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audio-src: test stream timings

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This commit is contained in:
George Kiagiadakis 2023-08-21 17:47:32 +03:00 • committed by Wim Taymans
parent ee05daea53
commit 150c1cc05d
1 changed files with 41 additions and 5 deletions
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46
src/examples/audio-src.c
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@ -19,6 +19,7 @@
#define M_PI_M2 ( M_PI + M_PI )
#define DEFAULT_QUANTUM 1764
#define DEFAULT_RATE 44100
#define DEFAULT_CHANNELS 2
#define DEFAULT_VOLUME 0.7
@ -28,6 +29,7 @@ struct data {
struct pw_stream *stream;
double accumulator;
uint64_t outputSample;
};
static void fill_f32(struct data *d, void *dest, int n_frames)
@ -63,17 +65,41 @@ static void on_process(void *userdata)
int n_frames, stride;
uint8_t *p;
struct pw_time streamTime;
uint64_t driverSample;
int64_t queued;
pw_stream_get_time_n(data->stream, &streamTime, sizeof(streamTime));
if ((b = pw_stream_dequeue_buffer(data->stream)) == NULL) {
pw_log_warn("out of buffers: %m");
return;
}
/* calculate the current sample in the driver's timeline */
driverSample = streamTime.ticks * DEFAULT_RATE * streamTime.rate.num / streamTime.rate.denom;
/* find how many samples we have queued but have not yet appeared in the driver's timeline;
these are queued in the audioadapter but pw_stream doesn't report them accurately */
queued = data->outputSample - driverSample;
if (queued < 0) {
/* XRun, resync our timeline */
pw_log_info("XRun! resync, data->outputSample += %"PRIi64, -queued);
data->outputSample += -queued;
queued = 0;
}
pw_log_info("process, b.req %"PRIu64", s.ticks %"PRIu64 ", sample(our) %"PRIu64", sample(dr) %"PRIu64",\n"
" s.buf %"PRIu64 ", s.queued %"PRIu64 ", queued (real) %"PRIi64 ", s.delay %"PRIu64,
b->requested, streamTime.ticks, data->outputSample, driverSample,
streamTime.buffered, streamTime.queued, queued, streamTime.delay);
buf = b->buffer;
if ((p = buf->datas[0].data) == NULL)
return;
stride = sizeof(float) * DEFAULT_CHANNELS;
n_frames = SPA_MIN(b->requested, buf->datas[0].maxsize / stride);
n_frames = SPA_MIN((uint64_t) DEFAULT_QUANTUM, buf->datas[0].maxsize / stride);
fill_f32(data, p, n_frames);
@ -81,6 +107,9 @@ static void on_process(void *userdata)
buf->datas[0].chunk->stride = stride;
buf->datas[0].chunk->size = n_frames * stride;
b->size = n_frames;
data->outputSample += n_frames;
pw_stream_queue_buffer(data->stream, b);
}
@ -98,7 +127,7 @@ static void do_quit(void *userdata, int signal_number)
int main(int argc, char *argv[])
{
struct data data = { 0, };
const struct spa_pod *params[1];
const struct spa_pod *params[2];
uint8_t buffer[1024];
struct pw_properties *props;
struct spa_pod_builder b = SPA_POD_BUILDER_INIT(buffer, sizeof(buffer));
@ -126,6 +155,7 @@ int main(int argc, char *argv[])
props = pw_properties_new(PW_KEY_MEDIA_TYPE, "Audio",
PW_KEY_MEDIA_CATEGORY, "Playback",
PW_KEY_MEDIA_ROLE, "Music",
PW_KEY_NODE_LATENCY, SPA_STRINGIFY (DEFAULT_QUANTUM) "/" SPA_STRINGIFY (DEFAULT_RATE),
NULL);
if (argc > 1)
/* Set stream target if given on command line */
@ -144,6 +174,13 @@ int main(int argc, char *argv[])
.format = SPA_AUDIO_FORMAT_F32,
.channels = DEFAULT_CHANNELS,
.rate = DEFAULT_RATE ));
params[1] = spa_pod_builder_add_object(&b,
SPA_TYPE_OBJECT_ParamBuffers, SPA_PARAM_Buffers,
SPA_PARAM_BUFFERS_buffers, SPA_POD_Int(4),
SPA_PARAM_BUFFERS_blocks, SPA_POD_Int(1),
SPA_PARAM_BUFFERS_size, SPA_POD_Int(DEFAULT_QUANTUM * sizeof(float) * DEFAULT_CHANNELS),
SPA_PARAM_BUFFERS_stride, SPA_POD_Int(sizeof(float) * DEFAULT_CHANNELS),
SPA_PARAM_BUFFERS_dataType, SPA_POD_CHOICE_FLAGS_Int((1<<SPA_DATA_MemPtr)));
/* Now connect this stream. We ask that our process function is
* called in a realtime thread. */
@ -151,9 +188,8 @@ int main(int argc, char *argv[])
PW_DIRECTION_OUTPUT,
PW_ID_ANY,
PW_STREAM_FLAG_AUTOCONNECT |
PW_STREAM_FLAG_MAP_BUFFERS |
PW_STREAM_FLAG_RT_PROCESS,
params, 1);
PW_STREAM_FLAG_MAP_BUFFERS,
params, 2);
/* and wait while we let things run */
pw_main_loop_run(data.loop);