player.c

/*
 * Slave-clocked ALAC stream player. This file is part of Shairport.
 * Copyright (c) James Laird 2011, 2013
 * All rights reserved.
 *
 * Modifications for audio synchronisation, AirPlay 2
 * and related work, copyright (c) Mike Brady 2014 -- 2022
 * All rights reserved.
 *
 * Permission is hereby granted, free of charge, to any person
 * obtaining a copy of this software and associated documentation
 * files (the "Software"), to deal in the Software without
 * restriction, including without limitation the rights to use,
 * copy, modify, merge, publish, distribute, sublicense, and/or
 * sell copies of the Software, and to permit persons to whom the
 * Software is furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be
 * included in all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
 * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
 * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
 * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
 * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
 * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
 * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
 * OTHER DEALINGS IN THE SOFTWARE.
 */

#include <assert.h>
#include <errno.h>
#include <fcntl.h>
#include <inttypes.h>
#include <limits.h>
#include <math.h>
#include <pthread.h>
#include <stdarg.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <sys/stat.h>
#include <sys/syslog.h>
#include <sys/types.h>
#include <unistd.h>

#include "config.h"

#ifdef CONFIG_MBEDTLS
#include <mbedtls/aes.h>
#include <mbedtls/havege.h>
#endif

#ifdef CONFIG_POLARSSL
#include <polarssl/aes.h>
#include <polarssl/havege.h>
#endif

#ifdef CONFIG_OPENSSL
#include <openssl/aes.h>
#endif

#ifdef CONFIG_SOXR
#include <soxr.h>
#endif

#ifdef CONFIG_CONVOLUTION
#include <FFTConvolver/convolver.h>
#endif

#ifdef CONFIG_METADATA_HUB
#include "metadata_hub.h"
#endif

#ifdef CONFIG_DACP_CLIENT
#include "dacp.h"
#endif

#include "common.h"
#include "mdns.h"
#include "player.h"
#include "rtp.h"
#include "rtsp.h"

#include "alac.h"

#ifdef CONFIG_APPLE_ALAC
#include "apple_alac.h"
#endif

#ifdef CONFIG_AIRPLAY_2
#include "ptp-utilities.h"
#endif

#include "loudness.h"

#include "activity_monitor.h"

// m<ake the first audio packet deliberately early to bias the sync error of
// the very first packet, making the error more likely to be too early
// rather than too late. It it's too early,
// a delay exactly compensating for it can be sent just before the
// first packet. This should exactly compensate for the error.

int64_t first_frame_early_bias = 8;

// default buffer size
// needs to be a power of 2 because of the way BUFIDX(seqno) works
//#define BUFFER_FRAMES 512
#define MAX_PACKET 2048

// DAC buffer occupancy stuff
#define DAC_BUFFER_QUEUE_MINIMUM_LENGTH 2500

// static abuf_t audio_buffer[BUFFER_FRAMES];
#define BUFIDX(seqno) ((seq_t)(seqno) % BUFFER_FRAMES)

int32_t modulo_32_offset(uint32_t from, uint32_t to) { return to - from; }

void do_flush(uint32_t timestamp, rtsp_conn_info *conn);

void ab_resync(rtsp_conn_info *conn) {
  int i;
  for (i = 0; i < BUFFER_FRAMES; i++) {
    conn->audio_buffer[i].ready = 0;
    conn->audio_buffer[i].resend_request_number = 0;
    conn->audio_buffer[i].resend_time =
        0; // this is either zero or the time the last resend was requested.
    conn->audio_buffer[i].initialisation_time =
        0; // this is either the time the packet was received or the time it was noticed the packet
           // was missing.
    conn->audio_buffer[i].sequence_number = 0;
  }
  conn->ab_synced = 0;
  conn->last_seqno_read = -1;
  conn->ab_buffering = 1;
}

// the sequence numbers will wrap pretty often.
// this returns true if the second arg is strictly after the first
static inline int is_after(seq_t a, seq_t b) {
  int16_t d = b - a;
  return d > 0;
}

void reset_input_flow_metrics(rtsp_conn_info *conn) {
  conn->play_number_after_flush = 0;
  conn->packet_count_since_flush = 0;
  conn->input_frame_rate_starting_point_is_valid = 0;
  conn->initial_reference_time = 0;
  conn->initial_reference_timestamp = 0;
}

void unencrypted_packet_decode(unsigned char *packet, int length, short *dest, int *outsize,
                               int size_limit, rtsp_conn_info *conn) {
  if (conn->stream.type == ast_apple_lossless) {
#ifdef CONFIG_APPLE_ALAC
    if (config.use_apple_decoder) {
      if (conn->decoder_in_use != 1 << decoder_apple_alac) {
        debug(2, "Apple ALAC Decoder used on encrypted audio.");
        conn->decoder_in_use = 1 << decoder_apple_alac;
      }
      apple_alac_decode_frame(packet, length, (unsigned char *)dest, outsize);
      *outsize = *outsize * 4; // bring the size to bytes
    } else
#endif
    {
      if (conn->decoder_in_use != 1 << decoder_hammerton) {
        debug(2, "Hammerton Decoder used on encrypted audio.");
        conn->decoder_in_use = 1 << decoder_hammerton;
      }
      alac_decode_frame(conn->decoder_info, packet, (unsigned char *)dest, outsize);
    }
  } else if (conn->stream.type == ast_uncompressed) {
    int length_to_use = length;
    if (length_to_use > size_limit) {
      warn("unencrypted_packet_decode: uncompressed audio packet too long (size: %d bytes) to "
           "process -- truncated",
           length);
      length_to_use = size_limit;
    }
    int i;
    short *source = (short *)packet;
    for (i = 0; i < (length_to_use / 2); i++) {
      *dest = ntohs(*source);
      dest++;
      source++;
    }
    *outsize = length_to_use;
  }
}

int audio_packet_decode(short *dest, int *destlen, uint8_t *buf, int len, rtsp_conn_info *conn) {
  // parameters: where the decoded stuff goes, its length in samples,
  // the incoming packet, the length of the incoming packet in bytes
  // destlen should contain the allowed max number of samples on entry

  if (len > MAX_PACKET) {
    warn("Incoming audio packet size is too large at %d; it should not exceed %d.", len,
         MAX_PACKET);
    return -1;
  }
  unsigned char packet[MAX_PACKET];
  // unsigned char packetp[MAX_PACKET];
  assert(len <= MAX_PACKET);
  int reply = 0;                                          // everything okay
  int outsize = conn->input_bytes_per_frame * (*destlen); // the size the output should be, in bytes
  int maximum_possible_outsize = outsize;

  if (conn->stream.encrypted) {
    unsigned char iv[16];
    int aeslen = len & ~0xf;
    memcpy(iv, conn->stream.aesiv, sizeof(iv));
#ifdef CONFIG_MBEDTLS
    mbedtls_aes_crypt_cbc(&conn->dctx, MBEDTLS_AES_DECRYPT, aeslen, iv, buf, packet);
#endif
#ifdef CONFIG_POLARSSL
    aes_crypt_cbc(&conn->dctx, AES_DECRYPT, aeslen, iv, buf, packet);
#endif
#ifdef CONFIG_OPENSSL
    AES_cbc_encrypt(buf, packet, aeslen, &conn->aes, iv, AES_DECRYPT);
#endif
    memcpy(packet + aeslen, buf + aeslen, len - aeslen);
    unencrypted_packet_decode(packet, len, dest, &outsize, maximum_possible_outsize, conn);
  } else {
    // not encrypted
    unencrypted_packet_decode(buf, len, dest, &outsize, maximum_possible_outsize, conn);
  }

  if (outsize > maximum_possible_outsize) {
    debug(2,
          "Output from alac_decode larger (%d bytes, not frames) than expected (%d bytes) -- "
          "truncated, but buffer overflow possible! Encrypted = %d.",
          outsize, maximum_possible_outsize, conn->stream.encrypted);
    reply = -1; // output packet is the wrong size
  }

  if (conn->input_bytes_per_frame != 0)
    *destlen = outsize / conn->input_bytes_per_frame;
  else
    die("Unexpectedly, conn->input_bytes_per_frame is zero.");
  if ((outsize % conn->input_bytes_per_frame) != 0)
    debug(1,
          "Number of audio frames (%d) does not correspond exactly to the number of bytes (%d) "
          "and the audio frame size (%d).",
          *destlen, outsize, conn->input_bytes_per_frame);
  return reply;
}

static int init_alac_decoder(int32_t fmtp[12], rtsp_conn_info *conn) {

  // clang-format off

  // This is a guess, but the format of the fmtp looks identical to the format of an
  // ALACSpecificCOnfig which is detailed in the file ALACMagicCookieDescription.txt
  // in the Apple ALAC sample implementation
  // Here it is:

  /*

    * ALAC Specific Info (24 bytes) (mandatory)
    __________________________________________________________________________________________________________________________________

    The Apple Lossless codec stores specific information about the encoded stream in the ALACSpecificConfig. This
    info is vended by the encoder and is used to setup the decoder for a given encoded bitstream.

    When read from and written to a file, the fields of this struct must be in big-endian order.
    When vended by the encoder (and received by the decoder) the struct values will be in big-endian order.


        struct      ALACSpecificConfig (defined in ALACAudioTypes.h)
        abstract    This struct is used to describe codec provided information about the encoded Apple Lossless bitstream.
                    It must accompany the encoded stream in the containing audio file and be provided to the decoder.

        field       frameLength             uint32_t        indicating the frames per packet when no explicit frames per packet setting is
                                                            present in the packet header. The encoder frames per packet can be explicitly set
                                                            but for maximum compatibility, the default encoder setting of 4096 should be used.

        field       compatibleVersion       uint8_t         indicating compatible version,
                                                            value must be set to 0

        field       bitDepth                uint8_t         describes the bit depth of the source PCM data (maximum value = 32)

        field       pb                      uint8_t         currently unused tuning parameter.
                                                            value should be set to 40

        field       mb                      uint8_t         currently unused tuning parameter.
                                                            value should be set to 10

        field       kb                      uint8_t         currently unused tuning parameter.
                                                            value should be set to 14

        field       numChannels             uint8_t         describes the channel count (1 = mono, 2 = stereo, etc...)
                                                            when channel layout info is not provided in the 'magic cookie', a channel count > 2
                                                            describes a set of discreet channels with no specific ordering

        field       maxRun                  uint16_t        currently unused.
                                                            value should be set to 255

        field       maxFrameBytes           uint32_t        the maximum size of an Apple Lossless packet within the encoded stream.
                                                            value of 0 indicates unknown

        field       avgBitRate              uint32_t        the average bit rate in bits per second of the Apple Lossless stream.
                                                            value of 0 indicates unknown

        field       sampleRate              uint32_t        sample rate of the encoded stream


    typedef struct ALACSpecificConfig
    {
            uint32_t        frameLength;
            uint8_t         compatibleVersion;
            uint8_t         bitDepth;
            uint8_t         pb;
            uint8_t         mb;
            uint8_t         kb;
            uint8_t         numChannels;
            uint16_t        maxRun;
            uint32_t        maxFrameBytes;
            uint32_t        avgBitRate;
            uint32_t        sampleRate;

    } ALACSpecificConfig;

   */

   // We are going to go on that basis

  // clang-format on

  alac_file *alac;

  alac = alac_create(conn->input_bit_depth,
                     conn->input_num_channels); // no pthread cancellation point in here
  if (!alac)
    return 1;
  conn->decoder_info = alac;

  alac->setinfo_max_samples_per_frame = conn->max_frames_per_packet;
  alac->setinfo_7a = fmtp[2];
  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];
  alac->setinfo_7f = fmtp[7];
  alac->setinfo_80 = fmtp[8];
  alac->setinfo_82 = fmtp[9];
  alac->setinfo_86 = fmtp[10];
  alac->setinfo_8a_rate = fmtp[11];
  alac_allocate_buffers(alac); // no pthread cancellation point in here

#ifdef CONFIG_APPLE_ALAC
  apple_alac_init(fmtp); // no pthread cancellation point in here
#endif

  return 0;
}

static void terminate_decoders(rtsp_conn_info *conn) {
  alac_free(conn->decoder_info);
#ifdef CONFIG_APPLE_ALAC
  apple_alac_terminate();
#endif
}

static void init_buffer(rtsp_conn_info *conn) {
  // debug(1,"input_bytes_per_frame: %d.", conn->input_bytes_per_frame);
  // debug(1,"input_bit_depth: %d.", conn->input_bit_depth);
  int i;
  for (i = 0; i < BUFFER_FRAMES; i++)
    //    conn->audio_buffer[i].data = malloc(conn->input_bytes_per_frame *
    //    conn->max_frames_per_packet);
    conn->audio_buffer[i].data = malloc(8 * conn->max_frames_per_packet); // todo
}

static void free_audio_buffers(rtsp_conn_info *conn) {
  int i;
  for (i = 0; i < BUFFER_FRAMES; i++)
    free(conn->audio_buffer[i].data);
}

int first_possibly_missing_frame = -1;

void reset_buffer(rtsp_conn_info *conn) {
  debug_mutex_lock(&conn->ab_mutex, 30000, 0);
  ab_resync(conn);
  debug_mutex_unlock(&conn->ab_mutex, 0);
  if (config.output->flush) {
    config.output->flush(); // no cancellation points
                            //            debug(1, "reset_buffer: flush output device.");
  }
}

void get_audio_buffer_size_and_occupancy(unsigned int *size, unsigned int *occupancy,
                                         rtsp_conn_info *conn) {
  debug_mutex_lock(&conn->ab_mutex, 30000, 0);
  *size = BUFFER_FRAMES;
  if (conn->ab_synced) {
    int16_t occ =
        conn->ab_write - conn->ab_read; // will be zero or positive if read and write are within
                                        // 2^15 of each other and write is at or after read
    *occupancy = occ;
  } else {
    *occupancy = 0;
  }
  debug_mutex_unlock(&conn->ab_mutex, 0);
}

void player_put_packet(int original_format, seq_t seqno, uint32_t actual_timestamp, uint8_t *data,
                       int len, rtsp_conn_info *conn) {

  // if it's original format, it has a valid seqno and must be decoded
  // otherwise, it can take the next seqno and doesn't need decoding.

  // ignore a request to flush that has been made before the first packet...
  if (conn->packet_count == 0) {
    debug_mutex_lock(&conn->flush_mutex, 1000, 1);
    conn->flush_requested = 0;
    conn->flush_rtp_timestamp = 0;
    debug_mutex_unlock(&conn->flush_mutex, 3);
  }

  debug_mutex_lock(&conn->ab_mutex, 30000, 0);
  uint64_t time_now = get_absolute_time_in_ns();
  conn->packet_count++;
  conn->packet_count_since_flush++;
  conn->time_of_last_audio_packet = time_now;
  if (conn->connection_state_to_output) { // if we are supposed to be processing these packets
    abuf_t *abuf = 0;
    if (!conn->ab_synced) {
      conn->ab_write = seqno;
      conn->ab_read = seqno;
      conn->ab_synced = 1;
      conn->first_packet_timestamp = 0;
      debug(2, "Connection %d: synced by first packet, seqno %u.", conn->connection_number, seqno);
    } else if (original_format == 0) {
      // if the packet is coming in original format, the sequence number is important
      // otherwise, ignore is by setting it equal to the expected sequence number in ab_write
      seqno = conn->ab_write;
    }
    if (conn->ab_write ==
        seqno) { // if this is the expected packet (which could be the first packet...)
      if (conn->input_frame_rate_starting_point_is_valid == 0) {
        if ((conn->packet_count_since_flush >= 500) && (conn->packet_count_since_flush <= 510)) {
          conn->frames_inward_measurement_start_time = time_now;
          conn->frames_inward_frames_received_at_measurement_start_time = actual_timestamp;
          conn->input_frame_rate_starting_point_is_valid = 1; // valid now
        }
      }
      conn->frames_inward_measurement_time = time_now;
      conn->frames_inward_frames_received_at_measurement_time = actual_timestamp;
      abuf = conn->audio_buffer + BUFIDX(seqno);
      conn->ab_write = seqno + 1;                 // move the write pointer to the next free space
    } else if (is_after(conn->ab_write, seqno)) { // newer than expected
      int32_t gap = seqno - conn->ab_write;
      if (gap <= 0)
        debug(1, "Unexpected gap size: %d.", gap);
      int i;
      for (i = 0; i < gap; i++) {
        abuf = conn->audio_buffer + BUFIDX(conn->ab_write + i);
        abuf->ready = 0; // to be sure, to be sure
        abuf->resend_request_number = 0;
        abuf->initialisation_time =
            time_now;          // this represents when the packet was noticed to be missing
        abuf->status = 1 << 0; // signifying missing
        abuf->resend_time = 0;
        abuf->given_timestamp = 0;
        abuf->sequence_number = 0;
      }
      abuf = conn->audio_buffer + BUFIDX(seqno);
      //        rtp_request_resend(ab_write, gap);
      //        resend_requests++;
      conn->ab_write = seqno + 1;
    } else if (is_after(conn->ab_read, seqno)) { // older than expected but not too late
      conn->late_packets++;
      abuf = conn->audio_buffer + BUFIDX(seqno);
    } else { // too late.
      conn->too_late_packets++;
    }

    if (abuf) {
      int datalen = conn->max_frames_per_packet;
      abuf->initialisation_time = time_now;
      abuf->resend_time = 0;
      if ((original_format != 0) &&
          (audio_packet_decode(abuf->data, &datalen, data, len, conn) == 0)) {
        abuf->ready = 1;
        abuf->status = 0; // signifying that it was received
        abuf->length = datalen;
        abuf->given_timestamp = actual_timestamp;
        abuf->sequence_number = seqno;
      } else if (original_format == 0) {
        memcpy(abuf->data, data, len * conn->input_bytes_per_frame);
        abuf->ready = 1;
        abuf->status = 0; // signifying that it was received
        abuf->length = len;
        abuf->given_timestamp = actual_timestamp;
        abuf->sequence_number = seqno;
      } else {
        debug(1, "Bad audio packet detected and discarded.");
        abuf->ready = 0;
        abuf->status = 1 << 1; // bad packet, discarded
        abuf->resend_request_number = 0;
        abuf->given_timestamp = 0;
        abuf->sequence_number = 0;
      }
    }

    int rc = pthread_cond_signal(&conn->flowcontrol);
    if (rc)
      debug(1, "Error signalling flowcontrol.");

    // resend checks
    {
      uint64_t minimum_wait_time =
          (uint64_t)(config.resend_control_first_check_time * (uint64_t)1000000000);
      uint64_t resend_repeat_interval =
          (uint64_t)(config.resend_control_check_interval_time * (uint64_t)1000000000);
      uint64_t minimum_remaining_time = (uint64_t)(
          (config.resend_control_last_check_time + config.audio_backend_buffer_desired_length) *
          (uint64_t)1000000000);
      uint64_t latency_time = (uint64_t)(conn->latency * (uint64_t)1000000000);
      latency_time = latency_time / (uint64_t)conn->input_rate;

      // find the first frame that is missing, if known
      int x = conn->ab_read;
      if (first_possibly_missing_frame >= 0) {
        // if it's within the range
        int16_t buffer_size = conn->ab_write - conn->ab_read; // must be positive
        if (buffer_size >= 0) {
          int16_t position_in_buffer = first_possibly_missing_frame - conn->ab_read;
          if ((position_in_buffer >= 0) && (position_in_buffer < buffer_size))
            x = first_possibly_missing_frame;
        }
      }

      first_possibly_missing_frame = -1; // has not been set

      int missing_frame_run_count = 0;
      int start_of_missing_frame_run = -1;
      int number_of_missing_frames = 0;
      while (x != conn->ab_write) {
        abuf_t *check_buf = conn->audio_buffer + BUFIDX(x);
        if (!check_buf->ready) {
          if (first_possibly_missing_frame < 0)
            first_possibly_missing_frame = x;
          number_of_missing_frames++;
          // debug(1, "frame %u's initialisation_time is 0x%" PRIx64 ", latency_time is 0x%"
          // PRIx64 ", time_now is 0x%" PRIx64 ", minimum_remaining_time is 0x%" PRIx64 ".", x,
          // check_buf->initialisation_time, latency_time, time_now, minimum_remaining_time);
          int too_late = ((check_buf->initialisation_time < (time_now - latency_time)) ||
                          ((check_buf->initialisation_time - (time_now - latency_time)) <
                           minimum_remaining_time));
          int too_early = ((time_now - check_buf->initialisation_time) < minimum_wait_time);
          int too_soon_after_last_request =
              ((check_buf->resend_time != 0) &&
               ((time_now - check_buf->resend_time) <
                resend_repeat_interval)); // time_now can never be less than the time_tag

          if (too_late)
            check_buf->status |= 1 << 2; // too late
          else
            check_buf->status &= 0xFF - (1 << 2); // not too late
          if (too_early)
            check_buf->status |= 1 << 3; // too early
          else
            check_buf->status &= 0xFF - (1 << 3); // not too early
          if (too_soon_after_last_request)
            check_buf->status |= 1 << 4; // too soon after last request
          else
            check_buf->status &= 0xFF - (1 << 4); // not too soon after last request

          if ((!too_soon_after_last_request) && (!too_late) && (!too_early)) {
            if (start_of_missing_frame_run == -1) {
              start_of_missing_frame_run = x;
              missing_frame_run_count = 1;
            } else {
              missing_frame_run_count++;
            }
            check_buf->resend_time = time_now; // setting the time to now because we are
                                               // definitely going to take action
            check_buf->resend_request_number++;
            debug(3, "Frame %d is missing with ab_read of %u and ab_write of %u.", x, conn->ab_read,
                  conn->ab_write);
          }
          // if (too_late) {
          //   debug(1,"too late to get missing frame %u.", x);
          // }
        }
        // if (number_of_missing_frames != 0)
        //  debug(1,"check with x = %u, ab_read = %u, ab_write = %u, first_possibly_missing_frame
        //  = %d.", x, conn->ab_read, conn->ab_write, first_possibly_missing_frame);
        x = (x + 1) & 0xffff;
        if (((check_buf->ready) || (x == conn->ab_write)) && (missing_frame_run_count > 0)) {
          // send a resend request
          if (missing_frame_run_count > 1)
            debug(3, "request resend of %d packets starting at seqno %u.", missing_frame_run_count,
                  start_of_missing_frame_run);
          if (config.disable_resend_requests == 0) {
            debug_mutex_unlock(&conn->ab_mutex, 3);
            rtp_request_resend(start_of_missing_frame_run, missing_frame_run_count, conn);
            debug_mutex_lock(&conn->ab_mutex, 20000, 1);
            conn->resend_requests++;
          }
          start_of_missing_frame_run = -1;
          missing_frame_run_count = 0;
        }
      }
      if (number_of_missing_frames == 0)
        first_possibly_missing_frame = conn->ab_write;
    }
  }
  debug_mutex_unlock(&conn->ab_mutex, 0);
}

int32_t rand_in_range(int32_t exclusive_range_limit) {
  static uint32_t lcg_prev = 12345;
  // returns a pseudo random integer in the range 0 to (exclusive_range_limit-1) inclusive
  int64_t sp = lcg_prev;
  int64_t rl = exclusive_range_limit;
  lcg_prev = lcg_prev * 69069 + 3; // crappy psrg
  sp = sp * rl; // 64 bit calculation. Interesting part is above the 32 rightmost bits;
  return sp >> 32;
}

static inline void process_sample(int32_t sample, char **outp, sps_format_t format, int volume,
                                  int dither, rtsp_conn_info *conn) {
  /*
  {
                static int old_volume = 0;
                if (volume != old_volume) {
                        debug(1,"Volume is now %d.",volume);
                        old_volume = volume;
                }
  }
  */

  int64_t hyper_sample = sample;
  int result = 0;

  if (config.loudness) {
    hyper_sample <<=
        32; // Do not apply volume as it has already been done with the Loudness DSP filter
  } else {
    int64_t hyper_volume = (int64_t)volume << 16;
    hyper_sample = hyper_sample * hyper_volume; // this is 64 bit bit multiplication -- we may need
                                                // to dither it down to its target resolution
  }

  // next, do dither, if necessary
  if (dither) {

    // add a TPDF dither -- see
    // http://educypedia.karadimov.info/library/DitherExplained.pdf
    // and the discussion around https://www.hydrogenaud.io/forums/index.php?showtopic=16963&st=25

    // I think, for a 32 --> 16 bits, the range of
    // random numbers needs to be from -2^16 to 2^16, i.e. from -65536 to 65536 inclusive, not from
    // -32768 to +32767

    // Actually, what would be generated here is from -65535 to 65535, i.e. one less on the limits.

    // See the original paper at
    // http://www.ece.rochester.edu/courses/ECE472/resources/Papers/Lipshitz_1992.pdf
    // by Lipshitz, Wannamaker and Vanderkooy, 1992.

    int64_t dither_mask = 0;
    switch (format) {
    case SPS_FORMAT_S32:
    case SPS_FORMAT_S32_LE:
    case SPS_FORMAT_S32_BE:
      dither_mask = (int64_t)1 << (64 - 32);
      break;
    case SPS_FORMAT_S24:
    case SPS_FORMAT_S24_LE:
    case SPS_FORMAT_S24_BE:
    case SPS_FORMAT_S24_3LE:
    case SPS_FORMAT_S24_3BE:
      dither_mask = (int64_t)1 << (64 - 24);
      break;
    case SPS_FORMAT_S16:
    case SPS_FORMAT_S16_LE:
    case SPS_FORMAT_S16_BE:
      dither_mask = (int64_t)1 << (64 - 16);
      break;
    case SPS_FORMAT_S8:
    case SPS_FORMAT_U8:
      dither_mask = (int64_t)1 << (64 - 8);
      break;
    case SPS_FORMAT_UNKNOWN:
      die("Unexpected SPS_FORMAT_UNKNOWN while calculating dither mask.");
      break;
    case SPS_FORMAT_AUTO:
      die("Unexpected SPS_FORMAT_AUTO while calculating dither mask.");
      break;
    case SPS_FORMAT_INVALID:
      die("Unexpected SPS_FORMAT_INVALID while calculating dither mask.");
      break;
    }
    dither_mask -= 1;
    int64_t r = r64i();

    int64_t tpdf = (r & dither_mask) - (conn->previous_random_number & dither_mask);
    conn->previous_random_number = r;
    // add dither, allowing for clipping
    if (tpdf >= 0) {
      if (INT64_MAX - tpdf >= hyper_sample)
        hyper_sample += tpdf;
      else
        hyper_sample = INT64_MAX;
    } else {
      if (INT64_MIN - tpdf <= hyper_sample)
        hyper_sample += tpdf;
      else
        hyper_sample = INT64_MIN;
    }
    // dither is complete here
  }

  // move the result to the desired position in the int64_t
  char *op = *outp;
  uint8_t byt;
  switch (format) {
  case SPS_FORMAT_S32_LE:
    hyper_sample >>= (64 - 32);
    byt = (uint8_t)hyper_sample;
    *op++ = byt;
    byt = (uint8_t)(hyper_sample >> 8);
    *op++ = byt;
    byt = (uint8_t)(hyper_sample >> 16);
    *op++ = byt;
    byt = (uint8_t)(hyper_sample >> 24);
    *op++ = byt;
    result = 4;
    break;
  case SPS_FORMAT_S32_BE:
    hyper_sample >>= (64 - 32);
    byt = (uint8_t)(hyper_sample >> 24);
    *op++ = byt;
    byt = (uint8_t)(hyper_sample >> 16);
    *op++ = byt;
    byt = (uint8_t)(hyper_sample >> 8);
    *op++ = byt;
    byt = (uint8_t)hyper_sample;
    *op++ = byt;
    result = 4;
    break;
  case SPS_FORMAT_S32:
    hyper_sample >>= (64 - 32);
    *(int32_t *)op = hyper_sample;
    result = 4;
    break;
  case SPS_FORMAT_S24_3LE:
    hyper_sample >>= (64 - 24);
    byt = (uint8_t)hyper_sample;
    *op++ = byt;
    byt = (uint8_t)(hyper_sample >> 8);
    *op++ = byt;
    byt = (uint8_t)(hyper_sample >> 16);
    *op++ = byt;
    result = 3;
    break;
  case SPS_FORMAT_S24_3BE:
    hyper_sample >>= (64 - 24);
    byt = (uint8_t)(hyper_sample >> 16);
    *op++ = byt;
    byt = (uint8_t)(hyper_sample >> 8);
    *op++ = byt;
    byt = (uint8_t)hyper_sample;
    *op++ = byt;
    result = 3;
    break;
  case SPS_FORMAT_S24_LE:
    hyper_sample >>= (64 - 24);
    byt = (uint8_t)hyper_sample;
    *op++ = byt;
    byt = (uint8_t)(hyper_sample >> 8);
    *op++ = byt;
    byt = (uint8_t)(hyper_sample >> 16);
    *op++ = byt;
    *op++ = 0;
    result = 4;
    break;
  case SPS_FORMAT_S24_BE:
    hyper_sample >>= (64 - 24);
    *op++ = 0;
    byt = (uint8_t)(hyper_sample >> 16);
    *op++ = byt;
    byt = (uint8_t)(hyper_sample >> 8);
    *op++ = byt;
    byt = (uint8_t)hyper_sample;
    *op++ = byt;
    result = 4;
    break;
  case SPS_FORMAT_S24:
    hyper_sample >>= (64 - 24);
    *(int32_t *)op = hyper_sample;
    result = 4;
    break;
  case SPS_FORMAT_S16_LE:
    hyper_sample >>= (64 - 16);
    byt = (uint8_t)hyper_sample;
    *op++ = byt;
    byt = (uint8_t)(hyper_sample >> 8);
    *op++ = byt;
    result = 2;
    break;
  case SPS_FORMAT_S16_BE:
    hyper_sample >>= (64 - 16);
    byt = (uint8_t)(hyper_sample >> 8);
    *op++ = byt;
    byt = (uint8_t)hyper_sample;
    *op++ = byt;
    result = 2;
    break;
  case SPS_FORMAT_S16:
    hyper_sample >>= (64 - 16);
    *(int16_t *)op = (int16_t)hyper_sample;
    result = 2;
    break;
  case SPS_FORMAT_S8:
    hyper_sample >>= (int8_t)(64 - 8);
    *op = hyper_sample;
    result = 1;
    break;
  case SPS_FORMAT_U8:
    hyper_sample >>= (uint8_t)(64 - 8);
    hyper_sample += 128;
    *op = hyper_sample;
    result = 1;
    break;
  case SPS_FORMAT_UNKNOWN:
    die("Unexpected SPS_FORMAT_UNKNOWN while outputting samples");
    break;
  case SPS_FORMAT_AUTO:
    die("Unexpected SPS_FORMAT_AUTO while outputting samples");
    break;
  case SPS_FORMAT_INVALID:
    die("Unexpected SPS_FORMAT_INVALID while outputting samples");
    break;
  }

  *outp += result;
}

void buffer_get_frame_cleanup_handler(void *arg) {
  rtsp_conn_info *conn = (rtsp_conn_info *)arg;
  debug_mutex_unlock(&conn->ab_mutex, 0);
}

// get the next frame, when available. return 0 if underrun/stream reset.
static abuf_t *buffer_get_frame(rtsp_conn_info *conn) {
  // int16_t buf_fill;
  uint64_t local_time_now;
  // struct timespec tn;
  abuf_t *curframe = NULL;
  int notified_buffer_empty = 0; // diagnostic only

  debug_mutex_lock(&conn->ab_mutex, 30000, 0);

  int wait;
  long dac_delay = 0; // long because alsa returns a long

  int have_sent_prefiller_silence =
      0; // set to true when we have sent at least one silent frame to the DAC

  pthread_cleanup_push(buffer_get_frame_cleanup_handler,
                       (void *)conn); // undo what's been done so far
  do {

    // get the time
    local_time_now = get_absolute_time_in_ns(); // type okay
    // debug(3, "buffer_get_frame is iterating");

    // we must have timing information before we can do anything here
    if (have_timestamp_timing_information(conn)) {

      int rco = get_requested_connection_state_to_output();

      if (conn->connection_state_to_output != rco) {
        conn->connection_state_to_output = rco;
        // change happening
        if (conn->connection_state_to_output == 0) { // going off
          debug(2, "request flush because connection_state_to_output is off");
          debug_mutex_lock(&conn->flush_mutex, 1000, 1);
          conn->flush_requested = 1;
          conn->flush_rtp_timestamp = 0;
          debug_mutex_unlock(&conn->flush_mutex, 3);
        }
      }

      if (config.output->is_running)
        if (config.output->is_running() != 0) { // if the back end isn't running for any reason
          debug(2, "request flush because back end is not running");
          debug_mutex_lock(&conn->flush_mutex, 1000, 0);
          conn->flush_requested = 1;
          conn->flush_rtp_timestamp = 0;
          debug_mutex_unlock(&conn->flush_mutex, 0);
        }

      debug_mutex_lock(&conn->flush_mutex, 1000, 0);
      pthread_cleanup_push(mutex_unlock, &conn->flush_mutex);
      if (conn->flush_requested == 1) {
        if (conn->flush_output_flushed == 0)
          if (config.output->flush) {
            config.output->flush(); // no cancellation points
            debug(2, "flush request: flush output device.");
          }
        conn->flush_output_flushed = 1;
      }
      // now check to see it the flush request is for frames in the buffer or not
      // if the first_packet_timestamp is zero, don't check
      int flush_needed = 0;
      int drop_request = 0;
      if (conn->flush_requested == 1) {
        if (conn->flush_rtp_timestamp == 0) {
          debug(1, "flush request: flush frame 0 -- flush assumed to be needed.");
          flush_needed = 1;
          drop_request = 1;
        } else {
          if ((conn->ab_synced) && ((conn->ab_write - conn->ab_read) > 0)) {
            abuf_t *firstPacket = conn->audio_buffer + BUFIDX(conn->ab_read);
            abuf_t *lastPacket = conn->audio_buffer + BUFIDX(conn->ab_write - 1);
            if ((firstPacket != NULL) && (firstPacket->ready)) {
              uint32_t first_frame_in_buffer = firstPacket->given_timestamp;
              int32_t offset_from_first_frame = conn->flush_rtp_timestamp - first_frame_in_buffer;
              if ((lastPacket != NULL) && (lastPacket->ready)) {
                // we have enough information to check if the flush is needed or can be discarded
                uint32_t last_frame_in_buffer =
                    lastPacket->given_timestamp + lastPacket->length - 1;

                // clang-format off
                // Now we have to work out if the flush frame is in the buffer.
                
                // If it is later than the end of the buffer, flush everything and keep the
                // request active.
                
                // If it is in the buffer, we need to flush part of the buffer.
                // (Actually we flush the entire buffer and drop the request.)
                
                // If it is before the buffer, no flush is needed. Drop the request.
                // clang-format on

                if (offset_from_first_frame > 0) {
                  int32_t offset_to_last_frame = last_frame_in_buffer - conn->flush_rtp_timestamp;
                  if (offset_to_last_frame >= 0) {
                    debug(2,
                          "flush request: flush frame %u active -- buffer contains %u frames, from "
                          "%u to %u.",
                          conn->flush_rtp_timestamp,
                          last_frame_in_buffer - first_frame_in_buffer + 1, first_frame_in_buffer,
                          last_frame_in_buffer);

                    // We need to drop all complete frames leading up to the frame containing
                    // the flush request frame.
                    int32_t offset_to_flush_frame = 0;
                    abuf_t *current_packet = NULL;
                    do {
                      current_packet = conn->audio_buffer + BUFIDX(conn->ab_read);
                      if (current_packet != NULL) {
                        uint32_t last_frame_in_current_packet =
                            current_packet->given_timestamp + current_packet->length - 1;
                        offset_to_flush_frame =
                            conn->flush_rtp_timestamp - last_frame_in_current_packet;
                        if (offset_to_flush_frame > 0) {
                          debug(2,
                                "flush to %u request: flush buffer %u, from "
                                "%u to %u. ab_write is: %u.",
                                conn->flush_rtp_timestamp, conn->ab_read,
                                current_packet->given_timestamp,
                                current_packet->given_timestamp + current_packet->length - 1,
                                conn->ab_write);
                          conn->ab_read++;
                        }
                      } else {
                        debug(1, "NULL current_packet");
                      }
                    } while ((current_packet == NULL) || (offset_to_flush_frame > 0));
                    // now remove any frames from the buffer that are before the flush frame itself.
                    int32_t frames_to_remove =
                        conn->flush_rtp_timestamp - current_packet->given_timestamp;
                    if (frames_to_remove > 0) {
                      debug(2, "%u frames to remove from current buffer", frames_to_remove);
                      void *dest = (void *)current_packet->data;
                      void *source = dest + conn->input_bytes_per_frame * frames_to_remove;
                      size_t frames_remaining = (current_packet->length - frames_to_remove);
                      memmove(dest, source, frames_remaining * conn->input_bytes_per_frame);
                      current_packet->given_timestamp = conn->flush_rtp_timestamp;
                      current_packet->length = frames_remaining;
                    }
                    debug(
                        2,
                        "flush request: flush frame %u complete -- buffer contains %u frames, from "
                        "%u to %u -- flushed to %u in buffer %u, with %u frames remaining.",
                        conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer + 1,
                        first_frame_in_buffer, last_frame_in_buffer,
                        current_packet->given_timestamp, conn->ab_read,
                        last_frame_in_buffer - current_packet->given_timestamp + 1);
                    drop_request = 1;
                  } else {
                    if (conn->flush_rtp_timestamp == last_frame_in_buffer + 1) {
                      debug(
                          2,
                          "flush request: flush frame %u completed -- buffer contained %u frames, "
                          "from "
                          "%u to %u",
                          conn->flush_rtp_timestamp,
                          last_frame_in_buffer - first_frame_in_buffer + 1, first_frame_in_buffer,
                          last_frame_in_buffer);
                      drop_request = 1;
                    } else {
                      debug(2,
                            "flush request: flush frame %u pending -- buffer contains %u frames, "
                            "from "
                            "%u to %u",
                            conn->flush_rtp_timestamp,
                            last_frame_in_buffer - first_frame_in_buffer + 1, first_frame_in_buffer,
                            last_frame_in_buffer);
                    }
                    flush_needed = 1;
                  }
                } else {
                  debug(2,
                        "flush request: flush frame %u expired -- buffer contains %u frames, "
                        "from %u "
                        "to %u",
                        conn->flush_rtp_timestamp, last_frame_in_buffer - first_frame_in_buffer + 1,
                        first_frame_in_buffer, last_frame_in_buffer);
                  drop_request = 1;
                }
              }
            }
          } else {
            debug(3,
                  "flush request: flush frame %u  -- buffer not synced or empty: synced: %d, "
                  "ab_read: "
                  "%u, ab_write: %u",
                  conn->flush_rtp_timestamp, conn->ab_synced, conn->ab_read, conn->ab_write);
            conn->flush_requested = 0; // remove the request
            // leave flush request pending and don't do a buffer flush, because there isn't one
          }
        }
      }
      if (flush_needed) {
        debug(2, "flush request: flush done.");
        ab_resync(conn); // no cancellation points
        conn->first_packet_timestamp = 0;
        conn->first_packet_time_to_play = 0;
        conn->time_since_play_started = 0;
        have_sent_prefiller_silence = 0;
        dac_delay = 0;
      }
      if (drop_request) {
        conn->flush_requested = 0;
        conn->flush_rtp_timestamp = 0;
        conn->flush_output_flushed = 0;
      }
      pthread_cleanup_pop(1); // unlock the conn->flush_mutex

      // skip out-of-date frames, and even more if we haven't seen the first frame
      int out_of_date = 1;
      uint32_t should_be_frame;

      uint64_t time_to_aim_for = local_time_now;
      uint64_t desired_lead_time = 120000000;
      if (conn->first_packet_timestamp == 0)
        time_to_aim_for = time_to_aim_for + desired_lead_time;

      while ((conn->ab_synced) && ((conn->ab_write - conn->ab_read) > 0) && (out_of_date != 0)) {
        abuf_t *thePacket = conn->audio_buffer + BUFIDX(conn->ab_read);
        if ((thePacket != NULL) && (thePacket->ready)) {
          local_time_to_frame(time_to_aim_for, &should_be_frame, conn);
          // debug(1,"should_be frame is %u.",should_be_frame);
          int32_t frame_difference = thePacket->given_timestamp - should_be_frame;
          if (frame_difference < 0) {
            debug(2, "Dropping out of date packet %u with timestamp %u. Lead time is %f seconds.",
                  conn->ab_read, thePacket->given_timestamp,
                  frame_difference * 1.0 / 44100.0 + desired_lead_time * 0.000000001);
            conn->ab_read++;
          } else {
            if (conn->first_packet_timestamp == 0)
              debug(2, "Accepting packet %u with timestamp %u. Lead time is %f seconds.",
                    conn->ab_read, thePacket->given_timestamp,
                    frame_difference * 1.0 / 44100.0 + desired_lead_time * 0.000000001);
            out_of_date = 0;
          }
        } else {
          debug(2, "Packet %u empty or not ready.", conn->ab_read);
          conn->ab_read++;
        }
      }

      if (conn->ab_synced) {
        curframe = conn->audio_buffer + BUFIDX(conn->ab_read);
        if (curframe != NULL) {
          uint64_t should_be_time;
          frame_to_local_time(curframe->given_timestamp, &should_be_time, conn);
          int64_t time_difference = should_be_time - local_time_now;
          debug(3, "Check packet from buffer %u, timestamp %u, %f seconds ahead.", conn->ab_read,
                curframe->given_timestamp, 0.000000001 * time_difference);
        } else {
          debug(3, "Check packet from buffer %u, empty.", conn->ab_read);
        }

        if ((conn->ab_read != conn->ab_write) &&
            (curframe->ready)) { // it could be synced and empty, under
                                 // exceptional circumstances, with the
                                 // frame unused, thus apparently ready

          if (curframe->sequence_number != conn->ab_read) {
            // some kind of sync problem has occurred.
            if (BUFIDX(curframe->sequence_number) == BUFIDX(conn->ab_read)) {
              // it looks like aliasing has happened
              // jump to the new incoming stuff...
              conn->ab_read = curframe->sequence_number;
              debug(1, "Aliasing of buffer index -- reset.");
            } else {
              debug(1, "Inconsistent sequence numbers detected");
            }
          }
        }

        if ((curframe) && (curframe->ready)) {
          notified_buffer_empty = 0; // at least one buffer now -- diagnostic only.
          if (conn->ab_buffering) {  // if we are getting packets but not yet forwarding them to the
                                     // player
            if (conn->first_packet_timestamp == 0) { // if this is the very first packet
              conn->first_packet_timestamp =
                  curframe->given_timestamp; // we will keep buffering until we are
                                             // supposed to start playing this

              // Here, calculate when we should start playing. We need to know when to allow the
              // packets to be sent to the player.

              // every second or so, we get a reference on when a particular packet should be
              // played.

              // It probably won't  be the timestamp of our first packet, however, so we might
              // have to do some calculations.

              // To calculate when the first packet will be played, we figure out the exact time
              // the packet should be played according to its timestamp and the reference time.
              // The desired latency, typically 88200 frames, will be calculated for in rtp.c,
              // and any desired backend latency offset included in it there.

              uint64_t should_be_time;

              frame_to_local_time(conn->first_packet_timestamp, // this will go modulo 2^32
                                  &should_be_time, conn);

              conn->first_packet_time_to_play = should_be_time;

              int64_t lt = conn->first_packet_time_to_play - local_time_now;

              // can't be too late because we skipped late packets already, FLW.
              debug(2, "Connection %d: Lead time for first frame %" PRId64 ": %f seconds.",
                    conn->connection_number, conn->first_packet_timestamp, lt * 0.000000001);
#ifdef CONFIG_METADATA
              // say we have started receiving frames here
              debug(2, "pffr");
              send_ssnc_metadata(
                  'pffr', NULL, 0,
                  0); // "first frame received", but don't wait if the queue is locked
#endif
            }

            if (conn->first_packet_time_to_play != 0) {
              // Now that we know the timing of the first packet...
              if (config.output->delay) {
                // and that the output device is capable of synchronization...

                // We may send packets of
                // silence from now until the time the first audio packet should be sent
                // and then we will send the first packet, which will be followed by
                // the subsequent packets.
                // here, we figure out whether and what silence to send.

                uint64_t should_be_time;

                // readjust first packet time to play
                frame_to_local_time(conn->first_packet_timestamp, // this will go modulo 2^32
                                    &should_be_time, conn);

                int64_t change_in_should_be_time =
                    (int64_t)(should_be_time - conn->first_packet_time_to_play);

                if (fabs(0.000001 * change_in_should_be_time) >
                    0.001) // the clock drift estimation might be nudging the estimate, and we can
                           // ignore this unless if's more than a microsecond
                  debug(2,
                        "Change in estimated first_packet_time: %f milliseconds for first_packet .",
                        0.000001 * change_in_should_be_time);

                conn->first_packet_time_to_play = should_be_time;

                int64_t lead_time =
                    conn->first_packet_time_to_play - local_time_now; // negative means late
                if (lead_time < 0) {
                  debug(1, "Gone past starting time for %u by %" PRId64 " nanoseconds.",
                        conn->first_packet_timestamp, -lead_time);
                  conn->ab_buffering = 0;
                } else {
                  // do some calculations
                  if ((config.audio_backend_silent_lead_in_time_auto == 1) ||
                      (lead_time <=
                       (int64_t)(config.audio_backend_silent_lead_in_time * (int64_t)1000000000))) {
                    // debug(1, "Lead time: %" PRId64 " nanoseconds.", lead_time);
                    int resp = 0;
                    dac_delay = 0;
                    if (have_sent_prefiller_silence != 0)
                      resp = config.output->delay(
                          &dac_delay); // we know the output device must have a delay function
                    if (resp == 0) {
                      int64_t gross_frame_gap =
                          ((conn->first_packet_time_to_play - local_time_now) *
                           config.output_rate) /
                          1000000000;
                      int64_t exact_frame_gap = gross_frame_gap - dac_delay;
                      int64_t frames_needed_to_maintain_desired_buffer =
                          (int64_t)(config.audio_backend_buffer_desired_length *
                                    config.output_rate) -
                          dac_delay;
                      // below, remember that exact_frame_gap and
                      // frames_needed_to_maintain_desired_buffer could both be negative
                      int64_t fs = frames_needed_to_maintain_desired_buffer;

                      // if there isn't enough time to have the desired buffer size
                      if (exact_frame_gap <= frames_needed_to_maintain_desired_buffer) {
                        fs = conn->max_frames_per_packet * 2;
                      }
                      // if we are very close to the end of buffering, i.e. within two
                      // frame-lengths, add the remaining silence needed and end buffering
                      if (exact_frame_gap <= conn->max_frames_per_packet * 2) {
                        fs = exact_frame_gap;
                        if (fs > first_frame_early_bias)
                          fs = fs - first_frame_early_bias; // deliberately make the first packet a
                                                            // tiny bit early so that the player may
                                                            // compensate for it at the last minute
                        conn->ab_buffering = 0;
                      }
                      void *silence;
                      if (fs > 0) {
                        silence = malloc(conn->output_bytes_per_frame * fs);
                        if (silence == NULL)
                          debug(1, "Failed to allocate %d byte silence buffer.", fs);
                        else {
                          // generate frames of silence with dither if necessary
                          conn->previous_random_number = generate_zero_frames(
                              silence, fs, config.output_format, conn->enable_dither,
                              conn->previous_random_number);
                          config.output->play(silence, fs, play_samples_are_untimed, 0, 0);
                          debug(3, "Sent %" PRId64 " frames of silence", fs);
                          free(silence);
                          have_sent_prefiller_silence = 1;
                        }
                      }
                    } else {

                      if (resp == sps_extra_code_output_stalled) {
                        if (conn->unfixable_error_reported == 0) {
                          conn->unfixable_error_reported = 1;
                          if (config.cmd_unfixable) {
                            command_execute(config.cmd_unfixable, "output_device_stalled", 1);
                          } else {
                            die("an unrecoverable error, \"output_device_stalled\", has been "
                                "detected.",
                                conn->connection_number);
                          }
                        }
                      } else {
                        debug(3, "Unexpected response to getting dac delay: %d.", resp);
                      }
                    }
                  }
                }
              } else {
                // if the output device doesn't have a delay, we simply send the lead-in
                int64_t lead_time =
                    conn->first_packet_time_to_play - local_time_now; // negative if we are late
                void *silence;
                int64_t frame_gap = (lead_time * config.output_rate) / 1000000000;
                // debug(1,"%d frames needed.",frame_gap);
                while (frame_gap > 0) {
                  ssize_t fs = config.output_rate / 10;
                  if (fs > frame_gap)
                    fs = frame_gap;

                  silence = malloc(conn->output_bytes_per_frame * fs);
                  if (silence == NULL)
                    debug(1, "Failed to allocate %d frame silence buffer.", fs);
                  else {
                    // debug(1, "No delay function -- outputting %d frames of silence.", fs);
                    conn->previous_random_number =
                        generate_zero_frames(silence, fs, config.output_format, conn->enable_dither,
                                             conn->previous_random_number);
                    config.output->play(silence, fs, play_samples_are_untimed, 0, 0);
                    free(silence);
                  }
                  frame_gap -= fs;
                }
                conn->ab_buffering = 0;
              }
            }
#ifdef CONFIG_METADATA
            if (conn->ab_buffering == 0) {
              debug(2, "prsm");
              send_ssnc_metadata('prsm', NULL, 0,
                                 0); // "resume", but don't wait if the queue is locked
            }
#endif
          }
        }
      }

      // Here, we work out whether to release a packet or wait
      // We release a packet when the time is right.

      // To work out when the time is right, we need to take account of (1) the actual time the
      // packet should be released, (2) the latency requested, (3) the audio backend latency offset
      // and (4) the desired length of the audio backend's buffer

      // The time is right if the current time is later or the same as
      // The packet time + (latency + latency offset - backend_buffer_length).
      // Note: the last three items are expressed in frames and must be converted to time.

      int do_wait = 0; // don't wait unless we can really prove we must
      if ((conn->ab_synced) && (curframe) && (curframe->ready) && (curframe->given_timestamp)) {
        do_wait = 1; // if the current frame exists and is ready, then wait unless it's time to let
                     // it go...

        // here, get the time to play the current frame.

        if (have_timestamp_timing_information(conn)) { // if we have a reference time

          uint64_t time_to_play;

          // we must enable packets to be released early enough for the
          // audio buffer to be filled to the desired length

          uint32_t buffer_latency_offset =
              (uint32_t)(config.audio_backend_buffer_desired_length * conn->input_rate);
          frame_to_local_time(curframe->given_timestamp -
                                  buffer_latency_offset, // this will go modulo 2^32
                              &time_to_play, conn);

          if (local_time_now >= time_to_play) {
            do_wait = 0;
          }
          // here, do a sanity check. if the time_to_play is not within a few seconds of the
          // time now, the frame is probably not meant to be there, so let it go.
          if (do_wait != 0) {
            // this is a hack.
            // we subtract two 2^n unsigned numbers and get a signed 2^n result.
            // If we think of the calculation as occurring in modulo 2^n arithmetic
            // then the signed result's magnitude represents the shorter distance around
            // the modulo wheel of values from one number to the other.
            // The sign indicates the direction: positive means clockwise (upwards) from the
            // second number to the first (i.e. the first number comes "after" the second).

            int64_t time_difference = local_time_now - time_to_play;
            if ((time_difference > 10000000000) || (time_difference < -10000000000)) {
              debug(2,
                    "crazy time interval of %f seconds between time now: 0x%" PRIx64
                    " and time of packet: %" PRIx64 ".",
                    0.000000001 * time_difference, local_time_now, time_to_play);
              debug(2, "packet rtptime: %u, reference_timestamp: %u", curframe->given_timestamp,
                    conn->anchor_rtptime);

              do_wait = 0; // let it go
            }
          }
        }
      }
      if (do_wait == 0)
        if ((conn->ab_synced != 0) && (conn->ab_read == conn->ab_write)) { // the buffer is empty!
          if (notified_buffer_empty == 0) {
            debug(2, "Connection %d: Buffer Empty", conn->connection_number);
            notified_buffer_empty = 1;
            // reset_input_flow_metrics(conn); // don't do a full flush parameters reset
            conn->initial_reference_time = 0;
            conn->initial_reference_timestamp = 0;
            conn->first_packet_timestamp = 0; // make sure the first packet isn't late
          }
          do_wait = 1;
        }
      wait = (conn->ab_buffering || (do_wait != 0) || (!conn->ab_synced));
    } else {
      wait = 1; // keep waiting until the timing information becomes available
    }
    if (wait) {
      if (conn->input_rate == 0)
        die("input_rate is zero -- should never happen!");
      uint64_t time_to_wait_for_wakeup_ns =
          1000000000 / conn->input_rate;      // this is time period of one frame
      time_to_wait_for_wakeup_ns *= 12 * 352; // two full 352-frame packets
      time_to_wait_for_wakeup_ns /= 3;        // two thirds of a packet time

#ifdef COMPILE_FOR_LINUX_AND_FREEBSD_AND_CYGWIN_AND_OPENBSD
      uint64_t time_of_wakeup_ns = get_realtime_in_ns() + time_to_wait_for_wakeup_ns;
      uint64_t sec = time_of_wakeup_ns / 1000000000;
      uint64_t nsec = time_of_wakeup_ns % 1000000000;

      struct timespec time_of_wakeup;
      time_of_wakeup.tv_sec = sec;
      time_of_wakeup.tv_nsec = nsec;

      int rc = pthread_cond_timedwait(&conn->flowcontrol, &conn->ab_mutex,
                                      &time_of_wakeup); // this is a pthread cancellation point
      if ((rc != 0) && (rc != ETIMEDOUT))
        debug(3, "pthread_cond_timedwait returned error code %d.", rc);
#endif
#ifdef COMPILE_FOR_OSX
      uint64_t sec = time_to_wait_for_wakeup_ns / 1000000000;
      uint64_t nsec = time_to_wait_for_wakeup_ns % 1000000000;
      struct timespec time_to_wait;
      time_to_wait.tv_sec = sec;
      time_to_wait.tv_nsec = nsec;
      pthread_cond_timedwait_relative_np(&conn->flowcontrol, &conn->ab_mutex, &time_to_wait);
#endif
    }
  } while (wait);

  // seq_t read = conn->ab_read;
  if (curframe) {
    if (!curframe->ready) {
      // debug(1, "Supplying a silent frame for frame %u", read);
      conn->missing_packets++;
      curframe->given_timestamp = 0; // indicate a silent frame should be substituted
    }
    curframe->ready = 0;
  }
  conn->ab_read++;
  pthread_cleanup_pop(1);
  return curframe;
}

static inline int32_t mean_32(int32_t a, int32_t b) {
  int64_t al = a;
  int64_t bl = b;
  int64_t mean = (al + bl) / 2;
  int32_t r = (int32_t)mean;
  if (r != mean)
    debug(1, "Error calculating average of two int32_t values: %d, %d.", a, b);
  return r;
}

// this takes an array of signed 32-bit integers and (a) removes or inserts a frame as specified in
// stuff,
// (b) multiplies each sample by the fixedvolume (a 16-bit quantity)
// (c) dithers the result to the output size 32/24/16/8 bits
// (d) outputs the result in the approprate format
// formats accepted so far include U8, S8, S16, S24, S24_3LE, S24_3BE and S32

// stuff: 1 means add 1; 0 means do nothing; -1 means remove 1
static int stuff_buffer_basic_32(int32_t *inptr, int length, sps_format_t l_output_format,
                                 char *outptr, int stuff, int dither, rtsp_conn_info *conn) {
  if (length < 3)
    die("buffer length expected to be 3 or more, but is %d!", length);
  int tstuff = stuff;
  char *l_outptr = outptr;
  if ((stuff > 1) || (stuff < -1) || (length < 100)) {
    // debug(1, "Stuff argument to stuff_buffer must be from -1 to +1 and length >100.");
    tstuff = 0; // if any of these conditions hold, don't stuff anything/
  }

  int i;
  int stuffsamp = length;
  if (tstuff)
    //      stuffsamp = rand() % (length - 1);
    stuffsamp =
        (rand() % (length - 2)) + 1; // ensure there's always a sample before and after the item

  for (i = 0; i < stuffsamp; i++) { // the whole frame, if no stuffing
    process_sample(*inptr++, &l_outptr, l_output_format, conn->fix_volume, dither, conn);
    process_sample(*inptr++, &l_outptr, l_output_format, conn->fix_volume, dither, conn);
  };
  if (tstuff) {
    if (tstuff == 1) {
      // debug(3, "+++++++++");
      // interpolate one sample
      process_sample(mean_32(inptr[-2], inptr[0]), &l_outptr, l_output_format, conn->fix_volume,
                     dither, conn);
      process_sample(mean_32(inptr[-1], inptr[1]), &l_outptr, l_output_format, conn->fix_volume,
                     dither, conn);
    } else if (stuff == -1) {
      // debug(3, "---------");
      inptr++;
      inptr++;
    }

    // if you're removing, i.e. stuff < 0, copy that much less over. If you're adding, do all the
    // rest.
    int remainder = length;
    if (tstuff < 0)
      remainder = remainder + tstuff; // don't run over the correct end of the output buffer

    for (i = stuffsamp; i < remainder; i++) {
      process_sample(*inptr++, &l_outptr, l_output_format, conn->fix_volume, dither, conn);
      process_sample(*inptr++, &l_outptr, l_output_format, conn->fix_volume, dither, conn);
    }
  }
  conn->amountStuffed = tstuff;
  return length + tstuff;
}

#ifdef CONFIG_SOXR
// this takes an array of signed 32-bit integers and
// (a) uses libsoxr to
// resample the array to have one more or one less frame, as specified in
// stuff,
// (b) multiplies each sample by the fixedvolume (a 16-bit quantity)
// (c) dithers the result to the output size 32/24/16/8 bits
// (d) outputs the result in the approprate format
// formats accepted so far include U8, S8, S16, S24, S24_3LE, S24_3BE and S32

int32_t stat_n = 0;
double stat_mean = 0.0;
double stat_M2 = 0.0;
double longest_soxr_execution_time = 0.0;
int64_t packets_processed = 0;

int stuff_buffer_soxr_32(int32_t *inptr, int32_t *scratchBuffer, int length,
                         sps_format_t l_output_format, char *outptr, int stuff, int dither,
                         rtsp_conn_info *conn) {
  if (scratchBuffer == NULL) {
    die("soxr scratchBuffer not initialised.");
  }
  packets_processed++;
  int tstuff = stuff;
  if ((stuff > 1) || (stuff < -1) || (length < 100)) {
    // debug(1, "Stuff argument to stuff_buffer must be from -1 to +1 and length >100.");
    tstuff = 0; // if any of these conditions hold, don't stuff anything/
  }

  if (tstuff) {
    // debug(1,"Stuff %d.",stuff);

    soxr_io_spec_t io_spec;
    io_spec.itype = SOXR_INT32_I;
    io_spec.otype = SOXR_INT32_I;
    io_spec.scale = 1.0; // this seems to crash if not = 1.0
    io_spec.e = NULL;
    io_spec.flags = 0;

    size_t odone;

    uint64_t soxr_start_time = get_absolute_time_in_ns();

    soxr_error_t error = soxr_oneshot(length, length + tstuff, 2, // Rates and # of chans.
                                      inptr, length, NULL,        // Input.
                                      scratchBuffer, length + tstuff, &odone, // Output.
                                      &io_spec,    // Input, output and transfer spec.
                                      NULL, NULL); // Default configuration.

    if (error)
      die("soxr error: %s\n", "error: %s\n", soxr_strerror(error));

    if (odone > (size_t)(length + 1))
      die("odone = %u!\n", odone);

    // mean and variance calculations from "online_variance" algorithm at
    // https://en.wikipedia.org/wiki/Algorithms_for_calculating_variance#Online_algorithm

    double soxr_execution_time = (get_absolute_time_in_ns() - soxr_start_time) * 0.000000001;
    // debug(1,"soxr_execution_time_us: %10.1f",soxr_execution_time_us);
    if (soxr_execution_time > longest_soxr_execution_time)
      longest_soxr_execution_time = soxr_execution_time;
    stat_n += 1;
    double stat_delta = soxr_execution_time - stat_mean;
    if (stat_n != 0)
      stat_mean += stat_delta / stat_n;
    else
      warn("calculation error for stat_n");
    stat_M2 += stat_delta * (soxr_execution_time - stat_mean);

    int i;
    int32_t *ip, *op;
    ip = inptr;
    op = scratchBuffer;

    const int gpm = 5;
    // keep the first (dpm) samples, to mitigate the Gibbs phenomenon
    for (i = 0; i < gpm; i++) {
      *op++ = *ip++;
      *op++ = *ip++;
    }

    // keep the last (dpm) samples, to mitigate the Gibbs phenomenon

    // pointer arithmetic, baby -- it's da bomb.
    op = scratchBuffer + (length + tstuff - gpm) * 2;
    ip = inptr + (length - gpm) * 2;
    for (i = 0; i < gpm; i++) {
      *op++ = *ip++;
      *op++ = *ip++;
    }

    // now, do the volume, dither and formatting processing
    ip = scratchBuffer;
    char *l_outptr = outptr;
    for (i = 0; i < length + tstuff; i++) {
      process_sample(*ip++, &l_outptr, l_output_format, conn->fix_volume, dither, conn);
      process_sample(*ip++, &l_outptr, l_output_format, conn->fix_volume, dither, conn);
    };

  } else { // the whole frame, if no stuffing

    // now, do the volume, dither and formatting processing
    int32_t *ip = inptr;
    char *l_outptr = outptr;
    int i;

    for (i = 0; i < length; i++) {
      process_sample(*ip++, &l_outptr, l_output_format, conn->fix_volume, dither, conn);
      process_sample(*ip++, &l_outptr, l_output_format, conn->fix_volume, dither, conn);
    };
  }

  if (packets_processed % 1250 == 0) {
    debug(3,
          "soxr_oneshot execution time in nanoseconds: mean, standard deviation and max "
          "for %" PRId32 " interpolations in the last "
          "1250 packets. %10.6f, %10.6f, %10.6f.",
          stat_n, stat_mean, stat_n <= 1 ? 0.0 : sqrtf(stat_M2 / (stat_n - 1)),
          longest_soxr_execution_time);
    stat_n = 0;
    stat_mean = 0.0;
    stat_M2 = 0.0;
    longest_soxr_execution_time = 0.0;
  }

  conn->amountStuffed = tstuff;
  return length + tstuff;
}
#endif

void player_thread_initial_cleanup_handler(__attribute__((unused)) void *arg) {
  rtsp_conn_info *conn = (rtsp_conn_info *)arg;
  debug(3, "Connection %d: player thread main loop exit via player_thread_initial_cleanup_handler.",
        conn->connection_number);
}

char line_of_stats[1024];
int statistics_row; // statistics_line 0 means print the headings; anything else 1 means print the
                    // values. Set to 0 the first time out.
int statistics_column; // used to index through the statistics_print_profile array to check if it
                       // should be printed
int was_a_previous_column;
int *statistics_print_profile;

// these arrays specify which of the statistics specified by the statistics_item calls will actually
// be printed -- 2 means print, 1 means print only in a debug mode, 0 means skip

// clang-format off
int ap1_synced_statistics_print_profile[] =                  {2, 2, 2, 0, 2, 1, 1, 2, 1, 1, 1, 0, 1, 1, 2, 2, 1, 1};
int ap1_nosync_statistics_print_profile[] =                  {2, 0, 0, 0, 2, 1, 1, 2, 1, 1, 1, 0, 1, 1, 0, 0, 1, 0};
int ap1_nodelay_statistics_print_profile[] =                 {0, 0, 0, 0, 2, 1, 1, 2, 0, 1, 1, 0, 1, 1, 0, 0, 1, 0};

int ap2_realtime_synced_stream_statistics_print_profile[] =  {2, 2, 2, 0, 2, 1, 1, 2, 1, 1, 1, 0, 0, 1, 2, 2, 0, 0};
int ap2_realtime_nosync_stream_statistics_print_profile[] =  {2, 0, 0, 0, 2, 1, 1, 2, 1, 1, 1, 0, 0, 1, 0, 0, 0, 0};
int ap2_realtime_nodelay_stream_statistics_print_profile[] = {0, 0, 0, 0, 2, 1, 1, 2, 0, 1, 1, 0, 0, 1, 0, 0, 0, 0};

int ap2_buffered_synced_stream_statistics_print_profile[] =  {2, 2, 2, 0, 0, 0, 0, 0, 1, 1, 0, 1, 0, 0, 2, 2, 0, 0};
int ap2_buffered_nosync_stream_statistics_print_profile[] =  {2, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 1, 0, 0, 0, 0, 0, 0};
int ap2_buffered_nodelay_stream_statistics_print_profile[] = {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0, 0, 0, 0};
// clang-format on

void statistics_item(const char *heading, const char *format, ...) {
  if (((statistics_print_profile[statistics_column] == 1) && (debuglev != 0)) ||
      (statistics_print_profile[statistics_column] == 2)) { // include this column?
    if (was_a_previous_column != 0) {
      if (statistics_row == 0)
        strcat(line_of_stats, " | ");
      else
        strcat(line_of_stats, "   ");
    }
    if (statistics_row == 0) {
      strcat(line_of_stats, heading);
    } else {
      char b[1024];
      b[0] = 0;
      va_list args;
      va_start(args, format);
      vsnprintf(b, sizeof(b), format, args);
      va_end(args);
      strcat(line_of_stats, b);
    }
    was_a_previous_column = 1;
  }
  statistics_column++;
}

void player_thread_cleanup_handler(void *arg) {
  rtsp_conn_info *conn = (rtsp_conn_info *)arg;
  if (pthread_mutex_trylock(&playing_conn_lock) == 0) {

    pthread_cleanup_push(mutex_unlock, &playing_conn_lock);
    if (playing_conn == conn) {
      if (config.output->stop) {
        debug(3, "Connection %d: Stop the output backend.", conn->connection_number);
        config.output->stop();
      }
    } else {
      debug(1, "This is not the playing conn.");
    }
    pthread_cleanup_pop(1); // unlock the mutex
  } else {
    debug(1, "Can not acquire play lock.");
  }

  int oldState;
  pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState);
  debug(3, "Connection %d: player thread main loop exit via player_thread_cleanup_handler.",
        conn->connection_number);

  if (config.statistics_requested) {
    int64_t time_playing = get_absolute_time_in_ns() - conn->playstart;
    time_playing = time_playing / 1000000000;
    int64_t elapsedHours = time_playing / 3600;
    int64_t elapsedMin = (time_playing / 60) % 60;
    int64_t elapsedSec = time_playing % 60;
    if (conn->frame_rate_valid)
      inform("Connection %d: Playback Stopped. Total playing time %02" PRId64 ":%02" PRId64
             ":%02" PRId64 ". "
             "Output: %0.2f (raw), %0.2f (corrected) "
             "frames per second.",
             conn->connection_number, elapsedHours, elapsedMin, elapsedSec, conn->raw_frame_rate,
             conn->corrected_frame_rate);
    else
      inform("Connection %d: Playback Stopped. Total playing time %02" PRId64 ":%02" PRId64
             ":%02" PRId64 ".",
             conn->connection_number, elapsedHours, elapsedMin, elapsedSec);
  }

#ifdef CONFIG_DACP_CLIENT
  relinquish_dacp_server_information(
      conn); // say it doesn't belong to this conversation thread any more...
#else
  mdns_dacp_monitor_set_id(NULL); // say we're not interested in following that DACP id any more
#endif

  // four possibilities
  // 1 -- regular AirPlay 1
  // 2 -- AirPlay 2 in AirPlay 1 mode
  // 3 -- AirPlay 2 in Buffered Audio Mode
  // 4 -- AirPlay 3 in Realtime Audio Mode.

#ifdef CONFIG_AIRPLAY_2
  if (conn->airplay_type == ap_2) {
    debug(2, "Cancelling AP2 timing, control and audio threads...");

    if (conn->airplay_stream_type == realtime_stream) {
      debug(2, "Connection %d: Delete Realtime Audio Stream thread", conn->connection_number);
      pthread_cancel(conn->rtp_realtime_audio_thread);
      pthread_join(conn->rtp_realtime_audio_thread, NULL);

    } else if (conn->airplay_stream_type == buffered_stream) {

      debug(2, "Connection %d: Delete Buffered Audio Stream thread", conn->connection_number);
      pthread_cancel(conn->rtp_buffered_audio_thread);
      pthread_join(conn->rtp_buffered_audio_thread, NULL);

    } else {
      die("Unrecognised Stream Type");
    }

    debug(2, "Connection %d: Delete AirPlay 2 Control thread");
    pthread_cancel(conn->rtp_ap2_control_thread);
    pthread_join(conn->rtp_ap2_control_thread, NULL);

  } else {
    debug(2, "Cancelling AP1-compatible timing, control and audio threads...");
#else
  debug(2, "Cancelling AP1 timing, control and audio threads...");
#endif
    debug(3, "Cancel timing thread.");
    pthread_cancel(conn->rtp_timing_thread);
    debug(3, "Join timing thread.");
    pthread_join(conn->rtp_timing_thread, NULL);
    debug(3, "Timing thread terminated.");
    debug(3, "Cancel control thread.");
    pthread_cancel(conn->rtp_control_thread);
    debug(3, "Join control thread.");
    pthread_join(conn->rtp_control_thread, NULL);
    debug(3, "Control thread terminated.");
    debug(3, "Cancel audio thread.");
    pthread_cancel(conn->rtp_audio_thread);
    debug(3, "Join audio thread.");
    pthread_join(conn->rtp_audio_thread, NULL);
    debug(3, "Audio thread terminated.");
#ifdef CONFIG_AIRPLAY_2
  }
  // ptp_send_control_message_string("T"); // remove all timing peers to force the master to 0
  reset_anchor_info(conn);
#endif

  if (conn->outbuf) {
    free(conn->outbuf);
    conn->outbuf = NULL;
  }
  if (conn->sbuf) {
    free(conn->sbuf);
    conn->sbuf = NULL;
  }
  if (conn->tbuf) {
    free(conn->tbuf);
    conn->tbuf = NULL;
  }

  if (conn->statistics) {
    free(conn->statistics);
    conn->statistics = NULL;
  }
  free_audio_buffers(conn);
  if (conn->stream.type == ast_apple_lossless)
    terminate_decoders(conn);

  // reset_anchor_info(conn);
  // release_play_lock(conn);
  conn->rtp_running = 0;
  pthread_setcancelstate(oldState, NULL);
  debug(2, "Connection %d: player terminated.", conn->connection_number);
}

void *player_thread_func(void *arg) {
  rtsp_conn_info *conn = (rtsp_conn_info *)arg;

  uint64_t previous_frames_played = 0; // initialised to avoid a "possibly uninitialised" warning
  uint64_t previous_raw_measurement_time =
      0; // initialised to avoid a "possibly uninitialised" warning
  uint64_t previous_corrected_measurement_time =
      0; // initialised to avoid a "possibly uninitialised" warning
  int previous_frames_played_valid = 0;

  // pthread_cleanup_push(player_thread_initial_cleanup_handler, arg);
  conn->latency_warning_issued =
      0; // be permitted to generate a warning each time a play is attempted
  conn->packet_count = 0;
  conn->packet_count_since_flush = 0;
  conn->previous_random_number = 0;
  conn->decoder_in_use = 0;
  conn->ab_buffering = 1;
  conn->ab_synced = 0;
  conn->first_packet_timestamp = 0;
  conn->flush_requested = 0;
  conn->flush_output_flushed = 0; // only send a flush command to the output device once
  conn->flush_rtp_timestamp = 0;  // it seems this number has a special significance -- it seems to
                                  // be used as a null operand, so we'll use it like that too
  conn->fix_volume = 0x10000;

#ifdef CONFIG_AIRPLAY_2
  conn->ap2_flush_requested = 0;
  conn->ap2_flush_from_valid = 0;
  conn->ap2_rate = 0;
  conn->ap2_play_enabled = 0;
#endif

  // reset_anchor_info(conn);

  if (conn->stream.type == ast_apple_lossless)
    init_alac_decoder((int32_t *)&conn->stream.fmtp,
                      conn); // this sets up incoming rate, bit depth, channels.
                             // No pthread cancellation point in here
  // This must be after init_alac_decoder
  init_buffer(conn); // will need a corresponding deallocation. No cancellation points in here
  ab_resync(conn);

  if (conn->stream.encrypted) {
#ifdef CONFIG_MBEDTLS
    memset(&conn->dctx, 0, sizeof(mbedtls_aes_context));
    mbedtls_aes_setkey_dec(&conn->dctx, conn->stream.aeskey, 128);
#endif

#ifdef CONFIG_POLARSSL
    memset(&conn->dctx, 0, sizeof(aes_context));
    aes_setkey_dec(&conn->dctx, conn->stream.aeskey, 128);
#endif

#ifdef CONFIG_OPENSSL
    AES_set_decrypt_key(conn->stream.aeskey, 128, &conn->aes);
#endif
  }

  conn->timestamp_epoch = 0; // indicate that the next timestamp will be the first one.
  conn->maximum_timestamp_interval = conn->input_rate * 60; // actually there shouldn't be more than
                                                            // about 13 seconds of a gap between
                                                            // successive rtptimes, at worst

  conn->output_sample_ratio = config.output_rate / conn->input_rate;

  // Sign extending rtptime calculations to 64 bit is needed from time to time.

  // The standard rtptime is unsigned 32 bits,
  // so you can do modulo 2^32 difference calculations
  // and get a signed result simply by typing the result as a signed 32-bit number.

  // So long as you can be sure the numbers are within 2^31 of each other,
  // the sign of the result calculated in this way indicates the order of the operands.
  // For example, if you subtract a from b and the result is positive, you can conclude
  // b is the same as or comes after a in module 2^32 order.

  // We want to do the same with the rtptime calculations for multiples of
  // the rtptimes (1, 2, 4 or 8 times), and we want to do this in signed 64-bit/
  // Therefore we need to sign extend these modulo 2^32, 2^33, 2^34, or 2^35 bit unsigned
  // numbers on the same basis.

  // That is what the output_rtptime_sign_bit, output_rtptime_mask, output_rtptime_mask_not and
  // output_rtptime_sign_mask are for -- see later, calculating the sync error.

  int output_rtptime_sign_bit;
  switch (conn->output_sample_ratio) {
  case 1:
    output_rtptime_sign_bit = 31;
    break;
  case 2:
    output_rtptime_sign_bit = 32;
    break;
  case 4:
    output_rtptime_sign_bit = 33;
    break;
  case 8:
    output_rtptime_sign_bit = 34;
    break;
  default:
    debug(1, "error with output ratio -- can't calculate sign bit number");
    output_rtptime_sign_bit = 31;
    break;
  }

  //  debug(1, "Output sample ratio is %d.", conn->output_sample_ratio);
  //  debug(1, "Output output_rtptime_sign_bit: %d.", output_rtptime_sign_bit);

  int64_t output_rtptime_mask = 1;
  output_rtptime_mask = output_rtptime_mask << (output_rtptime_sign_bit + 1);
  output_rtptime_mask = output_rtptime_mask - 1;

  int64_t output_rtptime_mask_not = output_rtptime_mask;
  output_rtptime_mask_not = ~output_rtptime_mask;

  int64_t output_rtptime_sign_mask = 1;
  output_rtptime_sign_mask = output_rtptime_sign_mask << output_rtptime_sign_bit;

  conn->max_frame_size_change =
      1 * conn->output_sample_ratio; // we add or subtract one frame at the nominal
                                     // rate, multiply it by the frame ratio.
                                     // but, on some occasions, more than one frame could be added

  switch (config.output_format) {
  case SPS_FORMAT_S24_3LE:
  case SPS_FORMAT_S24_3BE:
    conn->output_bytes_per_frame = 6;
    break;

  case SPS_FORMAT_S24:
  case SPS_FORMAT_S24_LE:
  case SPS_FORMAT_S24_BE:
    conn->output_bytes_per_frame = 8;
    break;
  case SPS_FORMAT_S32:
  case SPS_FORMAT_S32_LE:
  case SPS_FORMAT_S32_BE:
    conn->output_bytes_per_frame = 8;
    break;
  default:
    conn->output_bytes_per_frame = 4;
  }

  debug(3, "Output frame bytes is %d.", conn->output_bytes_per_frame);

  conn->dac_buffer_queue_minimum_length = (uint64_t)(
      config.audio_backend_buffer_interpolation_threshold_in_seconds * config.output_rate);
  debug(3, "dac_buffer_queue_minimum_length is %" PRIu64 " frames.",
        conn->dac_buffer_queue_minimum_length);

  conn->session_corrections = 0;
  conn->connection_state_to_output = get_requested_connection_state_to_output();
// this is about half a minute
//#define trend_interval 3758

// this is about 8 seconds
#define trend_interval 1003

  int number_of_statistics, oldest_statistic, newest_statistic;
  int at_least_one_frame_seen = 0;
  int at_least_one_frame_seen_this_session = 0;
  int64_t tsum_of_sync_errors, tsum_of_corrections, tsum_of_insertions_and_deletions,
      tsum_of_drifts;
  int64_t previous_sync_error = 0, previous_correction = 0;
  uint64_t minimum_dac_queue_size = UINT64_MAX;
  int32_t minimum_buffer_occupancy = INT32_MAX;
  int32_t maximum_buffer_occupancy = INT32_MIN;

#ifdef CONFIG_AIRPLAY_2
  conn->ap2_audio_buffer_minimum_size = -1;
#endif

  conn->raw_frame_rate = 0.0;
  conn->corrected_frame_rate = 0.0;
  conn->frame_rate_valid = 0;

  conn->input_frame_rate = 0.0;
  conn->input_frame_rate_starting_point_is_valid = 0;

  conn->buffer_occupancy = 0;

  int play_samples = 0;
  uint64_t current_delay;
  int play_number = 0;
  conn->play_number_after_flush = 0;
  //  int last_timestamp = 0; // for debugging only
  conn->time_of_last_audio_packet = 0;
  // conn->shutdown_requested = 0;
  number_of_statistics = oldest_statistic = newest_statistic = 0;
  tsum_of_sync_errors = tsum_of_corrections = tsum_of_insertions_and_deletions = tsum_of_drifts = 0;

  const int print_interval = trend_interval; // don't ask...
  // I think it's useful to keep this prime to prevent it from falling into a pattern with some
  // other process.

  static char rnstate[256];
  initstate(time(NULL), rnstate, 256);

  signed short *inbuf;
  int inbuflength;

  unsigned int output_bit_depth = 16; // default;

  switch (config.output_format) {
  case SPS_FORMAT_S8:
  case SPS_FORMAT_U8:
    output_bit_depth = 8;
    break;
  case SPS_FORMAT_S16:
  case SPS_FORMAT_S16_LE:
  case SPS_FORMAT_S16_BE:
    output_bit_depth = 16;
    break;
  case SPS_FORMAT_S24:
  case SPS_FORMAT_S24_LE:
  case SPS_FORMAT_S24_BE:
  case SPS_FORMAT_S24_3LE:
  case SPS_FORMAT_S24_3BE:
    output_bit_depth = 24;
    break;
  case SPS_FORMAT_S32:
  case SPS_FORMAT_S32_LE:
  case SPS_FORMAT_S32_BE:
    output_bit_depth = 32;
    break;
  case SPS_FORMAT_UNKNOWN:
    die("Unknown format choosing output bit depth");
    break;
  case SPS_FORMAT_AUTO:
    die("Invalid format -- SPS_FORMAT_AUTO -- choosing output bit depth");
    break;
  case SPS_FORMAT_INVALID:
    die("Invalid format -- SPS_FORMAT_INVALID -- choosing output bit depth");
    break;
  }

  debug(3, "Output bit depth is %d.", output_bit_depth);

  if (conn->input_bit_depth > output_bit_depth) {
    debug(3, "Dithering will be enabled because the input bit depth is greater than the output bit "
             "depth");
  }
  if (config.output->parameters == NULL) {
    debug(3, "Dithering will be enabled because the output volume is being altered in software");
  }

  if ((config.output->parameters == NULL) || (conn->input_bit_depth > output_bit_depth) ||
      (config.playback_mode == ST_mono))
    conn->enable_dither = 1;

  // remember, the output device may never have been initialised prior to this call
  config.output->start(config.output_rate, config.output_format); // will need a corresponding stop

  // we need an intermediate "transition" buffer

  conn->tbuf = malloc(
      sizeof(int32_t) * 2 *
      (conn->max_frames_per_packet * conn->output_sample_ratio + conn->max_frame_size_change));
  if (conn->tbuf == NULL)
    die("Failed to allocate memory for the transition buffer.");

  // initialise this, because soxr stuffing might be chosen later

  conn->sbuf = malloc(
      sizeof(int32_t) * 2 *
      (conn->max_frames_per_packet * conn->output_sample_ratio + conn->max_frame_size_change));
  if (conn->sbuf == NULL)
    die("Failed to allocate memory for the sbuf buffer.");

  // The size of these dependents on the number of frames, the size of each frame and the maximum
  // size change
  conn->outbuf = malloc(
      conn->output_bytes_per_frame *
      (conn->max_frames_per_packet * conn->output_sample_ratio + conn->max_frame_size_change));
  if (conn->outbuf == NULL)
    die("Failed to allocate memory for an output buffer.");
  conn->first_packet_timestamp = 0;
  conn->missing_packets = conn->late_packets = conn->too_late_packets = conn->resend_requests = 0;
  int sync_error_out_of_bounds =
      0; // number of times in a row that there's been a serious sync error

  conn->statistics = malloc(sizeof(stats_t) * trend_interval);
  if (conn->statistics == NULL)
    die("Failed to allocate a statistics buffer");

  conn->framesProcessedInThisEpoch = 0;
  conn->framesGeneratedInThisEpoch = 0;
  conn->correctionsRequestedInThisEpoch = 0;
  statistics_row = 0; // statistics_line 0 means print the headings; anything else 1 means print the
                      // values. Set to 0 the first time out.

  // decide on what statistics profile to use, if requested
#ifdef CONFIG_AIRPLAY_2
  if (conn->airplay_type == ap_2) {
    if (conn->airplay_stream_type == realtime_stream) {
      if (config.output->delay) {
        if (config.no_sync == 0)
          statistics_print_profile = ap2_realtime_synced_stream_statistics_print_profile;
        else
          statistics_print_profile = ap2_realtime_nosync_stream_statistics_print_profile;
      } else {
        statistics_print_profile = ap2_realtime_nodelay_stream_statistics_print_profile;
      }
    } else {
      if (config.output->delay) {
        if (config.no_sync == 0)
          statistics_print_profile = ap2_buffered_synced_stream_statistics_print_profile;
        else
          statistics_print_profile = ap2_buffered_nosync_stream_statistics_print_profile;
      } else {
        statistics_print_profile = ap2_buffered_nodelay_stream_statistics_print_profile;
      }
    }
  } else {
#endif
    if (config.output->delay) {
      if (config.no_sync == 0)
        statistics_print_profile = ap1_synced_statistics_print_profile;
      else
        statistics_print_profile = ap1_nosync_statistics_print_profile;
    } else {
      statistics_print_profile = ap1_nodelay_statistics_print_profile;
    }
// airplay 1 stuff here
#ifdef CONFIG_AIRPLAY_2
  }
#endif

#ifdef CONFIG_AIRPLAY_2
  if (conn->timing_type == ts_ntp) {
#endif

    // create and start the timing, control and audio receiver threads
    pthread_create(&conn->rtp_audio_thread, NULL, &rtp_audio_receiver, (void *)conn);
    pthread_create(&conn->rtp_control_thread, NULL, &rtp_control_receiver, (void *)conn);
    pthread_create(&conn->rtp_timing_thread, NULL, &rtp_timing_receiver, (void *)conn);

#ifdef CONFIG_AIRPLAY_2
  }
#endif

  pthread_cleanup_push(player_thread_cleanup_handler, arg); // undo what's been done so far

  // stop looking elsewhere for DACP stuff
  int oldState;
  pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState);

#ifdef CONFIG_DACP_CLIENT
  set_dacp_server_information(conn);
#else
  mdns_dacp_monitor_set_id(conn->dacp_id);
#endif

  pthread_setcancelstate(oldState, NULL);

  double initial_volume = config.airplay_volume; // default
  // set the default volume to whatever it was before, as stored in the config airplay_volume
  debug(2, "Set initial volume to %f.", initial_volume);
  player_volume(initial_volume, conn); // will contain a cancellation point if asked to wait

  debug(2, "Play begin");
  while (1) {

    // check a few parameters to ensure they are non-zero
    if (config.output_rate == 0)
      debug(1, "config.output_rate is zero!");
    if (conn->output_sample_ratio == 0)
      debug(1, "conn->output_sample_ratio is zero!");
    if (conn->input_rate == 0)
      debug(1, "conn->input_rate is zero!");
    if (conn->input_bytes_per_frame == 0)
      debug(1, "conn->input_bytes_per_frame is zero!");

    pthread_testcancel();                     // allow a pthread_cancel request to take effect.
    abuf_t *inframe = buffer_get_frame(conn); // this has cancellation point(s), but it's not
                                              // guaranteed that they'll always be executed
    uint64_t local_time_now = get_absolute_time_in_ns(); // types okay
    if (inframe) {
      inbuf = inframe->data;
      inbuflength = inframe->length;
      if (inbuf) {
        if (play_number == 0)
          conn->playstart = get_absolute_time_in_ns();
        play_number++;
        //        if (play_number % 100 == 0)
        //          debug(3, "Play frame %d.", play_number);
        conn->play_number_after_flush++;
        if (inframe->given_timestamp == 0) {
          debug(2,
                "Player has supplied a silent frame, (possibly frame %u) for play number %d, "
                "status 0x%X after %u resend requests.",
                conn->last_seqno_read + 1, play_number, inframe->status,
                inframe->resend_request_number);
          conn->last_seqno_read =
              ((conn->last_seqno_read + 1) & 0xffff); // manage the packet out of sequence minder

          void *silence = malloc(conn->output_bytes_per_frame * conn->max_frames_per_packet *
                                 conn->output_sample_ratio);
          if (silence == NULL) {
            debug(1, "Failed to allocate memory for a silent frame silence buffer.");
          } else {
            // the player may change the contents of the buffer, so it has to be zeroed each time;
            // might as well malloc and free it locally
            conn->previous_random_number = generate_zero_frames(
                silence, conn->max_frames_per_packet * conn->output_sample_ratio,
                config.output_format, conn->enable_dither, conn->previous_random_number);
            config.output->play(silence, conn->max_frames_per_packet * conn->output_sample_ratio,
                                play_samples_are_untimed, 0, 0);
            free(silence);
          }
        } else if (conn->play_number_after_flush < 10) {
          /*
          int64_t difference = 0;
          if (last_timestamp)
            difference = inframe->timestamp - last_timestamp;
          last_timestamp = inframe->timestamp;
          debug(1, "Play number %d, monotonic timestamp %llx, difference
          %lld.",conn->play_number_after_flush,inframe->timestamp,difference);
          */
          void *silence = malloc(conn->output_bytes_per_frame * conn->max_frames_per_packet *
                                 conn->output_sample_ratio);
          if (silence == NULL) {
            debug(1, "Failed to allocate memory for a flush silence buffer.");
          } else {
            // the player may change the contents of the buffer, so it has to be zeroed each time;
            // might as well malloc and free it locally
            conn->previous_random_number = generate_zero_frames(
                silence, conn->max_frames_per_packet * conn->output_sample_ratio,
                config.output_format, conn->enable_dither, conn->previous_random_number);
            config.output->play(silence, conn->max_frames_per_packet * conn->output_sample_ratio,
                                play_samples_are_untimed, 0, 0);
            free(silence);
          }
        } else {

          if (((config.output->parameters == NULL) && (config.ignore_volume_control == 0) &&
               (config.airplay_volume != 0.0)) ||
              (conn->input_bit_depth > output_bit_depth) || (config.playback_mode == ST_mono))
            conn->enable_dither = 1;
          else
            conn->enable_dither = 0;

          // here, let's transform the frame of data, if necessary

          switch (conn->input_bit_depth) {
          case 16: {
            int i, j;
            int16_t ls, rs;
            int32_t ll = 0, rl = 0;
            int16_t *inps = inbuf;
            // int16_t *outps = tbuf;
            int32_t *outpl = (int32_t *)conn->tbuf;
            for (i = 0; i < inbuflength; i++) {
              ls = *inps++;
              rs = *inps++;

              // here, do the mode stuff -- mono / reverse stereo / leftonly / rightonly
              // also, raise the 16-bit samples to 32 bits.

              switch (config.playback_mode) {
              case ST_mono: {
                int32_t lsl = ls;
                int32_t rsl = rs;
                int32_t both = lsl + rsl;
                both = both << (16 - 1); // keep all 17 bits of the sum of the 16bit left and right
                                         // -- the 17th bit will influence dithering later
                ll = both;
                rl = both;
              } break;
              case ST_reverse_stereo: {
                ll = rs;
                rl = ls;
                ll = ll << 16;
                rl = rl << 16;
              } break;
              case ST_left_only:
                rl = ls;
                ll = ls;
                ll = ll << 16;
                rl = rl << 16;
                break;
              case ST_right_only:
                ll = rs;
                rl = rs;
                ll = ll << 16;
                rl = rl << 16;
                break;
              case ST_stereo:
                ll = ls;
                rl = rs;
                ll = ll << 16;
                rl = rl << 16;
                break; // nothing extra to do
              }

              // here, replicate the samples if you're upsampling

              for (j = 0; j < conn->output_sample_ratio; j++) {
                *outpl++ = ll;
                *outpl++ = rl;
              }
            }
          } break;
          case 32: {
            int i, j;
            int32_t ls, rs;
            int32_t ll = 0, rl = 0;
            int32_t *inps = (int32_t *)inbuf;
            int32_t *outpl = (int32_t *)conn->tbuf;
            for (i = 0; i < inbuflength; i++) {
              ls = *inps++;
              rs = *inps++;

              // here, do the mode stuff -- mono / reverse stereo / leftonly / rightonly

              switch (config.playback_mode) {
              case ST_mono: {
                int64_t lsl = ls;
                int64_t rsl = rs;
                int64_t both = lsl + rsl;
                both = both >> 1;
                uint32_t both32 = both;
                ll = both32;
                rl = both32;
              } break;
              case ST_reverse_stereo: {
                ll = rs;
                rl = ls;
              } break;
              case ST_left_only:
                rl = ls;
                ll = ls;
                break;
              case ST_right_only:
                ll = rs;
                rl = rs;
                break;
              case ST_stereo:
                ll = ls;
                rl = rs;
                break; // nothing extra to do
              }

              // here, replicate the samples if you're upsampling

              for (j = 0; j < conn->output_sample_ratio; j++) {
                *outpl++ = ll;
                *outpl++ = rl;
              }
            }
          } break;

          default:
            die("Shairport Sync only supports 16 or 32 bit input");
          }

          inbuflength *= conn->output_sample_ratio;

          // We have a frame of data. We need to see if we want to add or remove a frame from it to
          // keep in sync.
          // So we calculate the timing error for the first frame in the DAC.
          // If it's ahead of time, we add one audio frame to this frame to delay a subsequent frame
          // If it's late, we remove an audio frame from this frame to bring a subsequent frame
          // forward in time

          // now, go back as far as the total latency less, say, 100 ms, and check the presence of
          // frames from then onwards

          at_least_one_frame_seen = 1;

          // This is the timing error for the next audio frame in the DAC, if applicable
          int64_t sync_error = 0;

          int amount_to_stuff = 0;

          // check sequencing
          if (conn->last_seqno_read == -1)
            conn->last_seqno_read =
                inframe->sequence_number; // int32_t from seq_t, i.e. uint16_t, so okay.
          else {
            conn->last_seqno_read =
                (conn->last_seqno_read + 1) & 0xffff; // int32_t from seq_t, i.e. uint16_t, so okay.
            if (inframe->sequence_number !=
                conn->last_seqno_read) { // seq_t, ei.e. uint16_t and int32_t, so okay
              debug(2,
                    "Player: packets out of sequence: expected: %u, got: %u, with ab_read: %u "
                    "and ab_write: %u.",
                    conn->last_seqno_read, inframe->sequence_number, conn->ab_read, conn->ab_write);
              conn->last_seqno_read = inframe->sequence_number; // reset warning...
            }
          }

          int16_t bo = conn->ab_write - conn->ab_read; // do this in 16 bits
          conn->buffer_occupancy = bo;                 // 32 bits

          if (conn->buffer_occupancy < minimum_buffer_occupancy)
            minimum_buffer_occupancy = conn->buffer_occupancy;

          if (conn->buffer_occupancy > maximum_buffer_occupancy)
            maximum_buffer_occupancy = conn->buffer_occupancy;

          // now, before outputting anything to the output device, check the stats

          if (play_number % print_interval == 0) {

            // here, calculate the input and output frame rates, where possible, even if statistics
            // have not been requested
            // this is to calculate them in case they are needed by the D-Bus interface or
            // elsewhere.

            if (conn->input_frame_rate_starting_point_is_valid) {
              uint64_t elapsed_reception_time, frames_received;
              elapsed_reception_time =
                  conn->frames_inward_measurement_time - conn->frames_inward_measurement_start_time;
              frames_received = conn->frames_inward_frames_received_at_measurement_time -
                                conn->frames_inward_frames_received_at_measurement_start_time;
              conn->input_frame_rate =
                  (1.0E9 * frames_received) /
                  elapsed_reception_time; // an IEEE double calculation with two 64-bit integers
            } else {
              conn->input_frame_rate = 0.0;
            }

            int stats_status = 0;
            if ((config.output->delay) && (config.no_sync == 0) && (config.output->stats)) {
              uint64_t frames_sent_for_play;
              uint64_t raw_measurement_time;
              uint64_t corrected_measurement_time;
              uint64_t actual_delay;
              stats_status =
                  config.output->stats(&raw_measurement_time, &corrected_measurement_time,
                                       &actual_delay, &frames_sent_for_play);
              // debug(1,"status: %d, actual_delay: %" PRIu64 ", frames_sent_for_play: %" PRIu64 ",
              // frames_played: %" PRIu64 ".", stats_status, actual_delay, frames_sent_for_play,
              // frames_sent_for_play - actual_delay);
              uint64_t frames_played = frames_sent_for_play - actual_delay;
              // If the status is zero, it means that there were no output problems since the
              // last time the stats call was made. Thus, the frame rate should be valid.
              if ((stats_status == 0) && (previous_frames_played_valid != 0)) {
                uint64_t frames_played_in_this_interval = frames_played - previous_frames_played;
                int64_t raw_interval = raw_measurement_time - previous_raw_measurement_time;
                int64_t corrected_interval =
                    corrected_measurement_time - previous_corrected_measurement_time;
                if (raw_interval != 0) {
                  conn->raw_frame_rate = (1e9 * frames_played_in_this_interval) / raw_interval;
                  conn->corrected_frame_rate =
                      (1e9 * frames_played_in_this_interval) / corrected_interval;
                  conn->frame_rate_valid = 1;
                  // debug(1,"frames_played_in_this_interval: %" PRIu64 ", interval: %" PRId64 ",
                  // rate: %f.",
                  //  frames_played_in_this_interval, interval, conn->frame_rate);
                }
              }

              // uncomment the if statement if your want to get as long a period for
              // calculating the frame rate as possible without an output break or error
              if ((stats_status != 0) || (previous_frames_played_valid == 0)) {
                // if we have just detected an outputting error, or if we have no
                // starting information
                if (stats_status != 0)
                  conn->frame_rate_valid = 0;
                previous_frames_played = frames_played;
                previous_raw_measurement_time = raw_measurement_time;
                previous_corrected_measurement_time = corrected_measurement_time;
                previous_frames_played_valid = 1;
              }
            }

            // we can now calculate running averages for sync error (frames), corrections (ppm),
            // insertions plus deletions (ppm), drift (ppm)
            double moving_average_sync_error = 0.0;
            double moving_average_correction = 0.0;
            double moving_average_insertions_plus_deletions = 0.0;
            if (number_of_statistics == 0) {
              debug(2, "number_of_statistics is zero!");
            } else {
              moving_average_sync_error = (1.0 * tsum_of_sync_errors) / number_of_statistics;
              moving_average_correction = (1.0 * tsum_of_corrections) / number_of_statistics;
              moving_average_insertions_plus_deletions =
                  (1.0 * tsum_of_insertions_and_deletions) / number_of_statistics;
              // double moving_average_drift = (1.0 * tsum_of_drifts) / number_of_statistics;
            }
            // if ((play_number/print_interval)%20==0)
            // figure out which statistics profile to use, depending on the kind of stream

            if (config.statistics_requested) {

              if (at_least_one_frame_seen) {
                do {
                  line_of_stats[0] = '\0';
                  statistics_column = 0;
                  was_a_previous_column = 0;
                  statistics_item("Sync Error ms", "%*.2f", 13,
                                  1000 * moving_average_sync_error / config.output_rate);
                  statistics_item("Net Sync PPM", "%*.1f", 12,
                                  moving_average_correction * 1000000 /
                                      (352 * conn->output_sample_ratio));
                  statistics_item("All Sync PPM", "%*.1f", 12,
                                  moving_average_insertions_plus_deletions * 1000000 /
                                      (352 * conn->output_sample_ratio));
                  statistics_item("    Packets", "%*d", 11, play_number);
                  statistics_item("Missing", "%*" PRIu64 "", 7, conn->missing_packets);
                  statistics_item("  Late", "%*" PRIu64 "", 6, conn->late_packets);
                  statistics_item("Too Late", "%*" PRIu64 "", 8, conn->too_late_packets);
                  statistics_item("Resend Reqs", "%*" PRIu64 "", 11, conn->resend_requests);
                  statistics_item("Min DAC Queue", "%*" PRIu64 "", 13, minimum_dac_queue_size);
                  statistics_item("Min Buffers", "%*" PRIu32 "", 11, minimum_buffer_occupancy);
                  statistics_item("Max Buffers", "%*" PRIu32 "", 11, maximum_buffer_occupancy);
#ifdef CONFIG_AIRPLAY_2
                  if (conn->ap2_audio_buffer_minimum_size > 10 * 1024)
                    statistics_item("Min Buffer Size", "%*" PRIu32 "k", 14,
                                    conn->ap2_audio_buffer_minimum_size / 1024);
                  else
                    statistics_item("Min Buffer Size", "%*" PRIu32 "", 15,
                                    conn->ap2_audio_buffer_minimum_size);
#else
                  statistics_item("N/A", "   "); // dummy -- should never be visible
#endif
                  statistics_item("Nominal FPS", "%*.2f", 11, conn->remote_frame_rate);
                  statistics_item("Received FPS", "%*.2f", 12, conn->input_frame_rate);
                  // only make the next two columns appear if we are getting stats information from
                  // the back end
                  if (config.output->stats) {
                    if (conn->frame_rate_valid) {
                      statistics_item("Output FPS (r)", "%*.2f", 14, conn->raw_frame_rate);
                      statistics_item("Output FPS (c)", "%*.2f", 14, conn->corrected_frame_rate);
                    } else {
                      statistics_item("Output FPS (r)", "           N/A");
                      statistics_item("Output FPS (c)", "           N/A");
                    }
                  } else {
                    statistics_column = statistics_column + 2;
                  }
                  statistics_item("Source Drift PPM", "%*.2f", 16,
                                  (conn->local_to_remote_time_gradient - 1.0) * 1000000);
                  statistics_item("Drift Samples", "%*d", 13,
                                  conn->local_to_remote_time_gradient_sample_count);
                  /*
                  statistics_item("estimated (unused) correction ppm", "%*.2f",
                                  strlen("estimated (unused) correction ppm"),
                                  (conn->frame_rate_valid != 0)
                                      ? ((conn->frame_rate -
                                          conn->remote_frame_rate * conn->output_sample_ratio *
                                              conn->local_to_remote_time_gradient) *
                                         1000000) /
                                            conn->frame_rate
                                      : 0.0);
                  */
                  statistics_row++;
                  inform(line_of_stats);
                } while (statistics_row < 2);
              } else {
                inform("No frames received in the last sampling interval.");
              }
            }
            minimum_dac_queue_size = UINT64_MAX;  // hack reset
            maximum_buffer_occupancy = INT32_MIN; // can't be less than this
            minimum_buffer_occupancy = INT32_MAX; // can't be more than this
#ifdef CONFIG_AIRPLAY_2
            conn->ap2_audio_buffer_minimum_size = -1;
#endif
            at_least_one_frame_seen = 0;
          }

          // here, we want to check (a) if we are meant to do synchronisation,
          // (b) if we have a delay procedure, (c) if we can get the delay.

          // If any of these are false, we don't do any synchronisation stuff

          int resp = -1; // use this as a flag -- if negative, we can't rely on a real known delay
          current_delay = -1; // use this as a failure flag

          if (config.output->delay) {
            long l_delay;
            resp = config.output->delay(&l_delay);
            if (resp == 0) { // no error
              current_delay = l_delay;
              if (l_delay >= 0)
                current_delay = l_delay;
              else {
                debug(2, "Underrun of %ld frames reported, but ignored.", l_delay);
                current_delay =
                    0; // could get a negative value if there was underrun, but ignore it.
              }
              if (current_delay < minimum_dac_queue_size) {
                minimum_dac_queue_size = current_delay; // update for display later
              }
            } else {
              current_delay = 0;
              if ((resp == sps_extra_code_output_stalled) &&
                  (conn->unfixable_error_reported == 0)) {
                conn->unfixable_error_reported = 1;
                if (config.cmd_unfixable) {
                  warn("Connection %d: An unfixable error has been detected -- output device is "
                       "stalled. Executing the "
                       "\"run_this_if_an_unfixable_error_is_detected\" command.",
                       conn->connection_number);
                  command_execute(config.cmd_unfixable, "output_device_stalled", 1);
                } else {
                  warn("Connection %d: An unfixable error has been detected -- output device is "
                       "stalled. \"No "
                       "run_this_if_an_unfixable_error_is_detected\" command provided -- nothing "
                       "done.",
                       conn->connection_number);
                }
              } else {
                if (resp != -EBUSY) // delay errors can be reported if the device is (hopefully
                                    // temporarily) busy
                  debug(1, "Delay error %d when checking running latency.", resp);
              }
            }
          }

          if (resp == 0) {

            uint32_t should_be_frame_32;
            // this is denominated in the frame rate of the incoming stream
            local_time_to_frame(local_time_now, &should_be_frame_32, conn);

            int64_t should_be_frame = should_be_frame_32;
            should_be_frame = should_be_frame * conn->output_sample_ratio;

            // current_delay is denominated in the frame rate of the outgoing stream
            int64_t will_be_frame = inframe->given_timestamp;
            will_be_frame = will_be_frame * conn->output_sample_ratio;
            will_be_frame = (will_be_frame - current_delay) &
                            output_rtptime_mask; // this is to make sure it's unsigned modulo 2^bits
                                                 // for the rtptime

            // Now we have a tricky piece of calculation to perform.
            // We know the rtptimes are unsigned in 32 or more bits -- call it r bits. We have to
            // calculate the difference between them. on the basis that they should be within
            // 2^(r-1) of one another, so that the unsigned subtraction result, modulo 2^r, if
            // interpreted as a signed number, should yield the difference _and_ the ordering.

            sync_error = should_be_frame - will_be_frame; // this is done in int64_t form

            // int64_t t_ping = should_be_frame - conn->anchor_rtptime;
            // if (t_ping < 0)
            //   debug(1, "Frame %" PRIu64 " is %" PRId64 " frames before anchor time %" PRIu64 ".",
            //   should_be_frame, -t_ping, conn->anchor_rtptime);

            // sign-extend the r-bit unsigned int calculation by treating it as an r-bit signed
            // integer
            if ((sync_error & output_rtptime_sign_mask) !=
                0) { // check what would be the sign bit in "r" bit unsigned arithmetic
                     // result is negative
              sync_error = sync_error | output_rtptime_mask_not;
            } else {
              // result is positive
              sync_error = sync_error & output_rtptime_mask;
            }

            if (at_least_one_frame_seen_this_session == 0) {
              at_least_one_frame_seen_this_session = 1;

              // debug(2,"first frame real sync error (positive --> late): %" PRId64 " frames.",
              // sync_error);

              // this is a sneaky attempt to make a final adjustment to the timing of the first
              // packet

              // the very first packet generally has a first_frame_early_bias subtracted from its
              // timing to make it more likely that it will be early than late, making it possible
              // to compensate for it be adding a few frames of silence.

              // debug(2,"first frame real sync error (positive --> late): %" PRId64 " frames.",
              // sync_error);

              // remove the bias when reporting the error to make it the true error
              debug(2,
                    "first frame sync error (positive --> late): %" PRId64
                    " frames, %.3f mS at %d frames per second output.",
                    sync_error + first_frame_early_bias,
                    (1000.0 * (sync_error + first_frame_early_bias)) / config.output_rate,
                    config.output_rate);

              // if the packet is early, add the frames needed to put it in sync.
              if (sync_error < 0) {
                size_t final_adjustment_length_sized = -sync_error;
                char *final_adjustment_silence =
                    malloc(conn->output_bytes_per_frame * final_adjustment_length_sized);
                if (final_adjustment_silence) {

                  conn->previous_random_number = generate_zero_frames(
                      final_adjustment_silence, final_adjustment_length_sized, config.output_format,
                      conn->enable_dither, conn->previous_random_number);
                  int final_adjustment = -sync_error;
                  final_adjustment = final_adjustment - first_frame_early_bias;
                  debug(2,
                        "final sync adjustment: %" PRId64
                        " silent frames added with a bias of %" PRId64 " frames.",
                        -sync_error, first_frame_early_bias);
                  config.output->play(final_adjustment_silence, final_adjustment_length_sized,
                                      play_samples_are_untimed, 0, 0);
                  free(final_adjustment_silence);
                } else {
                  warn("Failed to allocate memory for a final_adjustment_silence buffer of %d "
                       "frames for a "
                       "sync error of %d frames.",
                       final_adjustment_length_sized, sync_error);
                }
                sync_error = 0; // say the error was fixed!
              }
              // since this is the first frame of audio, inform the user if requested...
              if (config.statistics_requested) {
#ifdef CONFIG_AIRPLAY_2
                if (conn->airplay_stream_type == realtime_stream) {
                  if (conn->airplay_type == ap_1)
                    inform("Connection %d: Playback Started -- AirPlay 1 Compatible.",
                           conn->connection_number);
                  else
                    inform("Connection %d: Playback Started -- AirPlay 2 Realtime.",
                           conn->connection_number);
                } else {
                  inform("Connection %d: Playback Started -- AirPlay 2 Buffered.",
                         conn->connection_number);
                }
#else
                inform("Connection %d: Playback Started -- AirPlay 1.", conn->connection_number);
#endif
              }
            }
            // not too sure if abs() is implemented for int64_t, so we'll do it manually
            int64_t abs_sync_error = sync_error;
            if (abs_sync_error < 0)
              abs_sync_error = -abs_sync_error;

            if ((config.no_sync == 0) && (inframe->given_timestamp != 0) &&
                (config.resyncthreshold > 0.0) &&
                (abs_sync_error > config.resyncthreshold * config.output_rate)) {
              sync_error_out_of_bounds++;
            } else {
              sync_error_out_of_bounds = 0;
            }

            if (sync_error_out_of_bounds > 3) {
              // debug(1, "lost sync with source for %d consecutive packets -- flushing and "
              //          "resyncing. Error: %lld.",
              //        sync_error_out_of_bounds, sync_error);
              sync_error_out_of_bounds = 0;

              uint64_t frames_sent_for_play = 0;
              uint64_t actual_delay = 0;

              if ((config.output->delay) && (config.no_sync == 0) && (config.output->stats)) {
                uint64_t raw_measurement_time;
                uint64_t corrected_measurement_time;
                config.output->stats(&raw_measurement_time, &corrected_measurement_time,
                                     &actual_delay, &frames_sent_for_play);
              }

              int64_t filler_length =
                  (int64_t)(config.resyncthreshold * config.output_rate); // number of samples
              if ((sync_error > 0) && (sync_error > filler_length)) {
                debug(1,
                      "Large positive (i.e. late) sync error of %" PRId64
                      " frames (%f seconds), at frame: %" PRIu32 ".",
                      sync_error, (sync_error * 1.0) / config.output_rate,
                      inframe->given_timestamp);
                // debug(1, "%" PRId64 " frames sent to DAC. DAC buffer contains %" PRId64 "
                // frames.",
                //      frames_sent_for_play, actual_delay);
                // the sync error is output frames, but we have to work out how many source frames
                // to drop there may be a multiple (the conn->output_sample_ratio) of output frames
                // per input frame...
                int64_t source_frames_to_drop = sync_error;
                source_frames_to_drop = source_frames_to_drop / conn->output_sample_ratio;

                // add some time to give the pipeline a chance to recover -- a bit hacky
                double extra_time_to_drop = 0.1; // seconds
                int64_t extra_frames_to_drop = (int64_t)(conn->input_rate * extra_time_to_drop);
                source_frames_to_drop += extra_frames_to_drop;

                uint32_t frames_to_drop = source_frames_to_drop;
                uint32_t flush_to_frame = inframe->given_timestamp + frames_to_drop;

                do_flush(flush_to_frame, conn);

              } else if ((sync_error < 0) && ((-sync_error) > filler_length)) {
                debug(1,
                      "Large negative (i.e. early) sync error of %" PRId64
                      " frames (%f seconds), at frame: %" PRIu32 ".",
                      sync_error, (sync_error * 1.0) / config.output_rate,
                      inframe->given_timestamp);
                debug(3, "%" PRId64 " frames sent to DAC. DAC buffer contains %" PRId64 " frames.",
                      frames_sent_for_play, actual_delay);
                int64_t silence_length = -sync_error;
                if (silence_length > (filler_length * 5))
                  silence_length = filler_length * 5;
                size_t silence_length_sized = silence_length;
                char *long_silence = malloc(conn->output_bytes_per_frame * silence_length_sized);
                if (long_silence) {

                  conn->previous_random_number =
                      generate_zero_frames(long_silence, silence_length_sized, config.output_format,
                                           conn->enable_dither, conn->previous_random_number);

                  debug(2, "Play a silence of %d frames.", silence_length_sized);
                  config.output->play(long_silence, silence_length_sized, play_samples_are_untimed,
                                      0, 0);
                  free(long_silence);
                } else {
                  warn("Failed to allocate memory for a long_silence buffer of %d frames for a "
                       "sync error of %d frames.",
                       silence_length_sized, sync_error);
                }
                reset_input_flow_metrics(conn);
              }
            } else {

              /*
              // before we finally commit to this frame, check its sequencing and timing
              // require a certain error before bothering to fix it...
              if (sync_error > config.tolerance * config.output_rate) { // int64_t > int, okay
                amount_to_stuff = -1;
              }
              if (sync_error < -config.tolerance * config.output_rate) {
                amount_to_stuff = 1;
              }
              */

              if (amount_to_stuff == 0) {
                // use a "V" shaped function to decide if stuffing should occur
                int64_t s = r64i();
                s = s >> 31;
                s = s * config.tolerance * config.output_rate;
                s = (s >> 32) + config.tolerance * config.output_rate; // should be a number from 0
                                                                       // to config.tolerance *
                                                                       // config.output_rate;
                if ((sync_error > 0) && (sync_error > s)) {
                  // debug(1,"Extra stuff -1");
                  amount_to_stuff = -1;
                }
                if ((sync_error < 0) && (sync_error < (-s))) {
                  // debug(1,"Extra stuff +1");
                  amount_to_stuff = 1;
                }
              }

              // try to keep the corrections definitely below 1 in 1000 audio frames

              // calculate the time elapsed since the play session started.

              if (amount_to_stuff) {
                if ((local_time_now) && (conn->first_packet_time_to_play) &&
                    (local_time_now >= conn->first_packet_time_to_play)) {

                  int64_t tp =
                      (local_time_now - conn->first_packet_time_to_play) /
                      1000000000; // seconds int64_t from uint64_t which is always positive, so ok

                  if (tp < 5)
                    amount_to_stuff = 0; // wait at least five seconds
                  /*
                  else if (tp < 30) {
                    if ((random() % 1000) >
                        352) // keep it to about 1:1000 for the first thirty seconds
                      amount_to_stuff = 0;
                  }
                  */
                }
              }

              if (config.no_sync != 0)
                amount_to_stuff = 0; // no stuffing if it's been disabled

              // Apply DSP here

              // check the state of loudness and convolution flags here and don't change them for
              // the frame

              int do_loudness = config.loudness;

#ifdef CONFIG_CONVOLUTION
              int do_convolution = 0;
              if ((config.convolution) && (config.convolver_valid))
                do_convolution = 1;

              // we will apply the convolution gain if convolution is enabled, even if there is no
              // valid convolution happening

              int convolution_is_enabled = 0;
              if (config.convolution)
                convolution_is_enabled = 1;
#endif

              if (do_loudness
#ifdef CONFIG_CONVOLUTION
                  || convolution_is_enabled
#endif
              ) {
                int32_t *tbuf32 = (int32_t *)conn->tbuf;
                float fbuf_l[inbuflength];
                float fbuf_r[inbuflength];

                // Deinterleave, and convert to float
                int i;
                for (i = 0; i < inbuflength; ++i) {
                  fbuf_l[i] = tbuf32[2 * i];
                  fbuf_r[i] = tbuf32[2 * i + 1];
                }

#ifdef CONFIG_CONVOLUTION
                // Apply convolution
                if (do_convolution) {
                  convolver_process_l(fbuf_l, inbuflength);
                  convolver_process_r(fbuf_r, inbuflength);
                }
                if (convolution_is_enabled) {
                  float gain = pow(10.0, config.convolution_gain / 20.0);
                  for (i = 0; i < inbuflength; ++i) {
                    fbuf_l[i] *= gain;
                    fbuf_r[i] *= gain;
                  }
                }
#endif

                if (do_loudness) {
                  // Apply volume and loudness
                  // Volume must be applied here because the loudness filter will increase the
                  // signal level and it would saturate the int32_t otherwise
                  float gain = conn->fix_volume / 65536.0f;
                  // float gain_db = 20 * log10(gain);
                  // debug(1, "Applying soft volume dB: %f k: %f", gain_db, gain);

                  for (i = 0; i < inbuflength; ++i) {
                    fbuf_l[i] = loudness_process(&loudness_l, fbuf_l[i] * gain);
                    fbuf_r[i] = loudness_process(&loudness_r, fbuf_r[i] * gain);
                  }
                }

                // Interleave and convert back to int32_t
                for (i = 0; i < inbuflength; ++i) {
                  tbuf32[2 * i] = fbuf_l[i];
                  tbuf32[2 * i + 1] = fbuf_r[i];
                }
              }

#ifdef CONFIG_SOXR
              if ((current_delay < conn->dac_buffer_queue_minimum_length) ||
                  (config.packet_stuffing == ST_basic) ||
                  (config.soxr_delay_index == 0) || // not computed
                  ((config.packet_stuffing == ST_auto) &&
                   (config.soxr_delay_index >
                    config.soxr_delay_threshold)) // if the CPU is deemed too slow
              ) {
#endif
                play_samples =
                    stuff_buffer_basic_32((int32_t *)conn->tbuf, inbuflength, config.output_format,
                                          conn->outbuf, amount_to_stuff, conn->enable_dither, conn);
#ifdef CONFIG_SOXR
              } else { // soxr requested or auto requested with the index less or equal to the
                       // threshold
                play_samples = stuff_buffer_soxr_32((int32_t *)conn->tbuf, (int32_t *)conn->sbuf,
                                                    inbuflength, config.output_format, conn->outbuf,
                                                    amount_to_stuff, conn->enable_dither, conn);
              }
#endif

              /*
              {
                int co;
                int is_silent=1;
                short *p = outbuf;
                for (co=0;co<play_samples;co++) {
                  if (*p!=0)
                    is_silent=0;
                  p++;
                }
                if (is_silent)
                  debug(1,"Silence!");
              }
              */

              if (conn->outbuf == NULL)
                debug(1, "NULL outbuf to play -- skipping it.");
              else {
                if (play_samples == 0)
                  debug(1, "play_samples==0 skipping it (1).");
                else {
                  if (conn->software_mute_enabled) {
                    generate_zero_frames(conn->outbuf, play_samples, config.output_format,
                                         conn->enable_dither, conn->previous_random_number);
                  }
                  uint64_t should_be_time;
                  frame_to_local_time(inframe->given_timestamp, &should_be_time, conn);
                  config.output->play(conn->outbuf, play_samples, play_samples_are_timed,
                                      inframe->given_timestamp, should_be_time);
                }
              }

              // check for loss of sync
              // timestamp of zero means an inserted silent frame in place of a missing frame
              /*
              if ((config.no_sync == 0) && (inframe->timestamp != 0) &&
                  && (config.resyncthreshold > 0.0) &&
                  (abs_sync_error > config.resyncthreshold * config.output_rate)) {
                sync_error_out_of_bounds++;
                // debug(1,"Sync error out of bounds: Error: %lld; previous error: %lld; DAC: %lld;
                // timestamp: %llx, time now
                //
              %llx",sync_error,previous_sync_error,current_delay,inframe->timestamp,local_time_now);
                if (sync_error_out_of_bounds > 3) {
                  debug(1, "Lost sync with source for %d consecutive packets -- flushing and "
                           "resyncing. Error: %lld.",
                        sync_error_out_of_bounds, sync_error);
                  sync_error_out_of_bounds = 0;
                  player_flush(nt, conn);
                }
              } else {
                sync_error_out_of_bounds = 0;
              }
              */
            }
          } else {

            // if this is the first frame, see if it's close to when it's supposed to be
            // release, which will be its time plus latency and any offset_time
            if (at_least_one_frame_seen_this_session == 0) {
              at_least_one_frame_seen_this_session = 1;
            }

            play_samples =
                stuff_buffer_basic_32((int32_t *)conn->tbuf, inbuflength, config.output_format,
                                      conn->outbuf, 0, conn->enable_dither, conn);
            if (conn->outbuf == NULL)
              debug(1, "NULL outbuf to play -- skipping it.");
            else {
              if (conn->software_mute_enabled) {
                generate_zero_frames(conn->outbuf, play_samples, config.output_format,
                                     conn->enable_dither, conn->previous_random_number);
              }
              uint64_t should_be_time;
              frame_to_local_time(inframe->given_timestamp, &should_be_time, conn);
              config.output->play(conn->outbuf, play_samples, play_samples_are_timed,
                                  inframe->given_timestamp, should_be_time);
            }
          }

          // mark the frame as finished
          inframe->given_timestamp = 0;
          inframe->sequence_number = 0;
          inframe->resend_time = 0;
          inframe->initialisation_time = 0;

          // update the watchdog
          if ((config.dont_check_timeout == 0) && (config.timeout != 0)) {
            uint64_t time_now = get_absolute_time_in_ns();
            debug_mutex_lock(&conn->watchdog_mutex, 1000, 0);
            conn->watchdog_bark_time = time_now;
            debug_mutex_unlock(&conn->watchdog_mutex, 0);
          }

          // debug(1,"Sync error %lld frames. Amount to stuff %d." ,sync_error,amount_to_stuff);

          // new stats calculation. We want a running average of sync error, drift, adjustment,
          // number of additions+subtractions

          // this is a misleading hack -- the statistics should include some data on the number of
          // valid samples and the number of times sync wasn't checked due to non availability of a
          // delay figure.
          // for the present, stats are only updated when sync has been checked
          if (config.output->delay != NULL) {
            if (number_of_statistics == trend_interval) {
              // here we remove the oldest statistical data and take it from the summaries as well
              tsum_of_sync_errors -= conn->statistics[oldest_statistic].sync_error;
              tsum_of_drifts -= conn->statistics[oldest_statistic].drift;
              if (conn->statistics[oldest_statistic].correction > 0)
                tsum_of_insertions_and_deletions -= conn->statistics[oldest_statistic].correction;
              else
                tsum_of_insertions_and_deletions += conn->statistics[oldest_statistic].correction;
              tsum_of_corrections -= conn->statistics[oldest_statistic].correction;
              oldest_statistic = (oldest_statistic + 1) % trend_interval;
              number_of_statistics--;
            }

            conn->statistics[newest_statistic].sync_error = sync_error;
            conn->statistics[newest_statistic].correction = conn->amountStuffed;

            if (number_of_statistics == 0)
              conn->statistics[newest_statistic].drift = 0;
            else
              conn->statistics[newest_statistic].drift =
                  sync_error - previous_sync_error - previous_correction;

            previous_sync_error = sync_error;
            previous_correction = conn->amountStuffed;

            tsum_of_sync_errors += sync_error;
            tsum_of_drifts += conn->statistics[newest_statistic].drift;
            if (conn->amountStuffed > 0) {
              tsum_of_insertions_and_deletions += conn->amountStuffed;
            } else {
              tsum_of_insertions_and_deletions -= conn->amountStuffed;
            }
            tsum_of_corrections += conn->amountStuffed;
            conn->session_corrections += conn->amountStuffed;

            newest_statistic = (newest_statistic + 1) % trend_interval;
            number_of_statistics++;
          }
        }
      }
    }
  }

  debug(1, "This should never be called.");
  pthread_cleanup_pop(1); // pop the cleanup handler
                          //  debug(1, "This should never be called either.");
                          //  pthread_cleanup_pop(1); // pop the initial cleanup handler
  pthread_exit(NULL);
}

void player_volume_without_notification(double airplay_volume, rtsp_conn_info *conn) {
  debug_mutex_lock(&conn->volume_control_mutex, 5000, 1);
  // first, see if we are hw only, sw only, both with hw attenuation on the top or both with sw
  // attenuation on top

  enum volume_mode_type { vol_sw_only, vol_hw_only, vol_both } volume_mode;

  // take account of whether there is a hardware mixer, if a max volume has been specified and if a
  // range has been specified
  // the range might imply that both hw and software mixers are needed, so calculate this

  int32_t hw_max_db = 0, hw_min_db = 0; // zeroed to quieten an incorrect uninitialised warning
  int32_t sw_max_db = 0, sw_min_db = -9630;

  if (config.output->parameters) {
    volume_mode = vol_hw_only;
    audio_parameters audio_information;
    config.output->parameters(&audio_information);
    hw_max_db = audio_information.maximum_volume_dB;
    hw_min_db = audio_information.minimum_volume_dB;
    if (config.volume_max_db_set) {
      if (((config.volume_max_db * 100) <= hw_max_db) &&
          ((config.volume_max_db * 100) >= hw_min_db))
        hw_max_db = (int32_t)config.volume_max_db * 100;
      else if (config.volume_range_db) {
        hw_max_db = hw_min_db;
        sw_max_db = (config.volume_max_db * 100) - hw_min_db;
      } else {
        warn("The maximum output level is outside the range of the hardware mixer -- ignored");
      }
    }

    // here, we have set limits on the hw_max_db and the sw_max_db
    // but we haven't actually decided whether we need both hw and software attenuation
    // only if a range is specified could we need both
    if (config.volume_range_db) {
      // see if the range requested exceeds the hardware range available
      int32_t desired_range_db = (int32_t)trunc(config.volume_range_db * 100);
      if ((desired_range_db) > (hw_max_db - hw_min_db)) {
        volume_mode = vol_both;
        int32_t desired_sw_range = desired_range_db - (hw_max_db - hw_min_db);
        if ((sw_max_db - desired_sw_range) < sw_min_db)
          warn("The range requested is too large to accommodate -- ignored.");
        else
          sw_min_db = (sw_max_db - desired_sw_range);
      } else {
        hw_min_db = hw_max_db - desired_range_db;
      }
    }
  } else {
    // debug(1,"has no hardware mixer");
    volume_mode = vol_sw_only;
    if (config.volume_max_db_set) {
      if (((config.volume_max_db * 100) <= sw_max_db) &&
          ((config.volume_max_db * 100) >= sw_min_db))
        sw_max_db = (int32_t)config.volume_max_db * 100;
    }
    if (config.volume_range_db) {
      // see if the range requested exceeds the software range available
      int32_t desired_range_db = (int32_t)trunc(config.volume_range_db * 100);
      if ((desired_range_db) > (sw_max_db - sw_min_db))
        warn("The range requested is too large to accommodate -- ignored.");
      else
        sw_min_db = (sw_max_db - desired_range_db);
    }
  }

  // here, we know whether it's hw volume control only, sw only or both, and we have the hw and sw
  // limits.
  // if it's both, we haven't decided whether hw or sw should be on top
  // we have to consider the settings ignore_volume_control and mute.

  if (airplay_volume == -144.0) {

    if ((config.output->mute) && (config.output->mute(1) == 0))
      debug(2,
            "player_volume_without_notification: volume mode is %d, airplay_volume is %f, "
            "hardware mute is enabled.",
            volume_mode, airplay_volume);
    else {
      conn->software_mute_enabled = 1;
      debug(2,
            "player_volume_without_notification: volume mode is %d, airplay_volume is %f, "
            "software mute is enabled.",
            volume_mode, airplay_volume);
    }

  } else {
    int32_t max_db = 0, min_db = 0;
    switch (volume_mode) {
    case vol_hw_only:
      max_db = hw_max_db;
      min_db = hw_min_db;
      break;
    case vol_sw_only:
      max_db = sw_max_db;
      min_db = sw_min_db;
      break;
    case vol_both:
      // debug(1, "dB range passed is hw: %d, sw: %d, total: %d", hw_max_db - hw_min_db,
      //      sw_max_db - sw_min_db, (hw_max_db - hw_min_db) + (sw_max_db - sw_min_db));
      max_db =
          (hw_max_db - hw_min_db) + (sw_max_db - sw_min_db); // this should be the range requested
      min_db = 0;
      break;
    default:
      debug(1, "player_volume_without_notification: error: not in a volume mode");
      break;
    }
    double scaled_attenuation = max_db;
    if (config.ignore_volume_control == 0) {

      if (config.volume_control_profile == VCP_standard)
        scaled_attenuation = vol2attn(airplay_volume, max_db, min_db); // no cancellation points
      else if (config.volume_control_profile == VCP_flat)
        scaled_attenuation =
            flat_vol2attn(airplay_volume, max_db, min_db); // no cancellation points
      else
        debug(1, "player_volume_without_notification: unrecognised volume control profile");
    }
    // so here we have the scaled attenuation. If it's for hw or sw only, it's straightforward.
    double hardware_attenuation = 0.0;
    double software_attenuation = 0.0;

    switch (volume_mode) {
    case vol_hw_only:
      hardware_attenuation = scaled_attenuation;
      break;
    case vol_sw_only:
      software_attenuation = scaled_attenuation;
      break;
    case vol_both:
      // here, we now the attenuation required, so we have to apportion it to the sw and hw mixers
      // if we give the hw priority, that means when lowering the volume, set the hw volume to its
      // lowest
      // before using the sw attenuation.
      // similarly, if we give the sw priority, that means when lowering the volume, set the sw
      // volume to its lowest
      // before using the hw attenuation.
      // one imagines that hw priority is likely to be much better
      // if (config.volume_range_hw_priority) {
      if (config.volume_range_hw_priority != 0) {
        // hw priority
        if ((sw_max_db - sw_min_db) > scaled_attenuation) {
          software_attenuation = sw_min_db + scaled_attenuation;
          hardware_attenuation = hw_min_db;
        } else {
          software_attenuation = sw_max_db;
          hardware_attenuation = hw_min_db + scaled_attenuation - (sw_max_db - sw_min_db);
        }
      } else {
        // sw priority
        if ((hw_max_db - hw_min_db) > scaled_attenuation) {
          hardware_attenuation = hw_min_db + scaled_attenuation;
          software_attenuation = sw_min_db;
        } else {
          hardware_attenuation = hw_max_db;
          software_attenuation = sw_min_db + scaled_attenuation - (hw_max_db - hw_min_db);
        }
      }
      break;
    default:
      debug(1, "player_volume_without_notification: error: not in a volume mode");
      break;
    }

    if (((volume_mode == vol_hw_only) || (volume_mode == vol_both)) && (config.output->volume)) {
      config.output->volume(hardware_attenuation); // otherwise set the output to the lowest value
      // debug(1,"Hardware attenuation set to %f for airplay volume of
      // %f.",hardware_attenuation,airplay_volume);
      if (volume_mode == vol_hw_only)
        conn->fix_volume = 0x10000;
    }

    if ((volume_mode == vol_sw_only) || (volume_mode == vol_both)) {
      double temp_fix_volume = 65536.0 * pow(10, software_attenuation / 2000);

      if (config.ignore_volume_control == 0)
        debug(2, "Software attenuation set to %f, i.e %f out of 65,536, for airplay volume of %f",
              software_attenuation, temp_fix_volume, airplay_volume);
      else
        debug(2, "Software attenuation set to %f, i.e %f out of 65,536. Volume control is ignored.",
              software_attenuation, temp_fix_volume);

      conn->fix_volume = temp_fix_volume;

      // if (config.loudness)
      loudness_set_volume(software_attenuation / 100);
    }

    if (config.logOutputLevel) {
      inform("Output Level set to: %.2f dB.", scaled_attenuation / 100.0);
    }

#ifdef CONFIG_METADATA
    // here, send the 'pvol' metadata message when the airplay volume information
    // is being used by shairport sync to control the output volume
    char dv[128];
    memset(dv, 0, 128);
    if (config.ignore_volume_control == 0) {
      if (volume_mode == vol_both) {
        // normalise the maximum output to the hardware device's max output
        snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume,
                 (scaled_attenuation - max_db + hw_max_db) / 100.0,
                 (min_db - max_db + hw_max_db) / 100.0, (max_db - max_db + hw_max_db) / 100.0);
      } else {
        snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, scaled_attenuation / 100.0,
                 min_db / 100.0, max_db / 100.0);
      }
    } else {
      snprintf(dv, 127, "%.2f,%.2f,%.2f,%.2f", airplay_volume, 0.0, 0.0, 0.0);
    }
    send_ssnc_metadata('pvol', dv, strlen(dv), 1);
#endif

    if (config.output->mute)
      config.output->mute(0);
    conn->software_mute_enabled = 0;

    debug(2,
          "player_volume_without_notification: volume mode is %d, airplay volume is %.2f, "
          "software_attenuation dB: %.2f, hardware_attenuation dB: %.2f, muting "
          "is disabled.",
          volume_mode, airplay_volume, software_attenuation / 100.0, hardware_attenuation / 100.0);
  }
  // here, store the volume for possible use in the future
  config.airplay_volume = airplay_volume;
  debug_mutex_unlock(&conn->volume_control_mutex, 3);
}

void player_volume(double airplay_volume, rtsp_conn_info *conn) {
  command_set_volume(airplay_volume);
  player_volume_without_notification(airplay_volume, conn);
}

void do_flush(uint32_t timestamp, rtsp_conn_info *conn) {

  debug(3, "do_flush: flush to %u.", timestamp);
  debug_mutex_lock(&conn->flush_mutex, 1000, 1);
  conn->flush_requested = 1;
  conn->flush_rtp_timestamp = timestamp; // flush all packets up to, but not including, this one.
  reset_input_flow_metrics(conn);
  debug_mutex_unlock(&conn->flush_mutex, 3);
}

void player_flush(uint32_t timestamp, rtsp_conn_info *conn) {
  debug(3, "player_flush");
  do_flush(timestamp, conn);
#ifdef CONFIG_METADATA
  // only send a flush metadata message if the first packet has been seen -- it's a bogus message
  // otherwise
  if (conn->first_packet_timestamp) {
    debug(2, "pfls");
    char numbuf[32];
    snprintf(numbuf, sizeof(numbuf), "%u", timestamp);
    send_ssnc_metadata('pfls', numbuf, strlen(numbuf), 1); // contains cancellation points
  }
#endif
}

/*
void player_full_flush(rtsp_conn_info *conn) {
  debug(3, "player_full_flush");
  // this basically flushes everything from the player
  // here, find the rtptime of the last from in the buffer and add 1 to it
  // so as to ask to flush everything
  int flush_needed = 0;
  uint32_t rtpTimestamp;
  debug_mutex_lock(&conn->ab_mutex, 30000, 0);
  if ((conn->ab_synced != 0) && (conn->ab_write != conn->ab_read)) {
    abuf_t *abuf = NULL;
    seq_t last_seqno_written;
    do {
      last_seqno_written = conn->ab_write - 1;
      abuf = conn->audio_buffer + BUFIDX(last_seqno_written);
    } while ((abuf->ready == 0) && (last_seqno_written != conn->ab_read));
    if ((abuf != NULL) && (abuf->ready != 0)) {
      rtpTimestamp = abuf->given_timestamp + abuf->length + 1;
      debug(2, "full flush needed to %u", rtpTimestamp);
      flush_needed = 1;
    } else {
      debug(2, "full flush not needed");
    }
  } else {
    debug(2, "full flush not needed -- buffers empty or not synced");
  }
  debug_mutex_unlock(&conn->ab_mutex, 0);
  if (flush_needed)
    player_flush(rtpTimestamp, conn);
}
*/

// perpare_to_play and play are split so that we can get the capabilities of the
// dac etc. before initialising any decoders etc.
// for example, if we have 32-bit DACs, we can ask for 32 bit decodes

int player_prepare_to_play(rtsp_conn_info *conn) {
  // need to use conn in place of stream below. Need to put the stream as a parameter to he
  if (conn->player_thread != NULL)
    die("Trying to create a second player thread for this RTSP session");
  if (config.buffer_start_fill > BUFFER_FRAMES)
    die("specified buffer starting fill %d > buffer size %d", config.buffer_start_fill,
        BUFFER_FRAMES);
  // active, and should be before play's command hook, command_start()
  command_start();
  conn->input_bytes_per_frame = 4; // default -- may be changed later
  // call on the output device to prepare itself
  if ((config.output) && (config.output->prepare))
    config.output->prepare();
  return 0;
}

int player_play(rtsp_conn_info *conn) {
  pthread_t *pt = malloc(sizeof(pthread_t));
  if (pt == NULL)
    die("Couldn't allocate space for pthread_t");
  conn->player_thread = pt;
  int rc = pthread_create(pt, NULL, player_thread_func, (void *)conn);
  if (rc)
    debug(1, "Error creating player_thread: %s", strerror(errno));

#ifdef CONFIG_METADATA
  debug(2, "pbeg");
  send_ssnc_metadata('pbeg', NULL, 0, 1); // contains cancellation points
#endif
  return 0;
}

int player_stop(rtsp_conn_info *conn) {
  // note -- this may be called from another connection thread.
  // int dl = debuglev;
  // debuglev = 3;
  debug(3, "player_stop");
  if (conn->player_thread) {
    debug(3, "player_thread cancel...");
    pthread_cancel(*conn->player_thread);
    debug(3, "player_thread join...");
    if (pthread_join(*conn->player_thread, NULL) == -1) {
      char errorstring[1024];
      strerror_r(errno, (char *)errorstring, sizeof(errorstring));
      debug(1, "Connection %d: error %d joining player thread: \"%s\".", conn->connection_number,
            errno, (char *)errorstring);
    } else {
      debug(3, "player_thread joined.");
    }
    free(conn->player_thread);
    conn->player_thread = NULL;
#ifdef CONFIG_METADATA
    debug(2, "pend");
    send_ssnc_metadata('pend', NULL, 0, 1); // contains cancellation points
#endif
    // debuglev = dl;
    command_stop();
    return 0;
  } else {
    debug(3, "Connection %d: player thread already deleted.", conn->connection_number);
    // debuglev = dl;
    return -1;
  }
}