Merge "hal: add support to send compress offload fragment size"
diff --git a/hal/Android.mk b/hal/Android.mk
index 0545423..0edc5a3 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -54,7 +54,7 @@
ifeq ($(strip $(AUDIO_FEATURE_ENABLED_HDMI_EDID)),true)
LOCAL_SRC_FILES += edid.c
- LOCAL_CFLAGS := -DHDMI_EDID_ENABLED
+ LOCAL_CFLAGS += -DHDMI_EDID
endif
ifeq ($(strip $(AUDIO_USE_LL_AS_PRIMARY_OUTPUT)),true)
diff --git a/hal/audio_extn/fm.c b/hal/audio_extn/fm.c
index 0ef7c7e..ed3776c 100644
--- a/hal/audio_extn/fm.c
+++ b/hal/audio_extn/fm.c
@@ -243,20 +243,7 @@
if (ret >= 0) {
val = atoi(value);
if(val > 0)
- /*
- * Only when wsa does present and is in analog mode,
- * fm will stop/start here.
- * To-do: when the kernel codec type query change
- * is ready, enum of wsa mode should be checked here.
- * Currently, platform_get_wsa_mode will directly return
- * 1 when wsa is in analog mode.
- */
- if (platform_get_wsa_mode(adev) == 1) {
- fm_stop(adev);
- fm_start(adev);
- } else {
- select_devices(adev, USECASE_AUDIO_PLAYBACK_FM);
- }
+ select_devices(adev, USECASE_AUDIO_PLAYBACK_FM);
}
}
if (fmmod.restart_fm && (fmmod.scard_state == SND_CARD_STATE_ONLINE)) {
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index e64d318..e2db95b 100755
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -569,6 +569,9 @@
if (snd_device == SND_DEVICE_OUT_HDMI)
adev->mChannelStatusSet = false;
+ if (snd_device == SND_DEVICE_OUT_HDMI)
+ adev->mChannelStatusSet = false;
+
audio_extn_dev_arbi_release(snd_device);
audio_extn_sound_trigger_update_device_status(snd_device,
ST_EVENT_SND_DEVICE_FREE);
@@ -663,9 +666,10 @@
specified usecase to new snd devices */
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
- /* Update the out_snd_device only for the usecases that are enabled here */
- if (switch_device[usecase->id] && (usecase->type != VOICE_CALL)) {
- usecase->out_snd_device = snd_device;
+ /* Update the out_snd_device only before enabling the audio route */
+ if (switch_device[usecase->id] ) {
+ usecase->out_snd_device = snd_device;
+ if (usecase->type != VOICE_CALL)
enable_audio_route(adev, usecase);
}
}
@@ -700,7 +704,8 @@
usecase != uc_info &&
usecase->in_snd_device != snd_device &&
((uc_info->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
- ((usecase->devices & ~AUDIO_DEVICE_BIT_IN) & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND)) &&
+ (((usecase->devices & ~AUDIO_DEVICE_BIT_IN) & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND) ||
+ (usecase->type == VOICE_CALL))) &&
(usecase->id != USECASE_AUDIO_SPKR_CALIB_TX)) {
ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
__func__, use_case_table[usecase->id],
@@ -916,8 +921,14 @@
(usecase->in_snd_device != SND_DEVICE_NONE) &&
(usecase->out_snd_device != SND_DEVICE_NONE)) {
status = platform_switch_voice_call_device_pre(adev->platform);
- /* Disable sidetone only if voice call already exists */
- if (voice_is_call_state_active(adev))
+ }
+
+ if (((usecase->type == VOICE_CALL) ||
+ (usecase->type == VOIP_CALL)) &&
+ (usecase->out_snd_device != SND_DEVICE_NONE)) {
+ /* Disable sidetone only if voice/voip call already exists */
+ if (voice_is_call_state_active(adev) ||
+ voice_extn_compress_voip_is_started(adev))
voice_set_sidetone(adev, usecase->out_snd_device, false);
}
@@ -961,8 +972,9 @@
out_snd_device,
in_snd_device);
enable_audio_route_for_voice_usecases(adev, usecase);
- /* Enable sidetone only if voice call already exists */
- if (voice_is_call_state_active(adev))
+ /* Enable sidetone only if voice/voip call already exists */
+ if (voice_is_call_state_active(adev) ||
+ voice_extn_compress_voip_is_started(adev))
voice_set_sidetone(adev, out_snd_device, true);
}
diff --git a/hal/edid.c b/hal/edid.c
index 9b05950..06e1e05 100644
--- a/hal/edid.c
+++ b/hal/edid.c
@@ -662,12 +662,12 @@
ALOGV("AUDIO DESC BLOCK # %d\n",i);
info->audio_blocks_array[i].channels = channels[i];
- ALOGV("info->audio_blocks_array[i].channels %d\n",
+ ALOGD("info->audio_blocks_array[i].channels %d\n",
info->audio_blocks_array[i].channels);
ALOGV("Format Byte %d\n", formats[i]);
info->audio_blocks_array[i].format_id = (edid_audio_format_id)formats[i];
- ALOGV("info->audio_blocks_array[i].format_id %s",
+ ALOGD("info->audio_blocks_array[i].format_id %s",
edid_format_to_str(formats[i]));
ALOGV("Frequency Byte %d\n", frequency[i]);
diff --git a/hal/edid.h b/hal/edid.h
index ec83ec8..aa945bd 100644
--- a/hal/edid.h
+++ b/hal/edid.h
@@ -92,5 +92,10 @@
int channel_allocation;
} edid_audio_info;
+
+#ifndef HDMI_EDID
+#define edid_get_sink_caps(info, edid_data) (0)
+#else
bool edid_get_sink_caps(edid_audio_info* info, char *edid_data);
+#endif
#endif /* EDID_H */
diff --git a/hal/msm8916/platform.c b/hal/msm8916/platform.c
index 5d0eac5..50da713 100644
--- a/hal/msm8916/platform.c
+++ b/hal/msm8916/platform.c
@@ -1340,12 +1340,30 @@
ALOGE("%s: Could not send anc cal", __FUNCTION__);
}
+const char * get_snd_card_name_for_acdb_loader(const char *snd_card_name) {
+
+ if(snd_card_name == NULL)
+ return NULL;
+
+ // Both tasha & tasha-lite uses tasha ACDB files
+ // simulate sound card name for tasha lite, so that
+ // ACDB module loads tasha ACDB files for tasha lite
+ if(!strncmp(snd_card_name, "msm8976-tashalite-snd-card",
+ sizeof("msm8976-tashalite-snd-card"))) {
+ ALOGD("using tasha ACDB files for tasha-lite");
+ return "msm8976-tasha-snd-card";
+ } else {
+ return snd_card_name;
+ }
+}
+
+
int platform_acdb_init(void *platform)
{
struct platform_data *my_data = (struct platform_data *)platform;
char *cvd_version = NULL;
int key = 0;
- const char *snd_card_name;
+ const char *snd_card_name, *acdb_snd_card_name;
int result;
char value[PROPERTY_VALUE_MAX];
cvd_version = calloc(1, MAX_CVD_VERSION_STRING_SIZE);
@@ -1357,7 +1375,10 @@
property_get("audio.ds1.metainfo.key",value,"0");
key = atoi(value);
snd_card_name = mixer_get_name(my_data->adev->mixer);
- result = my_data->acdb_init(snd_card_name, cvd_version, key);
+ acdb_snd_card_name = get_snd_card_name_for_acdb_loader(snd_card_name);
+
+ result = my_data->acdb_init(acdb_snd_card_name, cvd_version, key);
+
if (cvd_version)
free(cvd_version);
if (!result) {
@@ -1376,7 +1397,7 @@
#define TZ_TYPE "/sys/class/thermal/thermal_zone%d/type"
#define TZ_WSA "/sys/class/thermal/thermal_zone%d/temp"
-bool is_wsa_found(void)
+static bool is_wsa_found(int *wsaCount)
{
DIR *tdir = NULL;
struct dirent *tdirent = NULL;
@@ -1385,6 +1406,7 @@
char cwd[MAX_PATH] = {0};
char file[10] = "wsa";
bool found = false;
+ int wsa_count = 0;
if (!getcwd(cwd, sizeof(cwd)))
return false;
@@ -1412,14 +1434,19 @@
ALOGD("Opening %s\n", name);
read_line_from_file(name, buf, sizeof(buf));
if (strstr(buf, file)) {
- found = true;
- break;
+ wsa_count++;
+ /*We support max only two WSA speakers*/
+ if (wsa_count == 2)
+ break;
}
tzn++;
}
closedir(tzdir);
- if (found == true)
- break;
+ }
+ if (wsa_count > 0){
+ ALOGD("Found %d WSA present on the platform", wsa_count);
+ found = true;
+ *wsaCount = wsa_count;
}
closedir(tdir);
chdir(cwd); /* Restore current working dir */
@@ -1439,6 +1466,7 @@
const char *mixer_ctl_name = "Set HPX ActiveBe";
struct mixer_ctl *ctl = NULL;
int idx;
+ int wsaCount =0;
my_data = calloc(1, sizeof(struct platform_data));
if (!my_data) {
@@ -1549,6 +1577,17 @@
my_data->fluence_mode = FLUENCE_BROADSIDE;
}
}
+
+ if (is_wsa_found(&wsaCount)) {
+ /*Set ACDB ID of Stereo speaker if two WSAs are present*/
+ /*Default ACDB ID for wsa speaker is that for mono*/
+ if (wsaCount == 2) {
+ platform_set_snd_device_acdb_id(SND_DEVICE_OUT_SPEAKER_WSA, 15);
+ platform_set_snd_device_acdb_id(SND_DEVICE_OUT_SPEAKER_VBAT, 15);
+ }
+ my_data->is_wsa_speaker = true;
+ }
+
property_get("persist.audio.FFSP.enable", ffspEnable, "");
if (!strncmp("true", ffspEnable, sizeof("true"))) {
acdb_device_table[SND_DEVICE_OUT_SPEAKER] = 131;
@@ -1631,9 +1670,6 @@
}
audio_extn_pm_vote();
- if (is_wsa_found())
- my_data->is_wsa_speaker = true;
-
/* Configure active back end for HPX*/
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (ctl) {
@@ -2392,7 +2428,7 @@
goto exit;
}
- if (popcount(devices) == 2 && !voice_is_in_call(adev)) {
+ if (popcount(devices) == 2) {
if (devices == (AUDIO_DEVICE_OUT_WIRED_HEADPHONE |
AUDIO_DEVICE_OUT_SPEAKER)) {
if (my_data->external_spk_1)
@@ -2492,7 +2528,8 @@
if (devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE ||
devices & AUDIO_DEVICE_OUT_WIRED_HEADSET) {
if (OUTPUT_SAMPLING_RATE_44100 == sample_rate &&
- platform_get_native_support()) {
+ platform_get_native_support() &&
+ !audio_extn_get_anc_enabled()) {
snd_device = SND_DEVICE_OUT_HEADPHONES_44_1;
}else if (devices & AUDIO_DEVICE_OUT_WIRED_HEADSET
&& audio_extn_get_anc_enabled()) {
@@ -4186,12 +4223,10 @@
edid_data[0] = count;
memcpy(&edid_data[1], block, count);
-#ifdef HDMI_EDID_ENABLED
if (!edid_get_sink_caps(info, edid_data)) {
ALOGE("%s: Failed to get HDMI sink capabilities", __func__);
goto fail;
}
-#endif
my_data->edid_valid = true;
return 0;
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 53f9833..bda6e45 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -2049,7 +2049,7 @@
goto exit;
}
- if (popcount(devices) == 2 && !voice_is_in_call(adev)) {
+ if (popcount(devices) == 2) {
if (devices == (AUDIO_DEVICE_OUT_WIRED_HEADPHONE |
AUDIO_DEVICE_OUT_SPEAKER)) {
if (my_data->external_spk_1)
diff --git a/hal/voice_extn/compress_voip.c b/hal/voice_extn/compress_voip.c
index 04e0cd0..bb1caf9 100644
--- a/hal/voice_extn/compress_voip.c
+++ b/hal/voice_extn/compress_voip.c
@@ -238,6 +238,7 @@
__func__, USECASE_COMPRESS_VOIP_CALL);
return -EINVAL;
}
+ voice_set_sidetone(adev, uc_info->out_snd_device, false);
/* 1. Close the PCM devices */
if (voip_data.pcm_rx) {
@@ -309,6 +310,19 @@
goto error_start_voip;
}
+ ALOGD("%s: Opening PCM capture device card_id(%d) device_id(%d)",
+ __func__, adev->snd_card, pcm_dev_tx_id);
+ voip_data.pcm_tx = pcm_open(adev->snd_card,
+ pcm_dev_tx_id,
+ PCM_IN, voip_config);
+ if (voip_data.pcm_tx && !pcm_is_ready(voip_data.pcm_tx)) {
+ ALOGE("%s: %s", __func__, pcm_get_error(voip_data.pcm_tx));
+ pcm_close(voip_data.pcm_tx);
+ voip_data.pcm_tx = NULL;
+ ret = -EIO;
+ goto error_start_voip;
+ }
+
ALOGD("%s: Opening PCM playback device card_id(%d) device_id(%d)",
__func__, adev->snd_card, pcm_dev_rx_id);
voip_data.pcm_rx = pcm_open(adev->snd_card,
@@ -318,35 +332,19 @@
ALOGE("%s: %s", __func__, pcm_get_error(voip_data.pcm_rx));
pcm_close(voip_data.pcm_rx);
voip_data.pcm_rx = NULL;
- ret = -EIO;
- goto error_start_voip;
- }
-
- ALOGD("%s: Opening PCM capture device card_id(%d) device_id(%d)",
- __func__, adev->snd_card, pcm_dev_tx_id);
- voip_data.pcm_tx = pcm_open(adev->snd_card,
- pcm_dev_tx_id,
- PCM_IN, voip_config);
- if (voip_data.pcm_tx && !pcm_is_ready(voip_data.pcm_tx)) {
- ALOGE("%s: %s", __func__, pcm_get_error(voip_data.pcm_tx));
- pcm_close(voip_data.pcm_rx);
- voip_data.pcm_tx = NULL;
- if (voip_data.pcm_rx) {
- pcm_close(voip_data.pcm_rx);
- voip_data.pcm_rx = NULL;
+ if (voip_data.pcm_tx) {
+ pcm_close(voip_data.pcm_tx);
+ voip_data.pcm_tx = NULL;
}
ret = -EIO;
goto error_start_voip;
}
- pcm_start(voip_data.pcm_rx);
+
pcm_start(voip_data.pcm_tx);
+ pcm_start(voip_data.pcm_rx);
+ voice_set_sidetone(adev, uc_info->out_snd_device, true);
voice_extn_compress_voip_set_volume(adev, adev->voice.volume);
-
- if (ret < 0) {
- ALOGE("%s: error %d\n", __func__, ret);
- goto error_start_voip;
- }
} else {
ALOGV("%s: voip usecase is already enabled", __func__);
if (voip_data.out_stream)
@@ -730,3 +728,13 @@
}
return ret;
}
+
+bool voice_extn_compress_voip_is_started(struct audio_device *adev)
+{
+ bool ret = false;
+ if (voice_extn_compress_voip_is_active(adev) &&
+ voip_data.pcm_tx && voip_data.pcm_rx)
+ ret = true;
+
+ return ret;
+}
diff --git a/hal/voice_extn/voice_extn.h b/hal/voice_extn/voice_extn.h
index 4a04adb..af0ad08 100644
--- a/hal/voice_extn/voice_extn.h
+++ b/hal/voice_extn/voice_extn.h
@@ -145,6 +145,7 @@
bool voice_extn_compress_voip_is_active(struct audio_device *adev);
bool voice_extn_compress_voip_is_format_supported(audio_format_t format);
bool voice_extn_compress_voip_is_config_supported(struct audio_config *config);
+bool voice_extn_compress_voip_is_started(struct audio_device *adev);
#else
static int voice_extn_compress_voip_close_output_stream(struct audio_stream *stream __unused)
{
@@ -264,6 +265,12 @@
ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
return true;
}
+
+static bool voice_extn_compress_voip_is_started(struct audio_device *adev __unused)
+{
+ ALOGE("%s: COMPRESS_VOIP_ENABLED is not defined", __func__);
+ return false;
+}
#endif
#endif //VOICE_EXTN_H
diff --git a/policy_hal/AudioPolicyManager.cpp b/policy_hal/AudioPolicyManager.cpp
index 2791b6d..e1405dc 100644
--- a/policy_hal/AudioPolicyManager.cpp
+++ b/policy_hal/AudioPolicyManager.cpp
@@ -1372,7 +1372,7 @@
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
} else if (/* stream == AUDIO_STREAM_MUSIC && */
flags == AUDIO_OUTPUT_FLAG_NONE &&
- property_get_bool("audio.deep_buffer.media", false /* default_value */)) {
+ property_get_bool("audio.deep_buffer.media", true/* default_value */)) {
flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
}
@@ -1794,6 +1794,37 @@
AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface)
: AudioPolicyManager(clientInterface)
{
+
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ ALOGV("Hw module %d", i);
+ for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) {
+ const sp<IOProfile> inProfile = mHwModules[i]->mInputProfiles[j];
+ ALOGV("\t Input profile ", j);
+ for (size_t k = 0; k < inProfile->mChannelMasks.size(); k++) {
+ audio_channel_mask_t channelMask =
+ inProfile->mChannelMasks.itemAt(k);
+ ALOGV("\t\tChannel Mask %x size %d", channelMask,
+ inProfile->mChannelMasks.size());
+ if (AUDIO_CHANNEL_IN_5POINT1 == channelMask) {
+ char ssr_enabled[PROPERTY_VALUE_MAX]={0};
+ if ((property_get("ro.qc.sdk.audio.ssr",
+ ssr_enabled, NULL) > 0) &&
+ (!strncmp("true", ssr_enabled, 4))) {
+ ALOGV("\t\t SSR supported, retain 5.1 channel"
+ " in input profile");
+ } else {
+ ALOGE("\t\t removing AUDIO_CHANNEL_IN_5POINT1 from"
+ " input profile as SSR(surround sound record)"
+ " is not supported on this chipset variant");
+ inProfile->mChannelMasks.removeItemsAt(k, 1);
+ ALOGV("\t\t Channel Mask size now %d",
+ inProfile->mChannelMasks.size());
+ }
+ }
+ }
+ }
+ }
+
#ifdef RECORD_PLAY_CONCURRENCY
mIsInputRequestOnProgress = false;
#endif