audio_hw: Add input record error log
Counts errors from the alsa device
Test: adb shell dumpsys media.audio_flinger
Bug: 62307033
Change-Id: I9ace0b758988d2717f471d125047b9ef5f3941ad
diff --git a/hal/audio_hw.c b/hal/audio_hw.c
index 1a8a650..065d1d0 100644
--- a/hal/audio_hw.c
+++ b/hal/audio_hw.c
@@ -93,6 +93,7 @@
#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.
@@ -2966,8 +2967,25 @@
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;
}
@@ -3100,6 +3118,7 @@
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);
@@ -3128,6 +3147,9 @@
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;
@@ -3166,17 +3188,21 @@
* 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(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 += frames;
@@ -4222,6 +4248,10 @@
in->config.channels = channel_count;
+ 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);
@@ -4246,12 +4276,17 @@
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 b1e5e45..1d8b624 100644
--- a/hal/audio_hw.h
+++ b/hal/audio_hw.h
@@ -66,6 +66,7 @@
enum {
ERROR_CODE_STANDBY = 1,
ERROR_CODE_WRITE,
+ ERROR_CODE_READ,
};
typedef enum card_status_t {
@@ -236,6 +237,7 @@
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;
@@ -255,6 +257,8 @@
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 {