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Skip the out-of-date frames at the start of a session in AirPLay 2 Buffered Streams in iOS 16.0 / Ventura beta / iOS 16.1 beta. Only skip two blocks to avoid AAC transients and substitute silent for them.

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This commit is contained in:
Mike Brady
2022-09-22 16:28:51 +01:00
parent a6e02449fb
commit b19dd9e93c
1 changed files with 35 additions and 25 deletions
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rtp.c
+35 -25
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@@ -1328,6 +1328,7 @@ void reset_ptp_anchor_info(rtsp_conn_info *conn) {
}
int long_time_notifcation_done = 0;
int get_ptp_anchor_local_time_info(rtsp_conn_info *conn, uint32_t *anchorRTP,
uint64_t *anchorLocalTime) {
int response = clock_not_valid;
@@ -2374,13 +2375,13 @@ void *rtp_buffered_audio_processor(void *arg) {
int pcm_buffer_occupancy = 0;
int pcm_buffer_read_point = 0; // offset to where the next buffer should come from
uint32_t pcm_buffer_read_point_rtptime = 0;
// uint32_t expected_pcm_buffer_read_point_rtptime = 0;
uint32_t expected_pcm_buffer_read_point_rtptime = 0;
uint64_t blocks_read = 0;
uint64_t blocks_read_since_flush = 0;
int flush_requested = 0;
uint32_t expected_timestamp = 0;
// int expected_timesamp_is_reasonable = 0;
int expected_timesamp_is_reasonable = 0;
uint32_t timestamp = 0; // initialised to avoid a "possibly uninitialised" warning.
int streaming_has_started = 0;
int play_enabled = 0;
@@ -2569,28 +2570,37 @@ void *rtp_buffered_audio_processor(void *arg) {
// debug(1,"block timestamp: %u, packet timestamp: %u.", timestamp, pcm_buffer_read_point_rtptime);
/*
int32_t timestamp_difference = pcm_buffer_read_point_rtptime - expected_pcm_buffer_read_point_rtptime;;
if (streaming_has_started != 0) {
int32_t timestamp_difference = pcm_buffer_read_point_rtptime - expected_pcm_buffer_read_point_rtptime;
if (timestamp_difference != 0)
debug(1,"Unexpected time difference between packets -- actual: %u, expected: %u, difference: %d. Packets played: %d. Blocks played since flush: %d. ",
pcm_buffer_read_point_rtptime, expected_pcm_buffer_read_point_rtptime, timestamp_difference, streaming_has_started, blocks_read_since_flush);
}
*/
// if it's not the very first block of AAC, but is from the first few blocks of a new AAC sequence,
// it will contain noisy transients, so replace it with silence.
if ((blocks_read != 1) && (blocks_read_since_flush <= 3)) {
// debug(1,"muting packet %u from block %u because it's not from a true starting block and blocks_read_since_flush is %" PRIu64 ".", pcm_buffer_read_point_rtptime, timestamp, blocks_read_since_flush);
conn->previous_random_number =
generate_zero_frames((char *)(pcm_buffer + pcm_buffer_read_point), 352, config.output_format, conn->enable_dither,
conn->previous_random_number);
if (streaming_has_started != 64) // don't log this situation because it's being dealt with below
debug(1,"Unexpected time difference between packets -- actual: %u, expected: %u, difference: %d. Packets played: %d. Blocks played since flush: %d. ",
pcm_buffer_read_point_rtptime, expected_pcm_buffer_read_point_rtptime, timestamp_difference, streaming_has_started, blocks_read_since_flush);
}
player_put_packet(0, 0, pcm_buffer_read_point_rtptime,
pcm_buffer + pcm_buffer_read_point, 352, conn);
streaming_has_started++;
// expected_pcm_buffer_read_point_rtptime = pcm_buffer_read_point_rtptime + 352;
// Very specific code to get around an apparent bug in AirPlay 2 from iOS 16 / Ventura 13.0
// It seems that the timestamp goes backwards by 2112 frames after the first 64 packets of 352 frames (64 * 352 = 22528 frames which is exactly 22 blocks)
// So, if the timestamp is before the expected timestamp when 64 packets have been played, then skip that packet.
// Put an arbitrary last-ditch limit of 64 blocks_read_since_flush in case it all goes wrong...
if ((timestamp_difference < 0) && (streaming_has_started == 64) && (blocks_read_since_flush < 64)) {
// debug(1,"skipping packet %u.", pcm_buffer_read_point_rtptime);
} else {
// if it's not the very first block of AAC, but is from the first few blocks of a new AAC sequence,
// it will contain noisy transients, so replace it with silence.
if ((blocks_read_since_flush <= 2) && (blocks_read_since_flush != blocks_read)) {
// debug(1,"Muting packet %u from block %u to avoid AAC transients because it's not from a true starting block. Blocks_read is %" PRIu64 ". Blocks_read_since_flush is %" PRIu64 ".", pcm_buffer_read_point_rtptime, timestamp, blocks_read, blocks_read_since_flush);
conn->previous_random_number =
generate_zero_frames((char *)(pcm_buffer + pcm_buffer_read_point), 352, config.output_format, conn->enable_dither,
conn->previous_random_number);
}
player_put_packet(0, 0, pcm_buffer_read_point_rtptime,
pcm_buffer + pcm_buffer_read_point, 352, conn);
streaming_has_started++;
expected_pcm_buffer_read_point_rtptime = pcm_buffer_read_point_rtptime + 352;
}
}
// }
} else {
@@ -2680,11 +2690,11 @@ void *rtp_buffered_audio_processor(void *arg) {
seq_no = packet[1] * (1 << 16) + packet[2] * (1 << 8) + packet[3];
timestamp = nctohl(&packet[4]);
// debug(1,"New block timestamp: %u.", timestamp);
// int32_t timestamp_difference = timestamp - expected_timestamp;
// if ((timestamp_difference != 0) && (expected_timesamp_is_reasonable != 0))
// debug(1, "Unexpected timestamp. Expected: %u, got: %u, difference: %d, blocks_read_since_flush: %" PRIu64 ".", expected_timestamp, timestamp, timestamp_difference, blocks_read_since_flush);
int32_t timestamp_difference = timestamp - expected_timestamp;
if ((timestamp_difference != 0) && (expected_timesamp_is_reasonable != 0))
debug(1, "Block with unexpected timestamp. Expected: %u, got: %u, difference: %d, blocks_read_since_flush: %" PRIu64 ".", expected_timestamp, timestamp, timestamp_difference, blocks_read_since_flush);
expected_timestamp = timestamp;
// expected_timesamp_is_reasonable = 0; // must be validated each time by decoding the frame
expected_timesamp_is_reasonable = 0; // must be validated each time by decoding the frame
// debug(1, "immediately: block %u, rtptime %u", seq_no, timestamp);
// uint32_t ssrc = nctohl(&packet[8]);
@@ -2885,7 +2895,7 @@ void *rtp_buffered_audio_processor(void *arg) {
} else {
memcpy(pcm_buffer + pcm_buffer_occupancy, pcm_audio, dst_bufsize);
expected_timestamp += (dst_bufsize/conn->input_bytes_per_frame);
// expected_timesamp_is_reasonable = 1;
expected_timesamp_is_reasonable = 1;
pcm_buffer_occupancy += dst_bufsize;
// debug(1,"frames added: pcm_buffer_read_point (frames): %u, pcm_buffer_occupancy (frames): %u",
// pcm_buffer_read_point/conn->input_bytes_per_frame, pcm_buffer_occupancy/conn->input_bytes_per_frame);