIO_COMPRESSED_OUT_DDP))
+ config.format = config.offload_info.format = AUDIO_FORMAT_E_AC3;
+ else
+ config.format = config.offload_info.format = AUDIO_FORMAT_AC3;
+
+ config.offload_info.bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
+ flags = AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_NON_BLOCKING;
+ devices = AUDIO_DEVICE_OUT_AUX_DIGITAL;
+
+ ret = adev_open_output_stream((struct audio_hw_device *) qaf_mod->adev, QAF_DEFAULT_COMPR_PASSTHROUGH_HANDLE, devices,
+ flags, &config, (struct audio_stream_out **) &(qaf_mod->qaf_compr_passthrough_out), NULL);
+ if (ret < 0) {
+ ALOGE("%s: adev_open_output_stream failed with ret = %d!", __func__, ret);
+ pthread_mutex_unlock(&qaf_module->lock);
+ return;
+ }
+ }
+
+ if (!qaf_mod->passthrough_enabled)
+ qaf_mod->passthrough_enabled = 1;
+
+ ret = qaf_mod->qaf_compr_passthrough_out->stream.write((struct audio_stream_out *) qaf_mod->qaf_compr_passthrough_out, buf, size);
+ } else {
+ if (device == AUDIO_DEVICE_OUT_AUX_DIGITAL && !qaf_mod->multi_ch_out_enabled) {
+ if (qaf_mod->qaf_compr_offload_out) {
+ adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev,
+ (struct audio_stream_out *) (qaf_mod->qaf_compr_offload_out));
+ qaf_mod->qaf_compr_offload_out = NULL;
+ }
+ qaf_mod->multi_ch_out_enabled = 1;
+ } else if (device == AUDIO_DEVICE_OUT_SPEAKER && qaf_mod->multi_ch_out_enabled) {
+ if (qaf_mod->qaf_compr_offload_out) {
+ adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev,
+ (struct audio_stream_out *) (qaf_mod->qaf_compr_offload_out));
+ qaf_mod->qaf_compr_offload_out = NULL;
+ }
+ qaf_mod->multi_ch_out_enabled = 0;
+ }
+
+ if (NULL == qaf_mod->qaf_compr_offload_out) {
+ struct audio_config config;
+ audio_devices_t devices;
+
+ config.sample_rate = config.offload_info.sample_rate = QAF_OUTPUT_SAMPLING_RATE;
+ config.offload_info.version = AUDIO_INFO_INITIALIZER.version;
+ config.offload_info.size = AUDIO_INFO_INITIALIZER.size;
+ config.offload_info.format = AUDIO_FORMAT_PCM_16_BIT_OFFLOAD;
+ config.offload_info.bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ config.format = AUDIO_FORMAT_PCM_16_BIT_OFFLOAD;
+ devices = AUDIO_DEVICE_NONE;
+
+ if (device == AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ if (qaf_mod->hdmi_sink_channels == 8) {
+ config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_7POINT1;
+ } else if (qaf_mod->hdmi_sink_channels == 6) {
+ config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
+ } else {
+ config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ }
+ devices = AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ } else {
+ config.offload_info.channel_mask = config.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ qaf_mod->multi_ch_out_enabled = 0;
+ }
+ flags = AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING;
+
+ /* TODO:: Need to Propagate errors to framework */
+ ret = adev_open_output_stream((struct audio_hw_device *) qaf_mod->adev, QAF_DEFAULT_COMPR_AUDIO_HANDLE, devices,
+ flags, &config, (struct audio_stream_out **) &(qaf_mod->qaf_compr_offload_out), NULL);
+ if (ret < 0) {
+ ALOGE("%s: adev_open_output_stream failed with ret = %d!", __func__, ret);
+ pthread_mutex_unlock(&qaf_module->lock);
+ return;
+ }
+ }
+
+ if (qaf_mod->passthrough_enabled) {
+ qaf_mod->passthrough_enabled = 0;
+ if (qaf_mod->qaf_compr_passthrough_out) {
+ adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev,
+ (struct audio_stream_out *) (qaf_mod->qaf_compr_passthrough_out));
+ qaf_mod->qaf_compr_passthrough_out = NULL;
+ }
+ }
+
+ /*
+ * TODO:: Since this is mixed data,
+ * need to identify to which stream the error should be sent
+ */
+ ret = qaf_mod->qaf_compr_offload_out->stream.write((struct audio_stream_out *) qaf_mod->qaf_compr_offload_out, buf, size);
+ }
+
+ ALOGVV("%s:%d stream write ret = %d for out handle[%p]", __func__, __LINE__, ret, qaf_mod->qaf_compr_offload_out);
+ pthread_mutex_unlock(&qaf_module->lock);
+ } else if (event_id == AUDIO_EOS_MAIN_DD_DDP_EVENT || event_id == AUDIO_EOS_MAIN_AAC_EVENT) {
+ /* TODO:: Only MAIN Stream EOS Event is added, need to add ASSOC stream EOS Event */
+ struct stream_out *out = qaf_module->stream_drain_main;
+ if (out != NULL) {
+ lock_output_stream(out);
+ out->offload_callback(STREAM_CBK_EVENT_DRAIN_READY, NULL, out->offload_cookie);
+ pthread_mutex_unlock(&out->lock);
+ qaf_module->stream_drain_main = NULL;
+ ALOGV("%s %d sent DRAIN_READY", __func__, __LINE__);
+ }
+ }
+ ALOGV("%s %d", __func__, __LINE__);
+}
+
+static int qaf_session_close()
+{
+ ALOGV("%s %d", __func__, __LINE__);
+ if (qaf_mod != NULL) {
+ if (!qaf_mod->qaf_audio_session_close)
+ return -EINVAL;
+
+ qaf_mod->qaf_audio_session_close(qaf_mod->session_handle);
+ qaf_mod->session_handle = NULL;
+ pthread_mutex_destroy(&qaf_mod->lock);
+ }
+ return 0;
+}
+
+static int qaf_stream_close(struct stream_out *out)
+{
+ int ret = 0;
+ ALOGV( "%s %d", __func__, __LINE__);
+ if (!qaf_mod->qaf_audio_stream_close)
+ return -EINVAL;
+ if (out->qaf_stream_handle) {
+ ALOGV( "%s %d output active flag is %x and stream handle %p", __func__, __LINE__, out->flags, out->qaf_stream_handle);
+ if ((out->flags & AUDIO_OUTPUT_FLAG_ASSOCIATED) && (out->flags & AUDIO_OUTPUT_FLAG_MAIN)) { /* Close for Stream with Main and Associated Content*/
+ qaf_mod->main_output_active = false;
+ qaf_mod->assoc_output_active = false;
+ } else if (out->flags & AUDIO_OUTPUT_FLAG_MAIN) {/*Close for Main Stream*/
+ qaf_mod->main_output_active = false;
+ qaf_mod->assoc_output_active = false; /* TODO to remove resetting associated stream active flag when main stream is closed*/
+ } else if (out->flags & AUDIO_OUTPUT_FLAG_ASSOCIATED) { /*Close for Associated Stream*/
+ qaf_mod->assoc_output_active = false;
+ } else { /*Close for Local Playback*/
+ qaf_mod->main_output_active = false;
+ }
+ ret = qaf_mod->qaf_audio_stream_close(out->qaf_stream_handle);
+ out->qaf_stream_handle = NULL;
+ }
+ ALOGV( "%s %d", __func__, __LINE__);
+ return ret;
+}
+
+static int qaf_stream_open(struct stream_out *out, struct audio_config *config, audio_output_flags_t flags, audio_devices_t devices)
+{
+ int status = 0;
+ ALOGV("%s %d", __func__, __LINE__);
+
+ if (!qaf_mod->qaf_audio_stream_open)
+ return -EINVAL;
+
+ audio_stream_config_t input_config;
+ input_config.sample_rate = config->sample_rate;
+ input_config.channel_mask = config->channel_mask;
+ input_config.format = config->format;
+
+ if ((config->format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) {
+ input_config.format = AUDIO_FORMAT_AAC;
+ } else if((config->format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS) {
+ input_config.format = AUDIO_FORMAT_AAC_ADTS;
+ }
+
+ ALOGV("%s %d audio_stream_open sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x) format(%#x)\
+ ",__func__, __LINE__, input_config.sample_rate, input_config.channel_mask, devices, flags, input_config.format);
+
+ /* TODO to send appropriated flags when support for system tones is added */
+ if (input_config.format == AUDIO_FORMAT_PCM_16_BIT) {
+ status = qaf_mod->qaf_audio_stream_open(qaf_mod->session_handle, &out->qaf_stream_handle, input_config, devices, /*flags*/AUDIO_STREAM_SYSTEM_TONE);
+ } else if (input_config.format == AUDIO_FORMAT_AC3 ||
+ input_config.format == AUDIO_FORMAT_E_AC3 ||
+ input_config.format == AUDIO_FORMAT_AAC ||
+ input_config.format == AUDIO_FORMAT_AAC_ADTS) {
+ if (qaf_mod->main_output_active == false) {
+ if ((flags & AUDIO_OUTPUT_FLAG_MAIN) && (flags & AUDIO_OUTPUT_FLAG_ASSOCIATED)) {
+ 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) {
+ ALOGV("%s %d Open stream for Input with both Main and Associated stream contents with flag [%x] and stream handle [%p]", __func__, __LINE__, flags, out->qaf_stream_handle);
+ qaf_mod->main_output_active = true;
+ qaf_mod->assoc_output_active = true;
+ }
+ } else if (flags & AUDIO_OUTPUT_FLAG_MAIN) {
+ 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) {
+ ALOGV("%s %d Open stream for Input with only Main flag [%x] stream handle [%p]", __func__, __LINE__, flags, out->qaf_stream_handle);
+ qaf_mod->main_output_active = true;
+ }
+ } else if (flags & AUDIO_OUTPUT_FLAG_ASSOCIATED) {
+ ALOGE("%s %d Error main input is not active", __func__, __LINE__);
+ 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) {
+ ALOGV("%s %d Open stream for Local playback with flag [%x] stream handle [%p] ", __func__, __LINE__, flags, out->qaf_stream_handle);
+ qaf_mod->main_output_active = true;
+ }
+ }
+ } else {
+ if (flags & AUDIO_OUTPUT_FLAG_MAIN) {
+ ALOGE("%s %d Error main input is already active", __func__, __LINE__);
+ return -EINVAL;
+ } else if (flags & AUDIO_OUTPUT_FLAG_ASSOCIATED) {
+ if (qaf_mod->assoc_output_active) {
+ ALOGE("%s %d Error assoc input is already active", __func__, __LINE__);
+ return -EINVAL;
+ } else {
+ 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) {
+ ALOGV("%s %d Open stream for Input with only Associated flag [%x] stream handle [%p]", __func__, __LINE__, flags, out->qaf_stream_handle);
+ qaf_mod->assoc_output_active = true;
+ }
+ }
+ } else {
+ ALOGE("%s %d Error main input is already active", __func__, __LINE__);
+ return -EINVAL;
+ }
+ }
+ }
+
+ return status;
+}
+
+static int qaf_deinit()
+{
+ ALOGV("%s %d", __func__, __LINE__);
+ if (qaf_mod != NULL) {
+ if (qaf_mod->qaf_compr_offload_out != NULL)
+ adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev, (struct audio_stream_out *) (qaf_mod->qaf_compr_offload_out));
+ if (qaf_mod->qaf_compr_passthrough_out != NULL)
+ adev_close_output_stream((struct audio_hw_device *) qaf_mod->adev, (struct audio_stream_out *) (qaf_mod->qaf_compr_passthrough_out));
+
+ if (qaf_mod->qaf_lib != NULL) {
+ dlclose(qaf_mod->qaf_lib);
+ qaf_mod->qaf_lib = NULL;
+ }
+ free(qaf_mod);
+ qaf_mod = NULL;
+ }
+ return 0;
+}
+
+static void *qaf_offload_thread_loop(void *context)
+{
+ struct stream_out *out = (struct stream_out *) context;
+ struct listnode *item;
+ int ret = 0;
+ struct str_parms *parms = NULL;
+ int value = 0;
+ char* kvpairs = NULL;
+
+ setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
+ set_sched_policy(0, SP_FOREGROUND);
+ prctl(PR_SET_NAME, (unsigned long)"Offload Callback", 0, 0, 0);
+
+ ALOGE("%s", __func__);
+ lock_output_stream(out);
+ for (;;) {
+ struct offload_cmd *cmd = NULL;
+ stream_callback_event_t event;
+ bool send_callback = false;
+
+ ALOGV("%s qaf_offload_cmd_list %d",
+ __func__, list_empty(&out->qaf_offload_cmd_list));
+ if (list_empty(&out->qaf_offload_cmd_list)) {
+ ALOGV("%s SLEEPING", __func__);
+ pthread_cond_wait(&out->qaf_offload_cond, &out->lock);
+ ALOGV("%s RUNNING", __func__);
+ continue;
+ }
+
+ item = list_head(&out->qaf_offload_cmd_list);
+ cmd = node_to_item(item, struct offload_cmd, node);
+ list_remove(item);
+
+ if (cmd->cmd == OFFLOAD_CMD_EXIT) {
+ free(cmd);
+ break;
+ }
+
+ pthread_mutex_unlock(&out->lock);
+ send_callback = false;
+ switch(cmd->cmd) {
+ case OFFLOAD_CMD_WAIT_FOR_BUFFER:
+ ALOGV("wait for ms12 buffer availability");
+ while (1) {
+ kvpairs = qaf_mod->qaf_audio_stream_get_param(out->qaf_stream_handle, "buf_available");
+ if (kvpairs) {
+ parms = str_parms_create_str(kvpairs);
+ ret = str_parms_get_int(parms, "buf_available", &value);
+ if (ret >= 0) {
+ if (value >= (int)out->compr_config.fragment_size) {
+ ALOGV("%s buffer available", __func__);
+ str_parms_destroy(parms);
+ parms = NULL;
+ break;
+ } else {
+ ALOGV("%s sleep", __func__);
+ str_parms_destroy(parms);
+ parms = NULL;
+ usleep(10000);
+ }
+ }
+ free(kvpairs);
+ kvpairs = NULL;
+ }
+ }
+ send_callback = true;
+ event = STREAM_CBK_EVENT_WRITE_READY;
+ break;
+ default:
+ ALOGV("%s unknown command received: %d", __func__, cmd->cmd);
+ break;
+ }
+ lock_output_stream(out);
+ if (send_callback && out->offload_callback) {
+ out->offload_callback(event, NULL, out->offload_cookie);
+ }
+ free(cmd);
+ }
+
+ while (!list_empty(&out->qaf_offload_cmd_list)) {
+ item = list_head(&out->qaf_offload_cmd_list);
+ list_remove(item);
+ free(node_to_item(item, struct offload_cmd, node));
+ }
+ pthread_mutex_unlock(&out->lock);
+
+ return NULL;
+}
+
+static int qaf_create_offload_callback_thread(struct stream_out *out)
+{
+ ALOGV("%s", __func__);
+ pthread_cond_init(&out->qaf_offload_cond, (const pthread_condattr_t *) NULL);
+ list_init(&out->qaf_offload_cmd_list);
+ pthread_create(&out->qaf_offload_thread, (const pthread_attr_t *) NULL,
+ qaf_offload_thread_loop, out);
+ return 0;
+}
+
+static int qaf_destroy_offload_callback_thread(struct stream_out *out)
+{
+ ALOGV("%s", __func__);
+ lock_output_stream(out);
+ qaf_send_offload_cmd_l(out, OFFLOAD_CMD_EXIT);
+ pthread_mutex_unlock(&out->lock);
+
+ pthread_join(out->qaf_offload_thread, (void **) NULL);
+ pthread_cond_destroy(&out->qaf_offload_cond);
+
+ return 0;
+}
+
+int audio_extn_qaf_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 __unused)
+{
+ int ret = 0;
+ struct stream_out *out;
+
+ ret = adev_open_output_stream(dev, handle, devices, flags, config, stream_out, address);
+ if (*stream_out == NULL) {
+ goto error_open;
+ }
+
+ out = (struct stream_out *) *stream_out;
+
+ /* Override function pointers based on qaf definitions */
+ out->stream.set_volume = qaf_out_set_volume;
+ out->stream.pause = qaf_out_pause;
+ out->stream.resume = qaf_out_resume;
+ out->stream.drain = qaf_out_drain;
+ out->stream.flush = qaf_out_flush;
+
+ out->stream.common.standby = qaf_out_standby;
+ out->stream.common.set_parameters = qaf_out_set_parameters;
+ out->stream.get_latency = qaf_out_get_latency;
+ out->stream.write = qaf_out_write;
+ out->stream.get_presentation_position = qaf_out_get_presentation_position;
+
+ ret = qaf_stream_open(out, config, flags, devices);
+ if (ret < 0) {
+ ALOGE("%s, Error opening QAF stream err[%d]!", __func__, ret);
+ adev_close_output_stream(dev, *stream_out);
+ goto error_open;
+ }
+
+ if (out->usecase == USECASE_AUDIO_PLAYBACK_LOW_LATENCY) {
+ out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
+ out->config.period_size = DEEP_BUFFER_OUTPUT_PERIOD_SIZE;
+ out->config.period_count = DEEP_BUFFER_OUTPUT_PERIOD_COUNT;
+ out->config.start_threshold = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4;
+ out->config.avail_min = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4;
+ }
+
+ *stream_out = &out->stream;
+ if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ qaf_create_offload_callback_thread(out);
+ }
+ ALOGV("%s: exit", __func__);
+ return 0;
+error_open:
+ *stream_out = NULL;
+ ALOGD("%s: exit: ret %d", __func__, ret);
+ return ret;
+}
+
+void audio_extn_qaf_close_output_stream(struct audio_hw_device *dev,
+ struct audio_stream_out *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+
+ ALOGV("%s: enter:stream_handle(%p) format = %x", __func__, out, out->format);
+ if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ qaf_destroy_offload_callback_thread(out);
+ }
+ qaf_mod->stream_drain_main = NULL;
+ lock_output_stream(out);
+ qaf_stream_close(out);
+ pthread_mutex_unlock(&out->lock);
+
+ adev_close_output_stream(dev, stream);
+ ALOGV("%s: exit", __func__);
+}
+
+bool audio_extn_qaf_is_enabled()
+{
+ bool prop_enabled = false;
+ char value[PROPERTY_VALUE_MAX] = {0};
+ property_get("audio.qaf.enabled", value, NULL);
+ prop_enabled = atoi(value) || !strncmp("true", value, 4);
+ return (prop_enabled);
+}
+
+int audio_extn_qaf_session_open(struct qaf *qaf_mod,
+ device_license_config_t* lic_config)
+{
+ ALOGV("%s %d", __func__, __LINE__);
+ int status = -ENOSYS;
+
+ pthread_mutex_init(&qaf_mod->lock, (const pthread_mutexattr_t *) NULL);
+
+ if (!qaf_mod->qaf_audio_session_open)
+ return -EINVAL;
+
+ status = qaf_mod->qaf_audio_session_open(&qaf_mod->session_handle,
+ (void *)(qaf_mod), (void *)lic_config);
+ if(status < 0)
+ return status;
+
+ if (qaf_mod->session_handle == NULL) {
+ ALOGE("%s %d QAF wrapper session handle is NULL", __func__, __LINE__);
+ return -ENOMEM;
+ }
+ if (qaf_mod->qaf_register_event_callback)
+ qaf_mod->qaf_register_event_callback(qaf_mod->session_handle,
+ qaf_mod, ¬ify_event_callback,
+ AUDIO_DATA_EVENT);
+ return status;
+}
+
+char* audio_extn_qaf_stream_get_param(struct stream_out *out __unused, const char *kv_pair __unused)
+{
+ return NULL;
+}
+
+int audio_extn_qaf_set_parameters(struct audio_device *adev, struct str_parms *parms)
+{
+ int status = 0, val = 0, channels = 0;
+ char *format_params, *kv_parirs;
+ struct str_parms *qaf_params;
+ char value[32];
+ bool passth_support = false;
+
+ ALOGV("%s %d ", __func__, __LINE__);
+ if (!qaf_mod || !qaf_mod->qaf_audio_session_set_param) {
+ return -EINVAL;
+ }
+
+ status = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value, sizeof(value));
+ if (status >= 0) {
+ val = atoi(value);
+ if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ if (property_get_bool("audio.offload.passthrough", false) &&
+ property_get_bool("audio.qaf.reencode", false)) {
+
+ qaf_params = str_parms_create();
+ if (platform_is_edid_supported_format(adev->platform, AUDIO_FORMAT_E_AC3)) {
+ passth_support = true;
+ if (qaf_params) {
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_RENDER_FORMAT,
+ AUDIO_QAF_PARAMETER_VALUE_REENCODE_EAC3);
+ }
+ } else if (platform_is_edid_supported_format(adev->platform, AUDIO_FORMAT_AC3)) {
+ passth_support = true;
+ if (qaf_params) {
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_RENDER_FORMAT,
+ AUDIO_QAF_PARAMETER_VALUE_REENCODE_AC3);
+ }
+ }
+
+ if (passth_support) {
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_DEVICE,
+ AUDIO_QAF_PARAMETER_VALUE_DEVICE_HDMI);
+ format_params = str_parms_to_str(qaf_params);
+
+ qaf_mod->qaf_audio_session_set_param(qaf_mod->session_handle, format_params);
+ }
+ str_parms_destroy(qaf_params);
+ }
+
+ if (!passth_support) {
+ channels = platform_edid_get_max_channels(adev->platform);
+
+ qaf_params = str_parms_create();
+ switch (channels) {
+ case 8:
+ ALOGV("%s: Switching Qaf output to 7.1 channels", __func__);
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_CHANNELS,
+ AUDIO_QAF_PARAMETER_VALUE_8_CHANNELS);
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_DEVICE,
+ AUDIO_QAF_PARAMETER_VALUE_DEVICE_HDMI);
+ qaf_mod->hdmi_sink_channels = channels;
+ break;
+ case 6:
+ ALOGV("%s: Switching Qaf output to 5.1 channels", __func__);
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_CHANNELS,
+ AUDIO_QAF_PARAMETER_VALUE_6_CHANNELS);
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_DEVICE,
+ AUDIO_QAF_PARAMETER_VALUE_DEVICE_HDMI);
+ qaf_mod->hdmi_sink_channels = channels;
+ break;
+ default:
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_DEVICE,
+ AUDIO_QAF_PARAMETER_VALUE_DEVICE_SPEAKER);
+ qaf_mod->hdmi_sink_channels = 2;
+ break;
+ }
+
+ format_params = str_parms_to_str(qaf_params);
+ qaf_mod->qaf_audio_session_set_param(qaf_mod->session_handle, format_params);
+ str_parms_destroy(qaf_params);
+ }
+ }
+ }
+
+ status = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, value, sizeof(value));
+ if (status >= 0) {
+ val = atoi(value);
+ if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ qaf_params = str_parms_create();
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_DEVICE,
+ AUDIO_QAF_PARAMETER_VALUE_DEVICE_SPEAKER);
+ str_parms_add_str(qaf_params, AUDIO_QAF_PARAMETER_KEY_RENDER_FORMAT,
+ AUDIO_QAF_PARAMETER_VALUE_PCM);
+ qaf_mod->hdmi_sink_channels = 0;
+
+ format_params = str_parms_to_str(qaf_params);
+ qaf_mod->qaf_audio_session_set_param(qaf_mod->session_handle, format_params);
+ str_parms_destroy(qaf_params);
+ }
+ }
+
+ kv_parirs = str_parms_to_str(parms);
+ qaf_mod->qaf_audio_session_set_param(qaf_mod->session_handle, kv_parirs);
+
+ return status;
+}
+
+char* audio_extn_qaf_get_param(struct audio_device *adev __unused, const char *kv_pair __unused)
+{
+ return 0;
+}
+
+int audio_extn_qaf_init(struct audio_device *adev)
+{
+ char value[PROPERTY_VALUE_MAX] = {0};
+ char lib_name[PROPERTY_VALUE_MAX] = {0};
+ unsigned char* license_data = NULL;
+ device_license_config_t* lic_config = NULL;
+ ALOGV("%s %d", __func__, __LINE__);
+ int ret = 0, size = 0;
+
+ qaf_mod = malloc(sizeof(struct qaf));
+ if(qaf_mod == NULL) {
+ ALOGE("%s, out of memory", __func__);
+ ret = -ENOMEM;
+ goto done;
+ }
+ memset(qaf_mod, 0, sizeof(struct qaf));
+ lic_config = (device_license_config_t*) calloc(1, sizeof(device_license_config_t));
+ if(lic_config == NULL) {
+ ALOGE("%s, out of memory", __func__);
+ ret = -ENOMEM;
+ goto done;
+ }
+ qaf_mod->adev = adev;
+ property_get("audio.qaf.library", value, NULL);
+ snprintf(lib_name, PROPERTY_VALUE_MAX, "%s", value);
+
+ license_data = platform_get_license((struct audio_hw_device *)(qaf_mod->adev->platform), &size);
+ if (!license_data) {
+ ALOGE("License is not present");
+ ret = -EINVAL;
+ goto done;
+ }
+ lic_config->p_license = (unsigned char* ) calloc(1, size);
+ if(lic_config->p_license == NULL) {
+ ALOGE("%s, out of memory", __func__);
+ ret = -ENOMEM;
+ goto done;
+ }
+ lic_config->l_size = size;
+ memcpy(lic_config->p_license, license_data, size);
+
+ if (property_get("audio.qaf.manufacturer", value, "") && atoi(value)) {
+ lic_config->manufacturer_id = (unsigned long) atoi (value);
+ } else {
+ ALOGE("audio.qaf.manufacturer id is not set");
+ ret = -EINVAL;
+ goto done;
+ }
+
+ ret = audio_extn_qaf_session_open(qaf_mod, lic_config);
+done:
+ if (license_data != NULL) {
+ free(license_data);
+ license_data = NULL;
+ }
+ if (lic_config->p_license != NULL) {
+ free(lic_config->p_license);
+ lic_config->p_license = NULL;
+ }
+ if (lic_config != NULL) {
+ free(lic_config);
+ lic_config = NULL;
+ }
+ if (ret != 0) {
+ if (qaf_mod != NULL) {
+ free(qaf_mod);
+ qaf_mod = NULL;
+ }
+ }
+ return ret;
+}
+
+void audio_extn_qaf_deinit()
+{
+ qaf_session_close();
+ qaf_deinit();
+}
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index 1f88c71..008130f 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -117,6 +117,11 @@
SPKR_PROTECTION_MODE_CALIBRATE = 1,
};
+struct spkr_prot_r0t0 {
+ int r0[SP_V2_NUM_MAX_SPKRS];
+ int t0[SP_V2_NUM_MAX_SPKRS];
+};
+
struct speaker_prot_session {
int spkr_prot_mode;
int spkr_processing_state;
@@ -142,6 +147,7 @@
bool spkr_prot_enable;
bool spkr_in_use;
struct timespec spkr_last_time_used;
+ struct spkr_prot_r0t0 sp_r0t0_cal;
bool wsa_found;
int spkr_1_tzn;
int spkr_2_tzn;
@@ -340,6 +346,7 @@
int ret = 0;
struct audio_cal_fb_spk_prot_cfg cal_data;
char value[PROPERTY_VALUE_MAX];
+ static int cal_done = 0;
if (cal_fd < 0) {
ALOGE("%s: Error: cal_fd = %d", __func__, cal_fd);
@@ -382,6 +389,13 @@
ret = -ENODEV;
goto done;
}
+ if (protCfg->mode == MSM_SPKR_PROT_CALIBRATED && !cal_done) {
+ handle.sp_r0t0_cal.r0[SP_V2_SPKR_1] = protCfg->r0[SP_V2_SPKR_1];
+ handle.sp_r0t0_cal.r0[SP_V2_SPKR_2] = protCfg->r0[SP_V2_SPKR_2];
+ handle.sp_r0t0_cal.t0[SP_V2_SPKR_1] = protCfg->t0[SP_V2_SPKR_1];
+ handle.sp_r0t0_cal.t0[SP_V2_SPKR_2] = protCfg->t0[SP_V2_SPKR_2];
+ cal_done = 1;
+ }
done:
return ret;
}
@@ -1347,12 +1361,48 @@
}
}
+int audio_extn_select_spkr_prot_cal_data(snd_device_t snd_device)
+{
+ struct audio_cal_info_spk_prot_cfg protCfg;
+ int acdb_fd = -1;
+ int ret = 0;
+
+ acdb_fd = open("/dev/msm_audio_cal", O_RDWR | O_NONBLOCK);
+ if (acdb_fd < 0) {
+ ALOGE("%s: open msm_acdb failed", __func__);
+ return -ENODEV;
+ }
+ switch(snd_device) {
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT:
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED:
+ protCfg.r0[SP_V2_SPKR_1] = handle.sp_r0t0_cal.r0[SP_V2_SPKR_2];
+ protCfg.r0[SP_V2_SPKR_2] = handle.sp_r0t0_cal.r0[SP_V2_SPKR_1];
+ protCfg.t0[SP_V2_SPKR_1] = handle.sp_r0t0_cal.t0[SP_V2_SPKR_2];
+ protCfg.t0[SP_V2_SPKR_2] = handle.sp_r0t0_cal.t0[SP_V2_SPKR_1];
+ break;
+ default:
+ protCfg.r0[SP_V2_SPKR_1] = handle.sp_r0t0_cal.r0[SP_V2_SPKR_1];
+ protCfg.r0[SP_V2_SPKR_2] = handle.sp_r0t0_cal.r0[SP_V2_SPKR_2];
+ protCfg.t0[SP_V2_SPKR_1] = handle.sp_r0t0_cal.t0[SP_V2_SPKR_1];
+ protCfg.t0[SP_V2_SPKR_2] = handle.sp_r0t0_cal.t0[SP_V2_SPKR_2];
+ break;
+ }
+ protCfg.mode = MSM_SPKR_PROT_CALIBRATED;
+ ret = set_spkr_prot_cal(acdb_fd, &protCfg);
+ if (ret)
+ ALOGE("%s: speaker protection cal data swap failed", __func__);
+
+ close(acdb_fd);
+ return ret;
+}
+
int audio_extn_spkr_prot_start_processing(snd_device_t snd_device)
{
struct audio_usecase *uc_info_tx;
struct audio_device *adev = handle.adev_handle;
int32_t pcm_dev_tx_id = -1, ret = 0;
bool disable_tx = false;
+ snd_device_t in_snd_device;
ALOGV("%s: Entry", __func__);
/* cancel speaker calibration */
@@ -1361,6 +1411,15 @@
return -EINVAL;
}
snd_device = platform_get_spkr_prot_snd_device(snd_device);
+ if (handle.spkr_prot_mode == MSM_SPKR_PROT_CALIBRATED) {
+ ret = audio_extn_select_spkr_prot_cal_data(snd_device);
+ if (ret) {
+ ALOGE("%s: Setting speaker protection cal data failed", __func__);
+ return ret;
+ }
+ }
+
+ in_snd_device = platform_get_vi_feedback_snd_device(snd_device);
spkr_prot_set_spkrstatus(true);
uc_info_tx = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
if (!uc_info_tx) {
@@ -1375,12 +1434,12 @@
if (handle.spkr_processing_state == SPKR_PROCESSING_IN_IDLE) {
uc_info_tx->id = USECASE_AUDIO_SPKR_CALIB_TX;
uc_info_tx->type = PCM_CAPTURE;
- uc_info_tx->in_snd_device = SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+ uc_info_tx->in_snd_device = in_snd_device;
uc_info_tx->out_snd_device = SND_DEVICE_NONE;
handle.pcm_tx = NULL;
list_add_tail(&adev->usecase_list, &uc_info_tx->list);
disable_tx = true;
- enable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+ enable_snd_device(adev, in_snd_device);
enable_audio_route(adev, uc_info_tx);
pcm_dev_tx_id = platform_get_pcm_device_id(uc_info_tx->id, PCM_CAPTURE);
@@ -1420,9 +1479,9 @@
list_remove(&uc_info_tx->list);
uc_info_tx->id = USECASE_AUDIO_SPKR_CALIB_TX;
uc_info_tx->type = PCM_CAPTURE;
- uc_info_tx->in_snd_device = SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+ uc_info_tx->in_snd_device = in_snd_device;
uc_info_tx->out_snd_device = SND_DEVICE_NONE;
- disable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+ disable_snd_device(adev, in_snd_device);
disable_audio_route(adev, uc_info_tx);
free(uc_info_tx);
} else
@@ -1436,17 +1495,20 @@
{
struct audio_usecase *uc_info_tx;
struct audio_device *adev = handle.adev_handle;
+ snd_device_t in_snd_device;
ALOGV("%s: Entry", __func__);
snd_device = platform_get_spkr_prot_snd_device(snd_device);
spkr_prot_set_spkrstatus(false);
+ in_snd_device = platform_get_vi_feedback_snd_device(snd_device);
+
pthread_mutex_lock(&handle.mutex_spkr_prot);
if (adev && handle.spkr_processing_state == SPKR_PROCESSING_IN_PROGRESS) {
uc_info_tx = get_usecase_from_list(adev, USECASE_AUDIO_SPKR_CALIB_TX);
if (handle.pcm_tx)
pcm_close(handle.pcm_tx);
handle.pcm_tx = NULL;
- disable_snd_device(adev, SND_DEVICE_IN_CAPTURE_VI_FEEDBACK);
+ disable_snd_device(adev, in_snd_device);
if (uc_info_tx) {
list_remove(&uc_info_tx->list);
disable_audio_route(adev, uc_info_tx);
diff --git a/hal/audio_extn/usb.c b/hal/audio_extn/usb.c
index a7b10d9..b3bd58f 100644
--- a/hal/audio_extn/usb.c
+++ b/hal/audio_extn/usb.c
@@ -190,33 +190,6 @@
}
}
-static int usb_set_channel_mixer_ctl(int channel,
- char *ch_mixer_ctl_name)
-{
- struct mixer_ctl *ctl;
-
- ctl = mixer_get_ctl_by_name(usbmod->adev->mixer, ch_mixer_ctl_name);
- if (!ctl) {
- ALOGE("%s: Could not get ctl for mixer cmd - %s",
- __func__, ch_mixer_ctl_name);
- return -EINVAL;
- }
- switch (channel) {
- case 1:
- mixer_ctl_set_enum_by_string(ctl, "One");
- break;
- case 2:
- mixer_ctl_set_enum_by_string(ctl, "Two");
- break;
- default:
- ALOGV("%s: channel(%d) not supported, set as default 2 channels",
- __func__, channel);
- mixer_ctl_set_enum_by_string(ctl, "Two");
- break;
- }
- return 0;
-}
-
static int usb_set_dev_id_mixer_ctl(unsigned int usb_usecase_type, int card,
char *dev_mixer_ctl_name)
{
@@ -472,8 +445,6 @@
int card)
{
int ret;
- struct listnode *node_d;
- struct usb_device_config *dev_info;
/* get capabilities */
if ((ret = usb_get_capability(USB_PLAYBACK, usb_card_info, card))) {
@@ -481,14 +452,6 @@
__func__);
goto exit;
}
- /* Currently only use the first profile using to configure channel for simplification */
- list_for_each(node_d, &usb_card_info->usb_device_conf_list) {
- dev_info = node_to_item(node_d, struct usb_device_config, list);
- if (dev_info != NULL) {
- usb_set_channel_mixer_ctl(dev_info->channels, "USB_AUDIO_RX Channels");
- break;
- }
- }
usb_set_dev_id_mixer_ctl(USB_PLAYBACK, card, "USB_AUDIO_RX dev_token");
exit:
@@ -500,8 +463,6 @@
int card)
{
int ret;
- struct listnode *node_d;
- struct usb_device_config *dev_info;
/* get capabilities */
if ((ret = usb_get_capability(USB_CAPTURE, usb_card_info, card))) {
@@ -509,14 +470,6 @@
__func__);
goto exit;
}
- /* Currently only use the first profile using to configure channel for simplification */
- list_for_each(node_d, &usb_card_info->usb_device_conf_list) {
- dev_info = node_to_item(node_d, struct usb_device_config, list);
- if (dev_info != NULL) {
- usb_set_channel_mixer_ctl(dev_info->channels, "USB_AUDIO_TX Channels");
- break;
- }
- }
usb_set_dev_id_mixer_ctl(USB_CAPTURE, card, "USB_AUDIO_TX dev_token");
exit:
@@ -909,14 +862,8 @@
"%s: card_dev_type (0x%x), card_no(%d)",
__func__, card_info->usb_device_type, card_info->usb_card);
/* Currently only apply the first playback sound card configuration */
- if (is_playback && card_info->usb_device_type == AUDIO_DEVICE_OUT_USB_DEVICE) {
- is_usb_supported = usb_audio_backend_apply_policy(
- &card_info->usb_device_conf_list,
- bit_width,
- sample_rate,
- ch);
- break;
- } else if (card_info->usb_device_type == AUDIO_DEVICE_IN_USB_DEVICE ) {
+ if ((is_playback && card_info->usb_device_type == AUDIO_DEVICE_OUT_USB_DEVICE) ||
+ ((!is_playback) && card_info->usb_device_type == AUDIO_DEVICE_IN_USB_DEVICE)){
is_usb_supported = usb_audio_backend_apply_policy(
&card_info->usb_device_conf_list,
bit_width,
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index df78f83..673c17e 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -1234,6 +1234,7 @@
(DSD_NATIVE_BACKEND == platform_get_backend_index(uc->out_snd_device))) {
active = true;
ALOGV("%s:DSD playback is active", __func__);
+ break;
}
}
return active;
@@ -2617,10 +2618,13 @@
static float AmpToDb(float amplification)
{
- if (amplification == 0) {
- return DSD_VOLUME_MIN_DB;
+ float db = DSD_VOLUME_MIN_DB;
+ if (amplification > 0) {
+ db = 20 * log10(amplification);
+ if(db < DSD_VOLUME_MIN_DB)
+ return DSD_VOLUME_MIN_DB;
}
- return 20 * log10(amplification);
+ return db;
}
static int out_set_volume(struct audio_stream_out *stream, float left,
@@ -2807,8 +2811,14 @@
if ( ret == (ssize_t)bytes && !out->non_blocking)
out->written += bytes;
- if (!out->playback_started && ret >= 0) {
- compress_start(out->compr);
+ /* Call compr start only when non-zero bytes of data is there to be rendered */
+ if (!out->playback_started && ret > 0) {
+ int status = compress_start(out->compr);
+ if (status < 0) {
+ ret = status;
+ ALOGE("%s: compr start failed with err %d", __func__, errno);
+ goto exit;
+ }
audio_extn_dts_eagle_fade(adev, true, out);
out->playback_started = 1;
out->offload_state = OFFLOAD_STATE_PLAYING;
@@ -2885,8 +2895,9 @@
out->standby = true;
}
out_standby(&out->stream.common);
- usleep((uint64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
- out_get_sample_rate(&out->stream.common));
+ if (!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))
+ usleep((uint64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
+ out_get_sample_rate(&out->stream.common));
}
return bytes;
}
@@ -3131,8 +3142,12 @@
if (is_offload_usecase(out->usecase)) {
ALOGD("copl(%p):calling compress flush", out);
lock_output_stream(out);
- stop_compressed_output_l(out);
- out->written = 0;
+ if (out->offload_state == OFFLOAD_STATE_PAUSED) {
+ stop_compressed_output_l(out);
+ out->written = 0;
+ } else {
+ ALOGW("%s called in invalid state %d", __func__, out->offload_state);
+ }
pthread_mutex_unlock(&out->lock);
ALOGD("copl(%p):out of compress flush", out);
return 0;
@@ -3489,7 +3504,7 @@
return add_remove_audio_effect(stream, effect, false);
}
-static int adev_open_output_stream(struct audio_hw_device *dev,
+int adev_open_output_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
audio_output_flags_t flags,
@@ -3774,13 +3789,13 @@
__func__, config->offload_info.version,
config->offload_info.bit_rate);
- /*Check if DSD audio format is supported in codec
- *and there is no active native DSD use case
+ /* Check if DSD audio format is supported in codec
+ * and there is no active native DSD use case
*/
if ((config->format == AUDIO_FORMAT_DSD) &&
- (!platform_check_codec_dsd_support(adev->platform) ||
- audio_is_dsd_native_stream_active(adev))) {
+ (!platform_check_codec_dsd_support(adev->platform) ||
+ audio_is_dsd_native_stream_active(adev))) {
ret = -EINVAL;
goto error_open;
}
@@ -3791,9 +3806,9 @@
* Direct PCM playback
*/
if (audio_extn_passthru_is_passthrough_stream(out) ||
- (config->format == AUDIO_FORMAT_DSD) ||
- config->offload_info.has_video ||
- out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
+ (config->format == AUDIO_FORMAT_DSD) ||
+ config->offload_info.has_video ||
+ out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) {
check_and_set_gapless_mode(adev, false);
} else
check_and_set_gapless_mode(adev, true);
@@ -3958,7 +3973,7 @@
return ret;
}
-static void adev_close_output_stream(struct audio_hw_device *dev __unused,
+void adev_close_output_stream(struct audio_hw_device *dev __unused,
struct audio_stream_out *stream)
{
struct stream_out *out = (struct stream_out *)stream;
@@ -4574,6 +4589,8 @@
if ((--audio_device_ref_count) == 0) {
audio_extn_sound_trigger_deinit(adev);
audio_extn_listen_deinit(adev);
+ if (audio_extn_qaf_is_enabled())
+ audio_extn_qaf_deinit();
audio_extn_utils_release_streams_output_cfg_list(&adev->streams_output_cfg_list);
audio_route_free(adev->audio_route);
audio_extn_gef_deinit();
@@ -4610,6 +4627,8 @@
static int adev_open(const hw_module_t *module, const char *name,
hw_device_t **device)
{
+ int ret;
+
ALOGD("%s: enter", __func__);
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL;
@@ -4684,9 +4703,26 @@
ALOGE("%s: Failed to init platform data, aborting.", __func__);
*device = NULL;
pthread_mutex_unlock(&adev_init_lock);
+ pthread_mutex_destroy(&adev->lock);
+ pthread_mutex_destroy(&adev->snd_card_status.lock);
return -EINVAL;
}
+ if (audio_extn_qaf_is_enabled()) {
+ ret = audio_extn_qaf_init(adev);
+ if (ret < 0) {
+ free(adev);
+ ALOGE("%s: Failed to init platform data, aborting.", __func__);
+ *device = NULL;
+ pthread_mutex_unlock(&adev_init_lock);
+ pthread_mutex_destroy(&adev->lock);
+ return ret;
+ }
+
+ adev->device.open_output_stream = audio_extn_qaf_open_output_stream;
+ adev->device.close_output_stream = audio_extn_qaf_close_output_stream;
+ }
+
adev->snd_card_status.state = SND_CARD_STATE_ONLINE;
if (access(VISUALIZER_LIBRARY_PATH, R_OK) == 0) {
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index c1797bd..0633eb0 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -241,6 +241,10 @@
bool routing_change;
struct audio_device *dev;
+ void* qaf_stream_handle;
+ pthread_cond_t qaf_offload_cond;
+ pthread_t qaf_offload_thread;
+ struct listnode qaf_offload_cmd_list;
};
struct stream_in {
@@ -427,6 +431,16 @@
audio_usecase_t get_usecase_id_from_usecase_type(const struct audio_device *adev,
usecase_type_t type);
+int adev_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 __unused);
+void adev_close_output_stream(struct audio_hw_device *dev __unused,
+ struct audio_stream_out *stream);
+
#define LITERAL_TO_STRING(x) #x
#define CHECK(condition) LOG_ALWAYS_FATAL_IF(!(condition), "%s",\
__FILE__ ":" LITERAL_TO_STRING(__LINE__)\
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 25f862e..47943da 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -118,6 +118,8 @@
#define AUDIO_PARAMETER_KEY_AUD_CALDATA "cal_data"
#define AUDIO_PARAMETER_KEY_AUD_CALRESULT "cal_result"
+#define AUDIO_PARAMETER_KEY_MONO_SPEAKER "mono_speaker"
+
/* Reload ACDB files from specified path */
#define AUDIO_PARAMETER_KEY_RELOAD_ACDB "reload_acdb"
@@ -221,6 +223,7 @@
/* Vbat monitor related flags */
bool is_vbat_speaker;
bool gsm_mode_enabled;
+ int mono_speaker;
/* Audio calibration related functions */
void *acdb_handle;
int voice_feature_set;
@@ -244,7 +247,6 @@
bool edid_valid;
int ext_disp_type;
codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
- codec_backend_cfg_t current_tx_backend_cfg[MAX_CODEC_TX_BACKENDS];
char ec_ref_mixer_path[64];
char codec_version[CODEC_VERSION_MAX_LENGTH];
int hw_dep_fd;
@@ -343,6 +345,9 @@
[SND_DEVICE_OUT_VOICE_SPEAKER] = "voice-speaker",
[SND_DEVICE_OUT_VOICE_SPEAKER_WSA] = "wsa-voice-speaker",
[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = "vbat-voice-speaker",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2] = "voice-speaker-2",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA] = "wsa-voice-speaker-2",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = "vbat-voice-speaker-2",
[SND_DEVICE_OUT_VOICE_HEADPHONES] = "voice-headphones",
[SND_DEVICE_OUT_VOICE_LINE] = "voice-line",
[SND_DEVICE_OUT_HDMI] = "hdmi",
@@ -369,8 +374,10 @@
[SND_DEVICE_OUT_ANC_HANDSET] = "anc-handset",
[SND_DEVICE_OUT_SPEAKER_PROTECTED] = "speaker-protected",
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = "voice-speaker-protected",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED] = "voice-speaker-2-protected",
[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = "speaker-protected-vbat",
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = "voice-speaker-protected-vbat",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT] = "voice-speaker-2-protected-vbat",
#ifdef RECORD_PLAY_CONCURRENCY
[SND_DEVICE_OUT_VOIP_HANDSET] = "voip-handset",
[SND_DEVICE_OUT_VOIP_SPEAKER] = "voip-speaker",
@@ -423,6 +430,8 @@
[SND_DEVICE_IN_HANDSET_STEREO_DMIC] = "handset-stereo-dmic-ef",
[SND_DEVICE_IN_SPEAKER_STEREO_DMIC] = "speaker-stereo-dmic-ef",
[SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = "vi-feedback",
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1] = "vi-feedback-mono-1",
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2] = "vi-feedback-mono-2",
[SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE] = "voice-speaker-dmic-broadside",
[SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE] = "speaker-dmic-broadside",
[SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE] = "speaker-dmic-broadside",
@@ -465,8 +474,11 @@
[SND_DEVICE_OUT_VOICE_HANDSET] = 7,
[SND_DEVICE_OUT_VOICE_LINE] = 10,
[SND_DEVICE_OUT_VOICE_SPEAKER] = 14,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2] = 14,
[SND_DEVICE_OUT_VOICE_SPEAKER_WSA] = 135,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA] = 135,
[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = 135,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = 135,
[SND_DEVICE_OUT_VOICE_HEADPHONES] = 10,
[SND_DEVICE_OUT_HDMI] = 18,
[SND_DEVICE_OUT_SPEAKER_AND_HDMI] = 14,
@@ -492,8 +504,10 @@
[SND_DEVICE_OUT_ANC_HANDSET] = 103,
[SND_DEVICE_OUT_SPEAKER_PROTECTED] = 124,
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = 101,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED] = 101,
[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = 124,
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = 101,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT] = 101,
#ifdef RECORD_PLAY_CONCURRENCY
[SND_DEVICE_OUT_VOIP_HANDSET] = 133,
[SND_DEVICE_OUT_VOIP_SPEAKER] = 132,
@@ -545,6 +559,8 @@
[SND_DEVICE_IN_HANDSET_STEREO_DMIC] = 34,
[SND_DEVICE_IN_SPEAKER_STEREO_DMIC] = 35,
[SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = 102,
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1] = 102,
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2] = 102,
[SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE] = 12,
[SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE] = 12,
[SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE] = 119,
@@ -591,6 +607,9 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_WSA)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HEADPHONES)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_LINE)},
{TO_NAME_INDEX(SND_DEVICE_OUT_HDMI)},
@@ -617,8 +636,10 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_ANC_HANDSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT)},
#ifdef RECORD_PLAY_CONCURRENCY
{TO_NAME_INDEX(SND_DEVICE_OUT_VOIP_HANDSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOIP_SPEAKER)},
@@ -669,6 +690,8 @@
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_STEREO_DMIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_STEREO_DMIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2)},
{TO_NAME_INDEX(SND_DEVICE_IN_VOICE_FLUENCE_DMIC_AANC)},
{TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE)},
{TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE)},
@@ -1224,6 +1247,7 @@
backend_tag_table[SND_DEVICE_OUT_TRANSMISSION_FM] = strdup("transmission-fm");
backend_tag_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("vbat-voice-speaker");
+ backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = strdup("vbat-voice-speaker-2");
backend_tag_table[SND_DEVICE_OUT_BT_A2DP] = strdup("bt-a2dp");
backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = strdup("speaker-and-bt-a2dp");
@@ -1689,6 +1713,7 @@
my_data->ext_disp_type = EXT_DISPLAY_TYPE_NONE;
my_data->is_wsa_speaker = false;
my_data->hw_dep_fd = -1;
+ my_data->mono_speaker = SPKR_1;
property_get("ro.qc.sdk.audio.fluencetype", my_data->fluence_cap, "");
if (!strncmp("fluencepro", my_data->fluence_cap, sizeof("fluencepro"))) {
@@ -1897,16 +1922,13 @@
my_data->current_backend_cfg[idx].sample_rate = OUTPUT_SAMPLING_RATE_44100;
my_data->current_backend_cfg[idx].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
my_data->current_backend_cfg[idx].channels = CODEC_BACKEND_DEFAULT_CHANNELS;
+ if (idx > MAX_RX_CODEC_BACKENDS)
+ my_data->current_backend_cfg[idx].channels = CODEC_BACKEND_DEFAULT_TX_CHANNELS;
my_data->current_backend_cfg[idx].bitwidth_mixer_ctl = NULL;
my_data->current_backend_cfg[idx].samplerate_mixer_ctl = NULL;
my_data->current_backend_cfg[idx].channels_mixer_ctl = NULL;
}
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].sample_rate =
- CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].bit_width =
- CODEC_BACKEND_DEFAULT_BIT_WIDTH;
-
if (is_external_codec) {
my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
strdup("SLIM_0_RX Format");
@@ -1923,9 +1945,9 @@
my_data->current_backend_cfg[HEADPHONE_BACKEND].samplerate_mixer_ctl =
strdup("SLIM_6_RX SampleRate");
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
+ my_data->current_backend_cfg[SLIMBUS_0_TX].bitwidth_mixer_ctl =
strdup("SLIM_0_TX Format");
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
+ my_data->current_backend_cfg[SLIMBUS_0_TX].samplerate_mixer_ctl =
strdup("SLIM_0_TX SampleRate");
} else {
my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].bitwidth_mixer_ctl =
@@ -1933,16 +1955,17 @@
my_data->current_backend_cfg[DEFAULT_CODEC_BACKEND].samplerate_mixer_ctl =
strdup("MI2S_RX SampleRate");
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_TX_BACKEND].bitwidth_mixer_ctl =
+ my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].bitwidth_mixer_ctl =
strdup("MI2S_TX Format");
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_TX_BACKEND].samplerate_mixer_ctl =
+ my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].samplerate_mixer_ctl =
strdup("MI2S_TX SampleRate");
-
- my_data->current_tx_backend_cfg[USB_AUDIO_TX_BACKEND].bitwidth_mixer_ctl =
- strdup("USB_AUDIO_TX Format");
- my_data->current_tx_backend_cfg[USB_AUDIO_TX_BACKEND].samplerate_mixer_ctl =
- strdup("USB_AUDIO_TX SampleRate");
}
+ my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].bitwidth_mixer_ctl =
+ strdup("USB_AUDIO_TX Format");
+ my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].samplerate_mixer_ctl =
+ strdup("USB_AUDIO_TX SampleRate");
+ my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].channels_mixer_ctl =
+ strdup("USB_AUDIO_TX Channels");
my_data->current_backend_cfg[USB_AUDIO_RX_BACKEND].bitwidth_mixer_ctl =
strdup("USB_AUDIO_RX Format");
@@ -2081,7 +2104,8 @@
return;
}
- if((snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+ if ((snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
!(usecase->type == VOICE_CALL || usecase->type == VOIP_CALL)) {
ALOGI("%s: Not adding vbat speaker device to non voice use cases", __func__);
return;
@@ -2485,7 +2509,7 @@
{
int32_t port = DEFAULT_CODEC_BACKEND;
- if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX) {
+ if (snd_device >= SND_DEVICE_OUT_BEGIN && snd_device < SND_DEVICE_OUT_END) {
if (backend_tag_table[snd_device] != NULL) {
if (strncmp(backend_tag_table[snd_device], "headphones-44.1",
sizeof("headphones-44.1")) == 0)
@@ -2500,29 +2524,17 @@
else if (strcmp(backend_tag_table[snd_device], "usb-headphones") == 0)
port = USB_AUDIO_RX_BACKEND;
}
- } else {
- ALOGV("%s:napb: Invalid device - %d ", __func__, snd_device);
- }
-
- ALOGV("%s:napb: backend port - %d device - %d ", __func__, port,
- snd_device);
- return port;
-}
-
-static int platform_get_capture_backend_index(snd_device_t snd_device)
-{
- int32_t port = DEFAULT_CODEC_TX_BACKEND;
-
- if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX) {
+ } else if (snd_device >= SND_DEVICE_IN_BEGIN && snd_device < SND_DEVICE_IN_END) {
+ port = DEFAULT_CODEC_TX_BACKEND;
if (backend_tag_table[snd_device] != NULL) {
if (strcmp(backend_tag_table[snd_device], "usb-headset-mic") == 0)
port = USB_AUDIO_TX_BACKEND;
}
} else {
- ALOGW("%s: Invalid device - %d ", __func__, snd_device);
+ ALOGW("%s:napb: Invalid device - %d ", __func__, snd_device);
}
- ALOGV("%s: backend port - %d snd_device %d", __func__, port, snd_device);
+ ALOGV("%s:napb: backend port - %d device - %d ", __func__, port, snd_device);
return port;
}
@@ -2605,7 +2617,9 @@
return ret;
if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
- out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
audio_extn_spkr_prot_is_enabled()) {
if (my_data->is_vbat_speaker)
acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
@@ -2640,9 +2654,18 @@
if (my_data->acdb_send_voice_cal == NULL) {
ALOGE("%s: dlsym error for acdb_send_voice_call", __func__);
} else {
- if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER &&
- audio_extn_spkr_prot_is_enabled())
- out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+ if (audio_extn_spkr_prot_is_enabled()) {
+ if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_WSA)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+ else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED;
+ else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT;
+ else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT;
+ }
acdb_rx_id = acdb_device_table[out_snd_device];
acdb_tx_id = acdb_device_table[in_snd_device];
@@ -2669,7 +2692,9 @@
return ret;
if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
- out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
audio_extn_spkr_prot_is_enabled()) {
if (my_data->is_vbat_speaker)
acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
@@ -3073,12 +3098,22 @@
} else if (devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
snd_device = SND_DEVICE_OUT_BT_A2DP;
} else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
- if (my_data->is_vbat_speaker)
- snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
- else if (my_data->is_wsa_speaker)
- snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_WSA;
- else
- snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+ if (my_data->is_vbat_speaker) {
+ if (my_data->mono_speaker == SPKR_1)
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
+ else
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT;
+ } else if (my_data->is_wsa_speaker) {
+ if (my_data->mono_speaker == SPKR_1)
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_WSA;
+ else
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA;
+ } else {
+ if (my_data->mono_speaker == SPKR_1)
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+ else
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2;
+ }
} else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
snd_device = SND_DEVICE_OUT_USB_HEADSET;
@@ -3784,6 +3819,16 @@
}
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_MONO_SPEAKER, value, len);
+ if (err >= 0) {
+ if (!strncmp("left", value, sizeof("left")))
+ my_data->mono_speaker = SPKR_1;
+ else if (!strncmp("right", value, sizeof("right")))
+ my_data->mono_speaker = SPKR_2;
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_MONO_SPEAKER);
+ }
+
#ifdef RECORD_PLAY_CONCURRENCY
err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_REC_PLAY_CONC, value, sizeof(value));
if (err >= 0) {
@@ -3987,6 +4032,13 @@
free(kv_pairs);
}
+unsigned char* platform_get_license(void *platform __unused, int *size __unused)
+{
+ ALOGE("%s: Not implemented", __func__);
+ return NULL;
+}
+
+
/* Delay in Us, only to be used for PCM formats */
int64_t platform_render_latency(audio_usecase_t usecase)
{
@@ -4020,7 +4072,9 @@
if ((snd_device >= SND_DEVICE_IN_BEGIN) &&
(snd_device < SND_DEVICE_IN_END) &&
(snd_device != SND_DEVICE_IN_CAPTURE_FM) &&
- (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK))
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2))
needs_event = true;
return needs_event;
@@ -4083,7 +4137,9 @@
if ((snd_device >= SND_DEVICE_IN_BEGIN) &&
(snd_device < SND_DEVICE_IN_END) &&
(snd_device != SND_DEVICE_IN_CAPTURE_FM) &&
- (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK))
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2))
needs_event = true;
return needs_event;
@@ -4232,7 +4288,7 @@
if (bit_width !=
my_data->current_backend_cfg[backend_idx].bit_width) {
- struct mixer_ctl *ctl;
+ struct mixer_ctl *ctl = NULL;
ctl = mixer_get_ctl_by_name(adev->mixer,
my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl);
if (!ctl) {
@@ -4289,14 +4345,24 @@
rate_str = "KHZ_44P1";
break;
case 64000:
- case 88200:
case 96000:
rate_str = "KHZ_96";
break;
+ case 88200:
+ rate_str = "KHZ_88P2";
+ break;
case 176400:
+ rate_str = "KHZ_176P4";
+ break;
case 192000:
rate_str = "KHZ_192";
break;
+ case 352800:
+ rate_str = "KHZ_352P8";
+ break;
+ case 384000:
+ rate_str = "KHZ_384";
+ break;
default:
rate_str = "KHZ_48";
break;
@@ -4336,6 +4402,9 @@
channel_cnt_str = "Four"; break;
case 3:
channel_cnt_str = "Three"; break;
+ case 1:
+ channel_cnt_str = "One"; break;
+ case 2:
default:
channel_cnt_str = "Two"; break;
}
@@ -4688,127 +4757,6 @@
}
/*
- * configures afe with bit width and Sample Rate
- */
-
-static int platform_set_capture_codec_backend_cfg(struct audio_device* adev,
- snd_device_t snd_device,
- struct audio_backend_cfg backend_cfg)
-{
- int ret = 0;
- int backend_idx = platform_get_capture_backend_index(snd_device);
- struct platform_data *my_data = (struct platform_data *)adev->platform;
-
- ALOGI("%s:txbecf: afe: bitwidth %d, samplerate %d, backend_idx %d device (%s)",
- __func__, backend_cfg.bit_width, backend_cfg.sample_rate, backend_idx,
- platform_get_snd_device_name(snd_device));
-
- if (backend_cfg.bit_width !=
- my_data->current_tx_backend_cfg[backend_idx].bit_width) {
-
- struct mixer_ctl *ctl = NULL;
- ctl = mixer_get_ctl_by_name(adev->mixer,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
- if (!ctl) {
- ALOGE("%s:txbecf: afe: Could not get ctl for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
- return -EINVAL;
- }
-
- if (backend_cfg.bit_width == 24) {
- if (backend_cfg.format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
- ret = mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
- else
- ret = mixer_ctl_set_enum_by_string(ctl, "S24_LE");
- } else {
- ret = mixer_ctl_set_enum_by_string(ctl, "S16_LE");
- }
-
- if (ret < 0) {
- ALOGE("%s:txbecf: afe: Could not set ctl for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
- return -EINVAL;
- }
-
- my_data->current_tx_backend_cfg[backend_idx].bit_width = backend_cfg.bit_width;
- ALOGD("%s:txbecf: afe: %s mixer set to %d bit", __func__,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl,
- backend_cfg.bit_width);
- }
-
- /*
- * Backend sample rate configuration follows:
- * 16 bit record - 48khz for streams at any valid sample rate
- * 24 bit record - 48khz for stream sample rate less than 48khz
- * 24 bit record - 96khz for sample rate range of 48khz to 96khz
- * 24 bit record - 192khz for sample rate range of 96khz to 192 khz
- * Upper limit is inclusive in the sample rate range.
- */
- // TODO: This has to be more dynamic based on policy file
-
- if (backend_cfg.sample_rate !=
- my_data->current_tx_backend_cfg[(int)backend_idx].sample_rate) {
- /*
- * sample rate update is needed only for hifi audio enabled platforms
- */
- char *rate_str = NULL;
- struct mixer_ctl *ctl = NULL;
-
- switch (backend_cfg.sample_rate) {
- case 8000:
- case 11025:
- case 16000:
- case 22050:
- case 32000:
- case 44100:
- case 48000:
- rate_str = "KHZ_48";
- break;
- case 64000:
- case 88200:
- case 96000:
- rate_str = "KHZ_96";
- break;
- case 176400:
- case 192000:
- rate_str = "KHZ_192";
- break;
- default:
- rate_str = "KHZ_48";
- break;
- }
-
- ctl = mixer_get_ctl_by_name(adev->mixer,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
-
- if (ctl < 0) {
- ALOGE("%s:txbecf: afe: Could not get ctl to set the Sample Rate for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
- return -EINVAL;
- }
-
- ALOGD("%s:txbecf: afe: %s set to %s", __func__,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl,
- rate_str);
- ret = mixer_ctl_set_enum_by_string(ctl, rate_str);
- if (ret < 0) {
- ALOGE("%s:txbecf: afe: Could not set ctl for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
- return -EINVAL;
- }
-
- my_data->current_tx_backend_cfg[backend_idx].sample_rate =
- backend_cfg.sample_rate;
- }
-
- return ret;
-}
-
-/*
* goes through all the current usecases and picks the highest
* bitwidth & samplerate
*/
@@ -4827,7 +4775,8 @@
channels = backend_cfg->channels;
ALOGI("%s:txbecf: afe: Codec selected backend: %d current bit width: %d and "
- "sample rate: %d",__func__,backend_idx, bit_width, sample_rate);
+ "sample rate: %d, channels %d",__func__,backend_idx, bit_width,
+ sample_rate, channels);
// For voice calls use default configuration i.e. 16b/48K, only applicable to
// default backend
@@ -4849,14 +4798,17 @@
"sample rate: %d", __func__, backend_idx, bit_width, sample_rate);
// Force routing if the expected bitwdith or samplerate
// is not same as current backend comfiguration
- if ((bit_width != my_data->current_tx_backend_cfg[backend_idx].bit_width) ||
- (sample_rate != my_data->current_tx_backend_cfg[backend_idx].sample_rate)) {
+ if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
+ (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
+ (channels != my_data->current_backend_cfg[backend_idx].channels)) {
backend_cfg->bit_width = bit_width;
backend_cfg->sample_rate= sample_rate;
+ backend_cfg->channels = channels;
backend_change = true;
ALOGI("%s:txbecf: afe: Codec backend needs to be updated. new bit width: %d "
- "new sample rate: %d", __func__, backend_cfg->bit_width,
- backend_cfg->sample_rate);
+ "new sample rate: %d new channel: %d",
+ __func__, backend_cfg->bit_width,
+ backend_cfg->sample_rate, backend_cfg->channels);
}
return backend_change;
@@ -4865,7 +4817,7 @@
bool platform_check_and_set_capture_codec_backend_cfg(struct audio_device* adev,
struct audio_usecase *usecase, snd_device_t snd_device)
{
- int backend_idx = platform_get_capture_backend_index(snd_device);
+ int backend_idx = platform_get_backend_index(snd_device);
int ret = 0;
struct audio_backend_cfg backend_cfg;
@@ -4891,8 +4843,8 @@
platform_get_snd_device_name(snd_device));
if (platform_check_capture_codec_backend_cfg(adev, backend_idx,
&backend_cfg)) {
- ret = platform_set_capture_codec_backend_cfg(adev, snd_device,
- backend_cfg);
+ ret = platform_set_codec_backend_cfg(adev, snd_device,
+ backend_cfg);
if(!ret)
return true;
}
@@ -5546,8 +5498,11 @@
snd_device == SND_DEVICE_OUT_SPEAKER_WSA ||
snd_device == SND_DEVICE_OUT_SPEAKER_VBAT ||
snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT ||
snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
- snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_WSA) {
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_WSA ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA) {
ret = true;
}
@@ -5567,12 +5522,19 @@
case SND_DEVICE_OUT_VOICE_SPEAKER_WSA:
acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED);
break;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2:
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA:
+ acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED);
+ break;
case SND_DEVICE_OUT_SPEAKER_VBAT:
acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT);
break;
case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT);
break;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT:
+ acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT);
+ break;
default:
acdb_id = -EINVAL;
break;
@@ -5592,15 +5554,37 @@
case SND_DEVICE_OUT_VOICE_SPEAKER:
case SND_DEVICE_OUT_VOICE_SPEAKER_WSA:
return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2:
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA:
+ return SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED;
case SND_DEVICE_OUT_SPEAKER_VBAT:
return SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT;
case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT:
+ return SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT;
default:
return snd_device;
}
}
+int platform_get_vi_feedback_snd_device(snd_device_t snd_device)
+{
+ switch(snd_device) {
+ case SND_DEVICE_OUT_SPEAKER_PROTECTED:
+ case SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED:
+ case SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED:
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2;
+ default:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+ }
+}
+
int platform_set_sidetone(struct audio_device *adev,
snd_device_t out_snd_device,
bool enable,
diff --git a/hal/msm8916/platform.h b/hal/msm8916/platform.h
index 1e54ee1..33be141 100644
--- a/hal/msm8916/platform.h
+++ b/hal/msm8916/platform.h
@@ -39,6 +39,11 @@
SOURCE_QUAD_MIC = 0x8, /* Target contains 4 mics */
};
+enum {
+ SPKR_1,
+ SPKR_2
+};
+
#define PLATFORM_IMAGE_NAME "modem"
/*
@@ -92,6 +97,9 @@
SND_DEVICE_OUT_VOICE_SPEAKER,
SND_DEVICE_OUT_VOICE_SPEAKER_WSA,
SND_DEVICE_OUT_VOICE_SPEAKER_VBAT,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT,
SND_DEVICE_OUT_VOICE_HEADPHONES,
SND_DEVICE_OUT_VOICE_LINE,
SND_DEVICE_OUT_HDMI,
@@ -118,8 +126,10 @@
SND_DEVICE_OUT_ANC_HANDSET,
SND_DEVICE_OUT_SPEAKER_PROTECTED,
SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED,
SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT,
SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT,
#ifdef RECORD_PLAY_CONCURRENCY
SND_DEVICE_OUT_VOIP_HANDSET,
SND_DEVICE_OUT_VOIP_SPEAKER,
@@ -178,6 +188,8 @@
SND_DEVICE_IN_HANDSET_STEREO_DMIC,
SND_DEVICE_IN_SPEAKER_STEREO_DMIC,
SND_DEVICE_IN_CAPTURE_VI_FEEDBACK,
+ SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1,
+ SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2,
SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE,
SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE,
SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE,
@@ -221,14 +233,14 @@
HDMI_RX_BACKEND,
DISP_PORT_RX_BACKEND,
USB_AUDIO_RX_BACKEND,
+ MAX_RX_CODEC_BACKENDS = USB_AUDIO_RX_BACKEND,
+ /* TX BE follows RX BE */
+ SLIMBUS_0_TX,
+ DEFAULT_CODEC_TX_BACKEND = SLIMBUS_0_TX,
+ USB_AUDIO_TX_BACKEND,
MAX_CODEC_BACKENDS
};
-enum {
- DEFAULT_CODEC_TX_BACKEND,
- SLIMBUS_0_TX = DEFAULT_CODEC_TX_BACKEND,
- USB_AUDIO_TX_BACKEND,
- MAX_CODEC_TX_BACKENDS
-};
+
#define AUDIO_PARAMETER_KEY_NATIVE_AUDIO "audio.nat.codec.enabled"
#define AUDIO_PARAMETER_KEY_NATIVE_AUDIO_MODE "native_audio_mode"
diff --git a/hal/msm8960/platform.c b/hal/msm8960/platform.c
index c1e8e7f..b687d96 100644
--- a/hal/msm8960/platform.c
+++ b/hal/msm8960/platform.c
@@ -1039,6 +1039,12 @@
return -ENOSYS;
}
+unsigned char* platform_get_license(void *platform, int *size)
+{
+ ALOGE("%s: Not implemented", __func__);
+ return NULL;
+}
+
/* Delay in Us */
int64_t platform_render_latency(audio_usecase_t usecase)
{
@@ -1223,6 +1229,11 @@
return -ENOSYS;
}
+int platform_get_vi_feedback_snd_device(snd_device_t snd_device __unused)
+{
+ return -ENOSYS;
+}
+
int platform_spkr_prot_is_wsa_analog_mode(void *adev __unused)
{
return 0;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 1f81973..6930286 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -42,6 +42,7 @@
#include "edid.h"
#include "sound/compress_params.h"
#include "sound/msmcal-hwdep.h"
+#include <linux/msm_audio_calibration.h>
#define SOUND_TRIGGER_DEVICE_HANDSET_MONO_LOW_POWER_ACDB_ID (100)
#define MIXER_XML_DEFAULT_PATH "/system/etc/mixer_paths.xml"
@@ -59,7 +60,8 @@
#define LIB_ACDB_LOADER "libacdbloader.so"
#define CVD_VERSION_MIXER_CTL "CVD Version"
-#define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
+#define FLAC_COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
+#define MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE (2 * 1024 * 1024)
#define MIN_COMPRESS_OFFLOAD_FRAGMENT_SIZE (2 * 1024)
#define COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING (2 * 1024)
#define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
@@ -107,6 +109,8 @@
#define AUDIO_PARAMETER_KEY_AUD_CALDATA "cal_data"
#define AUDIO_PARAMETER_KEY_AUD_CALRESULT "cal_result"
+#define AUDIO_PARAMETER_KEY_MONO_SPEAKER "mono_speaker"
+
#define AUDIO_PARAMETER_KEY_PERF_LOCK_OPTS "perf_lock_opts"
/* Reload ACDB files from specified path */
@@ -217,6 +221,7 @@
/* Vbat monitor related flags */
bool is_vbat_speaker;
bool gsm_mode_enabled;
+ int mono_speaker;
/* Audio calibration related functions */
void *acdb_handle;
int voice_feature_set;
@@ -239,7 +244,6 @@
int ext_disp_type;
char ec_ref_mixer_path[64];
codec_backend_cfg_t current_backend_cfg[MAX_CODEC_BACKENDS];
- codec_backend_cfg_t current_tx_backend_cfg[MAX_CODEC_TX_BACKENDS];
char codec_version[CODEC_VERSION_MAX_LENGTH];
int hw_dep_fd;
char cvd_version[MAX_CVD_VERSION_STRING_SIZE];
@@ -347,6 +351,8 @@
[SND_DEVICE_OUT_VOICE_HANDSET] = "voice-handset",
[SND_DEVICE_OUT_VOICE_SPEAKER] = "voice-speaker",
[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = "voice-speaker-vbat",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2] = "voice-speaker-2",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = "voice-speaker-2-vbat",
[SND_DEVICE_OUT_VOICE_HEADPHONES] = "voice-headphones",
[SND_DEVICE_OUT_VOICE_LINE] = "voice-line",
[SND_DEVICE_OUT_HDMI] = "hdmi",
@@ -373,8 +379,10 @@
[SND_DEVICE_OUT_ANC_HANDSET] = "anc-handset",
[SND_DEVICE_OUT_SPEAKER_PROTECTED] = "speaker-protected",
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = "voice-speaker-protected",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED] = "voice-speaker-2-protected",
[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = "speaker-protected-vbat",
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = "voice-speaker-protected-vbat",
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT] = "voice-speaker-2-protected-vbat",
/* Capture sound devices */
[SND_DEVICE_IN_HANDSET_MIC] = "handset-mic",
@@ -424,6 +432,8 @@
[SND_DEVICE_IN_HANDSET_STEREO_DMIC] = "handset-stereo-dmic-ef",
[SND_DEVICE_IN_SPEAKER_STEREO_DMIC] = "speaker-stereo-dmic-ef",
[SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = "vi-feedback",
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1] = "vi-feedback-mono-1",
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2] = "vi-feedback-mono-2",
[SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE] = "voice-speaker-dmic-broadside",
[SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE] = "speaker-dmic-broadside",
[SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE] = "speaker-dmic-broadside",
@@ -466,6 +476,8 @@
[SND_DEVICE_OUT_VOICE_HANDSET] = 7,
[SND_DEVICE_OUT_VOICE_SPEAKER] = 14,
[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = 14,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2] = 14,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = 14,
[SND_DEVICE_OUT_VOICE_HEADPHONES] = 10,
[SND_DEVICE_OUT_VOICE_LINE] = 10,
[SND_DEVICE_OUT_HDMI] = 18,
@@ -492,8 +504,10 @@
[SND_DEVICE_OUT_ANC_HANDSET] = 103,
[SND_DEVICE_OUT_SPEAKER_PROTECTED] = 124,
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED] = 101,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED] = 101,
[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT] = 124,
[SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT] = 101,
+ [SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT] = 101,
[SND_DEVICE_IN_HANDSET_MIC] = 4,
[SND_DEVICE_IN_HANDSET_MIC_EXTERNAL] = 4,
@@ -542,6 +556,8 @@
[SND_DEVICE_IN_HANDSET_STEREO_DMIC] = 34,
[SND_DEVICE_IN_SPEAKER_STEREO_DMIC] = 35,
[SND_DEVICE_IN_CAPTURE_VI_FEEDBACK] = 102,
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1] = 102,
+ [SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2] = 102,
[SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE] = 12,
[SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE] = 12,
[SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE] = 119,
@@ -587,6 +603,8 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HANDSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_HEADPHONES)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_LINE)},
{TO_NAME_INDEX(SND_DEVICE_OUT_HDMI)},
@@ -612,8 +630,10 @@
{TO_NAME_INDEX(SND_DEVICE_OUT_ANC_HANDSET)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED)},
{TO_NAME_INDEX(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT)},
{TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT)},
+ {TO_NAME_INDEX(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT)},
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_EXTERNAL)},
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_MIC_AEC)},
@@ -659,6 +679,8 @@
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_STEREO_DMIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_STEREO_DMIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2)},
{TO_NAME_INDEX(SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE)},
{TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE)},
{TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE)},
@@ -1120,6 +1142,7 @@
backend_tag_table[SND_DEVICE_OUT_HEADPHONES_DSD] = strdup("headphones-dsd");
backend_tag_table[SND_DEVICE_OUT_HEADPHONES_44_1] = strdup("headphones-44.1");
backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_VBAT] = strdup("voice-speaker-vbat");
+ backend_tag_table[SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT] = strdup("voice-speaker-2-vbat");
backend_tag_table[SND_DEVICE_OUT_BT_A2DP] = strdup("bt-a2dp");
backend_tag_table[SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP] = strdup("speaker-and-bt-a2dp");
@@ -1543,6 +1566,7 @@
my_data->edid_info = NULL;
my_data->ext_disp_type = EXT_DISPLAY_TYPE_NONE;
my_data->hw_dep_fd = -1;
+ my_data->mono_speaker = SPKR_1;
property_get("ro.qc.sdk.audio.fluencetype", my_data->fluence_cap, "");
if (!strncmp("fluencepro", my_data->fluence_cap, sizeof("fluencepro"))) {
@@ -1731,6 +1755,8 @@
my_data->current_backend_cfg[idx].sample_rate = OUTPUT_SAMPLING_RATE_44100;
my_data->current_backend_cfg[idx].bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
my_data->current_backend_cfg[idx].channels = CODEC_BACKEND_DEFAULT_CHANNELS;
+ if (idx > MAX_RX_CODEC_BACKENDS)
+ my_data->current_backend_cfg[idx].channels = CODEC_BACKEND_DEFAULT_TX_CHANNELS;
my_data->current_backend_cfg[idx].bitwidth_mixer_ctl = NULL;
my_data->current_backend_cfg[idx].samplerate_mixer_ctl = NULL;
my_data->current_backend_cfg[idx].channels_mixer_ctl = NULL;
@@ -1751,15 +1777,17 @@
my_data->current_backend_cfg[HEADPHONE_44_1_BACKEND].samplerate_mixer_ctl =
strdup("SLIM_5_RX SampleRate");
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_TX_BACKEND].bitwidth_mixer_ctl =
+ my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].bitwidth_mixer_ctl =
strdup("SLIM_0_TX Format");
- my_data->current_tx_backend_cfg[DEFAULT_CODEC_TX_BACKEND].samplerate_mixer_ctl =
+ my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].samplerate_mixer_ctl =
strdup("SLIM_0_TX SampleRate");
- my_data->current_tx_backend_cfg[USB_AUDIO_TX_BACKEND].bitwidth_mixer_ctl =
+ my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].bitwidth_mixer_ctl =
strdup("USB_AUDIO_TX Format");
- my_data->current_tx_backend_cfg[USB_AUDIO_TX_BACKEND].samplerate_mixer_ctl =
+ my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].samplerate_mixer_ctl =
strdup("USB_AUDIO_TX SampleRate");
+ my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].channels_mixer_ctl =
+ strdup("USB_AUDIO_TX Channels");
ret = audio_extn_utils_get_codec_version(snd_card_name,
my_data->adev->snd_card,
@@ -1910,7 +1938,8 @@
return;
}
- if ((snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+ if ((snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
!(usecase->type == VOICE_CALL || usecase->type == VOIP_CALL)) {
ALOGI("%s: Not adding vbat speaker device to non voice use cases", __func__);
return;
@@ -2334,7 +2363,7 @@
{
int32_t port = DEFAULT_CODEC_BACKEND;
- if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX) {
+ if (snd_device >= SND_DEVICE_OUT_BEGIN && snd_device < SND_DEVICE_OUT_END) {
if (backend_tag_table[snd_device] != NULL) {
if (strncmp(backend_tag_table[snd_device], "headphones-44.1",
sizeof("headphones-44.1")) == 0)
@@ -2352,28 +2381,17 @@
else if (strcmp(backend_tag_table[snd_device], "usb-headphones") == 0)
port = USB_AUDIO_RX_BACKEND;
}
- } else {
- ALOGV("%s:napb: Invalid device - %d ", __func__, snd_device);
- }
-
- ALOGV("%s:napb: backend port - %d snd_device %d", __func__, port, snd_device);
- return port;
-}
-
-static int platform_get_capture_backend_index(snd_device_t snd_device)
-{
- int32_t port = DEFAULT_CODEC_TX_BACKEND;
-
- if (snd_device >= SND_DEVICE_MIN && snd_device < SND_DEVICE_MAX) {
+ } else if (snd_device >= SND_DEVICE_IN_BEGIN && snd_device < SND_DEVICE_IN_END) {
+ port = DEFAULT_CODEC_TX_BACKEND;
if (backend_tag_table[snd_device] != NULL) {
if (strcmp(backend_tag_table[snd_device], "usb-headset-mic") == 0)
port = USB_AUDIO_TX_BACKEND;
}
} else {
- ALOGW("%s: Invalid device - %d ", __func__, snd_device);
+ ALOGW("%s:napb: Invalid device - %d ", __func__, snd_device);
}
- ALOGV("%s: backend port - %d snd_device %d", __func__, port, snd_device);
+ ALOGV("%s:napb: backend port - %d device - %d ", __func__, port, snd_device);
return port;
}
@@ -2458,7 +2476,9 @@
return ret;
if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
- out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
audio_extn_spkr_prot_is_enabled()) {
if (my_data->is_vbat_speaker)
acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
@@ -2493,9 +2513,16 @@
if (my_data->acdb_send_voice_cal == NULL) {
ALOGE("%s: dlsym error for acdb_send_voice_call", __func__);
} else {
- if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER &&
- audio_extn_spkr_prot_is_enabled())
- out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+ if (audio_extn_spkr_prot_is_enabled()) {
+ if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+ else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT;
+ else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED;
+ else if (out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT)
+ out_snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT;
+ }
acdb_rx_id = acdb_device_table[out_snd_device];
acdb_tx_id = acdb_device_table[in_snd_device];
@@ -2522,7 +2549,9 @@
return ret;
if ((out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
- out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT) &&
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2 ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
+ out_snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT) &&
audio_extn_spkr_prot_is_enabled()) {
if (my_data->is_vbat_speaker)
acdb_rx_id = acdb_device_table[SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT];
@@ -2901,10 +2930,17 @@
else
snd_device = SND_DEVICE_OUT_BT_SCO;
} else if (devices & AUDIO_DEVICE_OUT_SPEAKER) {
- if (my_data->is_vbat_speaker)
- snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
- else
- snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+ if (my_data->is_vbat_speaker) {
+ if (my_data->mono_speaker == SPKR_1)
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_VBAT;
+ else
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT;
+ } else {
+ if (my_data->mono_speaker == SPKR_1)
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER;
+ else
+ snd_device = SND_DEVICE_OUT_VOICE_SPEAKER_2;
+ }
} else if (devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
snd_device = SND_DEVICE_OUT_BT_A2DP;
} else if (devices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET ||
@@ -2926,7 +2962,8 @@
}
if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
- devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
+ devices & AUDIO_DEVICE_OUT_WIRED_HEADSET ||
+ devices & AUDIO_DEVICE_OUT_LINE) {
if (devices & AUDIO_DEVICE_OUT_WIRED_HEADSET
&& audio_extn_get_anc_enabled()) {
if (audio_extn_should_use_fb_anc())
@@ -3799,6 +3836,16 @@
}
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_MONO_SPEAKER, value, len);
+ if (err >= 0) {
+ if (!strncmp("left", value, sizeof("left")))
+ my_data->mono_speaker = SPKR_1;
+ else if (!strncmp("right", value, sizeof("right")))
+ my_data->mono_speaker = SPKR_2;
+
+ str_parms_del(parms, AUDIO_PARAMETER_KEY_MONO_SPEAKER);
+ }
+
err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_EXT_AUDIO_DEVICE,
value, len);
if (err >= 0) {
@@ -4116,6 +4163,54 @@
free(kv_pairs);
}
+unsigned char* platform_get_license(void *platform, int *size)
+{
+ struct platform_data *my_data = (struct platform_data *)platform;
+ char value[PROPERTY_VALUE_MAX] = {0};
+ acdb_audio_cal_cfg_t cal;
+ unsigned char *dptr = NULL;
+ int ret=0;
+ uint32_t param_len;
+
+ if (platform == NULL) {
+ ALOGE("[%s] received null pointer %d ",__func__, __LINE__);
+ ret = -EINVAL;
+ goto done;
+ }
+ memset(&cal, 0, sizeof(cal));
+ cal.persist = 1;
+ cal.cal_type = AUDIO_CORE_METAINFO_CAL_TYPE;
+ if (!property_get("audio.qaf.acdbid", value , "") && !atoi(value)) {
+ ALOGE("[%s] audio.qaf.acdbid is not set %d ",__func__, __LINE__);
+ ret = -EINVAL;
+ goto done;
+ }
+ cal.acdb_dev_id = (uint32_t) atoi (value);
+ param_len = MAX_SET_CAL_BYTE_SIZE;
+ dptr = (unsigned char*) calloc(param_len, sizeof(unsigned char*));
+ if (dptr == NULL) {
+ ALOGE("[%s] Memory allocation failed for length %d",__func__,param_len);
+ ret = -ENOMEM;
+ goto done;
+ }
+ if (my_data->acdb_get_audio_cal != NULL) {
+ ret = my_data->acdb_get_audio_cal((void*)&cal, (void*)dptr, ¶m_len);
+ ALOGE("%s, ret[%d], param_len[%d] line %d", __func__, ret, param_len, __LINE__);
+ if (ret == 0) {
+ *size = param_len;
+ return dptr;
+ } else {
+ *size = 0;
+ }
+ }
+done:
+ if (dptr != NULL)
+ free(dptr);
+
+ return NULL;
+}
+
+/* Delay in Us */
/* Delay in Us, only to be used for PCM formats */
int64_t platform_render_latency(audio_usecase_t usecase)
{
@@ -4149,7 +4244,9 @@
if ((snd_device >= SND_DEVICE_IN_BEGIN) &&
(snd_device < SND_DEVICE_IN_END) &&
(snd_device != SND_DEVICE_IN_CAPTURE_FM) &&
- (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK))
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2))
needs_event = true;
return needs_event;
@@ -4167,7 +4264,9 @@
if ((snd_device >= SND_DEVICE_IN_BEGIN) &&
(snd_device < SND_DEVICE_IN_END) &&
(snd_device != SND_DEVICE_IN_CAPTURE_FM) &&
- (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK))
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1) &&
+ (snd_device != SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2))
needs_event = true;
return needs_event;
@@ -4199,18 +4298,21 @@
fragment_size = info->offload_buffer_size;
}
- // For FLAC use max size since it is loss less, and has sampling rates
- // upto 192kHZ
- if (info != NULL && !info->has_video &&
- info->format == AUDIO_FORMAT_FLAC) {
- fragment_size = MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
- ALOGV("FLAC fragment size %d", fragment_size);
- }
-
- if (info != NULL && info->has_video && info->is_streaming) {
- fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING;
- ALOGV("%s: offload fragment size reduced for AV streaming to %d",
- __func__, fragment_size);
+ if (info != NULL && !info->has_video) {
+ if (info->is_streaming) {
+ fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE_FOR_AV_STREAMING;
+ ALOGV("%s: offload fragment size reduced for AV streaming to %d",
+ __func__, fragment_size);
+ } else if (info->format == AUDIO_FORMAT_FLAC) {
+ fragment_size = FLAC_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+ ALOGV("FLAC fragment size %d", fragment_size);
+ } else if (info->format == AUDIO_FORMAT_DSD) {
+ fragment_size = MAX_COMPRESS_OFFLOAD_FRAGMENT_SIZE;
+ if((property_get("audio.native.dsd.buffer.size.kb", value, "")) &&
+ atoi(value))
+ fragment_size = atoi(value) * 1024;
+ ALOGV("DSD fragment size %d", fragment_size);
+ }
}
fragment_size = ALIGN( fragment_size, 1024);
@@ -4254,7 +4356,7 @@
snd_device_t snd_device, struct audio_backend_cfg backend_cfg)
{
int ret = 0;
- int backend_idx = DEFAULT_CODEC_BACKEND;
+ int backend_idx = platform_get_backend_index(snd_device);
struct platform_data *my_data = (struct platform_data *)adev->platform;
backend_idx = platform_get_backend_index(snd_device);
unsigned int bit_width = backend_cfg.bit_width;
@@ -4264,13 +4366,14 @@
bool passthrough_enabled = backend_cfg.passthrough_enabled;
ALOGI("%s:becf: afe: bitwidth %d, samplerate %d channels %d"
- ", backend_idx %d device (%s)", __func__, bit_width, sample_rate, channels, backend_idx,
+ ", backend_idx %d device (%s)", __func__, bit_width,
+ sample_rate, channels, backend_idx,
platform_get_snd_device_name(snd_device));
if (bit_width !=
my_data->current_backend_cfg[backend_idx].bit_width) {
- struct mixer_ctl *ctl;
+ struct mixer_ctl *ctl = NULL;
ctl = mixer_get_ctl_by_name(adev->mixer,
my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl);
if (!ctl) {
@@ -4282,23 +4385,30 @@
if (bit_width == 24) {
if (format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
- mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
+ ret = mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
else
- mixer_ctl_set_enum_by_string(ctl, "S24_LE");
+ ret = mixer_ctl_set_enum_by_string(ctl, "S24_LE");
} else if (bit_width == 32) {
- mixer_ctl_set_enum_by_string(ctl, "S24_LE");
+ ret = mixer_ctl_set_enum_by_string(ctl, "S24_LE");
} else {
- mixer_ctl_set_enum_by_string(ctl, "S16_LE");
+ ret = mixer_ctl_set_enum_by_string(ctl, "S16_LE");
}
- my_data->current_backend_cfg[backend_idx].bit_width = bit_width;
- ALOGD("%s:becf: afe: %s mixer set to %d bit for %x format", __func__,
- my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
+ if ( ret < 0) {
+ ALOGE("%s:becf: afe: fail for %s mixer set to %d bit for %x format", __func__,
+ my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
+ } else {
+ my_data->current_backend_cfg[backend_idx].bit_width = bit_width;
+ ALOGD("%s:becf: afe: %s mixer set to %d bit for %x format", __func__,
+ my_data->current_backend_cfg[backend_idx].bitwidth_mixer_ctl, bit_width, format);
+ }
+ /* set the ret as 0 and not pass back to upper layer */
+ ret = 0;
}
if (sample_rate !=
my_data->current_backend_cfg[backend_idx].sample_rate) {
char *rate_str = NULL;
- struct mixer_ctl *ctl;
+ struct mixer_ctl *ctl = NULL;
switch (sample_rate) {
case 8000:
@@ -4352,7 +4462,7 @@
}
if ((my_data->current_backend_cfg[backend_idx].channels_mixer_ctl) &&
(channels != my_data->current_backend_cfg[backend_idx].channels)) {
- struct mixer_ctl *ctl;
+ struct mixer_ctl *ctl = NULL;
char *channel_cnt_str = NULL;
switch (channels) {
@@ -4368,6 +4478,9 @@
channel_cnt_str = "Four"; break;
case 3:
channel_cnt_str = "Three"; break;
+ case 1:
+ channel_cnt_str = "One"; break;
+ case 2:
default:
channel_cnt_str = "Two"; break;
}
@@ -4387,7 +4500,8 @@
platform_set_edid_channels_configuration(adev->platform, channels);
ALOGD("%s:becf: afe: %s set to %s", __func__,
- my_data->current_backend_cfg[backend_idx].channels_mixer_ctl, channel_cnt_str);
+ my_data->current_backend_cfg[backend_idx].channels_mixer_ctl,
+ channel_cnt_str);
}
bool set_ext_disp_format = false;
@@ -4737,126 +4851,6 @@
}
/*
- * configures afe with bit width and Sample Rate
- */
-
-static int platform_set_capture_codec_backend_cfg(struct audio_device* adev,
- snd_device_t snd_device,
- struct audio_backend_cfg backend_cfg)
-{
- int ret = 0;
- int backend_idx = platform_get_capture_backend_index(snd_device);
- struct platform_data *my_data = (struct platform_data *)adev->platform;
-
- ALOGI("%s:txbecf: afe: bitwidth %d, samplerate %d, backend_idx %d device (%s)",
- __func__, backend_cfg.bit_width, backend_cfg.sample_rate, backend_idx,
- platform_get_snd_device_name(snd_device));
-
- if (backend_cfg.bit_width!=
- my_data->current_tx_backend_cfg[backend_idx].bit_width) {
-
- struct mixer_ctl *ctl = NULL;
- ctl = mixer_get_ctl_by_name(adev->mixer,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
- if (!ctl) {
- ALOGE("%s:txbecf: afe: Could not get ctl for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
- return -EINVAL;
- }
- if (backend_cfg.bit_width == 24) {
- if (backend_cfg.format == AUDIO_FORMAT_PCM_24_BIT_PACKED)
- ret = mixer_ctl_set_enum_by_string(ctl, "S24_3LE");
- else
- ret = mixer_ctl_set_enum_by_string(ctl, "S24_LE");
- } else {
- ret = mixer_ctl_set_enum_by_string(ctl, "S16_LE");
- }
-
- if (ret < 0) {
- ALOGE("%s:txbecf: afe: Could not set ctl for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl);
- return -EINVAL;
- }
-
- my_data->current_tx_backend_cfg[backend_idx].bit_width = backend_cfg.bit_width;
- ALOGD("%s:txbecf: afe: %s mixer set to %d bit", __func__,
- my_data->current_tx_backend_cfg[backend_idx].bitwidth_mixer_ctl,
- backend_cfg.bit_width);
- }
-
- /*
- * Backend sample rate configuration follows:
- * 16 bit record - 48khz for streams at any valid sample rate
- * 24 bit record - 48khz for stream sample rate less than 48khz
- * 24 bit record - 96khz for sample rate range of 48khz to 96khz
- * 24 bit record - 192khz for sample rate range of 96khz to 192 khz
- * Upper limit is inclusive in the sample rate range.
- */
- // TODO: This has to be more dynamic based on policy file
-
- if (backend_cfg.sample_rate !=
- my_data->current_tx_backend_cfg[(int)backend_idx].sample_rate) {
- /*
- * sample rate update is needed only for hifi audio enabled platforms
- */
- char *rate_str = NULL;
- struct mixer_ctl *ctl = NULL;
-
- switch (backend_cfg.sample_rate) {
- case 8000:
- case 11025:
- case 16000:
- case 22050:
- case 32000:
- case 44100:
- case 48000:
- rate_str = "KHZ_48";
- break;
- case 64000:
- case 88200:
- case 96000:
- rate_str = "KHZ_96";
- break;
- case 176400:
- case 192000:
- rate_str = "KHZ_192";
- break;
- default:
- rate_str = "KHZ_48";
- break;
- }
-
- ctl = mixer_get_ctl_by_name(adev->mixer,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
-
- if (!ctl) {
- ALOGE("%s:txbecf: afe: Could not get ctl to set the Sample Rate for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
- return -EINVAL;
- }
-
- ALOGD("%s:txbecf: afe: %s set to %s", __func__,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl,
- rate_str);
- ret = mixer_ctl_set_enum_by_string(ctl, rate_str);
- if (ret < 0) {
- ALOGE("%s:txbecf: afe: Could not set ctl for mixer command - %s",
- __func__,
- my_data->current_tx_backend_cfg[backend_idx].samplerate_mixer_ctl);
- return -EINVAL;
- }
-
- my_data->current_tx_backend_cfg[backend_idx].sample_rate =
- backend_cfg.sample_rate;
- }
-
- return ret;
-}
-
-/*
* goes through all the current usecases and picks the highest
* bitwidth & samplerate
*/
@@ -4875,20 +4869,21 @@
channels = backend_cfg->channels;
ALOGI("%s:txbecf: afe: Codec selected backend: %d current bit width: %d and "
- "sample rate: %d",__func__,backend_idx, bit_width, sample_rate);
+ "sample rate: %d, channels %d",__func__,backend_idx, bit_width,
+ sample_rate, channels);
// For voice calls use default configuration i.e. 16b/48K, only applicable to
// default backend
// force routing is not required here, caller will do it anyway
if (voice_is_in_call(adev) || adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
- ALOGW("%s:txbecf: afe:Use default bw and sr for voice/voip calls and "
+ ALOGW("%s:txbecf: afe: Use default bw and sr for voice/voip calls and "
"for unprocessed/camera source", __func__);
bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
}
if (backend_idx == USB_AUDIO_TX_BACKEND) {
audio_extn_usb_is_config_supported(&bit_width, &sample_rate, &channels, false);
- ALOGV("%s: USB BE configured as bit_width(%d)sample_rate(%d)channels(%d)",
+ ALOGV("%s:txbecf: afe: USB BE configured as bit_width(%d)sample_rate(%d)channels(%d)",
__func__, bit_width, sample_rate, channels);
}
@@ -4896,14 +4891,17 @@
"sample rate: %d", __func__, backend_idx, bit_width, sample_rate);
// Force routing if the expected bitwdith or samplerate
// is not same as current backend comfiguration
- if ((bit_width != my_data->current_tx_backend_cfg[backend_idx].bit_width) ||
- (sample_rate != my_data->current_tx_backend_cfg[backend_idx].sample_rate)) {
+ if ((bit_width != my_data->current_backend_cfg[backend_idx].bit_width) ||
+ (sample_rate != my_data->current_backend_cfg[backend_idx].sample_rate) ||
+ (channels != my_data->current_backend_cfg[backend_idx].channels)) {
backend_cfg->bit_width = bit_width;
backend_cfg->sample_rate= sample_rate;
+ backend_cfg->channels = channels;
backend_change = true;
ALOGI("%s:txbecf: afe: Codec backend needs to be updated. new bit width: %d "
- "new sample rate: %d", __func__, backend_cfg->bit_width,
- backend_cfg->sample_rate);
+ "new sample rate: %d new channel: %d",
+ __func__, backend_cfg->bit_width,
+ backend_cfg->sample_rate, backend_cfg->channels);
}
return backend_change;
@@ -4912,7 +4910,7 @@
bool platform_check_and_set_capture_codec_backend_cfg(struct audio_device* adev,
struct audio_usecase *usecase, snd_device_t snd_device)
{
- int backend_idx = platform_get_capture_backend_index(snd_device);
+ int backend_idx = platform_get_backend_index(snd_device);
int ret = 0;
struct audio_backend_cfg backend_cfg;
@@ -4938,8 +4936,8 @@
platform_get_snd_device_name(snd_device));
if (platform_check_capture_codec_backend_cfg(adev, backend_idx,
&backend_cfg)) {
- ret = platform_set_capture_codec_backend_cfg(adev, snd_device,
- backend_cfg);
+ ret = platform_set_codec_backend_cfg(adev, snd_device,
+ backend_cfg);
if(!ret)
return true;
}
@@ -5497,7 +5495,9 @@
if (snd_device == SND_DEVICE_OUT_SPEAKER ||
snd_device == SND_DEVICE_OUT_SPEAKER_VBAT ||
snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_VBAT ||
- snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) {
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER_2) {
ret = true;
}
@@ -5515,12 +5515,18 @@
case SND_DEVICE_OUT_VOICE_SPEAKER:
acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED);
break;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2:
+ acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED);
+ break;
case SND_DEVICE_OUT_SPEAKER_VBAT:
acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT);
break;
case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT);
break;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT:
+ acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT);
+ break;
default:
acdb_id = -EINVAL;
break;
@@ -5538,14 +5544,34 @@
return SND_DEVICE_OUT_SPEAKER_PROTECTED;
case SND_DEVICE_OUT_VOICE_SPEAKER:
return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2:
+ return SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED;
case SND_DEVICE_OUT_SPEAKER_VBAT:
return SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT;
case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT:
+ return SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT;
default:
return snd_device;
}
}
+int platform_get_vi_feedback_snd_device(snd_device_t snd_device)
+{
+ switch(snd_device) {
+ case SND_DEVICE_OUT_SPEAKER_PROTECTED:
+ case SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED:
+ case SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1;
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED:
+ case SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2;
+ default:
+ return SND_DEVICE_IN_CAPTURE_VI_FEEDBACK;
+ }
+}
int platform_spkr_prot_is_wsa_analog_mode(void *adev __unused)
{
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index bcf5d93..2b65950 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -39,6 +39,11 @@
SOURCE_QUAD_MIC = 0x8, /* Target contains 4 mics */
};
+enum {
+ SPKR_1,
+ SPKR_2
+};
+
/*
* Below are the devices for which is back end is same, SLIMBUS_0_RX.
* All these devices are handled by the internal HW codec. We can
@@ -89,6 +94,8 @@
SND_DEVICE_OUT_VOICE_HANDSET,
SND_DEVICE_OUT_VOICE_SPEAKER,
SND_DEVICE_OUT_VOICE_SPEAKER_VBAT,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_VBAT,
SND_DEVICE_OUT_VOICE_HEADPHONES,
SND_DEVICE_OUT_VOICE_LINE,
SND_DEVICE_OUT_HDMI,
@@ -115,10 +122,13 @@
SND_DEVICE_OUT_ANC_HANDSET,
SND_DEVICE_OUT_SPEAKER_PROTECTED,
SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED,
SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT,
SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_PROTECTED_VBAT,
SND_DEVICE_OUT_SPEAKER_WSA,
SND_DEVICE_OUT_VOICE_SPEAKER_WSA,
+ SND_DEVICE_OUT_VOICE_SPEAKER_2_WSA,
SND_DEVICE_OUT_END,
/*
@@ -172,6 +182,8 @@
SND_DEVICE_IN_HANDSET_STEREO_DMIC,
SND_DEVICE_IN_SPEAKER_STEREO_DMIC,
SND_DEVICE_IN_CAPTURE_VI_FEEDBACK,
+ SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_1,
+ SND_DEVICE_IN_CAPTURE_VI_FEEDBACK_MONO_2,
SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE,
SND_DEVICE_IN_SPEAKER_DMIC_BROADSIDE,
SND_DEVICE_IN_SPEAKER_DMIC_AEC_BROADSIDE,
@@ -213,14 +225,12 @@
HDMI_RX_BACKEND,
DISP_PORT_RX_BACKEND,
USB_AUDIO_RX_BACKEND,
- MAX_CODEC_BACKENDS
-};
-
-enum {
- DEFAULT_CODEC_TX_BACKEND,
- SLIMBUS_0_TX = DEFAULT_CODEC_TX_BACKEND,
+ MAX_RX_CODEC_BACKENDS = USB_AUDIO_RX_BACKEND,
+ /* TX BE follows RX BE */
+ SLIMBUS_0_TX,
+ DEFAULT_CODEC_TX_BACKEND = SLIMBUS_0_TX,
USB_AUDIO_TX_BACKEND,
- MAX_CODEC_TX_BACKENDS
+ MAX_CODEC_BACKENDS
};
#define AUDIO_PARAMETER_KEY_NATIVE_AUDIO "audio.nat.codec.enabled"
diff --git a/hal/platform_api.h b/hal/platform_api.h
index e5f8f8a..7dcd1b6 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -24,6 +24,8 @@
#define CODEC_BACKEND_DEFAULT_BIT_WIDTH 16
#define CODEC_BACKEND_DEFAULT_SAMPLE_RATE 48000
#define CODEC_BACKEND_DEFAULT_CHANNELS 2
+#define CODEC_BACKEND_DEFAULT_TX_CHANNELS 1
+
enum {
NATIVE_AUDIO_MODE_SRC = 1,
@@ -143,6 +145,7 @@
bool platform_can_enable_spkr_prot_on_device(snd_device_t snd_device);
int platform_get_spkr_prot_acdb_id(snd_device_t snd_device);
int platform_get_spkr_prot_snd_device(snd_device_t snd_device);
+int platform_get_vi_feedback_snd_device(snd_device_t snd_device);
int platform_spkr_prot_is_wsa_analog_mode(void *adev);
bool platform_can_split_snd_device(void *platform,
snd_device_t snd_device,
@@ -177,4 +180,5 @@
int app_type, int topology_id, int sample_rate, uint32_t module_id, uint32_t param_id,
void* data, int* length);
+unsigned char* platform_get_license(void* platform, int* size);
#endif // AUDIO_PLATFORM_API_H
diff --git a/hal/voice.c b/hal/voice.c
index f86483e..b84c7b7 100644
--- a/hal/voice.c
+++ b/hal/voice.c
@@ -417,11 +417,13 @@
int err = 0;
adev->voice.mic_mute = state;
- if (adev->mode == AUDIO_MODE_IN_CALL)
+ if (audio_extn_hfp_is_active(adev)) {
+ err = hfp_set_mic_mute(adev, state);
+ } else if (adev->mode == AUDIO_MODE_IN_CALL) {
err = platform_set_mic_mute(adev->platform, state);
- if (adev->mode == AUDIO_MODE_IN_COMMUNICATION)
+ } else if (adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
err = voice_extn_compress_voip_set_mic_mute(adev, state);
-
+ }
return err;
}
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index 9b950d9..022a3c0 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -464,6 +464,11 @@
}
}
#endif
+ if (property_get_bool("voice.dsd.playback.conc.disabled", true) &&
+ isInCall() && (offloadInfo.format == AUDIO_FORMAT_DSD)) {
+ ALOGD("blocking DSD compress offload on call mode");
+ return false;
+ }
#ifdef RECORD_PLAY_CONCURRENCY
char recConcPropValue[PROPERTY_VALUE_MAX];
bool prop_rec_play_enabled = false;
@@ -846,6 +851,26 @@
}
#endif
+
+ sp<SwAudioOutputDescriptor> outputDesc = NULL;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ outputDesc = mOutputs.valueAt(i);
+ if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
+ ALOGD("voice_conc:ouput desc / profile is NULL");
+ continue;
+ }
+
+ if (property_get_bool("voice.dsd.playback.conc.disabled", true) &&
+ (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
+ (outputDesc->mFormat == AUDIO_FORMAT_DSD)) {
+ ALOGD("voice_conc:calling closeOutput on call mode for DSD COMPRESS output");
+ closeOutput(mOutputs.keyAt(i));
+ // call invalidate for music, so that DSD compress will fallback to deep-buffer.
+ mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
+ }
+
+ }
+
#ifdef RECORD_PLAY_CONCURRENCY
char recConcPropValue[PROPERTY_VALUE_MAX];
bool prop_rec_play_enabled = false;