| 1 | /* |
| 2 | * This file is part of DisOrder. |
| 3 | * Copyright (C) 2009 Richard Kettlewell |
| 4 | * |
| 5 | * This program is free software: you can redistribute it and/or modify |
| 6 | * it under the terms of the GNU General Public License as published by |
| 7 | * the Free Software Foundation, either version 3 of the License, or |
| 8 | * (at your option) any later version. |
| 9 | * |
| 10 | * This program is distributed in the hope that it will be useful, |
| 11 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| 12 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
| 13 | * GNU General Public License for more details. |
| 14 | * |
| 15 | * You should have received a copy of the GNU General Public License |
| 16 | * along with this program. If not, see <http://www.gnu.org/licenses/>. |
| 17 | */ |
| 18 | /** @file lib/uaudio-schedule.c |
| 19 | * @brief Scheduler for RTP and command backends |
| 20 | * |
| 21 | * These functions ensure that audio is only written at approximately the rate |
| 22 | * it should play at, allowing pause to function properly. |
| 23 | * |
| 24 | * OSS and ALSA we expect to be essentially synchronous (though we could use |
| 25 | * this code if they don't play nicely). Core Audio sorts out its own timing |
| 26 | * issues itself. |
| 27 | * |
| 28 | * The sequence numbers are intended for RTP's use but it's more convenient to |
| 29 | * maintain them here. |
| 30 | */ |
| 31 | |
| 32 | #include "common.h" |
| 33 | |
| 34 | #include <unistd.h> |
| 35 | #include <gcrypt.h> |
| 36 | |
| 37 | #include "uaudio.h" |
| 38 | #include "mem.h" |
| 39 | #include "log.h" |
| 40 | #include "syscalls.h" |
| 41 | #include "timeval.h" |
| 42 | |
| 43 | /** @brief Sample timestamp |
| 44 | * |
| 45 | * This is the timestamp that will be used on the next outbound packet. |
| 46 | * |
| 47 | * The timestamp in an RTP packet header is only 32 bits wide. With 44100Hz |
| 48 | * stereo, that only gives about half a day before wrapping, which is not |
| 49 | * particularly convenient for certain debugging purposes. Therefore the |
| 50 | * timestamp is maintained as a 64-bit integer, giving around six million years |
| 51 | * before wrapping, and truncated to 32 bits when transmitting. |
| 52 | */ |
| 53 | uint64_t uaudio_schedule_timestamp; |
| 54 | |
| 55 | /** @brief Actual time corresponding to @ref uaudio_schedule_timestamp |
| 56 | * |
| 57 | * This is the time, on this machine, at which the sample at @ref |
| 58 | * uaudio_schedule_timestamp ought to be sent, interpreted as the time the last |
| 59 | * packet was sent plus the time length of the packet. */ |
| 60 | static struct timeval uaudio_schedule_timeval; |
| 61 | |
| 62 | /** @brief Set when we (re-)activate, to provoke timestamp resync */ |
| 63 | int uaudio_schedule_reactivated; |
| 64 | |
| 65 | /** @brief Delay threshold in microseconds |
| 66 | * |
| 67 | * uaudio_schedule_play() never attempts to introduce a delay shorter than this. |
| 68 | */ |
| 69 | static int64_t uaudio_schedule_delay_threshold; |
| 70 | |
| 71 | /** @brief Time for current packet */ |
| 72 | static struct timeval uaudio_schedule_now; |
| 73 | |
| 74 | /** @brief Synchronize playback operations against real time |
| 75 | * |
| 76 | * This function sleeps as necessary to rate-limit playback operations to match |
| 77 | * the actual playback rate. It also maintains @ref uaudio_schedule_timestamp |
| 78 | * as an arbitrarily-based sample counter, for use by RTP. |
| 79 | * |
| 80 | * You should call this in your API's @ref uaudio_playcallback before writing |
| 81 | * and call uaudio_schedule_update() afterwards. |
| 82 | */ |
| 83 | void uaudio_schedule_synchronize(void) { |
| 84 | retry: |
| 85 | xgettimeofday(&uaudio_schedule_now, NULL); |
| 86 | if(uaudio_schedule_reactivated) { |
| 87 | /* We've been deactivated for some unknown interval. We need to advance |
| 88 | * rtp_timestamp to account for the dead air. */ |
| 89 | /* On the first run through we'll set the start time. */ |
| 90 | if(!uaudio_schedule_timeval.tv_sec) |
| 91 | uaudio_schedule_timeval = uaudio_schedule_now; |
| 92 | /* See how much time we missed. |
| 93 | * |
| 94 | * This will be 0 on the first run through, in which case we'll not modify |
| 95 | * anything. |
| 96 | * |
| 97 | * It'll be negative in the (rare) situation where the deactivation |
| 98 | * interval is shorter than the last packet we sent. In this case we wait |
| 99 | * for that much time and then return having sent no samples, which will |
| 100 | * cause uaudio_play_thread_fn() to retry. |
| 101 | * |
| 102 | * In the normal case it will be positive. |
| 103 | */ |
| 104 | const int64_t delay = tvsub_us(uaudio_schedule_now, |
| 105 | uaudio_schedule_timeval); /* microseconds */ |
| 106 | if(delay < 0) { |
| 107 | usleep(-delay); |
| 108 | goto retry; |
| 109 | } |
| 110 | /* Advance the RTP timestamp to the present. With 44.1KHz stereo this will |
| 111 | * overflow the intermediate value with a delay of a bit over 6 years. |
| 112 | * This seems acceptable. */ |
| 113 | uint64_t update = (delay * uaudio_rate * uaudio_channels) / 1000000; |
| 114 | /* Don't throw off channel synchronization */ |
| 115 | update -= update % uaudio_channels; |
| 116 | /* We log nontrivial changes */ |
| 117 | if(update) |
| 118 | info("advancing uaudio_schedule_timeval by %"PRIu64" samples", update); |
| 119 | uaudio_schedule_timestamp += update; |
| 120 | uaudio_schedule_timeval = uaudio_schedule_now; |
| 121 | uaudio_schedule_reactivated = 0; |
| 122 | } else { |
| 123 | /* Chances are we've been called right on the heels of the previous packet. |
| 124 | * If we just sent packets as fast as we got audio data we'd get way ahead |
| 125 | * of the player and some buffer somewhere would fill (or at least become |
| 126 | * unreasonably large). |
| 127 | * |
| 128 | * First find out how far ahead of the target time we are. |
| 129 | */ |
| 130 | const int64_t ahead = tvsub_us(uaudio_schedule_timeval, |
| 131 | uaudio_schedule_now); /* microseconds */ |
| 132 | /* Only delay at all if we are nontrivially ahead. */ |
| 133 | if(ahead > uaudio_schedule_delay_threshold) { |
| 134 | /* Don't delay by the full amount */ |
| 135 | usleep(ahead - uaudio_schedule_delay_threshold / 2); |
| 136 | /* Refetch time (so we don't get out of step with reality) */ |
| 137 | xgettimeofday(&uaudio_schedule_now, NULL); |
| 138 | } |
| 139 | } |
| 140 | } |
| 141 | |
| 142 | /** @brief Update schedule after writing |
| 143 | * |
| 144 | * Called by your API's @ref uaudio_playcallback after sending audio data (to a |
| 145 | * subprocess or network or whatever). A separate function so that the caller |
| 146 | * doesn't have to know how many samples they're going to write until they've |
| 147 | * done so. |
| 148 | */ |
| 149 | void uaudio_schedule_update(size_t written_samples) { |
| 150 | /* uaudio_schedule_timestamp and uaudio_schedule_timestamp are supposed to |
| 151 | * refer to the first sample of the next packet */ |
| 152 | uaudio_schedule_timestamp += written_samples; |
| 153 | const unsigned usec = (uaudio_schedule_timeval.tv_usec |
| 154 | + 1000000 * written_samples / (uaudio_rate |
| 155 | * uaudio_channels)); |
| 156 | /* ...will only overflow 32 bits if one packet is more than about half an |
| 157 | * hour long, which is not plausible. */ |
| 158 | uaudio_schedule_timeval.tv_sec += usec / 1000000; |
| 159 | uaudio_schedule_timeval.tv_usec = usec % 1000000; |
| 160 | } |
| 161 | |
| 162 | /** @brief Initialize audio scheduling |
| 163 | * |
| 164 | * Should be called from your API's @c start callback. |
| 165 | */ |
| 166 | void uaudio_schedule_init(void) { |
| 167 | gcry_create_nonce(&uaudio_schedule_timestamp, |
| 168 | sizeof uaudio_schedule_timestamp); |
| 169 | /* uaudio_schedule_play() will spot this and choose an initial value */ |
| 170 | uaudio_schedule_timeval.tv_sec = 0; |
| 171 | } |
| 172 | |
| 173 | /* |
| 174 | Local Variables: |
| 175 | c-basic-offset:2 |
| 176 | comment-column:40 |
| 177 | fill-column:79 |
| 178 | indent-tabs-mode:nil |
| 179 | End: |
| 180 | */ |