added functions to translate obs settings to pulse

master
fryshorts 2014-02-18 21:46:39 +01:00
parent f13ae77e00
commit 63d441e182
1 changed files with 184 additions and 54 deletions

View File

@ -17,6 +17,7 @@ along with this program. If not, see <http://www.gnu.org/licenses/>.
#include <util/bmem.h>
#include <util/threading.h>
#include <util/platform.h>
#include <pulse/mainloop.h>
#include <pulse/context.h>
@ -42,16 +43,100 @@ struct pulse_data {
event_t event;
obs_source_t source;
uint32_t frames;
uint32_t samples_per_sec;
uint32_t channels;
enum speaker_layout speakers;
pa_sample_format_t format;
pa_mainloop *mainloop;
pa_context *context;
pa_stream *stream;
pa_proplist *props;
};
/*
* get obs from pulse audio format
*/
static enum audio_format pulse_to_obs_audio_format(
pa_sample_format_t format)
{
switch (format) {
case PA_SAMPLE_U8:
return AUDIO_FORMAT_U8BIT;
case PA_SAMPLE_S16LE:
return AUDIO_FORMAT_16BIT;
case PA_SAMPLE_S24_32LE:
return AUDIO_FORMAT_32BIT;
case PA_SAMPLE_FLOAT32LE:
return AUDIO_FORMAT_FLOAT;
default:
return AUDIO_FORMAT_UNKNOWN;
}
return AUDIO_FORMAT_UNKNOWN;
}
/*
* get the number of frames from bytes and current format
*/
static uint32_t get_frames_from_bytes(struct pulse_data *data, size_t bytes)
{
uint32_t ret = bytes;
ret /= get_audio_bytes_per_channel(
pulse_to_obs_audio_format(data->format));
ret /= get_audio_channels(data->speakers);
return ret;
}
/*
* get the buffer size needed for length msec with current settings
*/
static uint32_t get_buffer_size(struct pulse_data *data, uint32_t length)
{
uint32_t ret = length;
ret *= data->samples_per_sec;
ret *= get_audio_bytes_per_channel(
pulse_to_obs_audio_format(data->format));
ret *= get_audio_channels(data->speakers);
ret /= 1000;
return ret;
}
/*
* Get latency for a pulse audio stream
*/
static int pulse_get_stream_latency(pa_stream *stream, int64_t *latency)
{
int ret;
int sign;
pa_usec_t abs;
ret = pa_stream_get_latency(stream, &abs, &sign);
*latency = (sign) ? -(int64_t) abs : (int64_t) abs;
return ret;
}
/*
* Iterate the mainloop
*
* The custom implementation gives better performance than the function
* provided by pulse audio, maybe due to the timeout set in prepare ?
*/
static void pulse_iterate(struct pulse_data *data)
{
if (pa_mainloop_prepare(data->mainloop, 1000) < 0) {
blog(LOG_ERROR, "Unable to prepare main loop");
return;
}
if (pa_mainloop_poll(data->mainloop) < 0) {
blog(LOG_ERROR, "Unable to poll main loop");
return;
}
if (pa_mainloop_dispatch(data->mainloop) < 0)
blog(LOG_ERROR, "Unable to dispatch main loop");
}
/*
* Create a new pulse audio main loop and connect to the server
*
@ -61,34 +146,35 @@ static int pulse_connect(struct pulse_data *data)
{
data->mainloop = pa_mainloop_new();
if (!data->mainloop) {
blog(LOG_ERROR, "Unable to create main loop");
blog(LOG_ERROR, "pulse-input: Unable to create main loop");
return -1;
}
data->context = pa_context_new(
pa_mainloop_get_api(data->mainloop), "OBS Studio");
data->context = pa_context_new_with_proplist(
pa_mainloop_get_api(data->mainloop), "OBS Studio", data->props);
if (!data->context) {
blog(LOG_ERROR, "Unable to create context");
blog(LOG_ERROR, "pulse-input: Unable to create context");
return -1;
}
int status = pa_context_connect(
data->context, NULL, PA_CONTEXT_NOAUTOSPAWN, NULL);
if (status < 0) {
blog(LOG_ERROR, "Unable to connect! Status: %d", status);
blog(LOG_ERROR, "pulse-input: Unable to connect! Status: %d",
status);
return -1;
}
// wait until connected
for (;;) {
pa_mainloop_iterate(data->mainloop, 0, NULL);
pulse_iterate(data);
pa_context_state_t state = pa_context_get_state(data->context);
if (state == PA_CONTEXT_READY) {
blog(LOG_DEBUG, "Context ready");
blog(LOG_DEBUG, "pulse-input: Context ready");
break;
}
if (!PA_CONTEXT_IS_GOOD(state)) {
blog(LOG_ERROR, "Connection attempt failed");
blog(LOG_ERROR, "pulse-input: Context connect failed");
return -1;
}
}
@ -118,21 +204,26 @@ static void pulse_disconnect(struct pulse_data *data)
static int pulse_connect_stream(struct pulse_data *data)
{
pa_sample_spec spec;
spec.format = PA_SAMPLE_U8;
spec.format = data->format;
spec.rate = data->samples_per_sec;
spec.channels = data->channels;
spec.channels = get_audio_channels(data->speakers);
if (!pa_sample_spec_valid(&spec)) {
blog(LOG_ERROR, "pulse-input: Sample spec is not valid");
return -1;
}
pa_buffer_attr attr;
attr.fragsize = data->frames * spec.channels;
attr.fragsize = get_buffer_size(data, 250);
attr.maxlength = (uint32_t) -1;
attr.minreq = (uint32_t) -1;
attr.prebuf = (uint32_t) -1;
attr.tlength = (uint32_t) -1;
data->stream = pa_stream_new(
data->context, "OBS Audio Input", &spec, NULL);
data->stream = pa_stream_new_with_proplist(data->context,
obs_source_getname(data->source), &spec, NULL, data->props);
if (!data->stream) {
blog(LOG_ERROR, "Unable to create stream");
blog(LOG_ERROR, "pulse-input: Unable to create stream");
return -1;
}
pa_stream_flags_t flags =
@ -140,19 +231,19 @@ static int pulse_connect_stream(struct pulse_data *data)
| PA_STREAM_AUTO_TIMING_UPDATE
| PA_STREAM_ADJUST_LATENCY;
if (pa_stream_connect_record(data->stream, NULL, &attr, flags) < 0) {
blog(LOG_ERROR, "Unable to connect to stream");
blog(LOG_ERROR, "pulse-input: Unable to connect to stream");
return -1;
}
for (;;) {
pa_mainloop_iterate(data->mainloop, 0, NULL);
pulse_iterate(data);
pa_stream_state_t state = pa_stream_get_state(data->stream);
if (state == PA_STREAM_READY) {
blog(LOG_DEBUG, "Stream ready");
blog(LOG_DEBUG, "pulse-input: Stream ready");
break;
}
if (!PA_STREAM_IS_GOOD(state)) {
blog(LOG_ERROR, "Stream connection failed");
blog(LOG_ERROR, "pulse-input: Stream connect failed");
return -1;
}
}
@ -171,6 +262,38 @@ static void pulse_diconnect_stream(struct pulse_data *data)
}
}
/*
* Loop to skip the first few samples of a stream
*/
static int pulse_skip(struct pulse_data *data)
{
uint64_t skip = 1;
const void *frames;
size_t bytes;
uint64_t pa_time;
while (event_try(&data->event) == EAGAIN) {
pulse_iterate(data);
pa_stream_peek(data->stream, &frames, &bytes);
if (!bytes)
continue;
if (!frames || pa_stream_get_time(data->stream, &pa_time) < 0) {
pa_stream_drop(data->stream);
continue;
}
if (skip == 1 && pa_time)
skip = pa_time;
if (skip + pulse_start_delay < pa_time)
return 0;
pa_stream_drop(data->stream);
}
return -1;
}
/*
* Worker thread to get audio data
*
@ -185,55 +308,51 @@ static void *pulse_thread(void *vptr)
if (pulse_connect_stream(data) < 0)
return NULL;
uint64_t skip = 1;
if (pulse_skip(data) < 0)
return NULL;
blog(LOG_DEBUG, "pulse-input: Start recording");
const void *frames;
size_t bytes;
uint64_t pa_time;
int64_t pa_latency;
struct source_audio out;
out.speakers = data->speakers;
out.samples_per_sec = data->samples_per_sec;
out.format = pulse_to_obs_audio_format(data->format);
while (event_try(&data->event) == EAGAIN) {
pa_mainloop_iterate(data->mainloop, 0, NULL);
pulse_iterate(data);
const void *frames;
size_t bytes;
pa_stream_peek(data->stream, &frames, &bytes);
// check if we got data
if (!bytes) {
if (!bytes)
continue;
if (!frames) {
blog(LOG_DEBUG,
"pulse-input: Got audio hole of %u bytes",
(unsigned int) bytes);
pa_stream_drop(data->stream);
continue;
}
uint64_t pa_time;
if (pa_stream_get_time(data->stream, &pa_time) < 0)
continue;
// skip the first frames received
if (skip) {
// start delay when we receive the first bytes
if (skip == 1 && bytes)
skip = pa_time;
if (skip + pulse_start_delay < pa_time)
skip = 0;
if (pa_stream_get_time(data->stream, &pa_time) < 0) {
blog(LOG_ERROR,
"pulse-input: Failed to get timing info !");
pa_stream_drop(data->stream);
continue;
}
// get stream latency
pa_usec_t l_abs;
int l_sign;
pa_stream_get_latency(data->stream, &l_abs, &l_sign);
int64_t latency = (l_sign) ? -(int64_t) l_abs : (int64_t) l_abs;
pulse_get_stream_latency(data->stream, &pa_latency);
// push structure
struct source_audio out;
out.data[0] = (uint8_t *) frames;
out.frames = bytes / data->channels;
out.speakers = data->speakers;
out.samples_per_sec = data->samples_per_sec;
out.timestamp = (pa_time - latency) * 1000;
out.format = AUDIO_FORMAT_U8BIT;
// send data to obs
out.frames = get_frames_from_bytes(data, bytes);
out.timestamp = (pa_time - pa_latency) * 1000;
obs_source_output_audio(data->source, &out);
// clear pulse audio buffer
pa_stream_drop(data->stream);
}
@ -270,6 +389,8 @@ static void pulse_destroy(void *vptr)
event_destroy(&data->event);
pa_proplist_free(data->props);
bfree(data);
}
@ -284,10 +405,19 @@ static void *pulse_create(obs_data_t settings, obs_source_t source)
memset(data, 0, sizeof(struct pulse_data));
data->source = source;
data->frames = 480;
data->samples_per_sec = 48000;
data->samples_per_sec = 44100;
data->speakers = SPEAKERS_STEREO;
data->channels = (data->speakers == SPEAKERS_STEREO) ? 2 : 1;
data->format = PA_SAMPLE_S16LE;
/* TODO: use obs-studio icon */
data->props = pa_proplist_new();
pa_proplist_sets(data->props, PA_PROP_APPLICATION_NAME,
"OBS Studio");
pa_proplist_sets(data->props, PA_PROP_APPLICATION_ICON_NAME,
"application-exit");
pa_proplist_sets(data->props, PA_PROP_MEDIA_ROLE,
"production");
if (event_init(&data->event, EVENT_TYPE_MANUAL) != 0)
goto fail;