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De-globalise max_frames_per_packet,input_num_channels,input_bit_depth,input_rate

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This commit is contained in:
Mike Brady
2017-03-09 14:26:28 +00:00
parent d90fd0df50
commit cae9337dc4
2 changed files with 27 additions and 28 deletions
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player.c
+26 -28
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@@ -76,8 +76,6 @@
#include "apple_alac.h"
#endif
static int input_rate, input_bit_depth, input_num_channels, max_frames_per_packet;
static uint32_t timestamp_epoch, last_timestamp,
maximum_timestamp_interval; // timestamp_epoch of zero means not initialised, could start at 2
// or 1.
@@ -367,7 +365,7 @@ static int alac_decode(short *dest, int *destlen, uint8_t *buf, int len, rtsp_co
return reply;
}
static int init_decoder(int32_t fmtp[12]) {
static int init_decoder(int32_t fmtp[12],rtsp_conn_info* conn) {
// This is a guess, but the format of the fmtp looks identical to the format of an
// ALACSpecificCOnfig
@@ -448,22 +446,22 @@ static int init_decoder(int32_t fmtp[12]) {
alac_file *alac;
max_frames_per_packet = fmtp[1]; // number of audio frames per packet.
conn->max_frames_per_packet = fmtp[1]; // number of audio frames per packet.
input_rate = fmtp[11];
input_num_channels = fmtp[7];
input_bit_depth = fmtp[3];
conn->input_rate = fmtp[11];
conn->input_num_channels = fmtp[7];
conn->input_bit_depth = fmtp[3];
input_bytes_per_frame = input_num_channels * ((input_bit_depth + 7) / 8);
input_bytes_per_frame = conn->input_num_channels * ((conn->input_bit_depth + 7) / 8);
alac = alac_create(input_bit_depth, input_num_channels);
alac = alac_create(conn->input_bit_depth, conn->input_num_channels);
if (!alac)
return 1;
decoder_info = alac;
alac->setinfo_max_samples_per_frame = max_frames_per_packet;
alac->setinfo_max_samples_per_frame = conn->max_frames_per_packet;
alac->setinfo_7a = fmtp[2];
alac->setinfo_sample_size = input_bit_depth;
alac->setinfo_sample_size = conn->input_bit_depth;
alac->setinfo_rice_historymult = fmtp[4];
alac->setinfo_rice_initialhistory = fmtp[5];
alac->setinfo_rice_kmodifier = fmtp[6];
@@ -492,7 +490,7 @@ static void init_buffer(rtsp_conn_info* conn) {
int i;
for (i = 0; i < BUFFER_FRAMES; i++)
conn->audio_buffer[i].data =
malloc(input_bytes_per_frame * (max_frames_per_packet + max_frame_size_change));
malloc(input_bytes_per_frame * (conn->max_frames_per_packet + max_frame_size_change));
ab_resync(conn);
}
@@ -580,7 +578,7 @@ void player_put_packet(seq_t seqno, int64_t timestamp, uint8_t *data, int len, r
// pthread_mutex_unlock(&ab_mutex);
if (abuf) {
int datalen = max_frames_per_packet;
int datalen = conn->max_frames_per_packet;
if (alac_decode(abuf->data, &datalen, data, len, conn) == 0) {
abuf->ready = 1;
abuf->length = datalen;
@@ -998,7 +996,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info* conn) {
max_dac_delay, dac_delay);
fs = 0;
}
if ((exact_frame_gap <= fs) || (exact_frame_gap <= max_frames_per_packet * 2)) {
if ((exact_frame_gap <= fs) || (exact_frame_gap <= conn->max_frames_per_packet * 2)) {
fs = exact_frame_gap;
// debug(1,"Exact frame gap is %llu; play %d frames of silence. Dac_delay is %d,
// with %d packets, ab_read is %04x, ab_write is
@@ -1105,7 +1103,7 @@ static abuf_t *buffer_get_frame(rtsp_conn_info* conn) {
if (wait) {
uint64_t time_to_wait_for_wakeup_fp =
((uint64_t)1 << 32) / input_rate; // this is time period of one frame
((uint64_t)1 << 32) / conn->input_rate; // this is time period of one frame
time_to_wait_for_wakeup_fp *= 4 * 352; // four full 352-frame packets
time_to_wait_for_wakeup_fp /= 3; // four thirds of a packet time
@@ -1473,7 +1471,7 @@ static void *player_thread_func(void *arg) {
conn->please_stop = 0;
conn->packet_count = 0;
init_decoder((int32_t *)&conn->stream.fmtp); // this sets up incoming rate, bit depth, channels
init_decoder((int32_t *)&conn->stream.fmtp,conn); // this sets up incoming rate, bit depth, channels
// must be after decoder init
init_buffer(conn);
@@ -1496,11 +1494,11 @@ static void *player_thread_func(void *arg) {
}
timestamp_epoch = 0; // indicate that the next timestamp will be the first one.
maximum_timestamp_interval = input_rate * 60; // actually there shouldn't be more than about 13v
maximum_timestamp_interval = conn->input_rate * 60; // actually there shouldn't be more than about 13v
// seconds of a gap between successive rtptimes, at
// worst
output_sample_ratio = config.output_rate / input_rate;
output_sample_ratio = config.output_rate / conn->input_rate;
debug(1, "Output sample ratio is %d.", output_sample_ratio);
@@ -1532,7 +1530,7 @@ static void *player_thread_func(void *arg) {
// 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 * input_rate;
int maximum_latency = config.latency + config.audio_backend_latency_offset * conn->input_rate;
if ((maximum_latency + (352 - 1)) / 352 + 10 > BUFFER_FRAMES)
die("Not enough buffers available for a total latency of %d frames. A maximum of %d 352-frame "
"packets may be accommodated.",
@@ -1599,7 +1597,7 @@ static void *player_thread_func(void *arg) {
debug(1, "Output bit depth is %d.", output_bit_depth);
if (input_bit_depth > output_bit_depth) {
if (conn->input_bit_depth > output_bit_depth) {
debug(1, "Dithering will be enabled because the input bit depth is greater than the output bit "
"depth");
}
@@ -1616,13 +1614,13 @@ static void *player_thread_func(void *arg) {
// debug(1,"Define tbuf of length
// %d.",output_bytes_per_frame*(max_frames_per_packet*output_sample_ratio+max_frame_size_change));
tbuf = malloc(sizeof(int32_t) * 2 *
(max_frames_per_packet * output_sample_ratio + max_frame_size_change));
(conn->max_frames_per_packet * output_sample_ratio + max_frame_size_change));
if (tbuf == NULL)
debug(1, "Failed to allocate memory for the transition buffer.");
sbuf = 0;
if (config.packet_stuffing == ST_soxr) { // needed for stuffing
sbuf = malloc(sizeof(int32_t) * 2 *
(max_frames_per_packet * output_sample_ratio + max_frame_size_change));
(conn->max_frames_per_packet * output_sample_ratio + max_frame_size_change));
if (sbuf == NULL)
debug(1, "Failed to allocate memory for the transition buffer.");
}
@@ -1634,13 +1632,13 @@ static void *player_thread_func(void *arg) {
// The size of these dependents on the number of frames, the size of each frame and the maximum
// size change
outbuf = malloc(output_bytes_per_frame *
(max_frames_per_packet * output_sample_ratio + max_frame_size_change));
(conn->max_frames_per_packet * output_sample_ratio + max_frame_size_change));
if (outbuf == NULL)
debug(1, "Failed to allocate memory for an output buffer.");
silence = malloc(output_bytes_per_frame * max_frames_per_packet * output_sample_ratio);
silence = malloc(output_bytes_per_frame * conn->max_frames_per_packet * output_sample_ratio);
if (silence == NULL)
debug(1, "Failed to allocate memory for a silence buffer.");
memset(silence, 0, output_bytes_per_frame * max_frames_per_packet * output_sample_ratio);
memset(silence, 0, output_bytes_per_frame * conn->max_frames_per_packet * output_sample_ratio);
late_packet_message_sent = 0;
first_packet_timestamp = 0;
missing_packets = late_packets = too_late_packets = resend_requests = 0;
@@ -1698,17 +1696,17 @@ static void *player_thread_func(void *arg) {
// debug(1,"Player has a supplied silent frame.");
last_seqno_read =
(SUCCESSOR(last_seqno_read) & 0xffff); // manage the packet out of sequence minder
config.output->play(silence, max_frames_per_packet * output_sample_ratio);
config.output->play(silence, conn->max_frames_per_packet * output_sample_ratio);
} else {
int enable_dither = 0;
if ((fix_volume != 0x10000) || (input_bit_depth > output_bit_depth))
if ((fix_volume != 0x10000) || (conn->input_bit_depth > output_bit_depth))
enable_dither = 1;
// here, let's transform the frame of data, if necessary
if (tbuf != NULL) { // this will be null if no changes are needed
switch (input_bit_depth) {
switch (conn->input_bit_depth) {
case 16: {
int i, j;
int16_t ls, rs;
player.h
+1
View File
@@ -50,6 +50,7 @@ typedef struct {
pthread_t player_thread;
abuf_t audio_buffer[BUFFER_FRAMES];
int max_frames_per_packet,input_num_channels,input_bit_depth,input_rate;
uint32_t please_stop;
uint64_t packet_count;
#ifdef HAVE_LIBMBEDTLS