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module-rtp: don't leak opus codec and ptp_sender

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Add a deinit() function and use it to free the opus codec we created in
init().

Also free the ptp_sender when it was created.
This commit is contained in:
Wim Taymans 2025-07-24 13:16:15 +02:00
parent a09bf57944
commit 42b779974c
2 changed files with 21 additions and 1 deletions
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11
src/modules/module-rtp/opus.c
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@ -320,6 +320,16 @@ static void rtp_opus_process_capture(void *data)
rtp_opus_flush_packets(impl);
}
static void rtp_opus_deinit(struct impl *impl, enum spa_direction direction)
{
if (impl->stream_data) {
if (direction == SPA_DIRECTION_INPUT)
opus_multistream_encoder_destroy(impl->stream_data);
else
opus_multistream_decoder_destroy(impl->stream_data);
}
}
static int rtp_opus_init(struct impl *impl, enum spa_direction direction)
{
int err;
@ -342,6 +352,7 @@ static int rtp_opus_init(struct impl *impl, enum spa_direction direction)
for (i = 0; i < impl->info.info.opus.channels; i++)
mapping[i] = i;
impl->deinit = rtp_opus_deinit;
impl->receive_rtp = rtp_opus_receive;
if (direction == SPA_DIRECTION_INPUT) {
impl->stream_events.process = rtp_opus_process_capture;