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This commit is contained in:
Mike Brady
2022-09-23 17:05:26 +01:00
parent c433e2e87d
commit 7dc077e260
1 changed files with 74 additions and 47 deletions
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rtp.c
+74 -47
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@@ -2588,48 +2588,66 @@ void *rtp_buffered_audio_processor(void *arg) {
not_first_time_out = 1;
}
*/
// clang-format on
// clang-format on
// debug(1,"block timestamp: %u, packet timestamp: %u.", timestamp, pcm_buffer_read_point_rtptime);
// 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 (packets_played_in_this_sequence != 0) {
if (timestamp_difference != 0)
debug(2,"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, packets_played_in_this_sequence, blocks_read_in_sequence);
}
// 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 not later than the 65th packet of 352 frames (64 * 352 = 22528 frames which is exactly 22 blocks)
// So, if that happens, we'll add 2112 to the timstamp passed to the player
if ((timestamp_difference == -2112) && (packets_played_in_this_sequence <= 64)) {
debug(1,"iOS 16.0 discontinuity detected with %d packets played in this sequence. Nothing done.", packets_played_in_this_sequence);
// pcm_buffer_read_point_rtptime_offset = 2112; // this pretends the timestamps after the discontinuity are 2112 frames later, but this just delays everything by 2112 frames, pushing stuff out of sync, and i think you can hear it.
}
// 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_in_sequence <= 2) && (blocks_read_in_sequence != 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_in_sequence is %" PRIu64 ".", pcm_buffer_read_point_rtptime, timestamp, blocks_read, blocks_read_in_sequence);
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_read_point_rtptime_offset,
pcm_buffer + pcm_buffer_read_point, 352, conn);
packets_played_in_this_sequence++;
expected_pcm_buffer_read_point_rtptime = pcm_buffer_read_point_rtptime + 352;
int32_t timestamp_difference =
pcm_buffer_read_point_rtptime - expected_pcm_buffer_read_point_rtptime;
;
if (packets_played_in_this_sequence != 0) {
if (timestamp_difference != 0)
debug(
2,
"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, packets_played_in_this_sequence,
blocks_read_in_sequence);
}
// 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 not
// later than the 65th packet of 352 frames (64 * 352 = 22528 frames which is
// exactly 22 blocks) So, if that happens, we'll add 2112 to the timstamp passed
// to the player
if ((timestamp_difference == -2112) && (packets_played_in_this_sequence <= 64)) {
debug(1,
"iOS 16.0 discontinuity detected with %d packets played in this "
"sequence. Nothing done.",
packets_played_in_this_sequence);
// pcm_buffer_read_point_rtptime_offset = 2112; // this pretends the timestamps
// after the discontinuity are 2112 frames later, but this just delays
// everything by 2112 frames, pushing stuff out of sync, and i think you can
// hear it.
}
// 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_in_sequence <= 2) && (blocks_read_in_sequence != 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_in_sequence is %" PRIu64 ".", pcm_buffer_read_point_rtptime,
// timestamp, blocks_read, blocks_read_in_sequence);
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_read_point_rtptime_offset,
pcm_buffer + pcm_buffer_read_point, 352, conn);
packets_played_in_this_sequence++;
expected_pcm_buffer_read_point_rtptime = pcm_buffer_read_point_rtptime + 352;
}
// }
} else {
debug(3,
"Dropping packet %u from block %u with out-of-range lead_time: %.3f seconds.",
pcm_buffer_read_point_rtptime, seq_no, 0.000000001 * lead_time);
expected_pcm_buffer_read_point_rtptime = pcm_buffer_read_point_rtptime + 352;
expected_pcm_buffer_read_point_rtptime = pcm_buffer_read_point_rtptime + 352;
}
pcm_buffer_read_point_rtptime += 352;
@@ -2642,13 +2660,16 @@ void *rtp_buffered_audio_processor(void *arg) {
usleep(20000); // wait before asking if play is enabled again
}
} else {
// debug(1,"new buffer needed for buffer starting at %u because pcm_buffer_read_point (frames) is %u and pcm_buffer_occupancy (frames) is %u.", pcm_buffer_read_point_rtptime, pcm_buffer_read_point/conn->input_bytes_per_frame,
// debug(1,"new buffer needed for buffer starting at %u because pcm_buffer_read_point
// (frames) is %u and pcm_buffer_occupancy (frames) is %u.",
// pcm_buffer_read_point_rtptime, pcm_buffer_read_point/conn->input_bytes_per_frame,
// pcm_buffer_occupancy/conn->input_bytes_per_frame);
new_buffer_needed = 1;
if (pcm_buffer_read_point != 0) {
// debug(1,"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); // if there is anything to move down
// pcm_buffer_read_point/conn->input_bytes_per_frame,
// pcm_buffer_occupancy/conn->input_bytes_per_frame); // if there is anything to move
// down
// to the front of the buffer, do it now;
if ((pcm_buffer_occupancy - pcm_buffer_read_point) > 0) {
// move the remaining frames down to the start of the buffer
@@ -2668,9 +2689,9 @@ void *rtp_buffered_audio_processor(void *arg) {
if ((flush_requested) || (new_buffer_needed)) {
// debug(1,"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);
// ok, so here we know we need material from the sender
// do we will get in a packet of audio
// pcm_buffer_read_point/conn->input_bytes_per_frame,
// pcm_buffer_occupancy/conn->input_bytes_per_frame); ok, so here we know we need material
// from the sender do we will get in a packet of audio
uint16_t data_len;
// here we read from the buffer that our thread has been reading
size_t bytes_remaining_in_buffer;
@@ -2714,13 +2735,17 @@ void *rtp_buffered_audio_processor(void *arg) {
previous_seq_no++;
seq_no = packet[1] * (1 << 16) + packet[2] * (1 << 8) + packet[3];
if (previous_seq_no != seq_no) {
debug(2,"block sequence number changed from expected %u to actual %u.", previous_seq_no, seq_no);
debug(2, "block sequence number changed from expected %u to actual %u.", previous_seq_no,
seq_no);
}
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(2, "Block with unexpected timestamp. Expected: %u, got: %u, difference: %d, blocks_read_in_sequence: %" PRIu64 ".", expected_timestamp, timestamp, timestamp_difference, blocks_read_in_sequence);
if ((timestamp_difference != 0) && (expected_timesamp_is_reasonable != 0))
debug(2,
"Block with unexpected timestamp. Expected: %u, got: %u, difference: %d, "
"blocks_read_in_sequence: %" PRIu64 ".",
expected_timestamp, timestamp, timestamp_difference, blocks_read_in_sequence);
expected_timestamp = timestamp;
expected_timesamp_is_reasonable = 0; // must be validated each time by decoding the frame
@@ -2922,12 +2947,14 @@ void *rtp_buffered_audio_processor(void *arg) {
pcm_buffer_size);
} else {
memcpy(pcm_buffer + pcm_buffer_occupancy, pcm_audio, dst_bufsize);
expected_timestamp += (dst_bufsize/conn->input_bytes_per_frame);
expected_timestamp += (dst_bufsize / conn->input_bytes_per_frame);
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);
}
// 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);
}
// debug(1,"decoded %d samples", decoded_frame->nb_samples);
// memcpy(sampleBuffer,outputBuffer16,dst_bufsize);
av_freep(&pcm_audio);