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softhddev.c
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///
/// @file softhddev.c @brief A software HD device plugin for VDR.
///
/// Copyright (c) 2011 - 2015 by Johns. All Rights Reserved.
///
/// Contributor(s):
///
/// License: AGPLv3
///
/// This program is free software: you can redistribute it and/or modify
/// it under the terms of the GNU Affero General Public License as
/// published by the Free Software Foundation, either version 3 of the
/// License.
///
/// This program is distributed in the hope that it will be useful,
/// but WITHOUT ANY WARRANTY; without even the implied warranty of
/// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
/// GNU Affero General Public License for more details.
///
/// $Id: 8881600a16f475cba7db8911ad88ce2234f72d14 $
//////////////////////////////////////////////////////////////////////////////
#define noUSE_SOFTLIMIT ///< add soft buffer limits to Play..
#define noUSE_PIP ///< include PIP support + new API
#define noDUMP_TRICKSPEED ///< dump raw trickspeed packets
#include <sys/stat.h>
#include <sys/types.h>
#ifdef __FreeBSD__
#include <signal.h>
#endif
#include <fcntl.h>
#include <inttypes.h>
#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include <libintl.h>
#define _(str) gettext(str) ///< gettext shortcut
#define _N(str) str ///< gettext_noop shortcut
#include <libavcodec/avcodec.h>
#include <libavutil/mem.h>
#ifndef __USE_GNU
#define __USE_GNU
#endif
#include <pthread.h>
#include "iatomic.h" // portable atomic_t
#include "misc.h"
#include "softhddev.h"
// clang-format off
#include "audio.h"
#include "video.h"
#include "codec.h"
// clang-format on
#ifdef DEBUG
static int DumpH264(const uint8_t *data, int size);
static void DumpMpeg(const uint8_t *data, int size);
#endif
//////////////////////////////////////////////////////////////////////////////
// Variables
//////////////////////////////////////////////////////////////////////////////
extern int ConfigAudioBufferTime; ///< config size ms of audio buffer
extern int ConfigVideoClearOnSwitch; //< clear decoder on channel switch
char ConfigStartX11Server; ///< flag start the x11 server
static signed char ConfigStartSuspended; ///< flag to start in suspend mode
static char ConfigFullscreen; ///< fullscreen modus
static const char *X11ServerArguments; ///< default command arguments
static char ConfigStillDecoder; ///< hw/sw decoder for still picture
static pthread_mutex_t SuspendLockMutex; ///< suspend lock mutex
static volatile char StreamFreezed; ///< stream freezed
//////////////////////////////////////////////////////////////////////////////
// Audio
//////////////////////////////////////////////////////////////////////////////
static volatile char NewAudioStream; ///< new audio stream
static volatile char SkipAudio; ///< skip audio stream
static AudioDecoder *MyAudioDecoder; ///< audio decoder
static enum AVCodecID AudioCodecID; ///< current codec id
static int AudioChannelID; ///< current audio channel id
static VideoStream *AudioSyncStream; ///< video stream for audio/video sync
/// Minimum free space in audio buffer 8 packets for 8 channels
#define AUDIO_MIN_BUFFER_FREE (3072 * 8 * 8)
#define AUDIO_BUFFER_SIZE (512 * 1024) ///< audio PES buffer default size
static AVPacket AudioAvPkt[1]; ///< audio a/v packet
int AudioDelay = 0;
//////////////////////////////////////////////////////////////////////////////
// Audio codec parser
//////////////////////////////////////////////////////////////////////////////
///
/// Mpeg bitrate table.
///
/// BitRateTable[Version][Layer][Index]
///
static const uint16_t BitRateTable[2][4][16] = {
// MPEG Version 1
{{},
{0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448, 0},
{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, 0},
{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 0}},
// MPEG Version 2 & 2.5
{{},
{0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256, 0},
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 0},
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 0}}};
///
/// Mpeg samperate table.
///
static const uint16_t SampleRateTable[4] = {44100, 48000, 32000, 0};
///
/// Fast check for Mpeg audio.
///
/// 4 bytes 0xFFExxxxx Mpeg audio
///
static inline int FastMpegCheck(const uint8_t *p) {
if (p[0] != 0xFF) { // 11bit frame sync
return 0;
}
if ((p[1] & 0xE0) != 0xE0) {
return 0;
}
if ((p[1] & 0x18) == 0x08) { // version ID - 01 reserved
return 0;
}
if (!(p[1] & 0x06)) { // layer description - 00 reserved
return 0;
}
if ((p[2] & 0xF0) == 0xF0) { // bitrate index - 1111 reserved
return 0;
}
if ((p[2] & 0x0C) == 0x0C) { // sampling rate index - 11 reserved
return 0;
}
return 1;
}
///
/// Check for Mpeg audio.
///
/// 0xFFEx already checked.
///
/// @param data incomplete PES packet
/// @param size number of bytes
///
/// @retval <0 possible mpeg audio, but need more data
/// @retval 0 no valid mpeg audio
/// @retval >0 valid mpeg audio
///
/// From: http://www.mpgedit.org/mpgedit/mpeg_format/mpeghdr.htm
///
/// AAAAAAAA AAABBCCD EEEEFFGH IIJJKLMM
///
/// o a 11x Frame sync
/// o b 2x Mpeg audio version (2.5, reserved, 2, 1)
/// o c 2x Layer (reserved, III, II, I)
/// o e 2x BitRate index
/// o f 2x SampleRate index (4100, 48000, 32000, 0)
/// o g 1x Paddding bit
/// o .. Doesn't care
///
/// frame length:
/// Layer I:
/// FrameLengthInBytes = (12 * BitRate / SampleRate + Padding) * 4
/// Layer II & III:
/// FrameLengthInBytes = 144 * BitRate / SampleRate + Padding
///
static int MpegCheck(const uint8_t *data, int size) {
int mpeg2;
int mpeg25;
int layer;
int bit_rate_index;
int sample_rate_index;
int padding;
int bit_rate;
int sample_rate;
int frame_size;
mpeg2 = !(data[1] & 0x08) && (data[1] & 0x10);
mpeg25 = !(data[1] & 0x08) && !(data[1] & 0x10);
layer = 4 - ((data[1] >> 1) & 0x03);
bit_rate_index = (data[2] >> 4) & 0x0F;
sample_rate_index = (data[2] >> 2) & 0x03;
padding = (data[2] >> 1) & 0x01;
sample_rate = SampleRateTable[sample_rate_index];
if (!sample_rate) { // no valid sample rate try next
// moved into fast check
abort();
return 0;
}
sample_rate >>= mpeg2; // mpeg 2 half rate
sample_rate >>= mpeg25; // mpeg 2.5 quarter rate
bit_rate = BitRateTable[mpeg2 | mpeg25][layer][bit_rate_index];
if (!bit_rate) { // no valid bit-rate try next
// FIXME: move into fast check?
return 0;
}
bit_rate *= 1000;
switch (layer) {
case 1:
frame_size = (12 * bit_rate) / sample_rate;
frame_size = (frame_size + padding) * 4;
break;
case 2:
case 3:
default:
frame_size = (144 * bit_rate) / sample_rate;
frame_size = frame_size + padding;
break;
}
if (0) {
Debug(3, "pesdemux: mpeg%s layer%d bitrate=%d samplerate=%d %d bytes\n",
mpeg25 ? "2.5"
: mpeg2 ? "2"
: "1",
layer, bit_rate, sample_rate, frame_size);
}
if (frame_size + 4 > size) {
return -frame_size - 4;
}
// check if after this frame a new mpeg frame starts
if (FastMpegCheck(data + frame_size)) {
return frame_size;
}
return 0;
}
///
/// Fast check for AAC LATM audio.
///
/// 3 bytes 0x56Exxx AAC LATM audio
///
static inline int FastLatmCheck(const uint8_t *p) {
if (p[0] != 0x56) { // 11bit sync
return 0;
}
if ((p[1] & 0xE0) != 0xE0) {
return 0;
}
return 1;
}
///
/// Check for AAC LATM audio.
///
/// 0x56Exxx already checked.
///
/// @param data incomplete PES packet
/// @param size number of bytes
///
/// @retval <0 possible AAC LATM audio, but need more data
/// @retval 0 no valid AAC LATM audio
/// @retval >0 valid AAC LATM audio
///
static int LatmCheck(const uint8_t *data, int size) {
int frame_size;
// 13 bit frame size without header
frame_size = ((data[1] & 0x1F) << 8) + data[2];
frame_size += 3;
if (frame_size + 2 > size) {
return -frame_size - 2;
}
// check if after this frame a new AAC LATM frame starts
if (FastLatmCheck(data + frame_size)) {
return frame_size;
}
return 0;
}
///
/// Possible AC-3 frame sizes.
///
/// from ATSC A/52 table 5.18 frame size code table.
///
const uint16_t Ac3FrameSizeTable[38][3] = {
{64, 69, 96}, {64, 70, 96}, {80, 87, 120}, {80, 88, 120}, {96, 104, 144},
{96, 105, 144}, {112, 121, 168}, {112, 122, 168}, {128, 139, 192}, {128, 140, 192},
{160, 174, 240}, {160, 175, 240}, {192, 208, 288}, {192, 209, 288}, {224, 243, 336},
{224, 244, 336}, {256, 278, 384}, {256, 279, 384}, {320, 348, 480}, {320, 349, 480},
{384, 417, 576}, {384, 418, 576}, {448, 487, 672}, {448, 488, 672}, {512, 557, 768},
{512, 558, 768}, {640, 696, 960}, {640, 697, 960}, {768, 835, 1152}, {768, 836, 1152},
{896, 975, 1344}, {896, 976, 1344}, {1024, 1114, 1536}, {1024, 1115, 1536}, {1152, 1253, 1728},
{1152, 1254, 1728}, {1280, 1393, 1920}, {1280, 1394, 1920},
};
///
/// Fast check for (E-)AC-3 audio.
///
/// 5 bytes 0x0B77xxxxxx AC-3 audio
///
static inline int FastAc3Check(const uint8_t *p) {
if (p[0] != 0x0B) { // 16bit sync
return 0;
}
if (p[1] != 0x77) {
return 0;
}
return 1;
}
///
/// Check for (E-)AC-3 audio.
///
/// 0x0B77xxxxxx already checked.
///
/// @param data incomplete PES packet
/// @param size number of bytes
///
/// @retval <0 possible AC-3 audio, but need more data
/// @retval 0 no valid AC-3 audio
/// @retval >0 valid AC-3 audio
///
/// o AC-3 Header
/// AAAAAAAA AAAAAAAA BBBBBBBB BBBBBBBB CCDDDDDD EEEEEFFF
///
/// o a 16x Frame sync, always 0x0B77
/// o b 16x CRC 16
/// o c 2x Samplerate
/// o d 6x Framesize code
/// o e 5x Bitstream ID
/// o f 3x Bitstream mode
///
/// o E-AC-3 Header
/// AAAAAAAA AAAAAAAA BBCCCDDD DDDDDDDD EEFFGGGH IIIII...
///
/// o a 16x Frame sync, always 0x0B77
/// o b 2x Frame type
/// o c 3x Sub stream ID
/// o d 10x Framesize - 1 in words
/// o e 2x Framesize code
/// o f 2x Framesize code 2
///
static int Ac3Check(const uint8_t *data, int size) {
int frame_size;
if (size < 5) { // need 5 bytes to see if AC-3/E-AC-3
return -5;
}
if (data[5] > (10 << 3)) { // E-AC-3
if ((data[4] & 0xF0) == 0xF0) { // invalid fscod fscod2
return 0;
}
frame_size = ((data[2] & 0x07) << 8) + data[3] + 1;
frame_size *= 2;
} else { // AC-3
int fscod;
int frmsizcod;
// crc1 crc1 fscod|frmsizcod
fscod = data[4] >> 6;
if (fscod == 0x03) { // invalid sample rate
return 0;
}
frmsizcod = data[4] & 0x3F;
if (frmsizcod > 37) { // invalid frame size
return 0;
}
// invalid is checked above
frame_size = Ac3FrameSizeTable[frmsizcod][fscod] * 2;
}
if (frame_size + 5 > size) {
return -frame_size - 5;
}
// FIXME: relaxed checks if codec is already detected
// check if after this frame a new AC-3 frame starts
if (FastAc3Check(data + frame_size)) {
return frame_size;
}
return 0;
}
///
/// Fast check for ADTS Audio Data Transport Stream.
///
/// 7/9 bytes 0xFFFxxxxxxxxxxx(xxxx) ADTS audio
///
static inline int FastAdtsCheck(const uint8_t *p) {
if (p[0] != 0xFF) { // 12bit sync
return 0;
}
if ((p[1] & 0xF6) != 0xF0) { // sync + layer must be 0
return 0;
}
if ((p[2] & 0x3C) == 0x3C) { // sampling frequency index != 15
return 0;
}
return 1;
}
///
/// Check for ADTS Audio Data Transport Stream.
///
/// 0xFFF already checked.
///
/// @param data incomplete PES packet
/// @param size number of bytes
///
/// @retval <0 possible ADTS audio, but need more data
/// @retval 0 no valid ADTS audio
/// @retval >0 valid AC-3 audio
///
/// AAAAAAAA AAAABCCD EEFFFFGH HHIJKLMM MMMMMMMM MMMOOOOO OOOOOOPP
/// (QQQQQQQQ QQQQQQQ)
///
/// o A*12 syncword 0xFFF
/// o B*1 MPEG Version: 0 for MPEG-4, 1 for MPEG-2
/// o C*2 layer: always 0
/// o ..
/// o F*4 sampling frequency index (15 is invalid)
/// o ..
/// o M*13 frame length
///
static int AdtsCheck(const uint8_t *data, int size) {
int frame_size;
if (size < 6) {
return -6;
}
frame_size = (data[3] & 0x03) << 11;
frame_size |= (data[4] & 0xFF) << 3;
frame_size |= (data[5] & 0xE0) >> 5;
if (frame_size + 3 > size) {
return -frame_size - 3;
}
// check if after this frame a new ADTS frame starts
if (FastAdtsCheck(data + frame_size)) {
return frame_size;
}
return 0;
}
//////////////////////////////////////////////////////////////////////////////
// PES Demux
//////////////////////////////////////////////////////////////////////////////
///
/// PES type.
///
enum {
PES_PROG_STREAM_MAP = 0xBC,
PES_PRIVATE_STREAM1 = 0xBD,
PES_PADDING_STREAM = 0xBE, ///< filler, padding stream
PES_PRIVATE_STREAM2 = 0xBF,
PES_AUDIO_STREAM_S = 0xC0,
PES_AUDIO_STREAM_E = 0xDF,
PES_VIDEO_STREAM_S = 0xE0,
PES_VIDEO_STREAM_E = 0xEF,
PES_ECM_STREAM = 0xF0,
PES_EMM_STREAM = 0xF1,
PES_DSM_CC_STREAM = 0xF2,
PES_ISO13522_STREAM = 0xF3,
PES_TYPE_E_STREAM = 0xF8, ///< ITU-T rec. h.222.1 type E stream
PES_PROG_STREAM_DIR = 0xFF,
};
#ifndef NO_TS_AUDIO
///
/// PES parser state.
///
enum {
PES_INIT, ///< unknown codec
PES_SKIP, ///< skip packet
PES_SYNC, ///< search packet sync byte
PES_HEADER, ///< copy header
PES_START, ///< pes packet start found
PES_PAYLOAD, ///< copy payload
PES_LPCM_HEADER, ///< copy lcpm header
PES_LPCM_PAYLOAD, ///< copy lcpm payload
};
#define PES_START_CODE_SIZE 6 ///< size of pes start code with length
#define PES_HEADER_SIZE 9 ///< size of pes header
#define PES_MAX_HEADER_SIZE (PES_HEADER_SIZE + 256) ///< maximal header size
#define PES_MAX_PAYLOAD (512 * 1024) ///< max pay load size
///
/// PES demuxer.
///
typedef struct _pes_demux_ {
// int Pid; ///< packet id
// int PcrPid; ///< program clock reference pid
// int StreamType; ///< stream type
int State; ///< parsing state
uint8_t Header[PES_MAX_HEADER_SIZE]; ///< buffer for pes header
int HeaderIndex; ///< header index
int HeaderSize; ///< size of pes header
uint8_t *Buffer; ///< payload buffer
int Index; ///< buffer index
int Skip; ///< buffer skip
int Size; ///< size of payload buffer
uint8_t StartCode; ///< pes packet start code
int64_t PTS; ///< presentation time stamp
int64_t DTS; ///< decode time stamp
} PesDemux;
///
/// Reset packetized elementary stream demuxer.
///
static void PesReset(PesDemux *pesdx) {
pesdx->State = PES_INIT;
pesdx->Index = 0;
pesdx->Skip = 0;
pesdx->StartCode = -1;
pesdx->PTS = AV_NOPTS_VALUE;
pesdx->DTS = AV_NOPTS_VALUE;
}
///
/// Initialize a packetized elementary stream demuxer.
///
/// @param pesdx packetized elementary stream demuxer
///
static void PesInit(PesDemux *pesdx) {
memset(pesdx, 0, sizeof(*pesdx));
pesdx->Size = PES_MAX_PAYLOAD;
pesdx->Buffer = av_malloc(PES_MAX_PAYLOAD + AV_INPUT_BUFFER_PADDING_SIZE);
if (!pesdx->Buffer) {
Fatal(_("pesdemux: out of memory\n"));
}
PesReset(pesdx);
}
///
/// Parse packetized elementary stream.
///
/// @param pesdx packetized elementary stream demuxer
/// @param data payload data of transport stream
/// @param size number of payload data bytes
/// @param is_start flag, start of pes packet
///
static void PesParse(PesDemux *pesdx, const uint8_t *data, int size, int is_start) {
const uint8_t *p;
const uint8_t *q;
if (is_start) { // start of pes packet
if (pesdx->Index && pesdx->Skip) {
// copy remaining bytes down
pesdx->Index -= pesdx->Skip;
memmove(pesdx->Buffer, pesdx->Buffer + pesdx->Skip, pesdx->Index);
pesdx->Skip = 0;
}
pesdx->State = PES_SYNC;
pesdx->HeaderIndex = 0;
pesdx->PTS = AV_NOPTS_VALUE; // reset if not yet used
pesdx->DTS = AV_NOPTS_VALUE;
}
// cleanup, if too much cruft
if (pesdx->Skip > PES_MAX_PAYLOAD / 2) {
// copy remaining bytes down
pesdx->Index -= pesdx->Skip;
memmove(pesdx->Buffer, pesdx->Buffer + pesdx->Skip, pesdx->Index);
pesdx->Skip = 0;
}
p = data;
do {
int n;
switch (pesdx->State) {
case PES_SKIP: // skip this packet
return;
case PES_START: // at start of pes packet payload
#if 0
// Played with PlayAudio
// FIXME: need 0x80 -- 0xA0 state
if (AudioCodecID == AV_CODEC_ID_NONE) {
if ((*p & 0xF0) == 0x80) { // AC-3 & DTS
Debug(3, "pesdemux: dvd ac-3\n");
} else if ((*p & 0xFF) == 0xA0) { // LPCM
Debug(3, "pesdemux: dvd lpcm\n");
pesdx->State = PES_LPCM_HEADER;
pesdx->HeaderIndex = 0;
pesdx->HeaderSize = 7;
// FIXME: need harder LPCM check
//break;
}
}
#endif
case PES_INIT: // find start of audio packet
// FIXME: increase if needed the buffer
// fill buffer
n = pesdx->Size - pesdx->Index;
if (n > size) {
n = size;
}
memcpy(pesdx->Buffer + pesdx->Index, p, n);
pesdx->Index += n;
p += n;
size -= n;
q = pesdx->Buffer + pesdx->Skip;
n = pesdx->Index - pesdx->Skip;
while (n >= 5) {
int r = 0;
unsigned codec_id = AV_CODEC_ID_NONE;
// 4 bytes 0xFFExxxxx Mpeg audio
// 5 bytes 0x0B77xxxxxx AC-3 audio
// 6 bytes 0x0B77xxxxxxxx E-AC-3 audio
// 3 bytes 0x56Exxx AAC LATM audio
// 7/9 bytes 0xFFFxxxxxxxxxxx ADTS audio
// PCM audio can't be found
// FIXME: simple+faster detection, if codec already known
if (FastMpegCheck(q)) {
r = MpegCheck(q, n);
codec_id = AV_CODEC_ID_MP2;
}
if (!r && FastAc3Check(q)) {
r = Ac3Check(q, n);
codec_id = AV_CODEC_ID_AC3;
if (r > 0 && q[5] > (10 << 3)) {
codec_id = AV_CODEC_ID_EAC3;
}
}
if (!r && FastLatmCheck(q)) {
r = LatmCheck(q, n);
codec_id = AV_CODEC_ID_AAC_LATM;
}
if (!r && FastAdtsCheck(q)) {
r = AdtsCheck(q, n);
codec_id = AV_CODEC_ID_AAC;
}
if (r < 0) { // need more bytes
break;
}
if (r > 0) {
AVPacket avpkt[1];
// new codec id, close and open new
if (AudioCodecID != codec_id) {
Debug(3, "pesdemux: new codec %#06x -> %#06x\n", AudioCodecID, codec_id);
CodecAudioClose(MyAudioDecoder);
CodecAudioOpen(MyAudioDecoder, codec_id);
AudioCodecID = codec_id;
}
av_init_packet(avpkt);
avpkt->data = (void *)q;
avpkt->size = r;
avpkt->pts = pesdx->PTS;
avpkt->dts = pesdx->DTS;
// FIXME: not aligned for ffmpeg
CodecAudioDecode(MyAudioDecoder, avpkt);
pesdx->PTS = AV_NOPTS_VALUE;
pesdx->DTS = AV_NOPTS_VALUE;
pesdx->Skip += r;
// FIXME: switch to decoder state
// pesdx->State = PES_MPEG_DECODE;
break;
}
if (AudioCodecID != AV_CODEC_ID_NONE) {
// shouldn't happen after we have a vaild codec
// detected
Debug(4, "pesdemux: skip @%d %02x\n", pesdx->Skip, q[0]);
}
// try next byte
++pesdx->Skip;
++q;
--n;
}
break;
case PES_SYNC: // wait for pes sync
n = PES_START_CODE_SIZE - pesdx->HeaderIndex;
if (n > size) {
n = size;
}
memcpy(pesdx->Header + pesdx->HeaderIndex, p, n);
pesdx->HeaderIndex += n;
p += n;
size -= n;
// have complete packet start code
if (pesdx->HeaderIndex >= PES_START_CODE_SIZE) {
unsigned code;
// bad mpeg pes packet start code prefix 0x00001xx
if (pesdx->Header[0] || pesdx->Header[1] || pesdx->Header[2] != 0x01) {
Debug(3, "pesdemux: bad pes packet\n");
pesdx->State = PES_SKIP;
return;
}
code = pesdx->Header[3];
if (code != pesdx->StartCode) {
Debug(3, "pesdemux: pes start code id %#02x\n", code);
// FIXME: need to save start code id?
pesdx->StartCode = code;
// we could have already detect a valid stream type
// don't switch to codec 'none'
}
pesdx->State = PES_HEADER;
pesdx->HeaderSize = PES_HEADER_SIZE;
}
break;
case PES_HEADER: // parse PES header
n = pesdx->HeaderSize - pesdx->HeaderIndex;
if (n > size) {
n = size;
}
memcpy(pesdx->Header + pesdx->HeaderIndex, p, n);
pesdx->HeaderIndex += n;
p += n;
size -= n;
// have header upto size bits
if (pesdx->HeaderIndex == PES_HEADER_SIZE) {
if ((pesdx->Header[6] & 0xC0) != 0x80) {
Error(_("pesdemux: mpeg1 pes packet unsupported\n"));
pesdx->State = PES_SKIP;
return;
}
// have pes extension
if (!pesdx->Header[8]) {
goto empty_header;
}
pesdx->HeaderSize += pesdx->Header[8];
// have complete header
} else if (pesdx->HeaderIndex == pesdx->HeaderSize) {
int64_t pts;
int64_t dts;
if ((pesdx->Header[7] & 0xC0) == 0x80) {
pts = (int64_t)(data[9] & 0x0E) << 29 | data[10] << 22 | (data[11] & 0xFE) << 14 |
data[12] << 7 | (data[13] & 0xFE) >> 1;
pesdx->PTS = pts;
pesdx->DTS = AV_NOPTS_VALUE;
} else if ((pesdx->Header[7] & 0xC0) == 0xC0) {
pts = (int64_t)(data[9] & 0x0E) << 29 | data[10] << 22 | (data[11] & 0xFE) << 14 |
data[12] << 7 | (data[13] & 0xFE) >> 1;
pesdx->PTS = pts;
dts = (int64_t)(data[14] & 0x0E) << 29 | data[15] << 22 | (data[16] & 0xFE) << 14 |
data[17] << 7 | (data[18] & 0xFE) >> 1;
pesdx->DTS = dts;
Debug(4, "pesdemux: pts %#012" PRIx64 " %#012" PRIx64 "\n", pts, dts);
}
empty_header:
pesdx->State = PES_INIT;
if (pesdx->StartCode == PES_PRIVATE_STREAM1) {
// only private stream 1, has sub streams
pesdx->State = PES_START;
}
}
break;
#if 0
// Played with PlayAudio
case PES_LPCM_HEADER: // lpcm header
n = pesdx->HeaderSize - pesdx->HeaderIndex;
if (n > size) {
n = size;
}
memcpy(pesdx->Header + pesdx->HeaderIndex, p, n);
pesdx->HeaderIndex += n;
p += n;
size -= n;
if (pesdx->HeaderIndex == pesdx->HeaderSize) {
static int samplerates[] = { 48000, 96000, 44100, 32000 };
int samplerate;
int channels;
int bits_per_sample;
const uint8_t *q;
if (AudioCodecID != AV_CODEC_ID_PCM_DVD) {
q = pesdx->Header;
Debug(3, "pesdemux: LPCM %d sr:%d bits:%d chan:%d\n", q[0], q[5] >> 4,
(((q[5] >> 6) & 0x3) + 4) * 4, (q[5] & 0x7) + 1);
CodecAudioClose(MyAudioDecoder);
bits_per_sample = (((q[5] >> 6) & 0x3) + 4) * 4;
if (bits_per_sample != 16) {
Error(_("softhddev: LPCM %d bits per sample aren't supported\n"), bits_per_sample);
// FIXME: handle unsupported formats.
}
samplerate = samplerates[q[5] >> 4];
channels = (q[5] & 0x7) + 1;
AudioSetup(&samplerate, &channels, 0);
if (samplerate != samplerates[q[5] >> 4]) {
Error(_("softhddev: LPCM %d sample-rate is unsupported\n"), samplerates[q[5] >> 4]);
// FIXME: support resample
}
if (channels != (q[5] & 0x7) + 1) {
Error(_("softhddev: LPCM %d channels are unsupported\n"), (q[5] & 0x7) + 1);
// FIXME: support resample
}
//CodecAudioOpen(MyAudioDecoder, AV_CODEC_ID_PCM_DVD);
AudioCodecID = AV_CODEC_ID_PCM_DVD;
}
pesdx->State = PES_LPCM_PAYLOAD;
pesdx->Index = 0;
pesdx->Skip = 0;
}
break;
case PES_LPCM_PAYLOAD: // lpcm payload
// fill buffer
n = pesdx->Size - pesdx->Index;
if (n > size) {
n = size;
}
memcpy(pesdx->Buffer + pesdx->Index, p, n);
pesdx->Index += n;
p += n;
size -= n;
if (pesdx->PTS != (int64_t) AV_NOPTS_VALUE) {
// FIXME: needs bigger buffer
AudioSetClock(pesdx->PTS);
pesdx->PTS = AV_NOPTS_VALUE;
}
swab(pesdx->Buffer, pesdx->Buffer, pesdx->Index);
AudioEnqueue(pesdx->Buffer, pesdx->Index);
pesdx->Index = 0;
break;
#endif
}
} while (size > 0);
}
//////////////////////////////////////////////////////////////////////////////
// Transport stream demux
//////////////////////////////////////////////////////////////////////////////
/// Transport stream packet size
#define TS_PACKET_SIZE 188
/// Transport stream packet sync byte
#define TS_PACKET_SYNC 0x47
///
/// transport stream demuxer typedef.
///
typedef struct _ts_demux_ TsDemux;
///
/// transport stream demuxer structure.
///
struct _ts_demux_ {
int Packets; ///< packets between PCR
};
static PesDemux PesDemuxAudio[1]; ///< audio demuxer
///
/// Transport stream demuxer.
///
/// @param tsdx transport stream demuxer
/// @param data buffer of transport stream packets
/// @param size size of buffer
///
/// @returns number of bytes consumed from buffer.
///
static int TsDemuxer(TsDemux *tsdx, const uint8_t *data, int size) {
const uint8_t *p;
p = data;
while (size >= TS_PACKET_SIZE) {
#ifdef DEBUG
int pid;
#endif
int payload;
if (p[0] != TS_PACKET_SYNC) {
Error(_("tsdemux: transport stream out of sync\n"));
// FIXME: kill all buffers
return size;
}
++tsdx->Packets;
if (p[1] & 0x80) { // error indicator
Debug(3, "tsdemux: transport error\n");
// FIXME: kill all buffers
goto next_packet;
}
#ifdef DEBUG
pid = (p[1] & 0x1F) << 8 | p[2];
Debug(4, "tsdemux: PID: %#04x%s%s\n", pid, p[1] & 0x40 ? " start" : "", p[3] & 0x10 ? " payload" : "");
#endif
// skip adaptation field
switch (p[3] & 0x30) { // adaption field
case 0x00: // reserved
case 0x20: // adaptation field only
default:
goto next_packet;
case 0x10: // only payload
payload = 4;
break;
case 0x30: // skip adapation field
payload = 5 + p[4];
// illegal length, ignore packet
if (payload >= TS_PACKET_SIZE) {
Debug(3, "tsdemux: illegal adaption field length\n");
goto next_packet;
}
break;
}
PesParse(PesDemuxAudio, p + payload, TS_PACKET_SIZE - payload, p[1] & 0x40);
#if 0
int tmp;
// check continuity
tmp = p[3] & 0x0F; // continuity counter
if (((tsdx->CC + 1) & 0x0F) != tmp) {
Debug(3, "tsdemux: OUT OF SYNC: %d %d\n", tmp, tsdx->CC);
// TS discontinuity (received 8, expected 0) for PID
}
tsdx->CC = tmp;
#endif
next_packet:
p += TS_PACKET_SIZE;
size -= TS_PACKET_SIZE;
}
return p - data;
}
#endif
/**
** Play audio packet.
**
** @param data data of exactly one complete PES packet
** @param size size of PES packet
** @param id PES packet type
*/
int PlayAudio(const uint8_t *data, int size, uint8_t id) {
int n;
const uint8_t *p;
// channel switch: SetAudioChannelDevice: SetDigitalAudioDevice:
if (SkipAudio || !MyAudioDecoder) { // skip audio
return size;
}
if (StreamFreezed) { // stream freezed
return 0;
}
if (AudioDelay) {
Debug(3, "AudioDelay %dms\n", AudioDelay);
usleep(AudioDelay / 90);
AudioDelay = 0;
return 0;
}
if (NewAudioStream) {
// this clears the audio ringbuffer indirect, open and setup does it
CodecAudioClose(MyAudioDecoder);
AudioFlushBuffers();
AudioSetBufferTime(ConfigAudioBufferTime);
AudioCodecID = AV_CODEC_ID_NONE;
AudioChannelID = -1;
NewAudioStream = 0;
}
// hard limit buffer full: don't overrun audio buffers on replay
if (AudioFreeBytes() < AUDIO_MIN_BUFFER_FREE) {
return 0;
}
#ifdef USE_SOFTLIMIT
// soft limit buffer full
if (AudioSyncStream && VideoGetBuffers(AudioSyncStream) > 3 && AudioUsedBytes() > AUDIO_MIN_BUFFER_FREE * 2) {
return 0;