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Disable using linear mixer scale for the time being.

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This commit is contained in:
Mike Brady
2015-10-03 21:33:10 +01:00
parent b82952dc13
commit 2a4f404953
2 changed files with 17 additions and 6 deletions
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audio_alsa.c
+4 -1
View File
@@ -236,7 +236,8 @@ static int init(int argc, char **argv) {
(1.0 * alsa_mix_maxdb) / 100.0);
} else {
// use the linear scale and do the db conversion ourselves
debug(1, "note: the hardware mixer specified -- \"%s\" -- does not have dB volume -- using the [hopefully] linear settings.",alsa_mix_ctrl);
debug(1, "note: the hardware mixer specified -- \"%s\" -- does not have a dB volume scale, so it can't be used.",alsa_mix_ctrl);
/*
debug(1, "Min and max volumes are %d and %d.",alsa_mix_minv,alsa_mix_maxv);
alsa_mix_maxdb = 0;
if ((alsa_mix_maxv!=0) && (alsa_mix_minv!=0))
@@ -246,6 +247,7 @@ static int init(int argc, char **argv) {
audio_alsa.volume = &linear_volume; // insert the linear volume function
audio_alsa.parameters = &parameters; // likewise the parameters stuff
debug(1,"Max and min dB calculated are %d and %d.",alsa_mix_maxdb,alsa_mix_mindb);
*/
}
}
if (snd_mixer_selem_has_playback_switch(alsa_mix_elem)) {
@@ -409,6 +411,7 @@ static void parameters(audio_parameters *info) {
}
static void volume(double vol) {
// debug(1,"Volume called %f.",vol);
set_volume = vol;
double vol_setting = vol2attn(vol, alsa_mix_maxdb, alsa_mix_mindb);
// debug(1,"Setting volume db to %f, for volume input of %f.",vol_setting/100,vol);
player.c
+13 -5
View File
@@ -492,12 +492,12 @@ static abuf_t *buffer_get_frame(void) {
if (curframe->ready) {
if (ab_buffering) { // if we are getting packets but not yet forwarding them to the player
uint32_t reference_timestamp;
uint64_t reference_timestamp_time,remote_reference_timestamp_time;
get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, &remote_reference_timestamp_time);
if (first_packet_timestamp == 0) { // if this is the very first packet
// debug(1,"First frame seen, time %u, with %d
// frames...",curframe->timestamp,seq_diff(ab_read, ab_write));
uint32_t reference_timestamp;
uint64_t reference_timestamp_time,remote_reference_timestamp_time;
get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, &remote_reference_timestamp_time);
if (reference_timestamp) { // if we have a reference time
// debug(1,"First frame seen with timestamp...");
first_packet_timestamp = curframe->timestamp; // we will keep buffering until we are
@@ -527,7 +527,6 @@ static abuf_t *buffer_get_frame(void) {
// -4410 frames.
int64_t delta = ((int64_t)first_packet_timestamp - (int64_t)reference_timestamp);
first_packet_time_to_play =
reference_timestamp_time +
((delta + (int64_t)config.latency + (int64_t)config.audio_backend_latency_offset)
@@ -544,6 +543,13 @@ static abuf_t *buffer_get_frame(void) {
}
if (first_packet_time_to_play != 0) {
// recalculate first_packet_time_to_play -- the latency might change
int64_t delta = ((int64_t)first_packet_timestamp - (int64_t)reference_timestamp);
first_packet_time_to_play =
reference_timestamp_time +
((delta + (int64_t)config.latency + (int64_t)config.audio_backend_latency_offset)
<< 32) /
44100;
uint32_t filler_size = frame_size;
uint32_t max_dac_delay = 4410;
@@ -1210,12 +1216,13 @@ void player_volume(double f) {
// it back to a number.
double scaled_volume = vol2attn(f, 0, -9630);
double linear_volume = pow(10, scaled_volume / 20*100);
double linear_volume = pow(10, scaled_volume / 2000);
if (f == -144.0)
linear_volume = 0.0;
if (config.output->volume) {
// debug(1,"Set volume to %f.",f);
config.output->volume(f); // volume will be sent as metadata by the config.output device
linear_volume = 1.0; // no attenuation needed -- this value is used as a flag to avoid calculations
}
@@ -1231,6 +1238,7 @@ void player_volume(double f) {
audio_information.is_muted = 0;
}
audio_information.valid = 1;
// debug(1,"Software volume set to %f on scale with a %f dB",f,linear_volume);
pthread_mutex_lock(&vol_mutex);
software_mixer_volume = linear_volume;
fix_volume = 65536.0 * software_mixer_volume;