Merge "hal: fix klockwork failures in audio hal"
diff --git a/hal/Android.mk b/hal/Android.mk
index 38873cf..05d3e82 100644
--- a/hal/Android.mk
+++ b/hal/Android.mk
@@ -52,7 +52,7 @@
LOCAL_C_INCLUDES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr/include
LOCAL_ADDITIONAL_DEPENDENCIES += $(TARGET_OUT_INTERMEDIATES)/KERNEL_OBJ/usr
-ifneq ($(filter msm8994 msm8992,$(TARGET_BOARD_PLATFORM)),)
+ifeq ($(strip $(AUDIO_FEATURE_ENABLED_HDMI_EDID)),true)
LOCAL_SRC_FILES += edid.c
endif
diff --git a/hal/audio_extn/dev_arbi.c b/hal/audio_extn/dev_arbi.c
index d3c01c5..d7ab5ff 100644
--- a/hal/audio_extn/dev_arbi.c
+++ b/hal/audio_extn/dev_arbi.c
@@ -128,7 +128,13 @@
{
static snd_aud_dev_mapping_t snd_aud_dev_map[] = {
{SND_DEVICE_OUT_HANDSET, AUDIO_DEVICE_OUT_EARPIECE},
- {SND_DEVICE_OUT_VOICE_HANDSET, AUDIO_DEVICE_OUT_EARPIECE}
+ {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_HEADPHONES, AUDIO_DEVICE_OUT_WIRED_HEADPHONE},
+ {SND_DEVICE_OUT_VOICE_HEADPHONES, AUDIO_DEVICE_OUT_WIRED_HEADPHONE},
+ {SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+ AUDIO_DEVICE_OUT_SPEAKER | AUDIO_DEVICE_OUT_WIRED_HEADPHONE}
};
audio_devices_t aud_device = AUDIO_DEVICE_NONE;
diff --git a/hal/audio_extn/dts_eagle.c b/hal/audio_extn/dts_eagle.c
index b4bbb9f..7fa2c9b 100644
--- a/hal/audio_extn/dts_eagle.c
+++ b/hal/audio_extn/dts_eagle.c
@@ -136,7 +136,19 @@
}
static void fade_node(bool need_data) {
- int fd = creat(FADE_NOTIFY_FILE, S_IRUSR|S_IWUSR|S_IRGRP|S_IROTH), n = 0;
+ char prop[PROPERTY_VALUE_MAX];
+ property_get("use.dts_eagle", prop, "0");
+ if (strncmp("true", prop, sizeof("true")))
+ return;
+ int fd, n = 0;
+ if ((fd = open(FADE_NOTIFY_FILE, O_RDONLY)) < 0)
+ ALOGV("No fade node");
+ else {
+ ALOGV("fade node exists, remove it before creating it");
+ close(fd);
+ remove(FADE_NOTIFY_FILE);
+ }
+ fd = creat(FADE_NOTIFY_FILE, S_IRUSR|S_IWUSR|S_IRGRP|S_IROTH);
char *str = need_data ? "need" : "have";
if (fd < 0) {
ALOGE("DTS_EAGLE_HAL (%s): opening fade notifier node failed", __func__);
@@ -301,7 +313,6 @@
if (fade_in > 0 && fade_in_data && fade_out_data)
fade_node(false);
}
-
ALOGV("DTS_EAGLE_HAL (%s): exit", __func__);
}
@@ -427,10 +438,9 @@
chmod(path, S_IRWXU|S_IRGRP|S_IXGRP|S_IROTH);
ALOGV("DTS_EAGLE_NODE_STREAM (%s): opening state notifier node successful", __func__);
close(fd);
+ if (!fade_in_data || !fade_out_data)
+ fade_node(true);
}
-
- if (!fade_in_data || !fade_out_data)
- fade_node(true);
}
void audio_extn_dts_notify_playback_state(int stream_out, int has_video, int sample_rate,
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index a72ae38..96f252c 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -75,7 +75,7 @@
#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
/* ToDo: Check and update a proper value in msec */
-#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 96
+#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 50
#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
#define PROXY_OPEN_RETRY_COUNT 100
@@ -465,8 +465,10 @@
adev->snd_dev_ref_cnt[snd_device]--;
return -EINVAL;
}
+ audio_extn_dev_arbi_acquire(snd_device);
if (audio_extn_spkr_prot_start_processing(snd_device)) {
ALOGE("%s: spkr_start_processing failed", __func__);
+ audio_extn_dev_arbi_release(snd_device);
return -EINVAL;
}
} else {
@@ -532,9 +534,9 @@
audio_extn_spkr_prot_stop_processing(snd_device);
} else {
audio_route_reset_and_update_path(adev->audio_route, device_name);
- audio_extn_dev_arbi_release(snd_device);
}
+ audio_extn_dev_arbi_release(snd_device);
audio_extn_sound_trigger_update_device_status(snd_device,
ST_EVENT_SND_DEVICE_FREE);
audio_extn_listen_update_device_status(snd_device,
@@ -1546,6 +1548,14 @@
if (out->offload_callback)
compress_nonblock(out->compr, out->non_blocking);
+ /* Since small bufs uses blocking writes, a write will be blocked
+ for the default max poll time (20s) in the event of an SSR.
+ Reduce the poll time to observe and deal with SSR faster.
+ */
+ if (out->use_small_bufs) {
+ compress_set_max_poll_wait(out->compr, 1000);
+ }
+
audio_extn_dts_create_state_notifier_node(out->usecase);
audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate,
popcount(out->channel_mask),
@@ -1995,13 +2005,7 @@
uint32_t latency = 0;
if (is_offload_usecase(out->usecase)) {
- if (out->use_small_bufs == true)
- latency = ((out->compr_config.fragments *
- out->compr_config.fragment_size * 1000) /
- (out->sample_rate * out->compr_config.codec->ch_in *
- audio_bytes_per_sample(out->format)));
- else
- latency = COMPRESS_OFFLOAD_PLAYBACK_LATENCY;
+ latency = COMPRESS_OFFLOAD_PLAYBACK_LATENCY;
} else {
latency = (out->config.period_count * out->config.period_size * 1000) /
(out->config.rate);
@@ -2093,7 +2097,7 @@
}
if (is_offload_usecase(out->usecase)) {
- ALOGD("copl(%p): writing buffer (%zu bytes) to compress device", out, bytes);
+ ALOGVV("copl(%p): writing buffer (%zu bytes) to compress device", out, bytes);
if (out->send_new_metadata) {
ALOGD("copl(%p):send new gapless metadata", out);
compress_set_gapless_metadata(out->compr, &out->gapless_mdata);
diff --git a/hal/msm8974/platform.c b/hal/msm8974/platform.c
index 3286ca5..d6261c1 100644
--- a/hal/msm8974/platform.c
+++ b/hal/msm8974/platform.c
@@ -79,7 +79,8 @@
#define MIN_COMPRESS_PASSTHROUGH_FRAGMENT_SIZE (2 * 1024)
#define MAX_COMPRESS_PASSTHROUGH_FRAGMENT_SIZE (8 * 1024)
-#define ALIGN( num, to ) (((num) + (to-1)) & (~(to-1)))
+#define DIV_ROUND_UP(x, y) (((x) + (y) - 1)/(y))
+#define ALIGN(x, y) ((y) * DIV_ROUND_UP((x), (y)))
/*
* This file will have a maximum of 38 bytes:
*
@@ -2942,8 +2943,11 @@
* info->sample_rate
* (bits_per_sample >> 3)
* popcount(info->channel_mask))/1000;
- // align with LCM of 2, 4, 6, 8
- fragment_size = ALIGN( fragment_size, 24 );
+ // To have same PCM samples for all channels, the buffer size requires to
+ // be multiple of (number of channels * bytes per sample)
+ // For writes to succeed, the buffer must be written at address which is multiple of 32
+ // Alignment of 96 satsfies both of the above requirements
+ fragment_size = ALIGN(fragment_size, 96);
if(fragment_size < MIN_PCM_OFFLOAD_FRAGMENT_SIZE)
fragment_size = MIN_PCM_OFFLOAD_FRAGMENT_SIZE;
else if(fragment_size > MAX_PCM_OFFLOAD_FRAGMENT_SIZE)
diff --git a/hal/voice_extn/compress_voip.c b/hal/voice_extn/compress_voip.c
index 26636db..e8ba1f9 100644
--- a/hal/voice_extn/compress_voip.c
+++ b/hal/voice_extn/compress_voip.c
@@ -328,6 +328,7 @@
int i, ret = 0;
struct audio_usecase *uc_info;
int pcm_dev_rx_id, pcm_dev_tx_id;
+ unsigned int flags = PCM_OUT | PCM_MONOTONIC;
ALOGD("%s: enter", __func__);
@@ -368,7 +369,7 @@
__func__, adev->snd_card, pcm_dev_rx_id);
voip_data.pcm_rx = pcm_open(adev->snd_card,
pcm_dev_rx_id,
- PCM_OUT, voip_config);
+ flags, voip_config);
if (voip_data.pcm_rx && !pcm_is_ready(voip_data.pcm_rx)) {
ALOGE("%s: %s", __func__, pcm_get_error(voip_data.pcm_rx));
pcm_close(voip_data.pcm_rx);