nils/shairport-sync
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Begin to add in calculations for expanded back end

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This commit is contained in:
Mike Brady
2016-08-19 18:25:59 +01:00
parent 0c72f73d77
commit 7bd82dad0e
1 changed files with 34 additions and 6 deletions
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player.c
+34 -6
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@@ -66,6 +66,7 @@
#include "rtsp.h"
#include "alac.h"
#include <alsa/pcm.h>
// parameters from the source
static unsigned char *aesiv;
@@ -75,9 +76,11 @@ static AES_KEY aes;
static int sampling_rate, frame_size;
static int bytes_per_audio_frame = 4;
static int bytes_per_output_audio_frame;
// The maximum frame size change there can be is +/- 1;
static int max_frame_size_change = 1;
static int max_frame_size_change;
// #define FRAME_BYTES(frame_size) (4 * frame_size)
// maximal resampling shift - conservative
//#define OUTFRAME_BYTES(frame_size) (4 * (frame_size + 3))
@@ -994,8 +997,28 @@ static void *player_thread_func(void *arg) {
pthread_create(&rtp_control_thread, NULL, &rtp_control_receiver, (void *)&itr);
pthread_create(&rtp_timing_thread, NULL, &rtp_timing_receiver, (void *)&itr);
session_corrections = 0;
play_segment_reference_frame = 0; // zero signals that we are not in a play segment
session_corrections = 0;
play_segment_reference_frame = 0; // zero signals that we are not in a play segment
int output_sample_ratio = 1;
if (config.output_rate!=0)
output_sample_ratio = config.output_rate/44100;
max_frame_size_change = 1*sample_ratio;
bytes_per_output_audio_frame = 4;
switch (config.output_format) {
case SND_PCM_FORMAT_S24_LE:
bytes_per_output_audio_frame=6;
break;
case SND_PCM_FORMAT_S32_LE:
bytes_per_output_audio_frame=8;
break;
}
// check that there are enough buffers to accommodate the desired latency and the latency offset
int maximum_latency = config.latency+config.audio_backend_latency_offset;
@@ -1034,13 +1057,16 @@ static void *player_thread_func(void *arg) {
initstate(time(NULL), rnstate, 256);
signed short *inbuf, *outbuf, *silence;
outbuf = malloc(bytes_per_audio_frame*(frame_size+max_frame_size_change));
// We might need an output buffer and a buffer of silence.
// The size of these dependents on the number of frames, the size of each frame and the maximum size change
outbuf = malloc(bytes_per_audio_frame*(frame_size*output_sample_ratio+max_frame_size_change));
if (outbuf==NULL)
debug(1,"Failed to allocate memory for an output buffer.");
silence = malloc(bytes_per_audio_frame*frame_size);
if (silence==NULL)
debug(1,"Failed to allocate memory for a silence buffer.");
memset(silence, 0, bytes_per_audio_frame*frame_size);
memset(silence, 0, bytes_per_audio_frame*frame_size*output_sample_ratio);
late_packet_message_sent = 0;
first_packet_timestamp = 0;
missing_packets = late_packets = too_late_packets = resend_requests = 0;
@@ -1242,7 +1268,7 @@ static void *player_thread_func(void *arg) {
if (config.no_sync!=0)
amount_to_stuff = 0 ; // no stuffing if it's been disabled
if ((amount_to_stuff == 0) && (fix_volume == 0x10000)) {
if ((amount_to_stuff == 0) && (fix_volume == 0x10000) && ((config.output_rate==0) || (config.output_rate==44100)) && ((config.output_format==0) || (config.output_format==SND_PCM_FORMAT_S16_LE))) {
// if no stuffing needed and no volume adjustment, then
// don't send to stuff_buffer_* and don't copy to outbuf; just send directly to the
// output device...
@@ -1255,6 +1281,8 @@ static void *player_thread_func(void *arg) {
config.output->play(inframe->data, inframe->length);
}
} else {
#ifdef HAVE_LIBSOXR
switch (config.packet_stuffing) {
case ST_basic: