Remove redundant (?) calls to 'prepare_decoding_chain()'.
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@@ -274,7 +274,7 @@ void *rtp_buffered_audio_processor(void *arg) {
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if (blocks_read_since_play_began == 1) {
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debug(2, "Preparing initial decoding chain for %s.", get_ssrc_name(payload_ssrc));
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prepare_decoding_chain(conn, payload_ssrc);
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prepare_decoding_chain(conn, payload_ssrc); // needed to set the input rate...
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sequence_number_for_player =
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seq_no & 0xffff; // this is arbitrary -- the sequence_number_for_player numbers will
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// be sequential irrespective of seq_no jumps...
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@@ -290,7 +290,7 @@ void *rtp_buffered_audio_processor(void *arg) {
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"Reading a block: new encoding: %s, old encoding: %s. Preparing a new "
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"decoding chain.",
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get_ssrc_name(payload_ssrc), get_ssrc_name(previous_ssrc));
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prepare_decoding_chain(conn, payload_ssrc);
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// prepare_decoding_chain(conn, payload_ssrc);
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}
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}
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@@ -441,7 +441,7 @@ void *rtp_buffered_audio_processor(void *arg) {
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int64_t lead_time = buffer_should_be_time - get_absolute_time_in_ns();
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payload_length = 0;
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if (ssrc_is_recognised(payload_ssrc) != 0) {
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prepare_decoding_chain(conn, payload_ssrc);
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// prepare_decoding_chain(conn, payload_ssrc);
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unsigned long long new_payload_length = 0;
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payload_pointer = m + leading_free_space_length;
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if ((lead_time < (int64_t)30000000000L) &&
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@@ -517,7 +517,7 @@ void *rtp_buffered_audio_processor(void *arg) {
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} else {
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timestamp_difference = timestamp - expected_timestamp;
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if (timestamp_difference != 0) {
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debug(2,
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debug(1,
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"Connection %d: "
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"unexpected timestamp in block %u. Actual: %u, expected: %u "
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"difference: %d, "
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@@ -542,7 +542,7 @@ void *rtp_buffered_audio_processor(void *arg) {
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int32_t abs_timestamp_difference = -timestamp_difference;
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if ((size_t)abs_timestamp_difference > get_ssrc_block_length(payload_ssrc)) {
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skip_this_block = 1;
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debug(2,
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debug(1,
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"skipping block %u because it is too old. Timestamp "
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"difference: %d, length of block: %u.",
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seq_no, timestamp_difference, get_ssrc_block_length(payload_ssrc));
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