Merge "hal: Fix test app crash issue"
diff --git a/configs/sdm845/sdm845.mk b/configs/sdm845/sdm845.mk
index 38a84c1..9d20d48 100644
--- a/configs/sdm845/sdm845.mk
+++ b/configs/sdm845/sdm845.mk
@@ -232,6 +232,7 @@
vendor.audio.adm.buffering.ms=2
#enable use of display-port for voice usecases
+PRODUCT_PROPERTY_OVERRIDES += \
vendor.audio.enable.dp.for.voice=false
# for HIDL related packages
diff --git a/hal/Makefile.am b/hal/Makefile.am
index 0096bf7..8ab3e7c 100644
--- a/hal/Makefile.am
+++ b/hal/Makefile.am
@@ -215,4 +215,5 @@
audio_primary_default_la_CFLAGS += -DINT_MAX=0x7fffffff
audio_primary_default_la_CFLAGS += -D__unused=__attribute__\(\(__unused__\)\)
audio_primary_default_la_CFLAGS += -DLINUX_ENABLED $(TARGET_CFLAGS) -DAUDIO_EXTN_FORMATS_ENABLED
+audio_primary_default_la_CFLAGS += -DNDEBUG
audio_primary_default_la_LDFLAGS = -module -shared -avoid-version
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index ea4d856..589a391 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -70,8 +70,8 @@
#define MIN_RESISTANCE_LOOKUP (3.2)
#define MAX_RESISTANCE_LOOKUP (8)
#define SPV3_LOOKUP_TABLE_ROWS (49)
-/* default limiter threshold is 0dB */
-#define DEFAULT_LIMITER_TH (0x0)
+/* default limiter threshold is 0dB(0x7FFFFFF in natural value) */
+#define DEFAULT_LIMITER_TH (0x07FFFFFF)
#define AFE_API_VERSION_SUPPORT_SPV3 (0x2)
enum spv3_boost_max_state {
BOOST_NO_MAX_STATE,
@@ -275,15 +275,15 @@
/* 3.2ohm : 0.1ohm : 8ohm lookup table */
static int spv3_limiter_th_q27_table[SPV3_LOOKUP_TABLE_ROWS] = {
- -526133494, -508685189, -491236884, -473788580, -457682452, -441576325,
- -426812375, -410706248, -395942298, -382520525, -367756575, -354334802,
- -340913029, -327491256, -315411661, -301989888, -289910292, -277830697,
- -265751101, -255013683, -242934088, -232196669, -221459251, -210721833,
- -199984415, -190589174, -179851756, -170456515, -159719096, -150323855,
- -140928614, -131533373, -122138132, -114085069, -104689828, -95294587,
- -87241523, -79188460, -69793219, -61740155, -53687091, -45634028,
- -37580964, -29527900, -22817014, -14763950, -6710886, 0,
- 0
+ 85469248, 86758070, 88066327, 89394311, 90637910, 91898809,
+ 93070036, 94364769, 95567425, 96674043, 97906130, 99039829,
+ 100186656, 101346763, 102402340, 103588104, 104667026, 105757185,
+ 106858699, 107847451, 108970736, 109979029, 110996653, 112023692,
+ 113060235, 113975074, 115029672, 115960448, 117033416, 117980405,
+ 118935056, 119897432, 120867596, 121705410, 122690202, 123682964,
+ 124540293, 125403565, 126418282, 127294571, 128176935, 129065415,
+ 129960054, 130860894, 131616362, 132528683, 133447328, 134217728,
+ 134217728
};
static struct speaker_prot_session handle;
static int vi_feed_no_channels;
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 4506115..2ff15a7 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -2709,7 +2709,13 @@
break;
}
- if (out->compr == NULL) {
+ // allow OFFLOAD_CMD_ERROR reporting during standby
+ // this is needed to handle failures during compress_open
+ // Note however that on a pause timeout, the stream is closed
+ // and no offload usecase will be active. Therefore this
+ // special case is needed for compress_open failures alone
+ if (cmd->cmd != OFFLOAD_CMD_ERROR &&
+ out->compr == NULL) {
ALOGE("%s: Compress handle is NULL", __func__);
free(cmd);
pthread_cond_signal(&out->cond);
@@ -3079,7 +3085,7 @@
COMPRESS_IN, &out->compr_config);
ATRACE_END();
if (out->compr && !is_compress_ready(out->compr)) {
- ALOGE("%s: %s", __func__, compress_get_error(out->compr));
+ ALOGE("%s: failed /w error %s", __func__, compress_get_error(out->compr));
compress_close(out->compr);
out->compr = NULL;
ret = -EIO;
@@ -3413,16 +3419,22 @@
static int out_on_error(struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
- bool do_standby = false;
lock_output_stream(out);
- if (!out->standby) {
- if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
- stop_compressed_output_l(out);
- send_offload_cmd_l(out, OFFLOAD_CMD_ERROR);
- } else
- do_standby = true;
+
+ // always send CMD_ERROR for offload streams, this
+ // is needed e.g. when SSR happens within compress_open
+ // since the stream is active, offload_callback_thread is also active.
+ if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ stop_compressed_output_l(out);
+ send_offload_cmd_l(out, OFFLOAD_CMD_ERROR);
}
+
+ // for compress streams , if the stream is not in standby
+ // it will be triggered eventually from AF.
+ bool do_standby = !out->standby &&
+ !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+
pthread_mutex_unlock(&out->lock);
if (do_standby)
@@ -6539,6 +6551,47 @@
return allowed;
}
+static int adev_update_voice_comm_input_stream(struct stream_in *in,
+ struct audio_config *config)
+{
+ bool valid_rate = (config->sample_rate == 8000 ||
+ config->sample_rate == 16000 ||
+ config->sample_rate == 32000 ||
+ config->sample_rate == 48000);
+ bool valid_ch = audio_channel_count_from_in_mask(in->channel_mask) == 1;
+
+#ifndef COMPRESS_VOIP_ENABLED
+ if (valid_rate && valid_ch) {
+ in->usecase = USECASE_AUDIO_RECORD_VOIP;
+ in->config = default_pcm_config_voip_copp;
+ in->config.period_size = VOIP_IO_BUF_SIZE(in->sample_rate,
+ DEFAULT_VOIP_BUF_DURATION_MS,
+ DEFAULT_VOIP_BIT_DEPTH_BYTE)/2;
+ } else {
+ if (!valid_ch) config->channel_mask = 1;
+ if (!valid_rate) config->sample_rate = 48000;
+ return -EINVAL;
+ }
+ in->config.rate = config->sample_rate;
+ in->sample_rate = config->sample_rate;
+#else
+ //XXX needed for voice_extn_compress_voip_open_input_stream
+ in->config.rate = config->sample_rate;
+ if ((in->dev->mode == AUDIO_MODE_IN_COMMUNICATION ||
+ voice_extn_compress_voip_is_active(in->dev)) &&
+ (voice_extn_compress_voip_is_format_supported(in->format)) &&
+ valid_rate && valid_ch) {
+ voice_extn_compress_voip_open_input_stream(in);
+ // update rate entries to match config from AF
+ in->config.rate = config->sample_rate;
+ in->sample_rate = config->sample_rate;
+ } else {
+ ALOGW("%s compress voip not active, use defaults", __func__);
+ }
+#endif
+ return 0;
+}
+
static int adev_open_input_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
@@ -6621,9 +6674,6 @@
goto err_open;
}
- /* restrict 24 bit capture for unprocessed source only
- * for other sources if 24 bit requested reject 24 and set 16 bit capture only
- */
if (is_usb_dev && may_use_hifi_record) {
/* HiFi record selects an appropriate format, channel, rate combo
depending on sink capabilities*/
@@ -6642,9 +6692,7 @@
goto err_open;
}
channel_count = audio_channel_count_from_in_mask(config->channel_mask);
- }
-
- if (config->format == AUDIO_FORMAT_DEFAULT) {
+ } else if (config->format == AUDIO_FORMAT_DEFAULT) {
config->format = AUDIO_FORMAT_PCM_16_BIT;
} else if ((config->format == AUDIO_FORMAT_PCM_FLOAT) ||
(config->format == AUDIO_FORMAT_PCM_32_BIT) ||
@@ -6706,6 +6754,7 @@
} else if (in->realtime) {
in->config = pcm_config_audio_capture_rt;
in->config.format = pcm_format_from_audio_format(config->format);
+ in->config.channels = channel_count;
in->sample_rate = in->config.rate;
in->af_period_multiplier = af_period_multiplier;
} else if (is_usb_dev && may_use_hifi_record) {
@@ -6760,40 +6809,6 @@
ret = audio_extn_cin_configure_input_stream(in);
if (ret)
goto err_open;
- } else if (in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
- bool valid_rate = (config->sample_rate == 8000 ||
- config->sample_rate == 16000 ||
- config->sample_rate == 32000 ||
- config->sample_rate == 48000);
- bool valid_ch = audio_channel_count_from_in_mask(in->channel_mask) == 1;
- //XXX needed for voice_extn_compress_voip_open_input_stream
- in->config.rate = config->sample_rate;
-#ifndef COMPRESS_VOIP_ENABLED
- if (valid_rate && valid_ch) {
- in->usecase = USECASE_AUDIO_RECORD_VOIP;
- in->config = default_pcm_config_voip_copp;
- in->config.period_size = VOIP_IO_BUF_SIZE(in->sample_rate,
- DEFAULT_VOIP_BUF_DURATION_MS,
- DEFAULT_VOIP_BIT_DEPTH_BYTE)/2;
- }
-#else
- if ((in->dev->mode == AUDIO_MODE_IN_COMMUNICATION ||
- voice_extn_compress_voip_is_active(in->dev)) &&
- (voice_extn_compress_voip_is_format_supported(in->format)) &&
- valid_rate && valid_ch) {
- voice_extn_compress_voip_open_input_stream(in);
- }
-#endif
- else {
- ALOGE("%s AUDIO_SOURCE_VOICE_COMMUNICATION invalid args", __func__);
- ret = -EINVAL;
- if (!valid_ch) config->channel_mask = 1;
- if (!valid_rate) config->sample_rate = 48000;
- goto err_open;
- }
- // update back to whatever was overwritten
- in->config.rate = config->sample_rate;
- in->sample_rate = config->sample_rate;
} else {
in->config = pcm_config_audio_capture;
in->config.rate = config->sample_rate;
@@ -6807,11 +6822,22 @@
channel_count,
is_low_latency);
in->config.period_size = buffer_size / frame_size;
+
+ if (in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
+ /* optionally use VOIP usecase depending on config(s) */
+ ret = adev_update_voice_comm_input_stream(in, config);
+ }
+
+ if (ret) {
+ ALOGE("%s AUDIO_SOURCE_VOICE_COMMUNICATION invalid args", __func__);
+ goto err_open;
+ }
}
audio_extn_utils_update_stream_input_app_type_cfg(adev->platform,
&adev->streams_input_cfg_list,
- devices, flags, in->format, in->sample_rate,
- in->bit_width, in->profile, &in->app_type_cfg);
+ devices, flags, in->format,
+ in->sample_rate, in->bit_width,
+ in->profile, &in->app_type_cfg);
/* This stream could be for sound trigger lab,
get sound trigger pcm if present */
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index c5ee848..7885b97 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -64,6 +64,9 @@
#define ANC_FLAG 0x00000001
#define DMIC_FLAG 0x00000002
#define QMIC_FLAG 0x00000004
+/* Include TMIC Flag after existing QMIC flag to avoid backward compatibility
+ * issues since they are bit masked */
+#define TMIC_FLAG 0x00000008
#define TTY_MODE_OFF 0x00000010
#define TTY_MODE_FULL 0x00000020
#define TTY_MODE_VCO 0x00000040
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 56d9aec..c31a8d8 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -6777,7 +6777,7 @@
*sample_rate = stream_sr;
if (snd_device == SND_DEVICE_OUT_HDMI)
- *sample_rate = platform_get_supported_sampling_rate_on_hdmi(stream_sr);
+ *sample_rate = platform_get_supported_copp_sampling_rate(stream_sr);
ALOGI("sn_device %d device sr %d stream sr %d copp sr %d", snd_device, device_sr, stream_sr
, *sample_rate);
@@ -7882,7 +7882,7 @@
return MAX_CODEC_BACKENDS;
}
-int platform_get_supported_sampling_rate_on_hdmi(uint32_t stream_sr)
+int platform_get_supported_copp_sampling_rate(uint32_t stream_sr)
{
int sample_rate;
switch (stream_sr){
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 891f262..4a6ba5b 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -525,7 +525,11 @@
[SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS] = "quad-mic",
[SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE] = "quad-mic",
[SND_DEVICE_IN_THREE_MIC] = "three-mic",
+ [SND_DEVICE_IN_HANDSET_TMIC_FLUENCE_PRO] = "three-mic",
[SND_DEVICE_IN_HANDSET_TMIC] = "three-mic",
+ [SND_DEVICE_IN_HANDSET_TMIC_AEC] = "three-mic",
+ [SND_DEVICE_IN_HANDSET_TMIC_NS] = "three-mic",
+ [SND_DEVICE_IN_HANDSET_TMIC_AEC_NS] = "three-mic",
[SND_DEVICE_IN_VOICE_REC_TMIC] = "three-mic",
[SND_DEVICE_IN_UNPROCESSED_MIC] = "unprocessed-mic",
[SND_DEVICE_IN_UNPROCESSED_STEREO_MIC] = "unprocessed-stereo-mic",
@@ -684,7 +688,11 @@
[SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS] = 129,
[SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE] = 125,
[SND_DEVICE_IN_THREE_MIC] = 46, /* for APSS Surround Sound Recording */
- [SND_DEVICE_IN_HANDSET_TMIC] = 125, /* for 3mic recording with fluence */
+ [SND_DEVICE_IN_HANDSET_TMIC_FLUENCE_PRO] = 125,
+ [SND_DEVICE_IN_HANDSET_TMIC] = 153,
+ [SND_DEVICE_IN_HANDSET_TMIC_AEC] = 154,
+ [SND_DEVICE_IN_HANDSET_TMIC_NS] = 155,
+ [SND_DEVICE_IN_HANDSET_TMIC_AEC_NS] = 156,
[SND_DEVICE_IN_VOICE_REC_TMIC] = 125,
[SND_DEVICE_IN_UNPROCESSED_MIC] = 143,
[SND_DEVICE_IN_UNPROCESSED_STEREO_MIC] = 144,
@@ -827,7 +835,11 @@
{TO_NAME_INDEX(SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS)},
{TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE)},
{TO_NAME_INDEX(SND_DEVICE_IN_THREE_MIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_TMIC_FLUENCE_PRO)},
{TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_TMIC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_TMIC_AEC)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_TMIC_NS)},
+ {TO_NAME_INDEX(SND_DEVICE_IN_HANDSET_TMIC_AEC_NS)},
{TO_NAME_INDEX(SND_DEVICE_IN_VOICE_REC_TMIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_MIC)},
{TO_NAME_INDEX(SND_DEVICE_IN_UNPROCESSED_STEREO_MIC)},
@@ -1507,7 +1519,11 @@
hw_interface_table[SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_THREE_MIC] = strdup("SLIMBUS_0_TX");
+ hw_interface_table[SND_DEVICE_IN_HANDSET_TMIC_FLUENCE_PRO] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_HANDSET_TMIC] = strdup("SLIMBUS_0_TX");
+ hw_interface_table[SND_DEVICE_IN_HANDSET_TMIC_AEC] = strdup("SLIMBUS_0_TX");
+ hw_interface_table[SND_DEVICE_IN_HANDSET_TMIC_NS] = strdup("SLIMBUS_0_TX");
+ hw_interface_table[SND_DEVICE_IN_HANDSET_TMIC_AEC_NS] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_VOICE_REC_TMIC] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_UNPROCESSED_MIC] = strdup("SLIMBUS_0_TX");
hw_interface_table[SND_DEVICE_IN_UNPROCESSED_STEREO_MIC] = strdup("SLIMBUS_0_TX");
@@ -2051,8 +2067,16 @@
property_get("ro.vendor.audio.sdk.fluencetype", my_data->fluence_cap, "");
if (!strncmp("fluencepro", my_data->fluence_cap, sizeof("fluencepro"))) {
my_data->fluence_type = FLUENCE_QUAD_MIC | FLUENCE_DUAL_MIC;
+
+ if (property_get_bool("persist.vendor.audio.fluence.tmic.enabled",false)) {
+ my_data->fluence_type |= FLUENCE_TRI_MIC;
+ }
} else if (!strncmp("fluence", my_data->fluence_cap, sizeof("fluence"))) {
my_data->fluence_type = FLUENCE_DUAL_MIC;
+
+ if (property_get_bool("persist.vendor.audio.fluence.tmic.enabled",false)) {
+ my_data->fluence_type |= FLUENCE_TRI_MIC;
+ }
} else {
my_data->fluence_type = FLUENCE_NONE;
}
@@ -2359,12 +2383,17 @@
my_data->current_backend_cfg[HEADPHONE_BACKEND].samplerate_mixer_ctl =
strdup("INT0_MI2S_RX SampleRate");
- }
+ } else {
- my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].bitwidth_mixer_ctl =
- strdup("SLIM_0_TX Format");
- my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].samplerate_mixer_ctl =
- strdup("SLIM_0_TX SampleRate");
+ my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].bitwidth_mixer_ctl =
+ strdup("SLIM_0_TX Format");
+ my_data->current_backend_cfg[DEFAULT_CODEC_TX_BACKEND].samplerate_mixer_ctl =
+ strdup("SLIM_0_TX SampleRate");
+ my_data->current_backend_cfg[HEADPHONE_BACKEND].bitwidth_mixer_ctl =
+ strdup("SLIM_6_RX Format");
+ my_data->current_backend_cfg[HEADPHONE_BACKEND].samplerate_mixer_ctl =
+ strdup("SLIM_6_RX SampleRate");
+ }
my_data->current_backend_cfg[USB_AUDIO_TX_BACKEND].bitwidth_mixer_ctl =
strdup("USB_AUDIO_TX Format");
@@ -2439,10 +2468,6 @@
}
}
- my_data->current_backend_cfg[HEADPHONE_BACKEND].bitwidth_mixer_ctl =
- strdup("SLIM_6_RX Format");
- my_data->current_backend_cfg[HEADPHONE_BACKEND].samplerate_mixer_ctl =
- strdup("SLIM_6_RX SampleRate");
my_data->current_backend_cfg[HDMI_RX_BACKEND].bitwidth_mixer_ctl =
strdup("HDMI_RX Bit Format");
my_data->current_backend_cfg[HDMI_RX_BACKEND].samplerate_mixer_ctl =
@@ -3926,7 +3951,11 @@
} else
snd_device = SND_DEVICE_IN_SPEAKER_MIC_AEC_NS;
} else if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- if ((my_data->fluence_type & FLUENCE_DUAL_MIC) &&
+ if ((my_data->fluence_type & FLUENCE_TRI_MIC) &&
+ (my_data->source_mic_type & SOURCE_THREE_MIC)) {
+ snd_device = SND_DEVICE_IN_HANDSET_TMIC_AEC_NS;
+ adev->acdb_settings |= TMIC_FLAG;
+ } else if ((my_data->fluence_type & FLUENCE_DUAL_MIC) &&
(my_data->source_mic_type & SOURCE_DUAL_MIC)) {
snd_device = SND_DEVICE_IN_HANDSET_DMIC_AEC_NS;
adev->acdb_settings |= DMIC_FLAG;
@@ -3966,7 +3995,11 @@
} else
snd_device = SND_DEVICE_IN_SPEAKER_MIC_AEC_NS;
} else if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- if ((my_data->fluence_type & FLUENCE_DUAL_MIC) &&
+ if ((my_data->fluence_type & FLUENCE_TRI_MIC) &&
+ (my_data->source_mic_type & SOURCE_THREE_MIC)) {
+ snd_device = SND_DEVICE_IN_HANDSET_TMIC_AEC_NS;
+ adev->acdb_settings |= TMIC_FLAG;
+ } else if ((my_data->fluence_type & FLUENCE_DUAL_MIC) &&
(my_data->source_mic_type & SOURCE_DUAL_MIC)) {
snd_device = SND_DEVICE_IN_HANDSET_DMIC_AEC_NS;
adev->acdb_settings |= DMIC_FLAG;
@@ -3996,7 +4029,11 @@
} else
snd_device = SND_DEVICE_IN_SPEAKER_MIC_AEC;
} else if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- if ((my_data->fluence_type & FLUENCE_DUAL_MIC) &&
+ if ((my_data->fluence_type & FLUENCE_TRI_MIC) &&
+ (my_data->source_mic_type & SOURCE_THREE_MIC)) {
+ snd_device = SND_DEVICE_IN_HANDSET_TMIC_AEC;
+ adev->acdb_settings |= TMIC_FLAG;
+ } else if ((my_data->fluence_type & FLUENCE_DUAL_MIC) &&
(my_data->source_mic_type & SOURCE_DUAL_MIC)) {
snd_device = SND_DEVICE_IN_HANDSET_DMIC_AEC;
adev->acdb_settings |= DMIC_FLAG;
@@ -4026,7 +4063,11 @@
} else
snd_device = SND_DEVICE_IN_SPEAKER_MIC_NS;
} else if (in_device & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- if ((my_data->fluence_type & FLUENCE_DUAL_MIC) &&
+ if ((my_data->fluence_type & FLUENCE_TRI_MIC) &&
+ (my_data->source_mic_type & SOURCE_THREE_MIC)) {
+ snd_device = SND_DEVICE_IN_HANDSET_TMIC_NS;
+ adev->acdb_settings |= TMIC_FLAG;
+ } else if ((my_data->fluence_type & FLUENCE_DUAL_MIC) &&
(my_data->source_mic_type & SOURCE_DUAL_MIC)) {
snd_device = SND_DEVICE_IN_HANDSET_DMIC_NS;
adev->acdb_settings |= DMIC_FLAG;
@@ -4142,8 +4183,14 @@
if (audio_extn_hfp_is_active(adev))
platform_set_echo_reference(adev, true, out_device);
} else {
- snd_device = SND_DEVICE_IN_VOICE_DMIC;
- adev->acdb_settings |= DMIC_FLAG;
+ if ((my_data->fluence_type & FLUENCE_TRI_MIC) &&
+ (my_data->source_mic_type & SOURCE_THREE_MIC)) {
+ snd_device = SND_DEVICE_IN_HANDSET_TMIC;
+ adev->acdb_settings |= TMIC_FLAG;
+ } else { /* for FLUENCE_DUAL_MIC and SOURCE_DUAL_MIC */
+ snd_device = SND_DEVICE_IN_VOICE_DMIC;
+ adev->acdb_settings |= DMIC_FLAG;
+ }
}
} else if (out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
snd_device = SND_DEVICE_IN_VOICE_HEADSET_MIC;
@@ -4314,6 +4361,9 @@
platform_set_echo_reference(adev, true, out_device);
} else if ((my_data->fluence_type & FLUENCE_QUAD_MIC) &&
(my_data->source_mic_type & SOURCE_THREE_MIC)) {
+ snd_device = SND_DEVICE_IN_HANDSET_TMIC_FLUENCE_PRO;
+ } else if ((my_data->fluence_type & FLUENCE_TRI_MIC) &&
+ (my_data->source_mic_type & SOURCE_THREE_MIC)) {
snd_device = SND_DEVICE_IN_HANDSET_TMIC;
} else if ((my_data->fluence_type & FLUENCE_DUAL_MIC) &&
(my_data->source_mic_type & SOURCE_DUAL_MIC)) {
@@ -6875,8 +6925,9 @@
} else
*sample_rate = stream_sr;
- if (snd_device == SND_DEVICE_OUT_HDMI)
- *sample_rate = platform_get_supported_sampling_rate_on_hdmi(stream_sr);
+ if ((snd_device == SND_DEVICE_OUT_HDMI) || (snd_device == SND_DEVICE_OUT_DISPLAY_PORT) ||
+ (snd_device == SND_DEVICE_OUT_USB_HEADSET))
+ *sample_rate = platform_get_supported_copp_sampling_rate(stream_sr);
ALOGI("sn_device %d device sr %d stream sr %d copp sr %d", snd_device, device_sr, stream_sr, *sample_rate);
@@ -7686,7 +7737,7 @@
return MAX_CODEC_BACKENDS;
}
-int platform_get_supported_sampling_rate_on_hdmi(uint32_t stream_sr)
+int platform_get_supported_copp_sampling_rate(uint32_t stream_sr)
{
int sample_rate;
switch (stream_sr){
diff --git a/hal/msm8974/platform.h b/hal/msm8974/platform.h
index db86cdc..c2fb810 100644
--- a/hal/msm8974/platform.h
+++ b/hal/msm8974/platform.h
@@ -24,8 +24,9 @@
enum {
FLUENCE_NONE,
FLUENCE_DUAL_MIC = 0x1,
- FLUENCE_QUAD_MIC = 0x2,
- FLUENCE_HEX_MIC = 0x4,
+ FLUENCE_TRI_MIC = 0x2,
+ FLUENCE_QUAD_MIC = 0x4,
+ FLUENCE_HEX_MIC = 0x8,
};
enum {
@@ -217,7 +218,11 @@
SND_DEVICE_IN_SPEAKER_QMIC_AEC_NS,
SND_DEVICE_IN_VOICE_REC_QMIC_FLUENCE,
SND_DEVICE_IN_THREE_MIC,
+ SND_DEVICE_IN_HANDSET_TMIC_FLUENCE_PRO,
SND_DEVICE_IN_HANDSET_TMIC,
+ SND_DEVICE_IN_HANDSET_TMIC_AEC,
+ SND_DEVICE_IN_HANDSET_TMIC_NS,
+ SND_DEVICE_IN_HANDSET_TMIC_AEC_NS,
SND_DEVICE_IN_VOICE_REC_TMIC,
SND_DEVICE_IN_UNPROCESSED_MIC,
SND_DEVICE_IN_UNPROCESSED_STEREO_MIC,
diff --git a/hal/platform_api.h b/hal/platform_api.h
index 0674761..e72c6e9 100644
--- a/hal/platform_api.h
+++ b/hal/platform_api.h
@@ -184,7 +184,7 @@
int platform_set_channel_allocation(void *platform, int channel_alloc);
int platform_get_edid_info(void *platform);
-int platform_get_supported_sampling_rate_on_hdmi(uint32_t stream_sr);
+int platform_get_supported_copp_sampling_rate(uint32_t stream_sr);
int platform_set_channel_map(void *platform, int ch_count, char *ch_map,
int snd_id);
int platform_set_stream_channel_map(void *platform, audio_channel_mask_t channel_mask,
diff --git a/mm-audio/aenc-aac/qdsp6/Makefile.am b/mm-audio/aenc-aac/qdsp6/Makefile.am
index a79ce70..08c9bee 100644
--- a/mm-audio/aenc-aac/qdsp6/Makefile.am
+++ b/mm-audio/aenc-aac/qdsp6/Makefile.am
@@ -5,6 +5,7 @@
AM_CFLAGS += -Wstrict-prototypes
AM_CFLAGS += -Wno-trigraphs
AM_CFLAGS += -g -O3
+AM_CFLAGS += -DNDEBUG
AM_CPPFLAGS = -D__packed__=
AM_CPPFLAGS += -DIMAGE_APPS_PROC
@@ -13,7 +14,7 @@
AM_CPPFLAGS += -DFEATURE_LINUX
AM_CPPFLAGS += -DFEATURE_NATIVELINUX
AM_CPPFLAGS += -DFEATURE_DSM_DUP_ITEMS
-AM_CPPFLAGS += -D_DEBUG
+AM_CPPFLAGS += -DNDEBUG
AM_CPPFLAGS += -Iinc
AM_CPPFLAGS += -I ${WORKSPACE}/hardware/qcom/media/mm-core/inc/
diff --git a/mm-audio/aenc-aac/qdsp6/inc/omx_aac_aenc.h b/mm-audio/aenc-aac/qdsp6/inc/omx_aac_aenc.h
index 6041ffe..38e1d9a 100644
--- a/mm-audio/aenc-aac/qdsp6/inc/omx_aac_aenc.h
+++ b/mm-audio/aenc-aac/qdsp6/inc/omx_aac_aenc.h
@@ -82,7 +82,7 @@
#define PrintFrameHdr(i,bufHdr) \
- DEBUG_PRINT("i=%d OMX bufHdr[%p]buf[%p]size[%d]TS[%lld]nFlags[0x%x]\n",\
+ DEBUG_DETAIL("i=%d OMX bufHdr[%p]buf[%p]size[%d]TS[%lld]nFlags[0x%x]\n",\
i,\
bufHdr, \
((OMX_BUFFERHEADERTYPE *)bufHdr)->pBuffer, \
diff --git a/mm-audio/aenc-amrnb/qdsp6/Makefile.am b/mm-audio/aenc-amrnb/qdsp6/Makefile.am
index 13379a3..8becd07 100644
--- a/mm-audio/aenc-amrnb/qdsp6/Makefile.am
+++ b/mm-audio/aenc-amrnb/qdsp6/Makefile.am
@@ -19,7 +19,7 @@
AM_CPPFLAGS += -I ${WORKSPACE}/hardware/qcom/media/mm-core/inc/
AM_CPPFLAGS += -g
-AM_CPPFLAGS += -D_DEBUG
+AM_CPPFLAGS += -DNDEBUG
AM_CPPFLAGS += -Iinc
c_sources = src/omx_amr_aenc.cpp \
diff --git a/mm-audio/aenc-amrnb/qdsp6/inc/omx_amr_aenc.h b/mm-audio/aenc-amrnb/qdsp6/inc/omx_amr_aenc.h
index 8236c03..54935c7 100644
--- a/mm-audio/aenc-amrnb/qdsp6/inc/omx_amr_aenc.h
+++ b/mm-audio/aenc-amrnb/qdsp6/inc/omx_amr_aenc.h
@@ -84,7 +84,7 @@
#define PrintFrameHdr(i,bufHdr) \
- DEBUG_PRINT("i=%d OMX bufHdr[%p]buf[%p]size[%d]TS[%lld]nFlags[0x%x]\n",\
+ DEBUG_DETAIL("i=%d OMX bufHdr[%p]buf[%p]size[%d]TS[%lld]nFlags[0x%x]\n",\
i,\
bufHdr, \
((OMX_BUFFERHEADERTYPE *)bufHdr)->pBuffer, \
diff --git a/mm-audio/aenc-amrnb/qdsp6/src/omx_amr_aenc.cpp b/mm-audio/aenc-amrnb/qdsp6/src/omx_amr_aenc.cpp
index 5532515..47b984c 100644
--- a/mm-audio/aenc-amrnb/qdsp6/src/omx_amr_aenc.cpp
+++ b/mm-audio/aenc-amrnb/qdsp6/src/omx_amr_aenc.cpp
@@ -512,13 +512,13 @@
m_amr_pb_stats.fbd_cnt++;
pthread_mutex_lock(&out_buf_count_lock);
nNumOutputBuf--;
- DEBUG_PRINT("FBD CB:: nNumOutputBuf=%d out_buf_len=%u fbd_cnt=%u\n",\
+ DEBUG_DETAIL("FBD CB:: nNumOutputBuf=%d out_buf_len=%u fbd_cnt=%u\n",\
nNumOutputBuf,
m_amr_pb_stats.tot_out_buf_len,
m_amr_pb_stats.fbd_cnt);
m_amr_pb_stats.tot_out_buf_len += bufHdr->nFilledLen;
m_amr_pb_stats.tot_pb_time = bufHdr->nTimeStamp;
- DEBUG_PRINT("FBD:in_buf_len=%u out_buf_len=%u\n",
+ DEBUG_DETAIL("FBD:in_buf_len=%u out_buf_len=%u\n",
m_amr_pb_stats.tot_in_buf_len,
m_amr_pb_stats.tot_out_buf_len);
@@ -3233,7 +3233,7 @@
return OMX_ErrorBadParameter;
}
*state = m_state;
- DEBUG_PRINT("Returning the state %d\n",*state);
+ DEBUG_DETAIL("Returning the state %d\n",*state);
return OMX_ErrorNone;
}
@@ -3975,12 +3975,12 @@
@return error status
*/
OMX_ERRORTYPE omx_amr_aenc::empty_this_buffer(
- OMX_IN OMX_HANDLETYPE hComp,
- OMX_IN OMX_BUFFERHEADERTYPE* buffer)
+ OMX_IN OMX_HANDLETYPE hComp,
+ OMX_IN OMX_BUFFERHEADERTYPE* buffer)
{
OMX_ERRORTYPE eRet = OMX_ErrorNone;
- DEBUG_PRINT("ETB:Buf:%p Len %u TS %lld numInBuf=%d\n", \
+ DEBUG_DETAIL("ETB:Buf:%p Len %u TS %lld numInBuf=%d\n", \
buffer, buffer->nFilledLen, buffer->nTimeStamp, (nNumInputBuf));
if (m_state == OMX_StateInvalid)
{
@@ -4076,7 +4076,7 @@
meta_in.nFlags |= OMX_BUFFERFLAG_EOS;
}
memcpy(data,&meta_in, meta_in.offsetVal);
- DEBUG_PRINT("meta_in.nFlags = %d\n",meta_in.nFlags);
+ DEBUG_DETAIL("meta_in.nFlags = %d\n",meta_in.nFlags);
} else {
DEBUG_PRINT_ERROR("temp meta is null buf\n");
return OMX_ErrorInsufficientResources;
@@ -4119,7 +4119,7 @@
if (true == search_output_bufhdr(buffer))
{
- DEBUG_PRINT("\nBefore Read..m_drv_fd = %d,\n",m_drv_fd);
+ DEBUG_DETAIL("\nBefore Read..m_drv_fd = %d,\n",m_drv_fd);
nReadbytes = read(m_drv_fd,buffer->pBuffer,output_buffer_size );
DEBUG_DETAIL("FTBP->Al_len[%lu]buf[%p]size[%d]numOutBuf[%d]\n",\
buffer->nAllocLen,buffer->pBuffer,
@@ -4131,7 +4131,7 @@
frame_done_cb((OMX_BUFFERHEADERTYPE *)buffer);
return OMX_ErrorNone;
} else
- DEBUG_PRINT("Read bytes %d\n",nReadbytes);
+ DEBUG_DETAIL("Read bytes %d\n",nReadbytes);
// Buffer from Driver will have
// 1 byte => Nr of frame field
// (sizeof(ENC_META_OUT) * Nr of frame) bytes => meta_out->offset_to_frame
@@ -4139,7 +4139,7 @@
meta_out = (ENC_META_OUT *)(buffer->pBuffer + sizeof(unsigned char));
buffer->nTimeStamp = (((OMX_TICKS)meta_out->msw_ts << 32)+
- meta_out->lsw_ts);
+ meta_out->lsw_ts);
buffer->nFlags |= meta_out->nflags;
buffer->nOffset = (OMX_U32)(meta_out->offset_to_frame +
sizeof(unsigned char));
@@ -4147,9 +4147,9 @@
ts += FRAMEDURATION;
buffer->nTimeStamp = ts;
nTimestamp = buffer->nTimeStamp;
- DEBUG_PRINT("nflags %d frame_size %d offset_to_frame %d \
- timestamp %lld\n", meta_out->nflags, meta_out->frame_size,
- meta_out->offset_to_frame, buffer->nTimeStamp);
+ DEBUG_DETAIL("nflags %d frame_size %d offset_to_frame %d \
+ timestamp %lld\n", meta_out->nflags, meta_out->frame_size,
+ meta_out->offset_to_frame, buffer->nTimeStamp);
if ((buffer->nFlags & OMX_BUFFERFLAG_EOS) == OMX_BUFFERFLAG_EOS )
{
@@ -4168,7 +4168,7 @@
return OMX_ErrorNone;
}
- DEBUG_PRINT("nState %d \n",nState );
+ DEBUG_DETAIL("nState %d \n",nState );
pthread_mutex_lock(&m_state_lock);
get_state(&m_cmp, &state);
@@ -4588,7 +4588,7 @@
========================================================================== */
bool omx_amr_aenc::release_done(OMX_U32 param1)
{
- DEBUG_PRINT("Inside omx_amr_aenc::release_done");
+ DEBUG_DETAIL("Inside omx_amr_aenc::release_done");
OMX_BOOL bRet = OMX_FALSE;
if (param1 == OMX_ALL)
diff --git a/mm-audio/aenc-g711/qdsp6/Makefile.am b/mm-audio/aenc-g711/qdsp6/Makefile.am
index f5ccca8..02b0d57 100644
--- a/mm-audio/aenc-g711/qdsp6/Makefile.am
+++ b/mm-audio/aenc-g711/qdsp6/Makefile.am
@@ -8,7 +8,7 @@
-D_ANDROID_ \
-D_ENABLE_QC_MSG_LOG_ \
-DVERBOSE \
- -D_DEBUG \
+ -DNDEBUG \
-DAUDIOV2 \
-I inc \
-I ${WORKSPACE}/hardware/qcom/media/mm-core/inc/
diff --git a/mm-audio/aenc-g711/qdsp6/inc/omx_g711_aenc.h b/mm-audio/aenc-g711/qdsp6/inc/omx_g711_aenc.h
index 4c8be66..0f415df 100644
--- a/mm-audio/aenc-g711/qdsp6/inc/omx_g711_aenc.h
+++ b/mm-audio/aenc-g711/qdsp6/inc/omx_g711_aenc.h
@@ -82,7 +82,7 @@
#define PrintFrameHdr(i,bufHdr) \
- DEBUG_PRINT("i=%d OMX bufHdr[%p]buf[%p]size[%d]TS[%lld]nFlags[0x%x]\n",\
+ DEBUG_DETAIL("i=%d OMX bufHdr[%p]buf[%p]size[%d]TS[%lld]nFlags[0x%x]\n",\
i,\
bufHdr, \
((OMX_BUFFERHEADERTYPE *)bufHdr)->pBuffer, \
diff --git a/qahw/Android.mk b/qahw/Android.mk
index c7df9e3..0b782f0 100644
--- a/qahw/Android.mk
+++ b/qahw/Android.mk
@@ -27,6 +27,7 @@
LOCAL_COPY_HEADERS += inc/qahw_effect_api.h
LOCAL_PRELINK_MODULE := false
+LOCAL_VENDOR_MODULE := true
include $(BUILD_SHARED_LIBRARY)
diff --git a/qahw/Makefile.am b/qahw/Makefile.am
index 2f33c9f..1375157 100644
--- a/qahw/Makefile.am
+++ b/qahw/Makefile.am
@@ -18,4 +18,5 @@
libqahwwrapper_la_CFLAGS += -Dstrlcpy=g_strlcpy $(GLIB_CFLAGS) -include glib.h
libqahwwrapper_la_CFLAGS += -D__unused=__attribute__\(\(__unused__\)\)
libqahwwrapper_la_CFLAGS += -Werror -Wall
+libqahwwrapper_la_CFLAGS += -DNDEBUG
libqahwwrapper_la_LDFLAGS = -shared -avoid-version -llog -lcutils -lhardware $(GLIB_LIBS)
diff --git a/qahw/inc/qahw_effect_api.h b/qahw/inc/qahw_effect_api.h
index de53cd3..dbd61e2 100644
--- a/qahw/inc/qahw_effect_api.h
+++ b/qahw/inc/qahw_effect_api.h
@@ -31,8 +31,6 @@
#include <system/audio.h>
-#include "qahw.h"
-
__BEGIN_DECLS
#define QAHW_EFFECT_API_VERSION_0_0 QAHW_MAKE_API_VERSION(0, 0)
diff --git a/qahw/src/qahw_effect.c b/qahw/src/qahw_effect.c
index cf7b3fd..2eff79f 100644
--- a/qahw/src/qahw_effect.c
+++ b/qahw/src/qahw_effect.c
@@ -38,6 +38,7 @@
#include <hardware/audio_effect.h>
#include <stdlib.h>
+#include "qahw.h"
#include "qahw_effect_api.h"
// The current effect API version.
diff --git a/qahw_api/Android.mk b/qahw_api/Android.mk
index ba402ba..fa4e6cb 100644
--- a/qahw_api/Android.mk
+++ b/qahw_api/Android.mk
@@ -34,6 +34,7 @@
LOCAL_COPY_HEADERS += inc/qahw_effect_visualizer.h
LOCAL_PRELINK_MODULE := false
+LOCAL_VENDOR_MODULE := true
include $(BUILD_SHARED_LIBRARY)
diff --git a/qahw_api/Makefile.am b/qahw_api/Makefile.am
index 13fe417..5892bb9 100644
--- a/qahw_api/Makefile.am
+++ b/qahw_api/Makefile.am
@@ -21,6 +21,7 @@
libqahw_la_CPPFLAGS := $(AM_CPPFLAGS)
libqahw_la_CPPFLAGS += -std=c++11 -DHAVE_PTHREADS -DHAVE_ANDROID_OS
libqahw_la_CPPFLAGS += -DDEBUG_REFS_CALLSTACK_ENABLED=0
+libqahw_la_CPPFLAGS += -DNDEBUG
libqahw_la_LDFLAGS = -ltinyalsa -lhardware -lexpat -lcutils -llog -ldl -lbinder -shared -avoid-version -llog -lcutils -lpthread -lutils
if QTI_AUDIO_SERVER_ENABLED
AM_CPPFLAGS += -DQTI_AUDIO_SERVER_ENABLED
diff --git a/qahw_api/test/Android.mk b/qahw_api/test/Android.mk
index ec4e698..06e8a5a 100644
--- a/qahw_api/test/Android.mk
+++ b/qahw_api/test/Android.mk
@@ -20,10 +20,10 @@
libutils \
libcutils
-LOCAL_LDLIBS := -lpthread
LOCAL_32_BIT_ONLY := true
LOCAL_C_INCLUDES += $(hal-play-inc)
+LOCAL_VENDOR_MODULE := true
include $(BUILD_EXECUTABLE)
@@ -43,4 +43,6 @@
hal-rec-inc = $(TARGET_OUT_HEADERS)/mm-audio/qahw_api/inc
LOCAL_C_INCLUDES += $(hal-rec-inc)
+LOCAL_VENDOR_MODULE := true
+
include $(BUILD_EXECUTABLE)
diff --git a/qahw_api/test/qahw_effect_test.c b/qahw_api/test/qahw_effect_test.c
index bc249f3..9ea362f 100644
--- a/qahw_api/test/qahw_effect_test.c
+++ b/qahw_api/test/qahw_effect_test.c
@@ -34,6 +34,7 @@
#include <unistd.h>
#include <string.h>
#include <errno.h>
+#include <signal.h>
#include "qahw_api.h"
#include "qahw_defs.h"
@@ -111,6 +112,12 @@
#define NUM_EQ_BANDS 5
const uint16_t qahw_equalizer_band_freqs[NUM_EQ_BANDS] = {60, 230, 910, 3600, 14000}; /* frequencies in HZ */
+/* Handler to handle input command_thread_func signal */
+void stop_effect_command_thread_handler(int signal __unused)
+{
+ pthread_exit(NULL);
+}
+
/* THREAD BODY OF BASSBOOST */
void *bassboost_thread_func(void* data) {
thread_data_t *thr_ctxt = (thread_data_t *)data;
@@ -489,6 +496,12 @@
qahw_effect_param_t *param = (qahw_effect_param_t *)buf32;
qahw_effect_param_t *param_2 = (qahw_effect_param_t *)buf32_2;
+ /* Register the SIGUSR1 to close this thread properly
+ as it is waiting for input in while loop */
+ if (signal(SIGUSR1, stop_effect_command_thread_handler) == SIG_ERR) {
+ fprintf(stderr, "Failed to register SIGUSR1:%d\n",errno);
+ }
+
while(!thr_ctxt->exit) {
if (fgets(cmd_str, sizeof(cmd_str), stdin) == NULL) {
fprintf(stderr, "read error\n");
diff --git a/qahw_api/test/qahw_multi_record_test.c b/qahw_api/test/qahw_multi_record_test.c
index d618101..2e0a396 100644
--- a/qahw_api/test/qahw_multi_record_test.c
+++ b/qahw_api/test/qahw_multi_record_test.c
@@ -588,7 +588,7 @@
printf(" For mono channel 16kHz rate for 30seconds\n\n");
}
-static void qti_audio_server_death_notify_cb(void *ctxt) {
+static void qti_audio_server_death_notify_cb(void *ctxt __unused) {
fprintf(log_file, "qas died\n");
fprintf(stderr, "qas died\n");
stop_record = true;
@@ -769,7 +769,7 @@
/* set global setparams entered by user.
* Also other global setparams can be concatenated if required.
*/
- if (params[0].kvpairs != NULL) {
+ if (params[0].kvpairs[0] != 0) {
size_t len;
len = strcspn(params[0].kvpairs, ",");
while (len < strlen(params[0].kvpairs)) {
diff --git a/qahw_api/test/qahw_playback_test.c b/qahw_api/test/qahw_playback_test.c
index fceff8b..2469b3c 100644
--- a/qahw_api/test/qahw_playback_test.c
+++ b/qahw_api/test/qahw_playback_test.c
@@ -854,10 +854,12 @@
// destory effect command thread
params->cmd_data.exit = true;
usleep(100000); // give a chance for thread to exit gracefully
- rc = pthread_cancel(params->cmd_data.cmd_thread);
+
+ //Send signal for input command_thread_func to stop
+ rc = pthread_kill(params->cmd_data.cmd_thread, SIGUSR1);
if (rc != 0) {
- fprintf(log_file, "Fail to cancel thread!\n");
- fprintf(stderr, "Fail to cancel thread!\n");
+ fprintf(log_file, "Fail to kill effect command thread!\n");
+ fprintf(stderr, "Fail to kill effect command thread!\n");
}
rc = pthread_join(params->cmd_data.cmd_thread, NULL);
if (rc < 0) {
@@ -1197,7 +1199,7 @@
int tigger_event(qahw_stream_handle_t* out_handle)
{
qahw_param_payload payload;
- struct event_data event_payload = {0};
+ struct event_data event_payload = {0, 0, 0, 0, 0, 0, 0};
int ret = 0;
event_payload.num_events = 1;
diff --git a/qahw_api/test/qahw_playback_test.h b/qahw_api/test/qahw_playback_test.h
index 3ec8f25..b643c1d 100644
--- a/qahw_api/test/qahw_playback_test.h
+++ b/qahw_api/test/qahw_playback_test.h
@@ -159,8 +159,12 @@
#define is_qap_session_active(argc, argv, kvp_string) (0)
#define get_play_list(fp, stream_param, num_of_streams, kvp_str) (0)
#define check_for_playlist(kvp_string) (0)
-#define start_playback_through_qap(kvp_string, num_of_streams,\
- qap_out_hal_handle_t) (0)
+inline int start_playback_through_qap(char * kvp_string __unused,
+ int num_of_streams __unused,
+ qahw_module_handle_t *qap_out_hal_handle_t __unused)
+{
+ return 0;
+}
#define start_playback_through_qap_playlist(cmd_kvp_str, num_of_streams,\
kvp_string, stream_param, qap_wrapper_session_active,\
qap_out_hal_handle_t) (0)