Merge "Improve stub hal read and write timing" into nyc-mr1-dev

This commit is contained in:
Andy Hung
2016-07-15 05:06:15 +00:00
committed by Android (Google) Code Review

View File

@@ -35,10 +35,12 @@ struct stub_audio_device {
struct stub_stream_out {
struct audio_stream_out stream;
int64_t last_write_time_us;
};
struct stub_stream_in {
struct audio_stream_in stream;
int64_t last_read_time_us;
};
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
@@ -79,7 +81,7 @@ static int out_set_format(struct audio_stream *stream, audio_format_t format)
static int out_standby(struct audio_stream *stream)
{
ALOGV("out_standby");
// out->last_write_time_us = 0; unnecessary as a stale write time has same effect
return 0;
}
@@ -118,9 +120,31 @@ static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
size_t bytes)
{
ALOGV("out_write: bytes: %d", bytes);
/* XXX: fake timing for audio output */
usleep((int64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
out_get_sample_rate(&stream->common));
struct stub_stream_out *out = (struct stub_stream_out *)stream;
struct timespec t = { .tv_sec = 0, .tv_nsec = 0 };
clock_gettime(CLOCK_MONOTONIC, &t);
const int64_t now = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000;
const int64_t elapsed_time_since_last_write = now - out->last_write_time_us;
int64_t sleep_time = bytes * 1000000LL / audio_stream_out_frame_size(stream) /
out_get_sample_rate(&stream->common) - elapsed_time_since_last_write;
if (sleep_time > 0) {
usleep(sleep_time);
} else {
// we don't sleep when we exit standby (this is typical for a real alsa buffer).
sleep_time = 0;
}
out->last_write_time_us = now + sleep_time;
// last_write_time_us is an approximation of when the (simulated) alsa
// buffer is believed completely full. The usleep above waits for more space
// in the buffer, but by the end of the sleep the buffer is considered
// topped-off.
//
// On the subsequent out_write(), we measure the elapsed time spent in
// the mixer. This is subtracted from the sleep estimate based on frames,
// thereby accounting for drain in the alsa buffer during mixing.
// This is a crude approximation; we don't handle underruns precisely.
return bytes;
}
@@ -189,6 +213,8 @@ static int in_set_format(struct audio_stream *stream, audio_format_t format)
static int in_standby(struct audio_stream *stream)
{
struct stub_stream_in *in = (struct stub_stream_in *)stream;
in->last_read_time_us = 0;
return 0;
}
@@ -217,9 +243,31 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
size_t bytes)
{
ALOGV("in_read: bytes %d", bytes);
/* XXX: fake timing for audio input */
usleep((int64_t)bytes * 1000000 / audio_stream_in_frame_size(stream) /
in_get_sample_rate(&stream->common));
struct stub_stream_in *in = (struct stub_stream_in *)stream;
struct timespec t = { .tv_sec = 0, .tv_nsec = 0 };
clock_gettime(CLOCK_MONOTONIC, &t);
const int64_t now = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000;
// we do a full sleep when exiting standby.
const bool standby = in->last_read_time_us == 0;
const int64_t elapsed_time_since_last_read = standby ?
0 : now - in->last_read_time_us;
int64_t sleep_time = bytes * 1000000LL / audio_stream_in_frame_size(stream) /
in_get_sample_rate(&stream->common) - elapsed_time_since_last_read;
if (sleep_time > 0) {
usleep(sleep_time);
} else {
sleep_time = 0;
}
in->last_read_time_us = now + sleep_time;
// last_read_time_us is an approximation of when the (simulated) alsa
// buffer is drained by the read, and is empty.
//
// On the subsequent in_read(), we measure the elapsed time spent in
// the recording thread. This is subtracted from the sleep estimate based on frames,
// thereby accounting for fill in the alsa buffer during the interim.
memset(buffer, 0, bytes);
return bytes;
}