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[cast] Add config option for max_volume, like for Airplay

This commit is contained in:
ejurgensen 2020-11-16 20:01:55 +01:00
parent c21266d26f
commit d5f5e312ab
3 changed files with 33 additions and 17 deletions
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4
forked-daapd.conf.in
View File

@ -317,6 +317,10 @@ audio {
# Chromecast settings
# (make sure you get the capitalization of the device name right)
#chromecast "My Chromecast device" {
# forked-daapd's volume goes to 11! If that's more than you can handle
# you can set a lower value here
# max_volume = 11
# Enable this option to exclude a particular device from the speaker
# list
# exclude = false

1
src/conffile.c
View File

@ -166,6 +166,7 @@ static cfg_opt_t sec_airplay[] =
/* Chromecast device section structure */
static cfg_opt_t sec_chromecast[] =
{
CFG_INT("max_volume", 11, CFGF_NONE),
CFG_BOOL("exclude", cfg_false, CFGF_NONE),
CFG_INT("offset_ms", 0, CFGF_NONE),
CFG_END()

45
src/outputs/cast.c
View File

@ -108,6 +108,9 @@
#define CAST_QUALITY_BITS_PER_SAMPLE_DEFAULT 16
#define CAST_QUALITY_CHANNELS_DEFAULT 2
// This is an arbitrary value which just needs to be kept in sync with the config
#define CAST_CONFIG_MAX_VOLUME 11
// This makes the rtp session buffer 4 seconds of audio (4 sec * 50 pkts/sec),
// which can be used for delayed transmission (and retransmission)
#define CAST_PACKET_BUFFER_SIZE 200
@ -666,15 +669,10 @@ cast_session_free(struct cast_session *cs)
gnutls_deinit(cs->tls_session);
if (cs->address)
free(cs->address);
if (cs->devname)
free(cs->devname);
if (cs->session_id)
free(cs->session_id);
if (cs->transport_id)
free(cs->transport_id);
free(cs->address);
free(cs->devname);
free(cs->session_id);
free(cs->transport_id);
free(cs);
}
@ -1148,12 +1146,13 @@ packet_send(struct cast_session *cs, uint16_t seqnum)
DPRINTF(E_WARN, L_CAST, "Partial send (%d) for '%s'\n", ret, cs->devname);
}
// DPRINTF(E_DBG, L_CAST, "Sent RTP PACKET seqnum %u, rtptime %u, payload 0x%x, pktbuf_s %zu\n",
// cs->master_session->rtp_session->seqnum,
// cs->master_session->rtp_session->pos,
// pkt->header[1],
// cs->master_session->rtp_session->pktbuf_len
/*
DPRINTF(E_DBG, L_CAST, "Sent RTP PACKET seqnum %u, have until %u, payload 0x%x, pktbuf_s %zu\n",
seqnum,
cs->master_session->rtp_session->seqnum,
pkt->header[1],
cs->master_session->rtp_session->pktbuf_len);
*/
return 0;
}
@ -1978,6 +1977,8 @@ master_session_make(struct media_quality *quality)
// Change the SSRC to be in the interval [CAST_SSRC_AUDIO_MIN, CAST_SSRC_AUDIO_MAX]
cms->rtp_session->ssrc_id = ((cms->rtp_session->ssrc_id + CAST_SSRC_AUDIO_MIN) % CAST_SSRC_AUDIO_MAX) + CAST_SSRC_AUDIO_MIN;
cms->rtp_session->seqnum = 0; // TODO test
cms->quality = *quality;
cms->samples_per_packet = CAST_SAMPLES_PER_PACKET;
cms->rawbuf_size = STOB(cms->samples_per_packet, quality->bits_per_sample, quality->channels);
@ -2318,12 +2319,22 @@ static int
cast_device_volume_set(struct output_device *device, int callback_id)
{
struct cast_session *cs = device->session;
cfg_t *cast_cfg;
int max_volume;
int ret;
if (!cs || !(cs->state & CAST_STATE_F_MEDIA_CONNECTED))
return 0;
cs->volume = 0.01 * device->volume;
cast_cfg = cfg_gettsec(cfg, "chromecast", device->name);
max_volume = cast_cfg ? cfg_getint(cast_cfg, "max_volume") : CAST_CONFIG_MAX_VOLUME;
if ((max_volume < 1) || (max_volume > CAST_CONFIG_MAX_VOLUME))
{
DPRINTF(E_LOG, L_CAST, "Config has bad max_volume (%d) for device '%s', using default instead\n", max_volume, device->name);
max_volume = CAST_CONFIG_MAX_VOLUME;
}
cs->volume = ((float)max_volume * (float)device->volume * 1.0) / (100.0 * CAST_CONFIG_MAX_VOLUME);
ret = cast_msg_send(cs, SET_VOLUME, cast_cb_volume);
if (ret < 0)
@ -2391,7 +2402,7 @@ cast_write(struct output_buffer *obuf)
cs->state = CAST_STATE_MEDIA_STREAMING;
}
// DPRINTF(E_DBG, L_CAST, "RTCP rtp %" PRIu16 ", next %" PRIu16 ", ack %" PRIu16 "\n", cast_master_session->rtp_session->seqnum, cs->seqnum_next, cs->ack_last);
// DPRINTF(E_DBG, L_CAST, "RTP last %u, have %u, ack %u\n", cs->seqnum_next - 1, cast_master_session->rtp_session->seqnum, cs->ack_last);
// We send packets to the device ping-pong style, meaning that we send the
// first packet, wait for an ack, then send the next, wait etc. This can