Merge "audio: Fix to return proper channel index masks supported for USB"
diff --git a/configs/msmsteppe/mixer_paths_wcd9375.xml b/configs/msmsteppe/mixer_paths_wcd9375.xml
index 85de3eb..680f445 100644
--- a/configs/msmsteppe/mixer_paths_wcd9375.xml
+++ b/configs/msmsteppe/mixer_paths_wcd9375.xml
@@ -371,8 +371,9 @@
<ctl name="COMP7 Switch" value="0" />
<ctl name="COMP8 Switch" value="0" />
- <!-- Headphone Default mode - AB -->
- <ctl name="RX HPH Mode" value="CLS_AB" />
+ <!-- Headphone class-H mode -->
+ <ctl name="RX_HPH_PWR_MODE" value="ULP" />
+ <ctl name="RX HPH Mode" value="CLS_H_ULP" />
<!-- IIR/voice anc -->
<ctl name="IIR0 Band1" id ="0" value="268435456" />
@@ -2130,9 +2131,13 @@
</path>
<path name="hph-highquality-mode">
+ <ctl name="RX_HPH_PWR_MODE" value="LOHIFI" />
+ <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
</path>
<path name="hph-lowpower-mode">
+ <ctl name="RX_HPH_PWR_MODE" value="ULP" />
+ <ctl name="RX HPH Mode" value="CLS_H_ULP" />
</path>
<path name="true-native-mode">
diff --git a/configs/msmsteppe/mixer_paths_wcd9375qrd.xml b/configs/msmsteppe/mixer_paths_wcd9375qrd.xml
index df6890b..2a9499e 100644
--- a/configs/msmsteppe/mixer_paths_wcd9375qrd.xml
+++ b/configs/msmsteppe/mixer_paths_wcd9375qrd.xml
@@ -371,8 +371,9 @@
<ctl name="COMP7 Switch" value="0" />
<ctl name="COMP8 Switch" value="0" />
- <!-- Headphone Default mode - AB -->
- <ctl name="RX HPH Mode" value="CLS_AB" />
+ <!-- Headphone class-H mode -->
+ <ctl name="RX_HPH_PWR_MODE" value="ULP" />
+ <ctl name="RX HPH Mode" value="CLS_H_ULP" />
<!-- IIR/voice anc -->
<ctl name="IIR0 Band1" id ="0" value="268435456" />
@@ -2210,9 +2211,13 @@
</path>
<path name="hph-highquality-mode">
+ <ctl name="RX_HPH_PWR_MODE" value="LOHIFI" />
+ <ctl name="RX HPH Mode" value="CLS_H_LOHIFI" />
</path>
<path name="hph-lowpower-mode">
+ <ctl name="RX_HPH_PWR_MODE" value="ULP" />
+ <ctl name="RX HPH Mode" value="CLS_H_ULP" />
</path>
<path name="true-native-mode">
diff --git a/configs/sdm710/audio_platform_info_intcodec.xml b/configs/sdm710/audio_platform_info_intcodec.xml
index 603ec57..4ae3bb1 100644
--- a/configs/sdm710/audio_platform_info_intcodec.xml
+++ b/configs/sdm710/audio_platform_info_intcodec.xml
@@ -27,6 +27,8 @@
<audio_platform_info>
<bit_width_configs>
<device name="SND_DEVICE_OUT_SPEAKER" bit_width="24"/>
+ <device name="SND_DEVICE_IN_HANDSET_MIC" bit_width="24"/>
+ <device name="SND_DEVICE_IN_HANDSET_GENERIC_QMIC" bit_width="24"/>
</bit_width_configs>
<interface_names>
<device name="AUDIO_DEVICE_IN_BUILTIN_MIC" interface="INT3_MI2S" codec_type="internal"/>
diff --git a/hal/audio_extn/audio_extn.c b/hal/audio_extn/audio_extn.c
index ba096c9..ae1fef7 100644
--- a/hal/audio_extn/audio_extn.c
+++ b/hal/audio_extn/audio_extn.c
@@ -64,6 +64,8 @@
#define MAX_SLEEP_RETRY 100
#define WIFI_INIT_WAIT_SLEEP 50
+#define MAX_NUM_CHANNELS 8
+#define Q14_GAIN_UNITY 0x4000
struct audio_extn_module {
bool anc_enabled;
@@ -301,6 +303,74 @@
ALOGV("%s: Setting custom stereo state success", __func__);
}
}
+
+void audio_extn_send_dual_mono_mixing_coefficients(struct stream_out *out)
+{
+ struct audio_device *adev = out->dev;
+ struct mixer_ctl *ctl;
+ char mixer_ctl_name[128];
+ int cust_ch_mixer_cfg[128], len = 0;
+ int ip_channel_cnt = audio_channel_count_from_out_mask(out->channel_mask);
+ int pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK);
+ int op_channel_cnt = 2;
+ int i, j, err;
+
+ ALOGV("%s", __func__);
+ if (!out->started) {
+ out->set_dual_mono = true;
+ goto exit;
+ }
+
+ ALOGD("%s: i/p channel count %d, o/p channel count %d, pcm id %d", __func__,
+ ip_channel_cnt, op_channel_cnt, pcm_device_id);
+
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+ "Audio Stream %d Channel Mix Cfg", pcm_device_id);
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: ERROR. Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ goto exit;
+ }
+
+ /* Output channel count corresponds to backend configuration channels.
+ * Input channel count corresponds to ASM session channels.
+ * Set params is called with channels that need to be selected from
+ * input to generate output.
+ * ex: "8,2" to downmix from 8 to 2 i.e. to downmix from 8 to 2,
+ *
+ * This mixer control takes values in the following sequence:
+ * - input channel count(m)
+ * - output channel count(n)
+ * - weight coeff for [out ch#1, in ch#1]
+ * ....
+ * - weight coeff for [out ch#1, in ch#m]
+ *
+ * - weight coeff for [out ch#2, in ch#1]
+ * ....
+ * - weight coeff for [out ch#2, in ch#m]
+ *
+ * - weight coeff for [out ch#n, in ch#1]
+ * ....
+ * - weight coeff for [out ch#n, in ch#m]
+ *
+ * To get dualmono ouptu weightage coeff is calculated as Unity gain
+ * divided by number of input channels.
+ */
+ cust_ch_mixer_cfg[len++] = ip_channel_cnt;
+ cust_ch_mixer_cfg[len++] = op_channel_cnt;
+ for (i = 0; i < op_channel_cnt; i++) {
+ for (j = 0; j < ip_channel_cnt; j++) {
+ cust_ch_mixer_cfg[len++] = Q14_GAIN_UNITY/ip_channel_cnt;
+ }
+ }
+
+ err = mixer_ctl_set_array(ctl, cust_ch_mixer_cfg, len);
+ if (err)
+ ALOGE("%s: ERROR. Mixer ctl set failed", __func__);
+exit:
+ return;
+}
#endif /* CUSTOM_STEREO_ENABLED */
#ifndef DTS_EAGLE
diff --git a/hal/audio_extn/audio_extn.h b/hal/audio_extn/audio_extn.h
index e74bd36..b2b0e73 100644
--- a/hal/audio_extn/audio_extn.h
+++ b/hal/audio_extn/audio_extn.h
@@ -43,6 +43,8 @@
#include "ip_hdlr_intf.h"
#include "battery_listener.h"
+#define AUDIO_PARAMETER_DUAL_MONO "dual_mono"
+
#ifndef AFE_PROXY_ENABLED
#define AUDIO_DEVICE_OUT_PROXY 0x40000
#endif
@@ -1132,4 +1134,9 @@
int audio_extn_ext_hw_plugin_set_audio_gain(void *plugin,
struct audio_usecase *usecase, uint32_t gain);
#endif
+#ifndef CUSTOM_STEREO_ENABLED
+#define audio_extn_send_dual_mono_mixing_coefficients(out) (0)
+#else
+void audio_extn_send_dual_mono_mixing_coefficients(struct stream_out *out);
+#endif
#endif /* AUDIO_EXTN_H */
diff --git a/hal/audio_extn/passthru.c b/hal/audio_extn/passthru.c
index b608043..9180391 100644
--- a/hal/audio_extn/passthru.c
+++ b/hal/audio_extn/passthru.c
@@ -546,6 +546,12 @@
struct stream_out *out)
{
struct audio_backend_cfg backend_cfg;
+ backend_cfg.sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ backend_cfg.channels = CODEC_BACKEND_DEFAULT_CHANNELS;
+ backend_cfg.bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ backend_cfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ backend_cfg.passthrough_enabled = false;
+
snd_device_t out_snd_device = SND_DEVICE_NONE;
int max_edid_channels = platform_edid_get_max_channels(out->dev->platform);
diff --git a/hal/audio_extn/spkr_protection.c b/hal/audio_extn/spkr_protection.c
index fc97cce..59fb5a3 100644
--- a/hal/audio_extn/spkr_protection.c
+++ b/hal/audio_extn/spkr_protection.c
@@ -747,7 +747,7 @@
struct timespec ts;
bool acquire_device = false;
- status.status = 0;
+ memset(&status, 0, sizeof(status));
memset(&protCfg, 0, sizeof(protCfg));
if (!adev) {
ALOGE("%s: Invalid params", __func__);
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 2053d3c..5c7fc9c 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -3210,6 +3210,8 @@
}
break;
}
+ platform_set_stream_channel_map(adev->platform, out->channel_mask,
+ out->pcm_device_id, &out->channel_map_param.channel_map[0]);
ALOGV("%s: pcm_prepare", __func__);
if (pcm_is_ready(out->pcm)) {
@@ -3223,8 +3225,6 @@
goto error_open;
}
}
- platform_set_stream_channel_map(adev->platform, out->channel_mask,
- out->pcm_device_id, &out->channel_map_param.channel_map[0]);
// apply volume for voip playback after path is set up
if (out->usecase == USECASE_AUDIO_PLAYBACK_VOIP)
out_set_voip_volume(&out->stream, out->volume_l, out->volume_r);
@@ -4020,6 +4020,13 @@
pthread_mutex_unlock(&out->lock);
}
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_DUAL_MONO, value,
+ sizeof(value));
+ if (err >= 0) {
+ if (!strncmp("true", value, sizeof("true")) || atoi(value))
+ audio_extn_send_dual_mono_mixing_coefficients(out);
+ }
+
err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_PROFILE, value, sizeof(value));
if (err >= 0) {
strlcpy(out->profile, value, sizeof(out->profile));
@@ -4630,6 +4637,8 @@
ret = -EINVAL;
goto exit;
}
+ if (out->set_dual_mono)
+ audio_extn_send_dual_mono_mixing_coefficients(out);
}
if (adev->is_channel_status_set == false && (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)){
@@ -6002,6 +6011,7 @@
out->a2dp_compress_mute = false;
out->hal_output_suspend_supported = 0;
out->dynamic_pm_qos_config_supported = 0;
+ out->set_dual_mono = false;
if ((flags & AUDIO_OUTPUT_FLAG_BD) &&
(property_get_bool("vendor.audio.matrix.limiter.enable", false)))
@@ -7273,6 +7283,7 @@
in->usecase = USECASE_AUDIO_RECORD_MMAP;
in->config = pcm_config_mmap_capture;
in->config.format = pcm_format_from_audio_format(config->format);
+ in->config.channels = channel_count;
in->stream.start = in_start;
in->stream.stop = in_stop;
in->stream.create_mmap_buffer = in_create_mmap_buffer;
@@ -7424,8 +7435,10 @@
// between the callback and close_stream
audio_extn_snd_mon_unregister_listener(stream);
- /* Disable echo reference while closing input stream */
- platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
+ // Disable echo reference if there are no active input and hfp call
+ // while closing input stream
+ if (!adev->active_input && !audio_extn_hfp_is_active(adev))
+ platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
if (in == NULL) {
ALOGE("%s: audio_stream_in ptr is NULL", __func__);
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index c13ca56..1675893 100755
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -354,6 +354,7 @@
bool stream_config_changed;
mix_matrix_params_t pan_scale_params;
mix_matrix_params_t downmix_params;
+ bool set_dual_mono;
};
struct stream_in {
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 2539ed1..30b9d10 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -2381,8 +2381,6 @@
ALOGD("ACDB initialization failed");
}
}
- /* init keep-alive for compress passthru */
- audio_extn_keep_alive_init(adev);
#ifdef DYNAMIC_LOG_ENABLED
log_utils_init();
#endif
@@ -6473,7 +6471,8 @@
*/
static bool platform_check_capture_codec_backend_cfg(struct audio_device* adev,
int backend_idx,
- struct audio_backend_cfg *backend_cfg)
+ struct audio_backend_cfg *backend_cfg,
+ snd_device_t snd_device)
{
bool backend_change = false;
unsigned int bit_width;
@@ -6494,14 +6493,18 @@
// 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)
- || (my_data->is_internal_codec)) {
+ 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 "
"for unprocessed/camera source", __func__);
bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
channels = CODEC_BACKEND_DEFAULT_TX_CHANNELS;
+ } else if (my_data->is_internal_codec && !audio_is_usb_in_device(snd_device)) {
+ sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ channels = CODEC_BACKEND_DEFAULT_TX_CHANNELS;
+ if (adev->active_input->bit_width == 24)
+ bit_width = platform_get_snd_device_bit_width(snd_device);
} else {
struct listnode *node;
struct audio_usecase *uc = NULL;
@@ -6593,7 +6596,7 @@
backend_idx, usecase->id,
platform_get_snd_device_name(snd_device));
if (platform_check_capture_codec_backend_cfg(adev, backend_idx,
- &backend_cfg)) {
+ &backend_cfg, snd_device)) {
ret = platform_set_codec_backend_cfg(adev, snd_device,
backend_cfg);
if(!ret)
diff --git a/qahw_api/test/qahw_playback_test.c b/qahw_api/test/qahw_playback_test.c
index 4160754..a048d3e 100644
--- a/qahw_api/test/qahw_playback_test.c
+++ b/qahw_api/test/qahw_playback_test.c
@@ -251,6 +251,7 @@
stream_param[i].ethread_data = nullptr;
stream_param[i].device_url = "stream";
stream_param[i].play_later = false;
+ stream_param[i].set_params = nullptr;
pthread_mutex_init(&stream_param[i].write_lock, (const pthread_mutexattr_t *)NULL);
pthread_cond_init(&stream_param[i].write_cond, (const pthread_condattr_t *) NULL);
@@ -797,6 +798,14 @@
if (bytes_to_read <= 0)
read_complete_file = true;
+ if (params->set_params) {
+ rc = qahw_out_set_parameters(params->out_handle, params->set_params);
+ if (rc) {
+ fprintf(log_file, "stream %s: failed to set kvpairs\n", params->set_params);
+ fprintf(stderr, "stream %s: failed to set kvpairs\n", params->set_params);
+ }
+ }
+
while (!exit && !stop_playback) {
if (!bytes_remaining) {
fprintf(log_file, "\nstream %d: reading bytes %zd\n", params->stream_index, bytes_wanted);
@@ -2095,7 +2104,7 @@
while ((opt = getopt_long(argc,
argv,
- "-f:r:c:b:d:s:v:V:l:t:a:w:k:PD:KF:Ee:A:u:m:S:C:p::x:y:qQh:i:h:g:",
+ "-f:r:c:b:d:s:v:V:l:t:a:w:k:PD:KF:Ee:A:u:m:S:C:p::x:y:qQh:i:h:g:O:",
long_options,
&option_index)) != -1) {
@@ -2233,6 +2242,9 @@
case 'K':
kpi_mode = true;
break;
+ case 'O':
+ stream_param[i].set_params = optarg;
+ break;
case 'F':
stream_param[i].flags = atoll(optarg);
stream_param[i].flags_set = true;
@@ -2533,6 +2545,7 @@
fprintf(log_file, "stream %d: Bitwidth:%d\n", stream->stream_index, stream->config.offload_info.bit_width);
fprintf(log_file, "stream %d: AAC Format Type:%d\n", stream->stream_index, stream->aac_fmt_type);
fprintf(log_file, "stream %d: Kvpair Values:%s\n", stream->stream_index, stream->kvpair_values);
+ fprintf(log_file, "stream %d: set params Values:%s\n", stream->stream_index, stream->set_params);
fprintf(log_file, "Log file:%s\n", log_filename);
fprintf(log_file, "Volume level:%f\n", vol_level);
diff --git a/qahw_api/test/qahw_playback_test.h b/qahw_api/test/qahw_playback_test.h
index 66229f5..b8bddbc 100644
--- a/qahw_api/test/qahw_playback_test.h
+++ b/qahw_api/test/qahw_playback_test.h
@@ -143,6 +143,7 @@
pthread_cond_t input_buffer_available_cond;
pthread_mutex_t input_buffer_available_lock;
uint32_t input_buffer_available_size;
+ char *set_params;
}stream_config;
qahw_module_handle_t * load_hal(audio_devices_t dev);