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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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305 lines
8.1 KiB
C++
305 lines
8.1 KiB
C++
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/*
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* lpcmstrm_in.c: LPCM Audio strem class members handling scanning and
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* buffering raw input stream.
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*
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* Copyright (C) 2001 Andrew Stevens <andrew.stevens@philips.com>
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* Copyright (C) 2000,2001 Brent Byeler for original header-structure
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* parsing code.
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*
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of version 2 of the GNU General Public License
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* as published by the Free Software Foundation.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*/
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#include <config.h>
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#include <math.h>
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#include <stdlib.h>
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#include <string.h>
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#include "audiostrm.hh"
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#include "outputstream.hh"
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#include <cassert>
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LPCMStream::LPCMStream (IBitStream & ibs, OutputStream & into):
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AudioStream (ibs, into)
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{
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}
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bool LPCMStream::Probe (IBitStream & bs)
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{
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return true;
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}
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/*************************************************************************
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*
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* Reads initial stream parameters and displays feedback banner to users
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*
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*************************************************************************/
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void
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LPCMStream::Init (const int stream_num)
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{
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MuxStream::Init (PRIVATE_STR_1, 1, // Buffer scale
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default_buffer_size,
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muxinto.vcd_zero_stuffing,
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muxinto.buffers_in_audio, muxinto.always_buffers_in_audio);
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mjpeg_info ("Scanning for header info: LPCM Audio stream %02x", stream_num);
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InitAUbuffer ();
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AU_start = bs.bitcount ();
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// This is a dummy debug version that simply assumes 48kHz
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// two channel 16 bit sample LPCM
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samples_per_second = 48000;
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channels = 2;
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bits_per_sample = 16;
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bytes_per_frame =
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samples_per_second * channels * bits_per_sample / 8 * ticks_per_frame_90kHz / 90000;
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frame_index = 0;
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dynamic_range_code = 0x80;
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/* Presentation/decoding time-stamping */
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access_unit.start = AU_start;
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access_unit.length = bytes_per_frame;
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access_unit.PTS = static_cast < clockticks > (decoding_order) *
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(CLOCKS_per_90Kth_sec * ticks_per_frame_90kHz);
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access_unit.DTS = access_unit.PTS;
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access_unit.dorder = decoding_order;
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decoding_order++;
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aunits.append (access_unit);
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OutputHdrInfo ();
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}
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unsigned int
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LPCMStream::NominalBitRate ()
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{
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return samples_per_second * channels * bits_per_sample;
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}
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void
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LPCMStream::FillAUbuffer (unsigned int frames_to_buffer)
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{
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last_buffered_AU += frames_to_buffer;
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mjpeg_debug ("Scanning %d MPEG LPCM audio frames to frame %d",
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frames_to_buffer, last_buffered_AU);
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static int header_skip = 0; // Initially skipped past 5 bytes of header
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int skip;
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bool bad_last_frame = false;
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while (!bs.eos () &&
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decoding_order < last_buffered_AU) {
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skip = access_unit.length - header_skip;
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mjpeg_debug ("Buffering frame %d (%d bytes)\n", decoding_order - 1, skip);
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if (skip & 0x1)
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bs.getbits (8);
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if (skip & 0x2)
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bs.getbits (16);
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skip = skip >> 2;
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for (int i = 0; i < skip; i++) {
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bs.getbits (32);
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}
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prev_offset = AU_start;
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AU_start = bs.bitcount ();
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if (AU_start - prev_offset != access_unit.length * 8) {
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bad_last_frame = true;
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break;
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}
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// Here we would check for header data but LPCM has no headers...
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if (bs.eos ())
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break;
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access_unit.start = AU_start;
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access_unit.length = bytes_per_frame;
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access_unit.PTS = static_cast < clockticks > (decoding_order) *
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(CLOCKS_per_90Kth_sec * ticks_per_frame_90kHz);
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access_unit.DTS = access_unit.PTS;
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access_unit.dorder = decoding_order;
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decoding_order++;
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aunits.append (access_unit);
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num_frames[0]++;
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num_syncword++;
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if (num_syncword >= old_frames + 10) {
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mjpeg_debug ("Got %d frame headers.", num_syncword);
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old_frames = num_syncword;
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}
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mjpeg_debug ("Got frame %d\n", decoding_order);
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}
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if (bad_last_frame) {
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mjpeg_error_exit1 ("Last LPCM frame ended prematurely!\n");
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}
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last_buffered_AU = decoding_order;
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eoscan = bs.eos ();
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}
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void
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LPCMStream::Close ()
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{
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stream_length = AU_start / 8;
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mjpeg_info ("AUDIO_STATISTICS: %02x", stream_id);
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mjpeg_info ("Audio stream length %lld bytes.", stream_length);
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mjpeg_info ("Frames : %8u ", num_frames[0]);
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bs.close ();
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}
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/*************************************************************************
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OutputAudioInfo
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gibt gesammelte Informationen zu den Audio Access Units aus.
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Prints information on audio access units
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*************************************************************************/
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void
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LPCMStream::OutputHdrInfo ()
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{
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mjpeg_info ("LPCM AUDIO STREAM:");
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mjpeg_info ("Bit rate : %8u bytes/sec (%3u kbit/sec)",
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NominalBitRate () / 8, NominalBitRate ());
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mjpeg_info ("Channels : %d\n", channels);
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mjpeg_info ("Bits per sample: %d\n", bits_per_sample);
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mjpeg_info ("Frequency : %d Hz", samples_per_second);
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}
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unsigned int
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LPCMStream::ReadPacketPayload (uint8_t * dst, unsigned int to_read)
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{
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unsigned int header_size = LPCMStream::StreamHeaderSize ();
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unsigned int bytes_read = bs.read_buffered_bytes (dst + header_size,
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to_read - header_size);
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clockticks decode_time;
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bool starting_frame_found = false;
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uint8_t starting_frame_index = 0;
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int starting_frame_offset = (new_au_next_sec || au_unsent > bytes_read)
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? 0 : au_unsent;
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unsigned int frames = 0;
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unsigned int bytes_muxed = bytes_read;
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if (bytes_muxed == 0 || MuxCompleted ()) {
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goto completion;
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}
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/* Work through what's left of the current frames and the
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following frames's updating the info until we reach a point where
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an frame had to be split between packets.
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The DTS/PTS field for the packet in this case would have been
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given the that for the first AU to start in the packet.
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*/
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decode_time = RequiredDTS ();
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while (au_unsent < bytes_muxed) {
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assert (bytes_muxed > 1);
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bufmodel.Queued (au_unsent, decode_time);
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bytes_muxed -= au_unsent;
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if (new_au_next_sec) {
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++frames;
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if (!starting_frame_found) {
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starting_frame_index = static_cast < uint8_t > (au->dorder % 20);
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starting_frame_found = true;
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}
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}
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if (!NextAU ()) {
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goto completion;
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}
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new_au_next_sec = true;
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decode_time = RequiredDTS ();
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};
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// We've now reached a point where the current AU overran or
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// fitted exactly. We need to distinguish the latter case so we
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// can record whether the next packet starts with the tail end of
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// // an already started frame or a new one. We need this info to
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// decide what PTS/DTS info to write at the start of the next
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// packet.
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if (au_unsent > bytes_muxed) {
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if (new_au_next_sec)
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++frames;
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bufmodel.Queued (bytes_muxed, decode_time);
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au_unsent -= bytes_muxed;
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new_au_next_sec = false;
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} else // if (au_unsent == bytes_muxed)
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{
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bufmodel.Queued (bytes_muxed, decode_time);
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if (new_au_next_sec)
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++frames;
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new_au_next_sec = NextAU ();
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}
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completion:
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// Generate the LPCM header...
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// Note the index counts from the low byte of the offset so
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// the smallest value is 1!
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dst[0] = LPCM_SUB_STR_0 + stream_num;
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dst[1] = frames;
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dst[2] = (starting_frame_offset + 1) >> 8;
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dst[3] = (starting_frame_offset + 1) & 0xff;
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unsigned int bps_code;
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switch (bits_per_sample) {
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case 16:
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bps_code = 0;
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break;
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case 20:
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bps_code = 1;
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break;
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case 24:
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bps_code = 2;
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break;
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default:
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bps_code = 3;
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break;
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}
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dst[4] = starting_frame_index;
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unsigned int bsf_code = (samples_per_second == 48000) ? 0 : 1;
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unsigned int channels_code = channels - 1;
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dst[5] = (bps_code << 6) | (bsf_code << 4) | channels_code;
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dst[6] = dynamic_range_code;
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return bytes_read + header_size;
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}
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/*
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* Local variables:
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* c-file-style: "stroustrup"
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* tab-width: 4
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* indent-tabs-mode: nil
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* End:
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*/
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