mirror of
https://gitlab.freedesktop.org/pipewire/pipewire.git
synced 2025-11-01 22:58:50 -04:00
module-rtp: add new rtp-session module
The module uses the apple session setup for managing peer connections. Make a generic rtp stream object, make midi and audio implementations.
This commit is contained in:
parent
c36d9bef38
commit
7da031c969
6 changed files with 2401 additions and 0 deletions
455
src/modules/module-rtp/stream.c
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455
src/modules/module-rtp/stream.c
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@ -0,0 +1,455 @@
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/* PipeWire */
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/* SPDX-FileCopyrightText: Copyright © 2023 Wim Taymans <wim.taymans@gmail.com> */
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/* SPDX-License-Identifier: MIT */
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#include <sys/socket.h>
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#include <arpa/inet.h>
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#include <spa/utils/result.h>
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#include <spa/utils/json.h>
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#include <spa/utils/ringbuffer.h>
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#include <spa/utils/dll.h>
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#include <spa/param/audio/format-utils.h>
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#include <spa/control/control.h>
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#include <spa/debug/types.h>
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#include <spa/debug/mem.h>
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#include <pipewire/pipewire.h>
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#include <pipewire/impl.h>
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#include <module-rtp/rtp.h>
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#include <module-rtp/stream.h>
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#include <module-rtp/apple-midi.h>
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#define DEFAULT_FORMAT "S16BE"
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#define DEFAULT_RATE 48000
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#define DEFAULT_CHANNELS 2
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#define DEFAULT_POSITION "[ FL FR ]"
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#define BUFFER_SIZE (1u<<22)
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#define BUFFER_MASK (BUFFER_SIZE-1)
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#define BUFFER_SIZE2 (BUFFER_SIZE>>1)
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#define BUFFER_MASK2 (BUFFER_SIZE2-1)
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#define DEFAULT_MTU 1280
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#define DEFAULT_MIN_PTIME 2
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#define DEFAULT_MAX_PTIME 20
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#define ERROR_MSEC 2
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#define DEFAULT_SESS_LATENCY 100
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#define rtp_stream_emit(s,m,v,...) spa_hook_list_call(&s->listener_list, \
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struct rtp_stream_events, m, v, ##__VA_ARGS__)
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#define rtp_stream_emit_destroy(s) rtp_stream_emit(s, destroy, 0)
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#define rtp_stream_emit_state_changed(s,n,e) rtp_stream_emit(s, state_changed,0,n,e)
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#define rtp_stream_emit_send_packet(s,i,l) rtp_stream_emit(s, send_packet,0,i,l)
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struct impl {
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struct spa_audio_info info;
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struct pw_stream *stream;
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struct spa_hook stream_listener;
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struct pw_stream_events stream_events;
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struct spa_hook_list listener_list;
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struct spa_hook listener;
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const struct format_info *format_info;
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uint32_t rate;
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uint32_t stride;
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uint8_t payload;
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uint32_t ssrc;
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uint16_t seq;
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unsigned have_ssrc:1;
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unsigned have_seq:1;
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uint32_t ts_offset;
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uint32_t psamples;
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uint32_t mtu;
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struct spa_ringbuffer ring;
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uint8_t buffer[BUFFER_SIZE];
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struct spa_io_rate_match *io_rate_match;
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struct spa_io_position *io_position;
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struct spa_dll dll;
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double corr;
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uint32_t target_buffer;
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float max_error;
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float last_timestamp;
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float last_time;
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unsigned direct_timestamp:1;
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unsigned always_process:1;
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unsigned started:1;
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unsigned have_sync:1;
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unsigned receiving:1;
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unsigned first:1;
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void (*receive_rtp)(struct impl *impl, uint8_t *buffer, ssize_t len);
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};
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#include "module-rtp/audio.c"
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#include "module-rtp/midi.c"
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struct format_info {
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uint32_t media_subtype;
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uint32_t format;
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uint32_t size;
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const char *mime;
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const char *media_type;
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};
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static const struct format_info audio_format_info[] = {
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{ SPA_MEDIA_SUBTYPE_raw, SPA_AUDIO_FORMAT_U8, 1, "L8", "audio" },
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{ SPA_MEDIA_SUBTYPE_raw, SPA_AUDIO_FORMAT_ALAW, 1, "PCMA", "audio" },
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{ SPA_MEDIA_SUBTYPE_raw, SPA_AUDIO_FORMAT_ULAW, 1, "PCMU", "audio" },
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{ SPA_MEDIA_SUBTYPE_raw, SPA_AUDIO_FORMAT_S16_BE, 2, "L16", "audio" },
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{ SPA_MEDIA_SUBTYPE_raw, SPA_AUDIO_FORMAT_S24_BE, 3, "L24", "audio" },
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{ SPA_MEDIA_SUBTYPE_control, 0, 1, "rtp-midi", "audio" },
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};
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static void stream_io_changed(void *data, uint32_t id, void *area, uint32_t size)
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{
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struct impl *impl = data;
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switch (id) {
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case SPA_IO_RateMatch:
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impl->io_rate_match = area;
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break;
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case SPA_IO_Position:
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impl->io_position = area;
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break;
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}
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}
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static void stream_destroy(void *d)
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{
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struct impl *impl = d;
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spa_hook_remove(&impl->stream_listener);
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impl->stream = NULL;
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}
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static int stream_start(struct impl *impl)
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{
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if (impl->started)
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return 0;
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rtp_stream_emit_state_changed(impl, true, NULL);
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impl->started = true;
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return 0;
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}
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static int stream_stop(struct impl *impl)
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{
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if (!impl->started)
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return 0;
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rtp_stream_emit_state_changed(impl, false, NULL);
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impl->started = false;
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return 0;
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}
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static void on_stream_state_changed(void *d, enum pw_stream_state old,
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enum pw_stream_state state, const char *error)
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{
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struct impl *impl = d;
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switch (state) {
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case PW_STREAM_STATE_UNCONNECTED:
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pw_log_info("stream disconnected");
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break;
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case PW_STREAM_STATE_ERROR:
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pw_log_error("stream error: %s", error);
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rtp_stream_emit_state_changed(impl, false, error);
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break;
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case PW_STREAM_STATE_STREAMING:
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if ((errno = -stream_start(impl)) < 0)
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pw_log_error("failed to start RTP stream: %m");
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break;
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case PW_STREAM_STATE_PAUSED:
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if (!impl->always_process)
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stream_stop(impl);
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impl->have_sync = false;
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break;
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default:
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break;
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}
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}
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static const struct pw_stream_events stream_events = {
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PW_VERSION_STREAM_EVENTS,
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.destroy = stream_destroy,
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.state_changed = on_stream_state_changed,
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.io_changed = stream_io_changed,
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};
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static const struct format_info *find_audio_format_info(const struct spa_audio_info *info)
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{
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SPA_FOR_EACH_ELEMENT_VAR(audio_format_info, f)
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if (f->media_subtype == info->media_subtype &&
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(f->format == 0 || f->format == info->info.raw.format))
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return f;
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return NULL;
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}
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static inline uint32_t format_from_name(const char *name, size_t len)
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{
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int i;
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for (i = 0; spa_type_audio_format[i].name; i++) {
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if (strncmp(name, spa_debug_type_short_name(spa_type_audio_format[i].name), len) == 0)
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return spa_type_audio_format[i].type;
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}
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return SPA_AUDIO_FORMAT_UNKNOWN;
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}
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static uint32_t channel_from_name(const char *name)
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{
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int i;
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for (i = 0; spa_type_audio_channel[i].name; i++) {
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if (spa_streq(name, spa_debug_type_short_name(spa_type_audio_channel[i].name)))
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return spa_type_audio_channel[i].type;
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}
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return SPA_AUDIO_CHANNEL_UNKNOWN;
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}
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static void parse_position(struct spa_audio_info_raw *info, const char *val, size_t len)
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{
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struct spa_json it[2];
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char v[256];
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spa_json_init(&it[0], val, len);
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if (spa_json_enter_array(&it[0], &it[1]) <= 0)
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spa_json_init(&it[1], val, len);
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info->channels = 0;
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while (spa_json_get_string(&it[1], v, sizeof(v)) > 0 &&
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info->channels < SPA_AUDIO_MAX_CHANNELS) {
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info->position[info->channels++] = channel_from_name(v);
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}
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}
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static void parse_audio_info(const struct pw_properties *props, struct spa_audio_info_raw *info)
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{
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const char *str;
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spa_zero(*info);
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if ((str = pw_properties_get(props, PW_KEY_AUDIO_FORMAT)) == NULL)
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str = DEFAULT_FORMAT;
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info->format = format_from_name(str, strlen(str));
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info->rate = pw_properties_get_uint32(props, PW_KEY_AUDIO_RATE, info->rate);
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if (info->rate == 0)
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info->rate = DEFAULT_RATE;
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info->channels = pw_properties_get_uint32(props, PW_KEY_AUDIO_CHANNELS, info->channels);
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info->channels = SPA_MIN(info->channels, SPA_AUDIO_MAX_CHANNELS);
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if ((str = pw_properties_get(props, SPA_KEY_AUDIO_POSITION)) != NULL)
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parse_position(info, str, strlen(str));
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if (info->channels == 0)
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parse_position(info, DEFAULT_POSITION, strlen(DEFAULT_POSITION));
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}
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static uint32_t msec_to_samples(struct impl *impl, uint32_t msec)
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{
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return msec * impl->rate / 1000;
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}
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struct rtp_stream *rtp_stream_new(struct pw_core *core,
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enum pw_direction direction, struct pw_properties *props,
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const struct rtp_stream_events *events, void *data)
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{
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struct impl *impl;
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const char *str;
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uint8_t buffer[1024];
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struct spa_pod_builder b;
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uint32_t n_params, min_samples, max_samples;
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float min_ptime, max_ptime;
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const struct spa_pod *params[1];
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enum pw_stream_flags flags;
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int latency_msec;
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int res;
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impl = calloc(1, sizeof(*impl));
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if (impl == NULL) {
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res = -errno;
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goto out;
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return NULL;
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}
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impl->first = true;
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spa_hook_list_init(&impl->listener_list);
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impl->stream_events = stream_events;
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impl->info.media_type = SPA_MEDIA_TYPE_audio;
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impl->info.media_subtype = SPA_MEDIA_SUBTYPE_raw;
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if ((str = pw_properties_get(props, "rtp.media")) != NULL) {
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if (spa_streq(str, "audio")) {
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impl->info.media_type = SPA_MEDIA_TYPE_audio;
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impl->info.media_subtype = SPA_MEDIA_SUBTYPE_raw;
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impl->payload = 127;
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}
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else if (spa_streq(str, "midi")) {
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impl->info.media_type = SPA_MEDIA_TYPE_application;
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impl->info.media_subtype = SPA_MEDIA_SUBTYPE_control;
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impl->payload = 0x61;
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}
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else {
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pw_log_error("unsupported media type:%s", str);
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res = -EINVAL;
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goto out;
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}
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}
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switch (impl->info.media_type) {
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case SPA_MEDIA_TYPE_audio:
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parse_audio_info(props, &impl->info.info.raw);
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impl->format_info = find_audio_format_info(&impl->info);
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if (impl->format_info == NULL) {
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pw_log_error("unsupported audio format:%d channels:%d",
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impl->info.info.raw.format, impl->info.info.raw.channels);
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res = -EINVAL;
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goto out;
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}
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impl->stride = impl->format_info->size * impl->info.info.raw.channels;
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impl->rate = impl->info.info.raw.rate;
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break;
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case SPA_MEDIA_TYPE_application:
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impl->format_info = find_audio_format_info(&impl->info);
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if (impl->format_info == NULL) {
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res = -EINVAL;
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goto out;
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}
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pw_properties_set(props, PW_KEY_FORMAT_DSP, "8 bit raw midi");
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impl->stride = impl->format_info->size;
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impl->rate = pw_properties_get_uint32(props, "midi.rate", 10000);
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if (impl->rate == 0)
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impl->rate = 10000;
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break;
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default:
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spa_assert_not_reached();
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break;
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}
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pw_properties_setf(props, "rtp.media", "%s", impl->format_info->media_type);
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pw_properties_setf(props, "rtp.mime", "%s", impl->format_info->mime);
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if (direction == PW_DIRECTION_INPUT)
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impl->ssrc = pw_properties_get_uint32(props, "rtp.sender-ssrc", pw_rand32());
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else
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impl->ssrc = pw_properties_get_uint32(props, "rtp.receiver-ssrc", pw_rand32());
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impl->payload = pw_properties_get_uint32(props, "rtp.payload", impl->payload);
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impl->mtu = pw_properties_get_uint32(props, "rtp.mtu", DEFAULT_MTU);
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str = pw_properties_get(props, "rtp.min-ptime");
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if (!spa_atof(str, &min_ptime))
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min_ptime = DEFAULT_MIN_PTIME;
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str = pw_properties_get(props, "rtp.max-ptime");
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if (!spa_atof(str, &max_ptime))
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max_ptime = DEFAULT_MAX_PTIME;
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min_samples = min_ptime * impl->rate / 1000;
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max_samples = max_ptime * impl->rate / 1000;
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impl->psamples = impl->mtu / impl->stride;
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impl->psamples = SPA_CLAMP(impl->psamples, min_samples, max_samples);
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latency_msec = pw_properties_get_uint32(props,
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"sess.latency.msec", DEFAULT_SESS_LATENCY);
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impl->target_buffer = msec_to_samples(impl, latency_msec);
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impl->max_error = msec_to_samples(impl, ERROR_MSEC);
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pw_properties_setf(props, PW_KEY_NODE_RATE, "1/%d", impl->rate);
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pw_properties_setf(props, PW_KEY_NODE_LATENCY, "%d/%d",
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impl->target_buffer / 2, impl->rate);
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spa_dll_init(&impl->dll);
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spa_dll_set_bw(&impl->dll, SPA_DLL_BW_MIN, 128, impl->rate);
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impl->corr = 1.0;
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impl->stream = pw_stream_new(core, "rtp-session", props);
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props = NULL;
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if (impl->stream == NULL) {
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res = -errno;
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pw_log_error("can't create stream: %m");
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goto out;
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}
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n_params = 0;
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spa_pod_builder_init(&b, buffer, sizeof(buffer));
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flags = PW_STREAM_FLAG_MAP_BUFFERS | PW_STREAM_FLAG_RT_PROCESS;
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switch (impl->info.media_type) {
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case SPA_MEDIA_TYPE_audio:
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params[n_params++] = spa_format_audio_build(&b,
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SPA_PARAM_EnumFormat, &impl->info);
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flags |= PW_STREAM_FLAG_AUTOCONNECT;
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if (direction == SPA_DIRECTION_INPUT)
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impl->stream_events.process = process_audio_capture;
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else
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impl->stream_events.process = process_audio_playback;
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impl->receive_rtp = receive_rtp_audio;
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break;
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case SPA_MEDIA_TYPE_application:
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params[n_params++] = spa_pod_builder_add_object(&b,
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SPA_TYPE_OBJECT_Format, SPA_PARAM_EnumFormat,
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SPA_FORMAT_mediaType, SPA_POD_Id(SPA_MEDIA_TYPE_application),
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SPA_FORMAT_mediaSubtype, SPA_POD_Id(SPA_MEDIA_SUBTYPE_control));
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if (direction == SPA_DIRECTION_INPUT)
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impl->stream_events.process = process_midi_capture;
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else
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impl->stream_events.process = process_midi_playback;
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impl->receive_rtp = receive_rtp_midi;
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break;
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default:
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res = -EINVAL;
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goto out;
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}
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pw_stream_add_listener(impl->stream,
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&impl->stream_listener,
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&impl->stream_events, impl);
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if ((res = pw_stream_connect(impl->stream,
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direction,
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PW_ID_ANY,
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flags,
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params, n_params)) < 0) {
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pw_log_error("can't connect stream: %s", spa_strerror(res));
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goto out;
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}
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if (impl->always_process &&
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(res = stream_start(impl)) < 0)
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goto out;
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spa_hook_list_append(&impl->listener_list, &impl->listener, events, data);
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return (struct rtp_stream*)impl;
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out:
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pw_properties_free(props);
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errno = -res;
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return NULL;
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}
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void rtp_stream_destroy(struct rtp_stream *s)
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{
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struct impl *impl = (struct impl*)s;
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rtp_stream_emit_destroy(impl);
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if (impl->stream)
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pw_stream_destroy(impl->stream);
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||||
|
||||
spa_hook_list_clean(&impl->listener_list);
|
||||
free(impl);
|
||||
}
|
||||
|
||||
int rtp_stream_receive_packet(struct rtp_stream *s, uint8_t *buffer, size_t len)
|
||||
{
|
||||
struct impl *impl = (struct impl*)s;
|
||||
impl->receive_rtp(impl, buffer, len);
|
||||
return 0;
|
||||
}
|
||||
Loading…
Add table
Add a link
Reference in a new issue