audio: Fix usecase list access am: 67486e2767
am: 50cdcc2f36
Change-Id: Idd416d12aa48bafd7447d8414802487ff893fc76
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index e5a8854..767798c 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -49,7 +49,6 @@
#include <audio_effects/effect_aec.h>
#include <audio_effects/effect_ns.h>
#include <audio_utils/clock.h>
-#include <audio_utils/power.h>
#include "audio_hw.h"
#include "audio_extn.h"
#include "platform_api.h"
@@ -98,6 +97,7 @@
#define MMAP_PERIOD_COUNT_MAX 512
#define MMAP_PERIOD_COUNT_DEFAULT (MMAP_PERIOD_COUNT_MAX)
+static const int64_t NANOS_PER_SECOND = 1000000000;
/* This constant enables extended precision handling.
* TODO The flag is off until more testing is done.
@@ -310,14 +310,6 @@
//cache last MBDRC cal step level
static int last_known_cal_step = -1 ;
-// TODO: Consider moving this to a pthread_once() if we have more
-// static initialization required.
-static bool is_userdebug_or_eng_build() {
- char value[PROPERTY_VALUE_MAX];
- (void)property_get("ro.build.type", value, "unknown"); // ignore actual length
- return strcmp(value, "userdebug") == 0 || strcmp(value, "eng") == 0;
-}
-
static bool may_use_noirq_mode(struct audio_device *adev, audio_usecase_t uc_id,
int flags __unused)
{
@@ -2070,9 +2062,7 @@
// dump error info
(void)error_log_dump(
out->error_log, fd, " " /* prefix */, 0 /* lines */, 0 /* limit_ns */);
- // dump power info (out->power_log may be null)
- (void)power_log_dump(
- out->power_log, fd, " " /* prefix */, POWER_LOG_LINES, 0 /* limit_ns */);
+
return 0;
}
@@ -2468,7 +2458,8 @@
lock_output_stream(out);
// this is always nonzero
- const int frame_size = audio_stream_out_frame_size(stream);
+ const size_t frame_size = audio_stream_out_frame_size(stream);
+ const size_t frames = bytes / frame_size;
if (out->usecase == USECASE_AUDIO_PLAYBACK_MMAP) {
error_code = ERROR_CODE_WRITE;
@@ -2559,22 +2550,16 @@
exit:
// For PCM we always consume the buffer and return #bytes regardless of ret.
if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
- out->written += bytes / (out->config.channels * sizeof(short));
+ out->written += frames;
}
long long sleeptime_us = 0;
- // only get time if needed for logging, as it is a system call on 32 bit devices.
- // TODO: Consider always enabling for 64 bit vDSO using compile time check on __LP64__.
- const int64_t now_ns = out->power_log != 0 || (ret != 0 && out->error_log != 0)
- ? audio_utils_get_real_time_ns() : 0;
-
if (ret != 0) {
- error_log_log(out->error_log, error_code, now_ns);
+ error_log_log(out->error_log, error_code, audio_utils_get_real_time_ns());
if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
ALOGE_IF(out->pcm != NULL,
"%s: error %zd - %s", __func__, ret, pcm_get_error(out->pcm));
- sleeptime_us = bytes * 1000000LL / frame_size /
- out_get_sample_rate(&out->stream.common);
+ sleeptime_us = frames * 1000000LL / out_get_sample_rate(&out->stream.common);
// usleep not guaranteed for values over 1 second but we don't limit here.
}
}
@@ -2585,9 +2570,6 @@
out_on_error(&out->stream.common);
if (sleeptime_us != 0)
usleep(sleeptime_us);
- } else {
- // only log if the data is properly written (out->power_log may be null)
- power_log_log(out->power_log, buffer, bytes / frame_size, now_ns);
}
return bytes;
}
@@ -2997,8 +2979,25 @@
return status;
}
-static int in_dump(const struct audio_stream *stream __unused, int fd __unused)
+static int in_dump(const struct audio_stream *stream, int fd)
{
+ struct stream_in *in = (struct stream_in *)stream;
+
+ // We try to get the lock for consistency,
+ // but it isn't necessary for these variables.
+ // If we're not in standby, we may be blocked on a read.
+ const bool locked = (pthread_mutex_trylock(&in->lock) == 0);
+ dprintf(fd, " Standby: %s\n", in->standby ? "yes" : "no");
+ dprintf(fd, " Frames read: %lld\n", (long long)in->frames_read);
+ dprintf(fd, " Frames muted: %lld\n", (long long)in->frames_muted);
+
+ if (locked) {
+ pthread_mutex_unlock(&in->lock);
+ }
+
+ // dump error info
+ (void)error_log_dump(
+ in->error_log, fd, " " /* prefix */, 0 /* lines */, 0 /* limit_ns */);
return 0;
}
@@ -3131,8 +3130,11 @@
struct audio_device *adev = in->dev;
int i, ret = -1;
int *int_buf_stream = NULL;
+ int error_code = ERROR_CODE_STANDBY; // initial errors are considered coming out of standby.
lock_input_stream(in);
+ const size_t frame_size = audio_stream_in_frame_size(stream);
+ const size_t frames = bytes / frame_size;
if (in->is_st_session) {
ALOGVV(" %s: reading on st session bytes=%zu", __func__, bytes);
@@ -3157,6 +3159,9 @@
in->standby = 0;
}
+ // errors that occur here are read errors.
+ error_code = ERROR_CODE_READ;
+
//what's the duration requested by the client?
long ns = pcm_bytes_to_frames(in->pcm, bytes)*1000000000LL/
in->config.rate;
@@ -3195,21 +3200,24 @@
* to always provide zeroes when muted.
* No need to acquire adev->lock to read mic_muted here as we don't change its state.
*/
- if (ret == 0 && adev->mic_muted && in->usecase != USECASE_AUDIO_RECORD_AFE_PROXY)
+ if (ret == 0 && adev->mic_muted && in->usecase != USECASE_AUDIO_RECORD_AFE_PROXY) {
memset(buffer, 0, bytes);
+ in->frames_muted += frames;
+ }
exit:
pthread_mutex_unlock(&in->lock);
if (ret != 0) {
+ error_log_log(in->error_log, error_code, audio_utils_get_real_time_ns());
in_standby(&in->stream.common);
ALOGV("%s: read failed - sleeping for buffer duration", __func__);
- usleep(bytes * 1000000 / audio_stream_in_frame_size(stream) /
- in_get_sample_rate(&in->stream.common));
+ usleep(frames * 1000000LL / in_get_sample_rate(&in->stream.common));
memset(buffer, 0, bytes); // clear return data
+ in->frames_muted += frames;
}
if (bytes > 0) {
- in->frames_read += bytes / audio_stream_in_frame_size(stream);
+ in->frames_read += frames;
}
return bytes;
}
@@ -3743,20 +3751,6 @@
ERROR_LOG_ENTRIES,
1000000000 /* aggregate consecutive identical errors within one second in ns */);
- // power_log may be null if the format is not supported
- // or not a userdebug or eng build.
- if (false /* is_userdebug_or_eng_build() */) {
- const size_t POWER_LOG_FRAMES_PER_ENTRY =
- (long long)config->sample_rate * POWER_LOG_SAMPLING_INTERVAL_MS / 1000;
-
- out->power_log = power_log_create(
- config->sample_rate,
- audio_channel_count_from_out_mask(config->channel_mask),
- config->format,
- POWER_LOG_ENTRIES,
- POWER_LOG_FRAMES_PER_ENTRY);
- }
-
/*
By locking output stream before registering, we allow the callback
to update stream's state only after stream's initial state is set to
@@ -3805,9 +3799,6 @@
if (adev->voice_tx_output == out)
adev->voice_tx_output = NULL;
- power_log_destroy(out->power_log);
- out->power_log = NULL;
-
error_log_destroy(out->error_log);
out->error_log = NULL;
@@ -4300,6 +4291,10 @@
in->config.channels = channel_count;
in->sample_rate = in->config.rate;
+ in->error_log = error_log_create(
+ ERROR_LOG_ENTRIES,
+ NANOS_PER_SECOND /* aggregate consecutive identical errors within one second */);
+
/* This stream could be for sound trigger lab,
get sound trigger pcm if present */
audio_extn_sound_trigger_check_and_get_session(in);
@@ -4326,12 +4321,17 @@
static void adev_close_input_stream(struct audio_hw_device *dev __unused,
struct audio_stream_in *stream)
{
+ struct stream_in *in = (struct stream_in *)stream;
ALOGV("%s", __func__);
// must deregister from sndmonitor first to prevent races
// between the callback and close_stream
audio_extn_snd_mon_unregister_listener(stream);
in_standby(&stream->common);
+
+ error_log_destroy(in->error_log);
+ in->error_log = NULL;
+
free(stream);
return;
diff --git a/hal/audio_hw.h b/hal/audio_hw.h
index 46384a2..a588eef 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -26,7 +26,6 @@
#include <audio_route/audio_route.h>
#include <audio_utils/ErrorLog.h>
-#include <audio_utils/PowerLog.h>
#include "voice.h"
// dlopen() does not go through default library path search if there is a "/" in the library name.
@@ -57,15 +56,11 @@
#define ERROR_LOG_ENTRIES 16
-#define POWER_LOG_LINES 40
-#define POWER_LOG_SAMPLING_INTERVAL_MS 50
-#define POWER_LOG_ENTRIES (1 /* minutes */ * 60 /* seconds */ * 1000 /* msec */ \
- / POWER_LOG_SAMPLING_INTERVAL_MS)
-
/* Error types for the error log */
enum {
ERROR_CODE_STANDBY = 1,
ERROR_CODE_WRITE,
+ ERROR_CODE_READ,
};
typedef enum card_status_t {
@@ -223,7 +218,6 @@
card_status_t card_status;
error_log_t *error_log;
- power_log_t *power_log;
struct stream_app_type_cfg app_type_cfg;
};
@@ -244,6 +238,7 @@
bool enable_aec;
bool enable_ns;
int64_t frames_read; /* total frames read, not cleared when entering standby */
+ int64_t frames_muted; /* total frames muted, not cleared when entering standby */
audio_io_handle_t capture_handle;
audio_input_flags_t flags;
@@ -263,6 +258,8 @@
audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1];
audio_format_t supported_formats[MAX_SUPPORTED_FORMATS + 1];
uint32_t supported_sample_rates[MAX_SUPPORTED_SAMPLE_RATES + 1];
+
+ error_log_t *error_log;
};
typedef enum usecase_type_t {
diff --git a/post_proc/Android.mk b/post_proc/Android.mk
index 493f2cc..f1174a0 100644
--- a/post_proc/Android.mk
+++ b/post_proc/Android.mk
@@ -30,6 +30,8 @@
external/tinyalsa/include \
$(call include-path-for, audio-effects)
+LOCAL_HEADER_LIBRARIES += libhardware_headers
+LOCAL_HEADER_LIBRARIES += libsystem_headers
include $(BUILD_SHARED_LIBRARY)
endif
@@ -60,6 +62,8 @@
hardware/qcom/audio/hal \
$(call include-path-for, audio-effects)
+LOCAL_HEADER_LIBRARIES += libhardware_headers
+LOCAL_HEADER_LIBRARIES += libsystem_headers
include $(BUILD_SHARED_LIBRARY)
endif
diff --git a/visualizer/Android.mk b/visualizer/Android.mk
index e318214..29a093e 100644
--- a/visualizer/Android.mk
+++ b/visualizer/Android.mk
@@ -36,4 +36,5 @@
external/tinyalsa/include \
$(call include-path-for, audio-effects)
+LOCAL_HEADER_LIBRARIES += libsystem_headers
include $(BUILD_SHARED_LIBRARY)
diff --git a/voice_processing/Android.mk b/voice_processing/Android.mk
index e8878ee..7ebeb70 100644
--- a/voice_processing/Android.mk
+++ b/voice_processing/Android.mk
@@ -23,4 +23,5 @@
LOCAL_CFLAGS += -fvisibility=hidden
+LOCAL_HEADER_LIBRARIES += libhardware_headers
include $(BUILD_SHARED_LIBRARY)