[pulseaudio-commits] [SCM] PulseAudio Sound Server branch, stable-queue, updated. v0.9.22-29-g2ee4ec5
Colin Guthrie
gitmailer-noreply at 0pointer.de
Fri Feb 25 03:18:38 PST 2011
This is an automated email from the git hooks/post-receive script. It was
generated because of a push to the "PulseAudio Sound Server" repository.
The stable-queue branch has been updated
from 67d188894c7e74fe78e14db566a11cfe2d500114 (commit)
- Log -----------------------------------------------------------------
2ee4ec5 module-rtp-recv: Remove smoother from write index
2bfc032 module-rtp-recv: Average the estimated real sample rate
4620039 module-rtp-recv: Use new algorithm for adjusting sample rate
90c5520 Limit rate adjustments to small, inaudible jumps
09770e5 module-loopback: Add adjust_time to valid args
-----------------------------------------------------------------------
Summary of changes:
src/modules/module-combine.c | 26 +++++---
src/modules/module-loopback.c | 19 +++++-
src/modules/rtp/module-rtp-recv.c | 114 ++++++++++++++++++++++---------------
3 files changed, 100 insertions(+), 59 deletions(-)
-----------------------------------------------------------------------
commit 09770e577991d49cf826bdf80b0f9559f1e67820
Author: Maarten Bosmans <mkbosmans at gmail.com>
Date: Sun Jan 16 01:42:20 2011 +0100
module-loopback: Add adjust_time to valid args
diff --git a/src/modules/module-loopback.c b/src/modules/module-loopback.c
index 265a469..ca06314 100644
--- a/src/modules/module-loopback.c
+++ b/src/modules/module-loopback.c
@@ -102,6 +102,7 @@ struct userdata {
static const char* const valid_modargs[] = {
"source",
"sink",
+ "adjust_time",
"latency_msec",
"format",
"rate",
commit 90c5520e03bbccf6c1d9f87221d3742cc70b53ed
Author: Maarten Bosmans <mkbosmans at gmail.com>
Date: Fri Jan 7 01:25:55 2011 +0100
Limit rate adjustments to small, inaudible jumps
The same logic is applied to the sample rate adjustments in module-rtp-recv,
module-loopback and module-combine:
- Each time an adjustment is made, the new rate can differ at most 2‰ from the
old rate. Such a step is equal to 3.5 cents (a cent is 1/100th of a
semitone) and as 5 cents is generally considered the smallest observable
difference in pitch, this results in inaudible adjustments.
- The sample rate of the stream can only differ from the rate of the
corresponding sink by 25%. As these adjustments are meant to account for
very small clock drifts, any large deviation from the base rate suggests
something is seriously wrong.
- If the calculated rate is within 20Hz of the base rate, set it to the base
rate. This saves CPU because no resampling is necessary.
diff --git a/src/modules/module-combine.c b/src/modules/module-combine.c
index bcea229..3104ed6 100644
--- a/src/modules/module-combine.c
+++ b/src/modules/module-combine.c
@@ -217,23 +217,29 @@ static void adjust_rates(struct userdata *u) {
base_rate = u->sink->sample_spec.rate;
PA_IDXSET_FOREACH(o, u->outputs, idx) {
- uint32_t r = base_rate;
+ uint32_t new_rate = base_rate;
+ uint32_t current_rate = o->sink_input->sample_spec.rate;
if (!o->sink_input || !PA_SINK_IS_OPENED(pa_sink_get_state(o->sink)))
continue;
- if (o->total_latency < target_latency)
- r -= (uint32_t) ((((double) (target_latency - o->total_latency))/(double)u->adjust_time)*(double)r);
- else if (o->total_latency > target_latency)
- r += (uint32_t) ((((double) (o->total_latency - target_latency))/(double)u->adjust_time)*(double)r);
+ if (o->total_latency != target_latency)
+ new_rate += (uint32_t) (((double) o->total_latency - (double) target_latency) / (double) u->adjust_time * (double) new_rate);
- if (r < (uint32_t) (base_rate*0.9) || r > (uint32_t) (base_rate*1.1)) {
- pa_log_warn("[%s] sample rates too different, not adjusting (%u vs. %u).", o->sink_input->sink->name, base_rate, r);
- pa_sink_input_set_rate(o->sink_input, base_rate);
+ if (new_rate < (uint32_t) (base_rate*0.8) || new_rate > (uint32_t) (base_rate*1.25)) {
+ pa_log_warn("[%s] sample rates too different, not adjusting (%u vs. %u).", o->sink_input->sink->name, base_rate, new_rate);
+ new_rate = base_rate;
} else {
- pa_log_info("[%s] new rate is %u Hz; ratio is %0.3f; latency is %0.0f usec.", o->sink_input->sink->name, r, (double) r / base_rate, (float) o->total_latency);
- pa_sink_input_set_rate(o->sink_input, r);
+ if (base_rate < new_rate + 20 && new_rate < base_rate + 20)
+ new_rate = base_rate;
+ /* Do the adjustment in small steps; 2‰ can be considered inaudible */
+ if (new_rate < (uint32_t) (current_rate*0.998) || new_rate > (uint32_t) (current_rate*1.002)) {
+ pa_log_info("[%s] new rate of %u Hz not within 2‰ of %u Hz, forcing smaller adjustment", o->sink_input->sink->name, new_rate, current_rate);
+ new_rate = PA_CLAMP(new_rate, (uint32_t) (current_rate*0.998), (uint32_t) (current_rate*1.002));
+ }
+ pa_log_info("[%s] new rate is %u Hz; ratio is %0.3f; latency is %0.2f msec.", o->sink_input->sink->name, new_rate, (double) new_rate / base_rate, (double) o->total_latency / PA_USEC_PER_MSEC);
}
+ pa_sink_input_set_rate(o->sink_input, new_rate);
}
pa_asyncmsgq_send(u->sink->asyncmsgq, PA_MSGOBJECT(u->sink), SINK_MESSAGE_UPDATE_LATENCY, NULL, (int64_t) avg_total_latency, NULL);
diff --git a/src/modules/module-loopback.c b/src/modules/module-loopback.c
index ca06314..e0277c1 100644
--- a/src/modules/module-loopback.c
+++ b/src/modules/module-loopback.c
@@ -167,13 +167,13 @@ static void adjust_rates(struct userdata *u) {
buffer_latency = pa_bytes_to_usec(buffer, &u->sink_input->sample_spec);
- pa_log_info("Loopback overall latency is %0.2f ms + %0.2f ms + %0.2f ms = %0.2f ms",
+ pa_log_debug("Loopback overall latency is %0.2f ms + %0.2f ms + %0.2f ms = %0.2f ms",
(double) u->latency_snapshot.sink_latency / PA_USEC_PER_MSEC,
(double) buffer_latency / PA_USEC_PER_MSEC,
(double) u->latency_snapshot.source_latency / PA_USEC_PER_MSEC,
((double) u->latency_snapshot.sink_latency + buffer_latency + u->latency_snapshot.source_latency) / PA_USEC_PER_MSEC);
- pa_log_info("Should buffer %zu bytes, buffered at minimum %zu bytes",
+ pa_log_debug("Should buffer %zu bytes, buffered at minimum %zu bytes",
u->latency_snapshot.max_request*2,
u->latency_snapshot.min_memblockq_length);
@@ -186,9 +186,21 @@ static void adjust_rates(struct userdata *u) {
else
new_rate = base_rate + (((u->latency_snapshot.min_memblockq_length - u->latency_snapshot.max_request*2) / fs) *PA_USEC_PER_SEC)/u->adjust_time;
- pa_log_info("Old rate %lu Hz, new rate %lu Hz", (unsigned long) old_rate, (unsigned long) new_rate);
+ if (new_rate < (uint32_t) (base_rate*0.8) || new_rate > (uint32_t) (base_rate*1.25)) {
+ pa_log_warn("Sample rates too different, not adjusting (%u vs. %u).", base_rate, new_rate);
+ new_rate = base_rate;
+ } else {
+ if (base_rate < new_rate + 20 && new_rate < base_rate + 20)
+ new_rate = base_rate;
+ /* Do the adjustment in small steps; 2‰ can be considered inaudible */
+ if (new_rate < (uint32_t) (old_rate*0.998) || new_rate > (uint32_t) (old_rate*1.002)) {
+ pa_log_info("New rate of %u Hz not within 2‰ of %u Hz, forcing smaller adjustment", new_rate, old_rate);
+ new_rate = PA_CLAMP(new_rate, (uint32_t) (old_rate*0.998), (uint32_t) (old_rate*1.002));
+ }
+ }
pa_sink_input_set_rate(u->sink_input, new_rate);
+ pa_log_debug("[%s] Updated sampling rate to %lu Hz.", u->sink_input->sink->name, (unsigned long) new_rate);
pa_core_rttime_restart(u->core, u->time_event, pa_rtclock_now() + u->adjust_time);
}
diff --git a/src/modules/rtp/module-rtp-recv.c b/src/modules/rtp/module-rtp-recv.c
index 1a05f57..491be4d 100644
--- a/src/modules/rtp/module-rtp-recv.c
+++ b/src/modules/rtp/module-rtp-recv.c
@@ -288,6 +288,9 @@ static int rtpoll_work_cb(pa_rtpoll_item *i) {
if (s->last_rate_update + RATE_UPDATE_INTERVAL < pa_timeval_load(&now)) {
pa_usec_t wi, ri, render_delay, sink_delay = 0, latency, fix;
unsigned fix_samples;
+ uint32_t base_rate = s->sink_input->sink->sample_spec.rate;
+ uint32_t current_rate = s->sink_input->sample_spec.rate;
+ uint32_t new_rate;
pa_log_debug("Updating sample rate");
@@ -309,7 +312,7 @@ static int rtpoll_work_cb(pa_rtpoll_item *i) {
else
latency = wi - ri;
- pa_log_debug("Write index deviates by %0.2f ms, expected %0.2f ms", (double) latency/PA_USEC_PER_MSEC, (double) s->intended_latency/PA_USEC_PER_MSEC);
+ pa_log_debug("Write index deviates by %0.2f ms, expected %0.2f ms", (double) latency/PA_USEC_PER_MSEC, (double) s->intended_latency/PA_USEC_PER_MSEC);
/* Calculate deviation */
if (latency < s->intended_latency)
@@ -320,19 +323,24 @@ static int rtpoll_work_cb(pa_rtpoll_item *i) {
/* How many samples is this per second? */
fix_samples = (unsigned) (fix * (pa_usec_t) s->sink_input->thread_info.sample_spec.rate / (pa_usec_t) RATE_UPDATE_INTERVAL);
- /* Check if deviation is in bounds */
- if (fix_samples > s->sink_input->sample_spec.rate*.50)
- pa_log_debug("Hmmm, rate fix is too large (%lu Hz), not applying.", (unsigned long) fix_samples);
- else {
- /* Fix up rate */
- if (latency < s->intended_latency)
- s->sink_input->sample_spec.rate -= fix_samples;
- else
- s->sink_input->sample_spec.rate += fix_samples;
-
- if (s->sink_input->sample_spec.rate > PA_RATE_MAX)
- s->sink_input->sample_spec.rate = PA_RATE_MAX;
+ if (latency < s->intended_latency)
+ new_rate = current_rate - fix_samples;
+ else
+ new_rate = current_rate + fix_samples;
+
+ if (new_rate < (uint32_t) (base_rate*0.8) || new_rate > (uint32_t) (base_rate*1.25)) {
+ pa_log_warn("Sample rates too different, not adjusting (%u vs. %u).", base_rate, new_rate);
+ new_rate = base_rate;
+ } else {
+ if (base_rate < new_rate + 20 && new_rate < base_rate + 20)
+ new_rate = base_rate;
+ /* Do the adjustment in small steps; 2‰ can be considered inaudible */
+ if (new_rate < (uint32_t) (current_rate*0.998) || new_rate > (uint32_t) (current_rate*1.002)) {
+ pa_log_info("New rate of %u Hz not within 2‰ of %u Hz, forcing smaller adjustment", new_rate, current_rate);
+ new_rate = PA_CLAMP(new_rate, (uint32_t) (current_rate*0.998), (uint32_t) (current_rate*1.002));
+ }
}
+ s->sink_input->sample_spec.rate = new_rate;
pa_assert(pa_sample_spec_valid(&s->sink_input->sample_spec));
commit 46200391f3a2f02b951cee40d7b6ddd2e7b9258a
Author: Maarten Bosmans <mkbosmans at gmail.com>
Date: Wed Jan 12 07:24:58 2011 +0100
module-rtp-recv: Use new algorithm for adjusting sample rate
diff --git a/src/modules/rtp/module-rtp-recv.c b/src/modules/rtp/module-rtp-recv.c
index 491be4d..20d7044 100644
--- a/src/modules/rtp/module-rtp-recv.c
+++ b/src/modules/rtp/module-rtp-recv.c
@@ -109,6 +109,7 @@ struct session {
pa_usec_t sink_latency;
pa_usec_t last_rate_update;
+ pa_usec_t last_latency;
};
struct userdata {
@@ -286,11 +287,11 @@ static int rtpoll_work_cb(pa_rtpoll_item *i) {
pa_atomic_store(&s->timestamp, (int) now.tv_sec);
if (s->last_rate_update + RATE_UPDATE_INTERVAL < pa_timeval_load(&now)) {
- pa_usec_t wi, ri, render_delay, sink_delay = 0, latency, fix;
- unsigned fix_samples;
+ pa_usec_t wi, ri, render_delay, sink_delay = 0, latency;
uint32_t base_rate = s->sink_input->sink->sample_spec.rate;
uint32_t current_rate = s->sink_input->sample_spec.rate;
uint32_t new_rate;
+ double estimated_rate;
pa_log_debug("Updating sample rate");
@@ -314,19 +315,31 @@ static int rtpoll_work_cb(pa_rtpoll_item *i) {
pa_log_debug("Write index deviates by %0.2f ms, expected %0.2f ms", (double) latency/PA_USEC_PER_MSEC, (double) s->intended_latency/PA_USEC_PER_MSEC);
- /* Calculate deviation */
- if (latency < s->intended_latency)
- fix = s->intended_latency - latency;
- else
- fix = latency - s->intended_latency;
-
- /* How many samples is this per second? */
- fix_samples = (unsigned) (fix * (pa_usec_t) s->sink_input->thread_info.sample_spec.rate / (pa_usec_t) RATE_UPDATE_INTERVAL);
-
- if (latency < s->intended_latency)
- new_rate = current_rate - fix_samples;
- else
- new_rate = current_rate + fix_samples;
+ /* The buffer is filling with some unknown rate RÌ‚ samples/second. If the rate of reading in
+ * the last T seconds was Râ¿, then the increase in buffer latency ΔLâ¿ = Lâ¿ - Lâ¿â»â± in that
+ * same period is ΔLâ¿ = (TRÌ‚ - TRâ¿) / RÌ‚, giving the estimated target rate
+ * T
+ * R̂ = ─────────────── R⿠. (1)
+ * T - (Lâ¿ - Lâ¿â»â±)
+ *
+ * Setting the sample rate to RÌ‚ results in the latency being constant (if the estimate of RÌ‚
+ * is correct). But there is also the requirement to keep the buffer at a predefined target
+ * latency LÌ‚. So instead of setting Râ¿âºâ± to RÌ‚ immediately, the strategy will be to reduce R
+ * from Râ¿âºâ± to RÌ‚ in a steps of T seconds, where Râ¿âºâ± is chosen such that in the total time
+ * aT the latency is reduced from Lâ¿ to LÌ‚. This strategy translates to the requirements
+ * â‚ RÌ‚ - Râ¿âºÊ² a-j+1 j-1
+ * Σ T ────────── = LÌ‚ - Lâ¿ with Râ¿âºÊ² = ───── Râ¿âºâ± + ───── RÌ‚ .
+ * ʲâ¼â± RÌ‚ a a
+ * Solving for Râ¿âºâ± gives
+ * T - ²∕â‚â‚Šâ‚(LÌ‚ - Lâ¿)
+ * Râ¿âºâ± = ───────────────── RÌ‚ . (2)
+ * T
+ * Together Equations (1) and (2) specify the algorithm used below, where a = 7 is used.
+ */
+ estimated_rate = (double) current_rate * (double) RATE_UPDATE_INTERVAL / (double) (RATE_UPDATE_INTERVAL + s->last_latency - latency);
+ pa_log_debug("Estimated target rate: %.0f Hz", estimated_rate);
+ new_rate = (uint32_t) ((double) (RATE_UPDATE_INTERVAL + latency/4 - s->intended_latency/4) / (double) RATE_UPDATE_INTERVAL * estimated_rate);
+ s->last_latency = latency;
if (new_rate < (uint32_t) (base_rate*0.8) || new_rate > (uint32_t) (base_rate*1.25)) {
pa_log_warn("Sample rates too different, not adjusting (%u vs. %u).", base_rate, new_rate);
@@ -488,6 +501,7 @@ static struct session *session_new(struct userdata *u, const pa_sdp_info *sdp_in
pa_timeval_load(&now),
TRUE);
s->last_rate_update = pa_timeval_load(&now);
+ s->last_latency = LATENCY_USEC;
pa_atomic_store(&s->timestamp, (int) now.tv_sec);
if ((fd = mcast_socket((const struct sockaddr*) &sdp_info->sa, sdp_info->salen)) < 0)
commit 2bfc0322c975f573e6aea136db1cf99c6dcb11fd
Author: Maarten Bosmans <mkbosmans at gmail.com>
Date: Sun Jan 16 01:27:29 2011 +0100
module-rtp-recv: Average the estimated real sample rate
diff --git a/src/modules/rtp/module-rtp-recv.c b/src/modules/rtp/module-rtp-recv.c
index 20d7044..baf5b50 100644
--- a/src/modules/rtp/module-rtp-recv.c
+++ b/src/modules/rtp/module-rtp-recv.c
@@ -110,6 +110,8 @@ struct session {
pa_usec_t last_rate_update;
pa_usec_t last_latency;
+ double estimated_rate;
+ double avg_estimated_rate;
};
struct userdata {
@@ -291,7 +293,7 @@ static int rtpoll_work_cb(pa_rtpoll_item *i) {
uint32_t base_rate = s->sink_input->sink->sample_spec.rate;
uint32_t current_rate = s->sink_input->sample_spec.rate;
uint32_t new_rate;
- double estimated_rate;
+ double estimated_rate, alpha = 0.02;
pa_log_debug("Updating sample rate");
@@ -334,11 +336,25 @@ static int rtpoll_work_cb(pa_rtpoll_item *i) {
* T - ²∕â‚â‚Šâ‚(LÌ‚ - Lâ¿)
* Râ¿âºâ± = ───────────────── RÌ‚ . (2)
* T
- * Together Equations (1) and (2) specify the algorithm used below, where a = 7 is used.
+ * In the code below a = 7 is used.
+ *
+ * Equation (1) is not directly used in (2), but instead an exponentially weighted average
+ * of the estimated rate RÌ‚ is used. This average RÌ… is defined as
+ * RÌ…â¿ = α RÌ‚â¿ + (1-α) RÌ…â¿â»â± .
+ * Because it is difficult to find a fixed value for the coefficient α such that the
+ * averaging is without significant lag but oscillations are filtered out, a heuristic is
+ * used. When the successive estimates R̂⿠do not change much then α→1, but when there is a
+ * sudden spike in the estimated rate α→0, such that the deviation is given little weight.
*/
estimated_rate = (double) current_rate * (double) RATE_UPDATE_INTERVAL / (double) (RATE_UPDATE_INTERVAL + s->last_latency - latency);
- pa_log_debug("Estimated target rate: %.0f Hz", estimated_rate);
- new_rate = (uint32_t) ((double) (RATE_UPDATE_INTERVAL + latency/4 - s->intended_latency/4) / (double) RATE_UPDATE_INTERVAL * estimated_rate);
+ if (fabs(s->estimated_rate - s->avg_estimated_rate) > 1) {
+ double ratio = (estimated_rate + s->estimated_rate - 2*s->avg_estimated_rate) / (s->estimated_rate - s->avg_estimated_rate);
+ alpha = PA_CLAMP(2 * (ratio + fabs(ratio)) / (4 + ratio*ratio), 0.02, 0.8);
+ }
+ s->avg_estimated_rate = alpha * estimated_rate + (1-alpha) * s->avg_estimated_rate;
+ s->estimated_rate = estimated_rate;
+ pa_log_debug("Estimated target rate: %.0f Hz, using average of %.0f Hz (α=%.3f)", estimated_rate, s->avg_estimated_rate, alpha);
+ new_rate = (uint32_t) ((double) (RATE_UPDATE_INTERVAL + latency/4 - s->intended_latency/4) / (double) RATE_UPDATE_INTERVAL * s->avg_estimated_rate);
s->last_latency = latency;
if (new_rate < (uint32_t) (base_rate*0.8) || new_rate > (uint32_t) (base_rate*1.25)) {
@@ -502,6 +518,8 @@ static struct session *session_new(struct userdata *u, const pa_sdp_info *sdp_in
TRUE);
s->last_rate_update = pa_timeval_load(&now);
s->last_latency = LATENCY_USEC;
+ s->estimated_rate = (double) sink->sample_spec.rate;
+ s->avg_estimated_rate = (double) sink->sample_spec.rate;
pa_atomic_store(&s->timestamp, (int) now.tv_sec);
if ((fd = mcast_socket((const struct sockaddr*) &sdp_info->sa, sdp_info->salen)) < 0)
commit 2ee4ec507cd4105fcddeaf706749524ddeb1ebf5
Author: Maarten Bosmans <mkbosmans at gmail.com>
Date: Wed Jan 12 07:31:26 2011 +0100
module-rtp-recv: Remove smoother from write index
It isn't necessary anymore with the new algorithm. The slow adjust of the
smoother was even detrimental to the accuracy of the rate estimate.
diff --git a/src/modules/rtp/module-rtp-recv.c b/src/modules/rtp/module-rtp-recv.c
index baf5b50..d214cbc 100644
--- a/src/modules/rtp/module-rtp-recv.c
+++ b/src/modules/rtp/module-rtp-recv.c
@@ -52,7 +52,6 @@
#include <pulsecore/macro.h>
#include <pulsecore/atomic.h>
#include <pulsecore/atomic.h>
-#include <pulsecore/time-smoother.h>
#include <pulsecore/socket-util.h>
#include <pulsecore/once.h>
@@ -104,7 +103,6 @@ struct session {
pa_atomic_t timestamp;
- pa_smoother *smoother;
pa_usec_t intended_latency;
pa_usec_t sink_latency;
@@ -197,10 +195,9 @@ static void sink_input_suspend_within_thread(pa_sink_input* i, pa_bool_t b) {
pa_sink_input_assert_ref(i);
pa_assert_se(s = i->userdata);
- if (b) {
- pa_smoother_pause(s->smoother, pa_rtclock_now());
+ if (b)
pa_memblockq_flush_read(s->memblockq);
- } else
+ else
s->first_packet = FALSE;
}
@@ -269,11 +266,6 @@ static int rtpoll_work_cb(pa_rtpoll_item *i) {
} else
pa_rtclock_from_wallclock(&now);
- pa_smoother_put(s->smoother, pa_timeval_load(&now), pa_bytes_to_usec((uint64_t) pa_memblockq_get_write_index(s->memblockq), &s->sink_input->sample_spec));
-
- /* Tell the smoother that we are rolling now, in case it is still paused */
- pa_smoother_resume(s->smoother, pa_timeval_load(&now), TRUE);
-
if (pa_memblockq_push(s->memblockq, &chunk) < 0) {
pa_log_warn("Queue overrun");
pa_memblockq_seek(s->memblockq, (int64_t) chunk.length, PA_SEEK_RELATIVE, TRUE);
@@ -297,7 +289,7 @@ static int rtpoll_work_cb(pa_rtpoll_item *i) {
pa_log_debug("Updating sample rate");
- wi = pa_smoother_get(s->smoother, pa_timeval_load(&now));
+ wi = pa_bytes_to_usec((uint64_t) pa_memblockq_get_write_index(s->memblockq), &s->sink_input->sample_spec);
ri = pa_bytes_to_usec((uint64_t) pa_memblockq_get_read_index(s->memblockq), &s->sink_input->sample_spec);
pa_log_debug("wi=%lu ri=%lu", (unsigned long) wi, (unsigned long) ri);
@@ -508,14 +500,6 @@ static struct session *session_new(struct userdata *u, const pa_sdp_info *sdp_in
s->sdp_info = *sdp_info;
s->rtpoll_item = NULL;
s->intended_latency = LATENCY_USEC;
- s->smoother = pa_smoother_new(
- PA_USEC_PER_SEC*5,
- PA_USEC_PER_SEC*2,
- TRUE,
- TRUE,
- 10,
- pa_timeval_load(&now),
- TRUE);
s->last_rate_update = pa_timeval_load(&now);
s->last_latency = LATENCY_USEC;
s->estimated_rate = (double) sink->sample_spec.rate;
@@ -619,8 +603,6 @@ static void session_free(struct session *s) {
pa_sdp_info_destroy(&s->sdp_info);
pa_rtp_context_destroy(&s->rtp_context);
- pa_smoother_free(s->smoother);
-
pa_xfree(s);
}
--
hooks/post-receive
PulseAudio Sound Server
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