Check for valid time information more completely.
This commit is contained in:
Mike Brady
+171
-172
@@ -495,196 +495,195 @@ void *rtp_buffered_audio_processor(void *arg) {
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// But it also means that Shairport Sync's buffers must be sufficient to hold all the
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// entire lead-time's amount of audio in case the device has a zero-sized buffer.
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if ((play_enabled != 0) &&
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(have_valid_time != 0)
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// (audio_decoded_buffer_below_desired_length != 0) &&
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&& (lead_time * 1E-9 < (config.audio_decoded_buffer_desired_length + 0.1))
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// && (audio_decoded_buffer_below_desired_length != 0)
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) {
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very_early_packets_signalled = 0; // reset very early packet warning signaller
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// try to identify blocks that are timed to before the last buffer, and drop 'em
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int64_t time_from_last_buffer_time =
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buffer_should_be_time - previous_buffer_should_be_time;
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if ((packets_played_in_this_sequence == 0) || (time_from_last_buffer_time > 0)) {
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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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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 >= 0) { // only decipher the packet if it's not too late
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int response = -1; // guess that there is a problem
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if (conn->session_key != NULL) {
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unsigned char nonce[12];
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memset(nonce, 0, sizeof(nonce));
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memcpy(
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nonce + 4, packet + nread - 8,
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8); // front-pad the 8-byte nonce received to get the 12-byte nonce expected
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// https://libsodium.gitbook.io/doc/secret-key_cryptography/aead/chacha20-poly1305/ietf_chacha20-poly1305_construction
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// Note: the eight-byte nonce must be front-padded out to 12 bytes.
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// Leave leading_free_space_length bytes at the start for possible headers like an
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// ADTS header (7 bytes)
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memset(m, 0, leading_free_space_length);
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response = crypto_aead_chacha20poly1305_ietf_decrypt(
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payload_pointer, // where the decrypted payload will start
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&new_payload_length, // mlen_p
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NULL, // nsec,
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packet +
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12, // the ciphertext starts 12 bytes in and is followed by the MAC tag,
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nread - (8 + 12), // clen -- the last 8 bytes are the nonce
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packet + 4, // authenticated additional data
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8, // authenticated additional data length
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nonce,
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conn->session_key); // *k
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if (response != 0)
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debug(1, "Error decrypting audio packet %u -- packet length %zd.", seq_no,
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nread);
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} else {
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debug(2, "No session key, so the audio packet can not be deciphered -- skipped.");
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}
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if ((response == 0) && (new_payload_length > 0)) {
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// now we have the deciphered block, so send it to the player if we can
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payload_length = new_payload_length;
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if (ssrc_is_aac(payload_ssrc)) {
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payload_pointer =
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payload_pointer - 7; // including the 7-byte leader for the ADTS
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payload_length = payload_length + 7;
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// now, fill in the 7-byte ADTS information, which seems to be needed by the
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// decoder we made room for it in the front of the buffer by filling from m + 7.
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int channelConfiguration = 2; // 2: 2 channels: front-left, front-right
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if (payload_ssrc == AAC_48000_F24_5P1)
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channelConfiguration = 6; // 6: 6 channels: front-center, front-left,
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// front-right, back-left, back-right, LFE-channel
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else if (payload_ssrc == AAC_48000_F24_7P1)
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channelConfiguration =
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7; // 7: 8 channels: front-center, front-left, front-right,
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// side-left, side-right, back-left, back-right, LFE-channel
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addADTStoPacket(payload_pointer, payload_length, conn->input_rate,
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channelConfiguration);
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}
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int mute =
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((packets_played_in_this_sequence == 0) && (ssrc_is_aac(payload_ssrc)));
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if (mute) {
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debug(2, "Connection %d: muting first AAC block -- block %u -- timestamp %u.",
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conn->connection_number, seq_no, timestamp);
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}
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int32_t timestamp_difference = 0;
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if (packets_played_in_this_sequence == 0) {
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// first_block_in_this_sequence = seq_no;
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first_timestamp_in_this_sequence = timestamp;
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debug(2,
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"Connection %d: "
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"first block %u, first timestamp %u.",
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conn->connection_number, seq_no, timestamp);
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if (have_valid_time != 0) {
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if ((play_enabled != 0) && (lead_time * 1E-9 < (config.audio_decoded_buffer_desired_length + 0.1))) {
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// && (audio_decoded_buffer_below_desired_length != 0)
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very_early_packets_signalled = 0; // reset very early packet warning signaller
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// try to identify blocks that are timed to before the last buffer, and drop 'em
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int64_t time_from_last_buffer_time =
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buffer_should_be_time - previous_buffer_should_be_time;
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if ((packets_played_in_this_sequence == 0) || (time_from_last_buffer_time > 0)) {
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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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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 >= 0) { // only decipher the packet if it's not too late
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int response = -1; // guess that there is a problem
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if (conn->session_key != NULL) {
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unsigned char nonce[12];
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memset(nonce, 0, sizeof(nonce));
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memcpy(
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nonce + 4, packet + nread - 8,
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8); // front-pad the 8-byte nonce received to get the 12-byte nonce expected
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// https://libsodium.gitbook.io/doc/secret-key_cryptography/aead/chacha20-poly1305/ietf_chacha20-poly1305_construction
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// Note: the eight-byte nonce must be front-padded out to 12 bytes.
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// Leave leading_free_space_length bytes at the start for possible headers like an
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// ADTS header (7 bytes)
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memset(m, 0, leading_free_space_length);
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response = crypto_aead_chacha20poly1305_ietf_decrypt(
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payload_pointer, // where the decrypted payload will start
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&new_payload_length, // mlen_p
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NULL, // nsec,
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packet +
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12, // the ciphertext starts 12 bytes in and is followed by the MAC tag,
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nread - (8 + 12), // clen -- the last 8 bytes are the nonce
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packet + 4, // authenticated additional data
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8, // authenticated additional data length
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nonce,
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conn->session_key); // *k
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if (response != 0)
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debug(1, "Error decrypting audio packet %u -- packet length %zd.", seq_no,
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nread);
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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, "No session key, so the audio packet can not be deciphered -- skipped.");
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}
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if ((response == 0) && (new_payload_length > 0)) {
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// now we have the deciphered block, so send it to the player if we can
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payload_length = new_payload_length;
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if (ssrc_is_aac(payload_ssrc)) {
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payload_pointer =
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payload_pointer - 7; // including the 7-byte leader for the ADTS
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payload_length = payload_length + 7;
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// now, fill in the 7-byte ADTS information, which seems to be needed by the
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// decoder we made room for it in the front of the buffer by filling from m + 7.
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int channelConfiguration = 2; // 2: 2 channels: front-left, front-right
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if (payload_ssrc == AAC_48000_F24_5P1)
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channelConfiguration = 6; // 6: 6 channels: front-center, front-left,
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// front-right, back-left, back-right, LFE-channel
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else if (payload_ssrc == AAC_48000_F24_7P1)
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channelConfiguration =
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7; // 7: 8 channels: front-center, front-left, front-right,
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// side-left, side-right, back-left, back-right, LFE-channel
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addADTStoPacket(payload_pointer, payload_length, conn->input_rate,
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channelConfiguration);
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}
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int mute =
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((packets_played_in_this_sequence == 0) && (ssrc_is_aac(payload_ssrc)));
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if (mute) {
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debug(2, "Connection %d: muting first AAC block -- block %u -- timestamp %u.",
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conn->connection_number, seq_no, timestamp);
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}
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int32_t timestamp_difference = 0;
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if (packets_played_in_this_sequence == 0) {
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// first_block_in_this_sequence = seq_no;
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first_timestamp_in_this_sequence = timestamp;
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debug(2,
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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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"%f ms. "
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"Positive means later, i.e. a gap. First timestamp was %u, payload "
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"type: \"%s\".",
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conn->connection_number, seq_no, timestamp, expected_timestamp,
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timestamp_difference, 1000.0 * timestamp_difference / conn->input_rate,
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first_timestamp_in_this_sequence, get_ssrc_name(payload_ssrc));
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// mute the first packet after a discontinuity
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if (ssrc_is_aac(payload_ssrc)) {
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"first block %u, first timestamp %u.",
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conn->connection_number, seq_no, timestamp);
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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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"Connection %d: muting first AAC block -- block %u -- following a "
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"timestamp discontinuity, timestamp %u.",
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conn->connection_number, seq_no, timestamp);
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mute = 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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"%f ms. "
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"Positive means later, i.e. a gap. First timestamp was %u, payload "
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"type: \"%s\".",
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conn->connection_number, seq_no, timestamp, expected_timestamp,
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timestamp_difference, 1000.0 * timestamp_difference / conn->input_rate,
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first_timestamp_in_this_sequence, get_ssrc_name(payload_ssrc));
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// mute the first packet after a discontinuity
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if (ssrc_is_aac(payload_ssrc)) {
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debug(2,
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"Connection %d: muting first AAC block -- block %u -- following a "
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"timestamp discontinuity, timestamp %u.",
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conn->connection_number, seq_no, timestamp);
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mute = 1;
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}
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}
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}
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}
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int skip_this_block = 0;
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if (timestamp_difference < 0) {
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// uncomment this to work back to replace buffers that have been already decoded
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// and placed in the player queue with the incoming new buffers this is a bit
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// trickier, but maybe the new buffers are better than the previous ones they
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// will replace (?)
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/*
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seq_t revised_seqno = get_revised_seqno(conn, timestamp);
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if (revised_seqno != sequence_number_for_player) {
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debug(1, "revised seqno calculated: conn->ab_read: %u, revised_seqno: %u,
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conn->ab_write: %u.", conn->ab_read, revised_seqno, conn->ab_write);
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clear_buffers_from(conn, revised_seqno);
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sequence_number_for_player = revised_seqno;
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timestamp_difference = 0;
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int skip_this_block = 0;
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if (timestamp_difference < 0) {
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// uncomment this to work back to replace buffers that have been already decoded
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// and placed in the player queue with the incoming new buffers this is a bit
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// trickier, but maybe the new buffers are better than the previous ones they
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// will replace (?)
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/*
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seq_t revised_seqno = get_revised_seqno(conn, timestamp);
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if (revised_seqno != sequence_number_for_player) {
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debug(1, "revised seqno calculated: conn->ab_read: %u, revised_seqno: %u,
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conn->ab_write: %u.", conn->ab_read, revised_seqno, conn->ab_write);
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clear_buffers_from(conn, revised_seqno);
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sequence_number_for_player = revised_seqno;
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timestamp_difference = 0;
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}
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*/
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// uncomment this to drop incoming new buffers that are too old and for whose
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// timings buffers have already been decoded and placed in the player queue this
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// is easier, but maybe the new late buffers are better than the previous ones
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// (?)
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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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"skipping block %u because it is too old. Timestamp "
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"difference: %d, length of block: %zu.",
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seq_no, timestamp_difference, get_ssrc_block_length(payload_ssrc));
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}
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}
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*/
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// uncomment this to drop incoming new buffers that are too old and for whose
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// timings buffers have already been decoded and placed in the player queue this
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// is easier, but maybe the new late buffers are better than the previous ones
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// (?)
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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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"skipping block %u because it is too old. Timestamp "
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"difference: %d, length of block: %zu.",
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seq_no, timestamp_difference, get_ssrc_block_length(payload_ssrc));
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if (skip_this_block == 0) {
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uint32_t packet_size = player_put_packet(
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payload_ssrc, sequence_number_for_player, timestamp, payload_pointer,
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payload_length, mute, timestamp_difference, conn);
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debug(4, "block %u, timestamp %u, length %u sent to the player.", seq_no,
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timestamp, packet_size);
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sequence_number_for_player++; // simply increment
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expected_timestamp = timestamp + packet_size; // for the next time
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packets_played_in_this_sequence++;
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}
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}
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if (skip_this_block == 0) {
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uint32_t packet_size = player_put_packet(
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payload_ssrc, sequence_number_for_player, timestamp, payload_pointer,
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payload_length, mute, timestamp_difference, conn);
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debug(4, "block %u, timestamp %u, length %u sent to the player.", seq_no,
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timestamp, packet_size);
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sequence_number_for_player++; // simply increment
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expected_timestamp = timestamp + packet_size; // for the next time
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packets_played_in_this_sequence++;
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} else {
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debug(3,
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"skipped deciphering block %u with timestamp %u because its lead time is "
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"out of range at %f "
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"seconds.",
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seq_no, timestamp, lead_time * 1.0E-9);
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uint32_t currentAnchorRTP = 0;
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uint64_t currentAnchorLocalTime = 0;
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if (get_ptp_anchor_local_time_info(conn, ¤tAnchorRTP,
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¤tAnchorLocalTime) == clock_ok) {
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debug(3, "anchorRTP: %u, anchorLocalTime: %" PRIu64 ".", currentAnchorRTP,
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currentAnchorLocalTime);
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} else {
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debug(3, "Clock not okay");
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}
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}
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} else {
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debug(3,
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"skipped deciphering block %u with timestamp %u because its lead time is "
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"out of range at %f "
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"seconds.",
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seq_no, timestamp, lead_time * 1.0E-9);
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uint32_t currentAnchorRTP = 0;
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uint64_t currentAnchorLocalTime = 0;
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if (get_ptp_anchor_local_time_info(conn, ¤tAnchorRTP,
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¤tAnchorLocalTime) == clock_ok) {
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debug(3, "anchorRTP: %u, anchorLocalTime: %" PRIu64 ".", currentAnchorRTP,
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currentAnchorLocalTime);
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} else {
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debug(3, "Clock not okay");
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}
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debug(2, "Unrecognised or invalid ssrc: %s.", get_ssrc_name(payload_ssrc));
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}
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} else {
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debug(2, "Unrecognised or invalid ssrc: %s.", get_ssrc_name(payload_ssrc));
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debug(1, "dropping buffer that should have played before the last actually played.");
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}
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new_audio_block_needed = 1; // the block has been used up and is no longer current
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} else {
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debug(1, "dropping buffer that should have played before the last actually played.");
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if ((very_early_packets_signalled == 0) && (lead_time * 1E-9 > (config.audio_decoded_buffer_desired_length + 0.2))) {
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debug(1,
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"incoming frame suddenly (?) has a lead time of %f seconds, with a desired "
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"decoded buffer length of %f.",
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1.0 * lead_time * 1E-9, config.audio_decoded_buffer_desired_length);
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very_early_packets_signalled = 1;
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}
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usleep(((1000000 * conn->frames_per_packet) / conn->input_rate) *
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2); // wait for approximately the length of two packets
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}
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new_audio_block_needed = 1; // the block has been used up and is no longer current
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} else {
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if ((have_valid_time != 0) && (very_early_packets_signalled == 0) &&
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(lead_time * 1E-9 > (config.audio_decoded_buffer_desired_length + 0.2))) {
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debug(1,
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"incoming frame suddenly (?) has a lead time of %f seconds, with a desired "
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"decoded buffer length of %f.",
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1.0 * lead_time * 1E-9, config.audio_decoded_buffer_desired_length);
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very_early_packets_signalled = 1;
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}
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usleep(((1000000 * conn->frames_per_packet) / conn->input_rate) *
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2); // wait for approximately the length of two packets
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debug(3, "just you wait, Henry Higgins, without valid timing information...");
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usleep(20000); // just you wait, Henry Higgins...
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}
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}
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}
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Block a user