uaudio.c uaudio-thread.c uaudio.h \
uaudio-oss.c uaudio-alsa.c \
uaudio-coreaudio.c \
- uaudio-rtp.c uaudio-command.c \
+ uaudio-rtp.c uaudio-command.c uaudio-schedule.c \
url.h url.c \
user.h user.c \
unicode.h unicode.c \
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
/** @file lib/uaudio-command.c
- * @brief Support for commmand backend */
+ * @brief Support for commmand backend
+ *
+ * We use the code in @ref lib/uaudio-schedule.c to ensure that we write at
+ * approximately the 'real' rate. For disorder-playrtp this isn't very useful
+ * (thought it might reduce the size of various buffers downstream of us) but
+ * when run from the speaker it means that pausing stands a chance of working.
+ */
#include "common.h"
#include <errno.h>
/** @brief Send audio data to subprocess */
static size_t command_play(void *buffer, size_t nsamples) {
+ uaudio_schedule_synchronize();
const size_t bytes = nsamples * uaudio_sample_size;
int written = write(command_fd, buffer, bytes);
if(written < 0) {
fatal(errno, "error writing to audio command subprocess");
}
}
- return written / uaudio_sample_size;
+ const size_t written_samples = written / uaudio_sample_size;
+ uaudio_schedule_update(written_samples);
+ return written_samples;
}
static void command_start(uaudio_callback *callback,
void *userdata) {
command_open();
+ uaudio_schedule_init();
uaudio_thread_start(callback,
userdata,
command_play,
}
static void command_activate(void) {
+ uaudio_schedule_reactivated = 1;
uaudio_thread_activate();
}
/** @brief RTP sequence number */
static uint16_t rtp_sequence;
-/** @brief RTP timestamp
- *
- * This is the timestamp that will be used on the next outbound packet.
- *
- * The timestamp in the packet header is only 32 bits wide. With 44100Hz
- * stereo, that only gives about half a day before wrapping, which is not
- * particularly convenient for certain debugging purposes. Therefore the
- * timestamp is maintained as a 64-bit integer, giving around six million years
- * before wrapping, and truncated to 32 bits when transmitting.
- */
-static uint64_t rtp_timestamp;
-
-/** @brief Actual time corresponding to @ref rtp_timestamp
- *
- * This is the time, on this machine, at which the sample at @ref rtp_timestamp
- * ought to be sent, interpreted as the time the last packet was sent plus the
- * time length of the packet. */
-static struct timeval rtp_timeval;
-
-/** @brief Set when we (re-)activate, to provoke timestamp resync */
-static int rtp_reactivated;
-
/** @brief Network error count
*
* If too many errors occur in too short a time, we give up.
"rtp-source-port",
"multicast-ttl",
"multicast-loop",
- "rtp-delay-threshold",
+ "delay-threshold",
NULL
};
static size_t rtp_play(void *buffer, size_t nsamples) {
struct rtp_header header;
struct iovec vec[2];
- struct timeval now;
/* We do as much work as possible before checking what time it is */
/* Fill out header */
header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
header.seq = htons(rtp_sequence++);
header.ssrc = rtp_id;
- header.mpt = (rtp_reactivated ? 0x80 : 0x00) | rtp_payload;
+ header.mpt = (uaudio_schedule_reactivated ? 0x80 : 0x00) | rtp_payload;
#if !WORDS_BIGENDIAN
/* Convert samples to network byte order */
uint16_t *u = buffer, *const limit = u + nsamples;
vec[0].iov_len = sizeof header;
vec[1].iov_base = buffer;
vec[1].iov_len = nsamples * uaudio_sample_size;
-retry:
- xgettimeofday(&now, NULL);
- if(rtp_reactivated) {
- /* We've been deactivated for some unknown interval. We need to advance
- * rtp_timestamp to account for the dead air. */
- /* On the first run through we'll set the start time. */
- if(!rtp_timeval.tv_sec)
- rtp_timeval = now;
- /* See how much time we missed.
- *
- * This will be 0 on the first run through, in which case we'll not modify
- * anything.
- *
- * It'll be negative in the (rare) situation where the deactivation
- * interval is shorter than the last packet we sent. In this case we wait
- * for that much time and then return having sent no samples, which will
- * cause uaudio_play_thread_fn() to retry.
- *
- * In the normal case it will be positive.
- */
- const int64_t delay = tvsub_us(now, rtp_timeval); /* microseconds */
- if(delay < 0) {
- usleep(-delay);
- goto retry;
- }
- /* Advance the RTP timestamp to the present. With 44.1KHz stereo this will
- * overflow the intermediate value with a delay of a bit over 6 years.
- * This seems acceptable. */
- uint64_t update = (delay * uaudio_rate * uaudio_channels) / 1000000;
- /* Don't throw off channel synchronization */
- update -= update % uaudio_channels;
- /* We log nontrivial changes */
- if(update)
- info("advancing rtp_time by %"PRIu64" samples", update);
- rtp_timestamp += update;
- rtp_timeval = now;
- rtp_reactivated = 0;
- } else {
- /* Chances are we've been called right on the heels of the previous packet.
- * If we just sent packets as fast as we got audio data we'd get way ahead
- * of the player and some buffer somewhere would fill (or at least become
- * unreasonably large).
- *
- * First find out how far ahead of the target time we are.
- */
- const int64_t ahead = tvsub_us(now, rtp_timeval); /* microseconds */
- /* Only delay at all if we are nontrivially ahead. */
- if(ahead > rtp_delay_threshold) {
- /* Don't delay by the full amount */
- usleep(ahead - rtp_delay_threshold / 2);
- /* Refetch time (so we don't get out of step with reality) */
- xgettimeofday(&now, NULL);
- }
- }
- header.timestamp = htonl((uint32_t)rtp_timestamp);
+ uaudio_schedule_synchronize();
+ header.timestamp = htonl((uint32_t)uaudio_schedule_timestamp);
int written_bytes;
do {
written_bytes = writev(rtp_fd, vec, 2);
} else
rtp_errors /= 2; /* gradual decay */
written_bytes -= sizeof (struct rtp_header);
- size_t written_samples = written_bytes / uaudio_sample_size;
- /* rtp_timestamp and rtp_timestamp are supposed to refer to the first sample
- * of the next packet */
- rtp_timestamp += written_samples;
- const unsigned usec = (rtp_timeval.tv_usec
- + 1000000 * written_samples / (uaudio_rate
- * uaudio_channels));
- /* ...will only overflow 32 bits if one packet is more than about half an
- * hour long, which is not plausible. */
- rtp_timeval.tv_sec += usec / 1000000;
- rtp_timeval.tv_usec = usec % 1000000;
+ const size_t written_samples = written_bytes / uaudio_sample_size;
+ uaudio_schedule_update(written_samples);
return written_samples;
}
fatal(errno, "error binding broadcast socket to %s", ssockname);
if(connect(rtp_fd, res->ai_addr, res->ai_addrlen) < 0)
fatal(errno, "error connecting broadcast socket to %s", sockname);
- /* Various fields are required to have random initial values by RFC3550. The
- * packet contents are highly public so there's no point asking for very
- * strong randomness. */
- gcry_create_nonce(&rtp_id, sizeof rtp_id);
- gcry_create_nonce(&rtp_sequence, sizeof rtp_sequence);
- gcry_create_nonce(&rtp_timestamp, sizeof rtp_timestamp);
- /* rtp_play() will spot this and choose an initial value */
- rtp_timeval.tv_sec = 0;
}
static void rtp_start(uaudio_callback *callback,
else
fatal(0, "asked for %d/%d/%d 16/44100/1 and 16/44100/2",
uaudio_bits, uaudio_rate, uaudio_channels);
+ /* Various fields are required to have random initial values by RFC3550. The
+ * packet contents are highly public so there's no point asking for very
+ * strong randomness. */
+ gcry_create_nonce(&rtp_id, sizeof rtp_id);
+ gcry_create_nonce(&rtp_sequence, sizeof rtp_sequence);
rtp_open();
+ uaudio_schedule_init();
uaudio_thread_start(callback,
userdata,
rtp_play,
}
static void rtp_activate(void) {
- rtp_reactivated = 1;
+ uaudio_schedule_reactivated = 1;
uaudio_thread_activate();
}
--- /dev/null
+/*
+ * This file is part of DisOrder.
+ * Copyright (C) 2009 Richard Kettlewell
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
+ */
+/** @file lib/uaudio-schedule.c
+ * @brief Scheduler for RTP and command backends
+ *
+ * These functions ensure that audio is only written at approximately the rate
+ * it should play at, allowing pause to function properly.
+ *
+ * OSS and ALSA we expect to be essentially synchronous (though we could use
+ * this code if they don't play nicely). Core Audio sorts out its own timing
+ * issues itself.
+ *
+ * The sequence numbers are intended for RTP's use but it's more convenient to
+ * maintain them here.
+ */
+
+#include "common.h"
+
+#include <unistd.h>
+#include <gcrypt.h>
+
+#include "uaudio.h"
+#include "mem.h"
+#include "log.h"
+#include "syscalls.h"
+#include "timeval.h"
+
+/** @brief Sample timestamp
+ *
+ * This is the timestamp that will be used on the next outbound packet.
+ *
+ * The timestamp in an RTP packet header is only 32 bits wide. With 44100Hz
+ * stereo, that only gives about half a day before wrapping, which is not
+ * particularly convenient for certain debugging purposes. Therefore the
+ * timestamp is maintained as a 64-bit integer, giving around six million years
+ * before wrapping, and truncated to 32 bits when transmitting.
+ */
+uint64_t uaudio_schedule_timestamp;
+
+/** @brief Actual time corresponding to @ref uaudio_schedule_timestamp
+ *
+ * This is the time, on this machine, at which the sample at @ref
+ * uaudio_schedule_timestamp ought to be sent, interpreted as the time the last
+ * packet was sent plus the time length of the packet. */
+static struct timeval uaudio_schedule_timeval;
+
+/** @brief Set when we (re-)activate, to provoke timestamp resync */
+int uaudio_schedule_reactivated;
+
+/** @brief Delay threshold in microseconds
+ *
+ * uaudio_schedule_play() never attempts to introduce a delay shorter than this.
+ */
+static int64_t uaudio_schedule_delay_threshold;
+
+/** @brief Time for current packet */
+static struct timeval uaudio_schedule_now;
+
+/** @brief Synchronize playback operations against real time
+ *
+ * This function sleeps as necessary to rate-limit playback operations to match
+ * the actual playback rate. It also maintains @ref uaudio_schedule_timestamp
+ * as an arbitrarily-based sample counter, for use by RTP.
+ *
+ * You should call this in your API's @ref uaudio_playcallback before writing
+ * and call uaudio_schedule_update() afterwards.
+ */
+void uaudio_schedule_synchronize(void) {
+retry:
+ xgettimeofday(&uaudio_schedule_now, NULL);
+ if(uaudio_schedule_reactivated) {
+ /* We've been deactivated for some unknown interval. We need to advance
+ * rtp_timestamp to account for the dead air. */
+ /* On the first run through we'll set the start time. */
+ if(!uaudio_schedule_timeval.tv_sec)
+ uaudio_schedule_timeval = uaudio_schedule_now;
+ /* See how much time we missed.
+ *
+ * This will be 0 on the first run through, in which case we'll not modify
+ * anything.
+ *
+ * It'll be negative in the (rare) situation where the deactivation
+ * interval is shorter than the last packet we sent. In this case we wait
+ * for that much time and then return having sent no samples, which will
+ * cause uaudio_play_thread_fn() to retry.
+ *
+ * In the normal case it will be positive.
+ */
+ const int64_t delay = tvsub_us(uaudio_schedule_now,
+ uaudio_schedule_timeval); /* microseconds */
+ if(delay < 0) {
+ usleep(-delay);
+ goto retry;
+ }
+ /* Advance the RTP timestamp to the present. With 44.1KHz stereo this will
+ * overflow the intermediate value with a delay of a bit over 6 years.
+ * This seems acceptable. */
+ uint64_t update = (delay * uaudio_rate * uaudio_channels) / 1000000;
+ /* Don't throw off channel synchronization */
+ update -= update % uaudio_channels;
+ /* We log nontrivial changes */
+ if(update)
+ info("advancing uaudio_schedule_timeval by %"PRIu64" samples", update);
+ uaudio_schedule_timestamp += update;
+ uaudio_schedule_timeval = uaudio_schedule_now;
+ uaudio_schedule_reactivated = 0;
+ } else {
+ /* Chances are we've been called right on the heels of the previous packet.
+ * If we just sent packets as fast as we got audio data we'd get way ahead
+ * of the player and some buffer somewhere would fill (or at least become
+ * unreasonably large).
+ *
+ * First find out how far ahead of the target time we are.
+ */
+ const int64_t ahead = tvsub_us(uaudio_schedule_timeval,
+ uaudio_schedule_now); /* microseconds */
+ /* Only delay at all if we are nontrivially ahead. */
+ if(ahead > uaudio_schedule_delay_threshold) {
+ /* Don't delay by the full amount */
+ usleep(ahead - uaudio_schedule_delay_threshold / 2);
+ /* Refetch time (so we don't get out of step with reality) */
+ xgettimeofday(&uaudio_schedule_now, NULL);
+ }
+ }
+}
+
+/** @brief Update schedule after writing
+ *
+ * Called by your API's @ref uaudio_playcallback after sending audio data (to a
+ * subprocess or network or whatever). A separate function so that the caller
+ * doesn't have to know how many samples they're going to write until they've
+ * done so.
+ */
+void uaudio_schedule_update(size_t written_samples) {
+ /* uaudio_schedule_timestamp and uaudio_schedule_timestamp are supposed to
+ * refer to the first sample of the next packet */
+ uaudio_schedule_timestamp += written_samples;
+ const unsigned usec = (uaudio_schedule_timeval.tv_usec
+ + 1000000 * written_samples / (uaudio_rate
+ * uaudio_channels));
+ /* ...will only overflow 32 bits if one packet is more than about half an
+ * hour long, which is not plausible. */
+ uaudio_schedule_timeval.tv_sec += usec / 1000000;
+ uaudio_schedule_timeval.tv_usec = usec % 1000000;
+}
+
+/** @brief Initialize audio scheduling
+ *
+ * Should be called from your API's @c start callback.
+ */
+void uaudio_schedule_init(void) {
+ gcry_create_nonce(&uaudio_schedule_timestamp,
+ sizeof uaudio_schedule_timestamp);
+ /* uaudio_schedule_play() will spot this and choose an initial value */
+ uaudio_schedule_timeval.tv_sec = 0;
+}
+
+/*
+Local Variables:
+c-basic-offset:2
+comment-column:40
+fill-column:79
+indent-tabs-mode:nil
+End:
+*/
void uaudio_thread_stop(void);
void uaudio_thread_activate(void);
void uaudio_thread_deactivate(void);
+void uaudio_schedule_synchronize(void);
+void uaudio_schedule_update(size_t written_samples);
+void uaudio_schedule_init(void);
+
+extern uint64_t uaudio_schedule_timestamp;
+extern int uaudio_schedule_reactivated;
#if HAVE_COREAUDIO_AUDIOHARDWARE_H
extern const struct uaudio uaudio_coreaudio;
~mdw / disorder / commitdiff