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Add new SPDIFEncoder class. This class uses ffmpeg muxer to encode IE…
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…C958 packets. This replace myth's own IEC958 encoder. By doing so, we now have the basis for E-AC3, TrueHD and DTS-HD passthrough
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@jyavenard
jyavenard committed Dec 14, 2010
1 parent 1e0ef88 commit 7531972
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Showing 9 changed files with 292 additions and 495 deletions.
293 changes: 24 additions & 269 deletions mythtv/libs/libmyth/audio/audiooutputdigitalencoder.cpp
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Expand Up @@ -8,27 +8,7 @@
// libav headers // libav headers
extern "C" { extern "C" {
#include "libavutil/mem.h" // for av_free #include "libavutil/mem.h" // for av_free

// copied from libavutil/internal.h
#include "libavutil/common.h" // for AV_GCC_VERSION_AT_LEAST()
#ifndef av_alias
#if HAVE_ATTRIBUTE_MAY_ALIAS && (!defined(__ICC) || __ICC > 1110) && AV_GCC_VERSION_AT_LEAST(3,3)
# define av_alias __attribute__((may_alias))
#else
# define av_alias
#endif
#endif

#include "libavcodec/avcodec.h" #include "libavcodec/avcodec.h"
#if CONFIG_AC3_DECODER
#include "libavcodec/ac3.h"
#ifndef INT_BIT
#define INT_BIT (CHAR_BIT * sizeof(int))
#endif
#include "libavcodec/ac3_parser.h"
#else
#include <a52dec/a52.h>
#endif
} }


// MythTV headers // MythTV headers
Expand All @@ -40,16 +20,15 @@ extern "C" {
#define LOC QString("DEnc: ") #define LOC QString("DEnc: ")
#define LOC_ERR QString("DEnc, Error: ") #define LOC_ERR QString("DEnc, Error: ")


#define MAX_AC3_FRAME_SIZE 6144

AudioOutputDigitalEncoder::AudioOutputDigitalEncoder(void) : AudioOutputDigitalEncoder::AudioOutputDigitalEncoder(void) :
bytes_per_sample(0), bytes_per_sample(0),
av_context(NULL), av_context(NULL),
outlen(0), outlen(0),
inlen(0), inlen(0),
one_frame_bytes(0) one_frame_bytes(0),
m_spdifenc(NULL)
{ {
in = (char *)(((long)&inbuf + 15) & ~15); in = (unsigned char *)(((long)&inbuf + 15) & ~15);
} }


AudioOutputDigitalEncoder::~AudioOutputDigitalEncoder() AudioOutputDigitalEncoder::~AudioOutputDigitalEncoder()
Expand All @@ -65,9 +44,13 @@ void AudioOutputDigitalEncoder::Dispose()
av_free(av_context); av_free(av_context);
av_context = NULL; av_context = NULL;
} }
if (m_spdifenc)
{
delete m_spdifenc;
m_spdifenc = NULL;
}
} }


//CODEC_ID_AC3
bool AudioOutputDigitalEncoder::Init( bool AudioOutputDigitalEncoder::Init(
CodecID codec_id, int bitrate, int samplerate, int channels) CodecID codec_id, int bitrate, int samplerate, int channels)
{ {
Expand All @@ -83,7 +66,6 @@ bool AudioOutputDigitalEncoder::Init(
// We need to do this when called from mythmusic // We need to do this when called from mythmusic
avcodec_init(); avcodec_init();
avcodec_register_all(); avcodec_register_all();
// always AC3 as there is no DTS encoder at the moment 2005/1/9
codec = avcodec_find_encoder(CODEC_ID_AC3); codec = avcodec_find_encoder(CODEC_ID_AC3);
if (!codec) if (!codec)
{ {
Expand All @@ -107,6 +89,12 @@ bool AudioOutputDigitalEncoder::Init(
return false; return false;
} }


if (m_spdifenc)
{
delete m_spdifenc;
}
m_spdifenc = new SPDIFEncoder("spdif", av_context);

bytes_per_sample = av_context->channels * sizeof(short); bytes_per_sample = av_context->channels * sizeof(short);
one_frame_bytes = bytes_per_sample * av_context->frame_size; one_frame_bytes = bytes_per_sample * av_context->frame_size;


Expand All @@ -118,242 +106,10 @@ bool AudioOutputDigitalEncoder::Init(
return true; return true;
} }


static int DTS_SAMPLEFREQS[16] =
{
0, 8000, 16000, 32000, 64000, 128000, 11025, 22050,
44100, 88200, 176400, 12000, 24000, 48000, 96000, 192000
};

static int DTS_BITRATES[30] =
{
32000, 56000, 64000, 96000, 112000, 128000,
192000, 224000, 256000, 320000, 384000, 448000,
512000, 576000, 640000, 768000, 896000, 1024000,
1152000, 1280000, 1344000, 1408000, 1411200, 1472000,
1536000, 1920000, 2048000, 3072000, 3840000, 4096000
};

static int dts_decode_header(uint8_t *indata_ptr, int *rate,
int *nblks, int *sfreq)
{
uint id = ((indata_ptr[0] << 24) | (indata_ptr[1] << 16) |
(indata_ptr[2] << 8) | (indata_ptr[3]));

if (id != 0x7ffe8001)
return -1;

int ftype = indata_ptr[4] >> 7;

int surp = (indata_ptr[4] >> 2) & 0x1f;
surp = (surp + 1) % 32;

*nblks = (indata_ptr[4] & 0x01) << 6 | (indata_ptr[5] >> 2);
++*nblks;

int fsize = (indata_ptr[5] & 0x03) << 12 |
(indata_ptr[6] << 4) | (indata_ptr[7] >> 4);
++fsize;

*sfreq = (indata_ptr[8] >> 2) & 0x0f;
*rate = (indata_ptr[8] & 0x03) << 3 | ((indata_ptr[9] >> 5) & 0x07);

if (ftype != 1)
{
VERBOSE(VB_IMPORTANT, LOC +
QString("DTS: Termination frames not handled (ftype %1)")
.arg(ftype));
return -1;
}

if (*sfreq != 13)
{
VERBOSE(VB_IMPORTANT, LOC +
QString("DTS: Only 48kHz supported (sfreq %1)").arg(*sfreq));
return -1;
}

if ((fsize > 8192) || (fsize < 96))
{
VERBOSE(VB_IMPORTANT, LOC +
QString("DTS: fsize: %1 invalid").arg(fsize));
return -1;
}

if (*nblks != 8 && *nblks != 16 && *nblks != 32 &&
*nblks != 64 && *nblks != 128 && ftype == 1)
{
VERBOSE(VB_IMPORTANT, LOC +
QString("DTS: nblks %1 not valid for normal frame")
.arg(*nblks));
return -1;
}

return fsize;
}

static int dts_syncinfo(uint8_t *indata_ptr, int *flags,
int *sample_rate, int *bit_rate)
{
(void) flags;

int nblks;
int rate;
int sfreq;

int fsize = dts_decode_header(indata_ptr, &rate, &nblks, &sfreq);
if (fsize >= 0)
{
*bit_rate = (rate >= 0 && rate <= 29) ? DTS_BITRATES[rate] : 0;
*sample_rate = (sfreq >= 1 && sfreq <= 15)? DTS_SAMPLEFREQS[sfreq] : 0;
}

return fsize;
}

// from http://www.ebu.ch/CMSimages/en/tec_AES-EBU_eg_tcm6-11890.pdf
// http://en.wikipedia.org/wiki/S/PDIF
typedef struct {
// byte 0
unsigned professional_consumer:1;
unsigned non_data:1;
// 4 - no emphasis
// 6 - 50/15us
// 7 - CCITT J17
unsigned audio_signal_emphasis:3;
unsigned SSFL:1;
// 0
// 1 - 48k
// 2 - 44.1k
// 3 - 32k
unsigned sample_frequency:2;
// byte 1
// 0
// 1 - 2 ch
// 2 - mono
// 3 - prim/sec
// 4 - stereo
unsigned channel_mode:4;
// 0
// 1 - 192 bit block
// 2 - AES18
// 3 - user def
unsigned user_bit_management:4;
// byte 2
// 1 - audio data
// 2 - co-ordn
unsigned auxiliary_bits:3;
// 4 - 16 bits
// 5-7 - redither to 16 bits
unsigned source_word_length:3;
unsigned reserved:2;
// byte 3
unsigned multi_channel_function_description:8;
// byte 4
unsigned digital_audio_reference_signal:2;
unsigned reserved2:6;

} AESHeader;

static int encode_frame(
bool dts,
unsigned char *data,
size_t enc_len)
{
unsigned char *payload = data + 8; // skip header, currently 52 or 54bits
int flags, sample_rate, bit_rate;

// we don't do any length/crc validation of the AC3 frame here; presumably
// the receiver will have enough sense to do that. if someone has a
// receiver that doesn't, here would be a good place to put in a call
// to a52_crc16_block(samples+2, data_size-2) - but what do we do if the
// packet is bad? we'd need to send something that the receiver would
// ignore, and if so, may as well just assume that it will ignore
// anything with a bad CRC...

uint nr_samples = 0, block_len = 0;

if (dts)
{
enc_len = dts_syncinfo(payload, &flags, &sample_rate, &bit_rate);
int rate, sfreq, nblks;
dts_decode_header(payload, &rate, &nblks, &sfreq);
nr_samples = nblks * 32;
block_len = nr_samples * 2 * 2;
}
else if (CONFIG_AC3_DECODER)
{
int err;
AC3HeaderInfo hdr;
GetBitContext gbc;

init_get_bits(&gbc, payload, 54);
err = ff_ac3_parse_header(&gbc, &hdr);

if(err < 0)
enc_len = 0;
else
{
sample_rate = hdr.sample_rate;
bit_rate = hdr.bit_rate;
enc_len = hdr.frame_size;
block_len = AC3_FRAME_SIZE * 4;
}
}

enc_len = std::min((uint)enc_len, block_len - 8);

swab((const char *)payload, (char *)payload, enc_len);

// the following values come from libmpcodecs/ad_hwac3.c in mplayer.
// they form a valid IEC958 AC3 header.
data[0] = 0x72;
data[1] = 0xF8;
data[2] = 0x1F;
data[3] = 0x4E;
data[4] = 0x01;
if (dts)
{
switch(nr_samples)
{
case 256:
data[4] = 0x0A; /* DTS-? (256-sample bursts) */
break;

case 512:
data[4] = 0x0B; /* DTS-1 (512-sample bursts) */
break;

case 1024:
data[4] = 0x0C; /* DTS-2 (1024-sample bursts) */
break;

case 2048:
data[4] = 0x0D; /* DTS-3 (2048-sample bursts) */
break;

case 4096:
data[4] = 0x0E; /* DTS-? (4096-sample bursts) */
break;

default:
VERBOSE(VB_IMPORTANT, LOC +
QString("DTS: %1-sample bursts not supported")
.arg(nr_samples));
data[4] = 0x00;
break;
}
}
data[5] = 0x00;
data[6] = (enc_len << 3) & 0xFF;
data[7] = (enc_len >> 5) & 0xFF;
memset(payload + enc_len, 0, block_len - 8 - enc_len);

return enc_len;
}

size_t AudioOutputDigitalEncoder::Encode(void *buf, int len, bool isFloat) size_t AudioOutputDigitalEncoder::Encode(void *buf, int len, bool isFloat)
{ {
size_t outsize = 0; size_t outsize = 0;
int data_size;


if (isFloat) if (isFloat)
inlen += AudioOutputUtil::fromFloat(FORMAT_S16, in + inlen, buf, len); inlen += AudioOutputUtil::fromFloat(FORMAT_S16, in + inlen, buf, len);
Expand All @@ -368,24 +124,23 @@ size_t AudioOutputDigitalEncoder::Encode(void *buf, int len, bool isFloat)


while (i < frames) while (i < frames)
{ {
// put data in the correct spot for encode frame
outsize = avcodec_encode_audio(av_context, outsize = avcodec_encode_audio(av_context,
((uchar*)out) + outlen + 8, m_encodebuffer,
OUTBUFSIZE - outlen - 8, FF_MIN_BUFFER_SIZE,
(short *)(in + i * one_frame_bytes)); (short *)(in + i * one_frame_bytes));
if (outsize < 0) if (outsize < 0)
{ {
VERBOSE(VB_AUDIO, LOC_ERR + "AC-3 encode error"); VERBOSE(VB_AUDIO, LOC_ERR + "AC-3 encode error");
return outlen; return outlen;
} }


encode_frame( if (!m_spdifenc)
/*av_context->codec_id==CODEC_ID_DTS*/ false, {
(uchar*)out + outlen, outsize m_spdifenc = new SPDIFEncoder("spdif", av_context);
); }

m_spdifenc->WriteFrame(m_encodebuffer, outsize);

m_spdifenc->GetData(out + outlen, data_size);
outlen += MAX_AC3_FRAME_SIZE; outlen += data_size;
inlen -= one_frame_bytes; inlen -= one_frame_bytes;
i++; i++;
} }
Expand Down
10 changes: 7 additions & 3 deletions mythtv/libs/libmyth/audio/audiooutputdigitalencoder.h
View file
Expand Up @@ -5,6 +5,8 @@ extern "C" {
#include "libavcodec/avcodec.h" #include "libavcodec/avcodec.h"
}; };


#include "spdifencoder.h"

#define INBUFSIZE 131072 #define INBUFSIZE 131072
#define OUTBUFSIZE 98304 #define OUTBUFSIZE 98304
#define ENCODER_INBUFSIZE INBUFSIZE #define ENCODER_INBUFSIZE INBUFSIZE
Expand All @@ -27,12 +29,14 @@ class AudioOutputDigitalEncoder


private: private:
AVCodecContext *av_context; AVCodecContext *av_context;
char out[OUTBUFSIZE]; unsigned char out[OUTBUFSIZE];
char inbuf[INBUFSIZE+16]; unsigned char inbuf[INBUFSIZE+16];
char *in; unsigned char *in;
int outlen; int outlen;
int inlen; int inlen;
size_t one_frame_bytes; size_t one_frame_bytes;
uint8_t m_encodebuffer[FF_MIN_BUFFER_SIZE];
SPDIFEncoder *m_spdifenc;
}; };


#endif #endif

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