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-rw-r--r--libhb/bd.c12
-rw-r--r--libhb/common.c2
-rw-r--r--libhb/common.h19
-rw-r--r--libhb/decavcodec.c83
-rw-r--r--libhb/dvd.c4
-rw-r--r--libhb/dvdnav.c4
-rw-r--r--libhb/muxmkv.c2
-rw-r--r--libhb/stream.c64
-rw-r--r--libhb/sync.c6
-rw-r--r--libhb/work.c36
10 files changed, 133 insertions, 99 deletions
diff --git a/libhb/bd.c b/libhb/bd.c
index 536769307..9231436f6 100644
--- a/libhb/bd.c
+++ b/libhb/bd.c
@@ -103,7 +103,7 @@ static void add_audio(int track, hb_list_t *list_audio, BLURAY_STREAM_INFO *bdau
strlen(lang->native_name) ? lang->native_name : lang->eng_name,
audio->config.in.codec == HB_ACODEC_AC3 ? "AC3" :
( audio->config.in.codec == HB_ACODEC_DCA ? "DTS" :
- ( audio->config.in.codec == HB_ACODEC_MPGA ?
+ ( ( audio->config.in.codec & HB_ACODEC_FF_MASK ) ?
( stream_type == BLURAY_STREAM_TYPE_AUDIO_LPCM ? "BD LPCM" :
( stream_type == BLURAY_STREAM_TYPE_AUDIO_AC3PLUS ? "E-AC3" :
( stream_type == BLURAY_STREAM_TYPE_AUDIO_TRUHD ? "TrueHD" :
@@ -352,7 +352,7 @@ hb_title_t * hb_bd_title_scan( hb_bd_t * d, int tt, uint64_t min_duration )
add_audio(ii, title->list_audio, bdaudio,
HB_SUBSTREAM_BD_AC3, HB_ACODEC_AC3, 0);
add_audio(ii, title->list_audio, bdaudio,
- HB_SUBSTREAM_BD_TRUEHD, HB_ACODEC_MPGA, CODEC_ID_TRUEHD);
+ HB_SUBSTREAM_BD_TRUEHD, HB_ACODEC_FFMPEG, CODEC_ID_TRUEHD);
break;
case BLURAY_STREAM_TYPE_AUDIO_DTS:
@@ -362,17 +362,17 @@ hb_title_t * hb_bd_title_scan( hb_bd_t * d, int tt, uint64_t min_duration )
case BLURAY_STREAM_TYPE_AUDIO_MPEG2:
case BLURAY_STREAM_TYPE_AUDIO_MPEG1:
add_audio(ii, title->list_audio, bdaudio, 0,
- HB_ACODEC_MPGA, CODEC_ID_MP2);
+ HB_ACODEC_FFMPEG, CODEC_ID_MP2);
break;
case BLURAY_STREAM_TYPE_AUDIO_AC3PLUS:
add_audio(ii, title->list_audio, bdaudio, 0,
- HB_ACODEC_MPGA, CODEC_ID_EAC3);
+ HB_ACODEC_FFMPEG, CODEC_ID_EAC3);
break;
case BLURAY_STREAM_TYPE_AUDIO_LPCM:
add_audio(ii, title->list_audio, bdaudio, 0,
- HB_ACODEC_MPGA, CODEC_ID_PCM_BLURAY);
+ HB_ACODEC_FFMPEG, CODEC_ID_PCM_BLURAY);
break;
case BLURAY_STREAM_TYPE_AUDIO_AC3:
@@ -388,7 +388,7 @@ hb_title_t * hb_bd_title_scan( hb_bd_t * d, int tt, uint64_t min_duration )
// DTS-core followed by DTS-hd-extensions. Setting
// a substream id of 0 says use all substreams.
add_audio(ii, title->list_audio, bdaudio, 0,
- HB_ACODEC_MPGA, CODEC_ID_DTS);
+ HB_ACODEC_DCA_HD, CODEC_ID_DTS);
break;
default:
diff --git a/libhb/common.c b/libhb/common.c
index 2d5963d76..5f3ce49f8 100644
--- a/libhb/common.c
+++ b/libhb/common.c
@@ -1226,7 +1226,7 @@ int hb_audio_add(const hb_job_t * job, const hb_audio_config_t * audiocfg)
*/
audio->config.out.track = hb_list_count(job->list_audio) + 1;
audio->config.out.codec = audiocfg->out.codec;
- if( (audiocfg->out.codec & HB_ACODEC_MASK) == audio->config.in.codec &&
+ if( (audiocfg->out.codec & HB_ACODEC_MASK) == (audio->config.in.codec & HB_ACODEC_MASK) &&
(audiocfg->out.codec & HB_ACODEC_PASS_FLAG ) )
{
/* Pass-through, copy from input. */
diff --git a/libhb/common.h b/libhb/common.h
index 492e39750..d350a6bf1 100644
--- a/libhb/common.h
+++ b/libhb/common.h
@@ -316,17 +316,20 @@ struct hb_job_s
#define HB_ACODEC_LAME 0x00000200
#define HB_ACODEC_VORBIS 0x00000400
#define HB_ACODEC_AC3 0x00000800
-#define HB_ACODEC_MPGA 0x00001000
-#define HB_ACODEC_LPCM 0x00002000
-#define HB_ACODEC_DCA 0x00004000
-#define HB_ACODEC_FFMPEG 0x00008000
-#define HB_ACODEC_CA_AAC 0x00010000
-#define HB_ACODEC_CA_HAAC 0x00020000
-#define HB_ACODEC_FFAAC 0x00040000
+#define HB_ACODEC_LPCM 0x00001000
+#define HB_ACODEC_DCA 0x00002000
+#define HB_ACODEC_CA_AAC 0x00004000
+#define HB_ACODEC_CA_HAAC 0x00008000
+#define HB_ACODEC_FFAAC 0x00010000
+#define HB_ACODEC_FFMPEG 0x00020000
+#define HB_ACODEC_DCA_HD 0x00040000
+#define HB_ACODEC_FF_MASK 0x00060000
+#define HB_ACODEC_FF_I_FLAG 0x80000000
#define HB_ACODEC_PASS_FLAG 0x40000000
-#define HB_ACODEC_PASS_MASK (HB_ACODEC_AC3 | HB_ACODEC_DCA)
+#define HB_ACODEC_PASS_MASK (HB_ACODEC_DCA_HD | HB_ACODEC_AC3 | HB_ACODEC_DCA)
#define HB_ACODEC_AC3_PASS (HB_ACODEC_AC3 | HB_ACODEC_PASS_FLAG)
#define HB_ACODEC_DCA_PASS (HB_ACODEC_DCA | HB_ACODEC_PASS_FLAG)
+#define HB_ACODEC_DCA_HD_PASS (HB_ACODEC_DCA_HD | HB_ACODEC_PASS_FLAG)
#define HB_ACODEC_ANY (HB_ACODEC_MASK | HB_ACODEC_PASS_FLAG)
#define HB_SUBSTREAM_BD_TRUEHD 0x72
diff --git a/libhb/decavcodec.c b/libhb/decavcodec.c
index 610ec1120..63b1c6dff 100644
--- a/libhb/decavcodec.c
+++ b/libhb/decavcodec.c
@@ -74,7 +74,7 @@ static int decavcodecBSInfo( hb_work_object_t *, const hb_buffer_t *, hb_work_in
hb_work_object_t hb_decavcodec =
{
WORK_DECAVCODEC,
- "MPGA decoder (libavcodec)",
+ "Audio decoder (libavcodec)",
decavcodecInit,
decavcodecWork,
decavcodecClose,
@@ -1327,23 +1327,6 @@ static void init_ffmpeg_context( hb_work_object_t *w )
pv->brokenByMicrosoft = 1;
}
-static void prepare_ffmpeg_buffer( hb_buffer_t * in )
-{
- // ffmpeg requires an extra 8 bytes of zero at the end of the buffer and
- // will seg fault in odd, data dependent ways if it's not there. (my guess
- // is this is a case of a local performance optimization creating a global
- // performance degradation since all the time wasted by extraneous data
- // copies & memory zeroing has to be huge compared to the minor reduction
- // in inner-loop instructions this affords - modern cpus bottleneck on
- // memory bandwidth not instruction bandwidth).
- if ( in->size + FF_INPUT_BUFFER_PADDING_SIZE > in->alloc )
- {
- // have to realloc to add the padding
- hb_buffer_realloc( in, in->size + FF_INPUT_BUFFER_PADDING_SIZE );
- }
- memset( in->data + in->size, 0, FF_INPUT_BUFFER_PADDING_SIZE );
-}
-
static int decavcodecviInit( hb_work_object_t * w, hb_job_t * job )
{
@@ -1429,7 +1412,6 @@ static int decavcodecviWork( hb_work_object_t * w, hb_buffer_t ** buf_in,
pv->new_chap = in->new_chap;
pv->chap_time = pts >= 0? pts : pv->pts_next;
}
- prepare_ffmpeg_buffer( in );
decodeFrame( pv, in->data, in->size, in->sequence, in->start, in->renderOffset );
hb_buffer_close( &in );
*buf_out = link_buf_list( pv );
@@ -1475,7 +1457,7 @@ static hb_buffer_t * downmixAudio(
else
{
buf = hb_buffer_init( nsamples * sizeof(float) );
- memcpy(buf->data, buffer, nsamples * sizeof(float) );
+ memcpy( buf->data, buffer, nsamples * sizeof(float) );
}
return buf;
@@ -1524,7 +1506,34 @@ static void decodeAudio( hb_audio_t * audio, hb_work_private_t *pv, uint8_t *dat
pos += len;
if( out_size > 0 )
{
- // We require signed 16-bit ints for the output format. If
+ int isamp = av_get_bytes_per_sample( context->sample_fmt );
+ nsamples = out_size / isamp;
+ double pts_next = pv->pts_next + nsamples * pv->duration;
+
+ // DTS-HD can be passed through to mkv
+ if( audio->config.out.codec & HB_ACODEC_PASS_FLAG )
+ {
+ // Note that even though we are doing passthru, we had
+ // to decode so that we know the stop time and the
+ // pts of the next audio packet.
+ hb_buffer_t * buf;
+
+ buf = hb_buffer_init( avp.size );
+ memcpy( buf->data, avp.data, len );
+ buf->start = pv->pts_next;
+ buf->stop = pts_next;
+ hb_list_add( pv->list, buf );
+
+ if ( hb_list_count( audio->priv.ff_audio_list ) == 0 )
+ {
+ pv->pts_next = pts_next;
+ continue;
+ }
+ // Fall through and process the list of other audio
+ // pipelines that use this ffmpeg audio stream.
+ }
+
+ // We require signed floats for the output format. If
// we got something different convert it.
if ( context->sample_fmt != AV_SAMPLE_FMT_FLT )
{
@@ -1535,16 +1544,13 @@ static void decodeAudio( hb_audio_t * audio, hb_work_private_t *pv, uint8_t *dat
// anything more complicated than a one-for-one format
// conversion we'd probably want to cache the converter
// context in the pv.
- int isamp;
AVAudioConvert *ctx;
- isamp = av_get_bytes_per_sample( context->sample_fmt );
ctx = av_audio_convert_alloc( AV_SAMPLE_FMT_FLT, 1,
context->sample_fmt, 1,
NULL, 0 );
// get output buffer size then malloc a buffer
- nsamples = out_size / isamp;
buffer = av_malloc( nsamples * sizeof(hb_sample_t) );
// we're doing straight sample format conversion which
@@ -1557,16 +1563,11 @@ static void decodeAudio( hb_audio_t * audio, hb_work_private_t *pv, uint8_t *dat
av_audio_convert( ctx, obuf, ostride, ibuf, istride, nsamples );
av_audio_convert_free( ctx );
}
- else
- {
- nsamples = out_size / sizeof(hb_sample_t);
- }
- hb_buffer_t * buf;
- double pts_next = pv->pts_next + nsamples * pv->duration;
- buf = downmixAudio( audio, pv, buffer, context->channels, nsamples );
- if ( buf )
+ if ( !( audio->config.out.codec & HB_ACODEC_PASS_FLAG ) )
{
+ hb_buffer_t * buf;
+ buf = downmixAudio( audio, pv, buffer, context->channels, nsamples );
buf->start = pv->pts_next;
buf->stop = pts_next;
hb_list_add( pv->list, buf );
@@ -1579,13 +1580,20 @@ static void decodeAudio( hb_audio_t * audio, hb_work_private_t *pv, uint8_t *dat
hb_work_private_t *ff_pv = hb_list_item( pv->ff_audio_list, i );
if ( ff_pv )
{
- buf = downmixAudio( ff_audio, ff_pv, buffer, context->channels, nsamples );
- if ( buf )
+ hb_buffer_t * buf;
+
+ if ( !( ff_audio->config.out.codec & HB_ACODEC_PASS_FLAG ) )
+ {
+ buf = downmixAudio( ff_audio, ff_pv, buffer, context->channels, nsamples );
+ }
+ else
{
- buf->start = pv->pts_next;
- buf->stop = pts_next;
- hb_list_add( ff_pv->list, buf );
+ buf = hb_buffer_init( avp.size );
+ memcpy( buf->data, avp.data, len );
}
+ buf->start = pv->pts_next;
+ buf->stop = pts_next;
+ hb_list_add( ff_pv->list, buf );
}
}
pv->pts_next = pts_next;
@@ -1639,7 +1647,6 @@ static int decavcodecaiWork( hb_work_object_t *w, hb_buffer_t **buf_in,
{
pv->pts_next = in->start;
}
- prepare_ffmpeg_buffer( in );
decodeAudio( w->audio, pv, in->data, in->size, pv->pts_next );
writeAudioFifos( w );
*buf_out = link_buf_list( pv );
diff --git a/libhb/dvd.c b/libhb/dvd.c
index c5c052223..2a36eb4f2 100644
--- a/libhb/dvd.c
+++ b/libhb/dvd.c
@@ -360,7 +360,7 @@ static hb_title_t * hb_dvdread_title_scan( hb_dvd_t * e, int t, uint64_t min_dur
case 0x02:
case 0x03:
audio->id = 0xc0 + position;
- audio->config.in.codec = HB_ACODEC_MPGA;
+ audio->config.in.codec = HB_ACODEC_FFMPEG;
break;
case 0x04:
@@ -411,7 +411,7 @@ static hb_title_t * hb_dvdread_title_scan( hb_dvd_t * e, int t, uint64_t min_dur
strlen(lang->native_name) ? lang->native_name : lang->eng_name,
audio->config.in.codec == HB_ACODEC_AC3 ? "AC3" : ( audio->config.in.codec ==
HB_ACODEC_DCA ? "DTS" : ( audio->config.in.codec ==
- HB_ACODEC_MPGA ? "MPEG" : "LPCM" ) ) );
+ HB_ACODEC_FFMPEG ? "MPEG" : "LPCM" ) ) );
snprintf( audio->config.lang.simple, sizeof( audio->config.lang.simple ), "%s",
strlen(lang->native_name) ? lang->native_name : lang->eng_name );
snprintf( audio->config.lang.iso639_2, sizeof( audio->config.lang.iso639_2 ), "%s",
diff --git a/libhb/dvdnav.c b/libhb/dvdnav.c
index d65b6abbb..eda1988bc 100644
--- a/libhb/dvdnav.c
+++ b/libhb/dvdnav.c
@@ -508,7 +508,7 @@ static hb_title_t * hb_dvdnav_title_scan( hb_dvd_t * e, int t, uint64_t min_dura
case 0x02:
case 0x03:
audio->id = 0xc0 + position;
- audio->config.in.codec = HB_ACODEC_MPGA;
+ audio->config.in.codec = HB_ACODEC_FFMPEG;
break;
case 0x04:
@@ -559,7 +559,7 @@ static hb_title_t * hb_dvdnav_title_scan( hb_dvd_t * e, int t, uint64_t min_dura
strlen(lang->native_name) ? lang->native_name : lang->eng_name,
audio->config.in.codec == HB_ACODEC_AC3 ? "AC3" : ( audio->config.in.codec ==
HB_ACODEC_DCA ? "DTS" : ( audio->config.in.codec ==
- HB_ACODEC_MPGA ? "MPEG" : "LPCM" ) ) );
+ HB_ACODEC_FFMPEG ? "MPEG" : "LPCM" ) ) );
snprintf( audio->config.lang.simple, sizeof( audio->config.lang.simple ), "%s",
strlen(lang->native_name) ? lang->native_name : lang->eng_name );
snprintf( audio->config.lang.iso639_2, sizeof( audio->config.lang.iso639_2 ), "%s",
diff --git a/libhb/muxmkv.c b/libhb/muxmkv.c
index 654c51bc0..f73139b98 100644
--- a/libhb/muxmkv.c
+++ b/libhb/muxmkv.c
@@ -191,6 +191,8 @@ static int MKVInit( hb_mux_object_t * m )
{
case HB_ACODEC_DCA:
case HB_ACODEC_DCA_PASS:
+ case HB_ACODEC_DCA_HD:
+ case HB_ACODEC_DCA_HD_PASS:
track->codecPrivate = NULL;
track->codecPrivateSize = 0;
track->codecID = MK_ACODEC_DTS;
diff --git a/libhb/stream.c b/libhb/stream.c
index fdfc8b6a7..ddf20602b 100644
--- a/libhb/stream.c
+++ b/libhb/stream.c
@@ -45,8 +45,8 @@ typedef struct {
static const stream2codec_t st2codec[256] = {
st(0x01, V, WORK_DECMPEG2, 0, "MPEG1"),
st(0x02, V, WORK_DECMPEG2, 0, "MPEG2"),
- st(0x03, A, HB_ACODEC_MPGA, CODEC_ID_MP2, "MPEG1"),
- st(0x04, A, HB_ACODEC_MPGA, CODEC_ID_MP2, "MPEG2"),
+ st(0x03, A, HB_ACODEC_FFMPEG, CODEC_ID_MP2, "MPEG1"),
+ st(0x04, A, HB_ACODEC_FFMPEG, CODEC_ID_MP2, "MPEG2"),
st(0x05, N, 0, 0, "ISO 13818-1 private section"),
st(0x06, U, 0, 0, "ISO 13818-1 PES private data"),
st(0x07, N, 0, 0, "ISO 13522 MHEG"),
@@ -57,23 +57,23 @@ static const stream2codec_t st2codec[256] = {
st(0x0c, N, 0, 0, "ISO 13818-6 Stream descriptors"),
st(0x0d, N, 0, 0, "ISO 13818-6 Sections"),
st(0x0e, N, 0, 0, "ISO 13818-1 auxiliary"),
- st(0x0f, A, HB_ACODEC_MPGA, CODEC_ID_AAC, "AAC"),
+ st(0x0f, A, HB_ACODEC_FFMPEG, CODEC_ID_AAC, "AAC"),
st(0x10, V, WORK_DECAVCODECV, CODEC_ID_MPEG4, "MPEG4"),
- st(0x11, A, HB_ACODEC_MPGA, CODEC_ID_AAC_LATM, "LATM AAC"),
+ st(0x11, A, HB_ACODEC_FFMPEG, CODEC_ID_AAC_LATM, "LATM AAC"),
st(0x12, U, 0, 0, "MPEG4 generic"),
st(0x14, N, 0, 0, "ISO 13818-6 DSM-CC download"),
st(0x1b, V, WORK_DECAVCODECV, CODEC_ID_H264, "H.264"),
- st(0x80, N, HB_ACODEC_MPGA, CODEC_ID_PCM_BLURAY, "Digicipher II Video"),
+ st(0x80, N, HB_ACODEC_FFMPEG, CODEC_ID_PCM_BLURAY, "Digicipher II Video"),
st(0x81, A, HB_ACODEC_AC3, 0, "AC3"),
st(0x82, A, HB_ACODEC_DCA, 0, "DTS"),
st(0x83, A, HB_ACODEC_LPCM, 0, "LPCM"),
st(0x84, U, 0, 0, "SDDS"),
st(0x85, U, 0, 0, "ATSC Program ID"),
st(0x86, A, HB_ACODEC_DCA, 0, "DTS-HD MA"),
- st(0x87, A, HB_ACODEC_MPGA, CODEC_ID_EAC3, "E-AC3"),
+ st(0x87, A, HB_ACODEC_FFMPEG, CODEC_ID_EAC3, "E-AC3"),
st(0x8a, A, HB_ACODEC_DCA, 0, "DTS"),
@@ -1705,7 +1705,8 @@ static void set_ts_audio_description( hb_audio_t *audio, iso639_lang_t *lang )
// For streams demuxed and decoded by ffmpeg, we have a cached context.
// Use it to get the name and profile information. Obtaining
// the profile requires that ffmpeg has already probed the stream.
- else if ( audio->config.in.codec == HB_ACODEC_FFMPEG &&
+ else if ( ( audio->config.in.codec & HB_ACODEC_FF_MASK ) &&
+ ( audio->config.in.codec & HB_ACODEC_FF_I_FLAG ) &&
( cc = hb_ffmpeg_context( audio->config.in.codec_param ) ) &&
avcodec_find_decoder( cc->codec_id ) )
{
@@ -1726,7 +1727,8 @@ static void set_ts_audio_description( hb_audio_t *audio, iso639_lang_t *lang )
}
// For streams demuxed by us and decoded by ffmpeg, we can lookup the
// decoder name.
- else if ( audio->config.in.codec == HB_ACODEC_MPGA &&
+ else if ( ( audio->config.in.codec & HB_ACODEC_FF_MASK ) &&
+ !( audio->config.in.codec & HB_ACODEC_FF_I_FLAG ) &&
avcodec_find_decoder( audio->config.in.codec_param ) )
{
codec_name = avcodec_find_decoder( audio->config.in.codec_param )->name;
@@ -1735,9 +1737,9 @@ static void set_ts_audio_description( hb_audio_t *audio, iso639_lang_t *lang )
{
codec_name = audio->config.in.codec == HB_ACODEC_AC3 ? "AC3" :
audio->config.in.codec == HB_ACODEC_DCA ? "DTS" :
- audio->config.in.codec == HB_ACODEC_MPGA ? "MPEG" :
+ audio->config.in.codec == HB_ACODEC_DCA_HD ? "DTS-HD" :
audio->config.in.codec == HB_ACODEC_LPCM ? "LPCM" :
- audio->config.in.codec == HB_ACODEC_FFMPEG ? "FFmpeg" :
+ (audio->config.in.codec & HB_ACODEC_FF_MASK) ? "FFmpeg" :
"Unknown";
}
snprintf( audio->config.lang.description,
@@ -1745,7 +1747,8 @@ static void set_ts_audio_description( hb_audio_t *audio, iso639_lang_t *lang )
strlen(lang->native_name) ? lang->native_name : lang->eng_name,
codec_name );
- if (audio->config.in.codec == HB_ACODEC_FFMPEG)
+ if ( ( audio->config.in.codec & HB_ACODEC_FF_MASK) &&
+ ( audio->config.in.codec & HB_ACODEC_FF_I_FLAG ) )
{
int layout = audio->config.in.channel_layout;
char *desc = audio->config.lang.description +
@@ -1776,7 +1779,8 @@ static void set_audio_description( hb_audio_t *audio, iso639_lang_t *lang )
// For streams demuxed and decoded by ffmpeg, we have a cached context.
// Use it to get the name and profile information. Obtaining
// the profile requires that ffmpeg has already probed the stream.
- if ( audio->config.in.codec == HB_ACODEC_FFMPEG &&
+ if ( ( audio->config.in.codec & HB_ACODEC_FF_MASK ) &&
+ ( audio->config.in.codec & HB_ACODEC_FF_I_FLAG ) &&
( cc = hb_ffmpeg_context( audio->config.in.codec_param ) ) &&
avcodec_find_decoder( cc->codec_id ) )
{
@@ -1792,7 +1796,8 @@ static void set_audio_description( hb_audio_t *audio, iso639_lang_t *lang )
}
// For streams demuxed by us and decoded by ffmpeg, we can lookup the
// decoder name.
- else if ( audio->config.in.codec == HB_ACODEC_MPGA &&
+ else if ( ( audio->config.in.codec & HB_ACODEC_FF_MASK ) &&
+ !( audio->config.in.codec & HB_ACODEC_FF_I_FLAG ) &&
avcodec_find_decoder( audio->config.in.codec_param ) )
{
codec_name = avcodec_find_decoder( audio->config.in.codec_param )->name;
@@ -1801,9 +1806,8 @@ static void set_audio_description( hb_audio_t *audio, iso639_lang_t *lang )
{
codec_name = audio->config.in.codec == HB_ACODEC_AC3 ? "AC3" :
audio->config.in.codec == HB_ACODEC_DCA ? "DTS" :
- audio->config.in.codec == HB_ACODEC_MPGA ? "MPEG" :
audio->config.in.codec == HB_ACODEC_LPCM ? "LPCM" :
- audio->config.in.codec == HB_ACODEC_FFMPEG ? "FFmpeg" :
+ (audio->config.in.codec & HB_ACODEC_FF_MASK) ? "FFmpeg" :
"Unknown";
}
snprintf( audio->config.lang.description,
@@ -1811,7 +1815,8 @@ static void set_audio_description( hb_audio_t *audio, iso639_lang_t *lang )
strlen(lang->native_name) ? lang->native_name : lang->eng_name,
codec_name );
- if (audio->config.in.codec == HB_ACODEC_FFMPEG)
+ if ( ( audio->config.in.codec & HB_ACODEC_FF_MASK) &&
+ ( audio->config.in.codec & HB_ACODEC_FF_I_FLAG ) )
{
int layout = audio->config.in.channel_layout;
char *desc = audio->config.lang.description +
@@ -1875,7 +1880,7 @@ static void hb_ts_stream_set_audio_list(
// To distinguish, Bluray streams have a reg_desc of HDMV
stream->ts[i].stream_kind = A;
add_audio(i, list_audio, stream, 0,
- HB_ACODEC_MPGA, CODEC_ID_PCM_BLURAY );
+ HB_ACODEC_FFMPEG, CODEC_ID_PCM_BLURAY );
continue;
}
@@ -1896,7 +1901,7 @@ static void hb_ts_stream_set_audio_list(
HB_ACODEC_AC3, 0 );
stream->ts[i].substream_type[1] = HB_SUBSTREAM_BD_TRUEHD;
add_audio(i, list_audio, stream, HB_SUBSTREAM_BD_TRUEHD,
- HB_ACODEC_MPGA, CODEC_ID_TRUEHD );
+ HB_ACODEC_FFMPEG, CODEC_ID_TRUEHD );
stream->ts[i].number_substreams = 2;
continue;
}
@@ -1910,7 +1915,7 @@ static void hb_ts_stream_set_audio_list(
add_audio(i, list_audio, stream, HB_SUBSTREAM_BD_DTS,
HB_ACODEC_DCA, 0 );
stream->ts[i].substream_type[1] = 0;
- add_audio(i, list_audio, stream, 0, HB_ACODEC_MPGA, CODEC_ID_DTS );
+ add_audio(i, list_audio, stream, 0, HB_ACODEC_DCA_HD, CODEC_ID_DTS );
stream->ts[i].number_substreams = 2;
continue;
}
@@ -1927,7 +1932,7 @@ static void hb_ts_stream_set_audio_list(
add_audio(i, list_audio, stream, HB_SUBSTREAM_BD_DTS,
HB_ACODEC_DCA, 0 );
stream->ts[i].substream_type[1] = 0;
- add_audio(i, list_audio, stream, 0, HB_ACODEC_MPGA, CODEC_ID_DTS );
+ add_audio(i, list_audio, stream, 0, HB_ACODEC_DCA_HD, CODEC_ID_DTS );
stream->ts[i].number_substreams = 2;
continue;
}
@@ -1938,7 +1943,7 @@ static void hb_ts_stream_set_audio_list(
// which conflicts with SDDS
// To distinguish, Bluray streams have a reg_desc of HDMV
stream->ts[i].stream_kind = A;
- add_audio(i, list_audio, stream, 0, HB_ACODEC_MPGA, CODEC_ID_EAC3 );
+ add_audio(i, list_audio, stream, 0, HB_ACODEC_FFMPEG, CODEC_ID_EAC3 );
continue;
}
@@ -1982,7 +1987,7 @@ static void hb_ts_stream_set_audio_list(
// 0xC0 - 0xCF are the normal containers for ISO-standard
// media (mpeg2 audio and mpeg4 AAC).
add_audio(i, list_audio, stream, 0,
- HB_ACODEC_MPGA, CODEC_ID_MP2 );
+ HB_ACODEC_FFMPEG, CODEC_ID_MP2 );
}
else
{
@@ -2010,7 +2015,7 @@ static void add_audio_to_title(hb_title_t *title, int id)
switch ( id >> 12 )
{
case 0x0:
- audio->config.in.codec = HB_ACODEC_MPGA;
+ audio->config.in.codec = HB_ACODEC_FFMPEG;
hb_log("add_audio_to_title: added MPEG audio stream 0x%x", id);
break;
case 0x2:
@@ -3189,7 +3194,7 @@ static void ffmpeg_remap_stream( hb_stream_t *stream, hb_title_t *title )
hb_audio_t *audio;
for ( i = 0; ( audio = hb_list_item( title->list_audio, i ) ); ++i )
{
- if ( audio->config.in.codec == HB_ACODEC_FFMPEG )
+ if ( audio->config.in.codec & HB_ACODEC_FF_MASK )
{
ffmpeg_add_codec( stream,
audio->config.in.codec_param >> ffmpeg_sl_bits );
@@ -3330,7 +3335,16 @@ static void add_ffmpeg_audio( hb_title_t *title, hb_stream_t *stream, int id )
}
else
{
- audio->config.in.codec = HB_ACODEC_FFMPEG;
+ if ( codec->codec_id == CODEC_ID_DTS &&
+ ( codec->profile == FF_PROFILE_DTS_HD_MA ||
+ codec->profile == FF_PROFILE_DTS_HD_HRA ) )
+ {
+ audio->config.in.codec = HB_ACODEC_DCA_HD | HB_ACODEC_FF_I_FLAG;
+ }
+ else
+ {
+ audio->config.in.codec = HB_ACODEC_FFMPEG | HB_ACODEC_FF_I_FLAG;
+ }
audio->config.in.codec_param = ffmpeg_codec_param( stream, id );
audio->config.in.bitrate = codec->bit_rate? codec->bit_rate : 1;
diff --git a/libhb/sync.c b/libhb/sync.c
index 09e88538c..b754a54cc 100644
--- a/libhb/sync.c
+++ b/libhb/sync.c
@@ -1033,7 +1033,8 @@ static int syncAudioWork( hb_work_object_t * w, hb_buffer_t ** buf_in,
* Or in the case of DCA, skip some frames from the
* other streams.
*/
- if( w->audio->config.out.codec == HB_ACODEC_DCA_PASS )
+ if( w->audio->config.out.codec == HB_ACODEC_DCA_PASS ||
+ w->audio->config.out.codec == HB_ACODEC_DCA_HD_PASS )
{
hb_log( "sync: audio gap %d ms. Skipping frames. Audio 0x%x"
" start %"PRId64", next %"PRId64,
@@ -1093,8 +1094,7 @@ static void InitAudio( hb_job_t * job, hb_sync_common_t * common, int i )
w->audio = hb_list_item( title->list_audio, i );
w->fifo_in = w->audio->priv.fifo_raw;
- if( w->audio->config.out.codec == HB_ACODEC_AC3_PASS ||
- w->audio->config.out.codec == HB_ACODEC_DCA_PASS )
+ if ( w->audio->config.out.codec & HB_ACODEC_PASS_FLAG )
{
w->fifo_out = w->audio->priv.fifo_out;
}
diff --git a/libhb/work.c b/libhb/work.c
index f0a81fa75..90209dbd2 100644
--- a/libhb/work.c
+++ b/libhb/work.c
@@ -112,9 +112,16 @@ hb_work_object_t * hb_codec_decoder( int codec )
{
case HB_ACODEC_AC3: return hb_get_work( WORK_DECA52 );
case HB_ACODEC_DCA: return hb_get_work( WORK_DECDCA );
- case HB_ACODEC_MPGA: return hb_get_work( WORK_DECAVCODEC );
case HB_ACODEC_LPCM: return hb_get_work( WORK_DECLPCM );
- case HB_ACODEC_FFMPEG: return hb_get_work( WORK_DECAVCODECAI );
+ default:
+ if ( codec & HB_ACODEC_FF_MASK )
+ {
+ if ( codec & HB_ACODEC_FF_I_FLAG )
+ return hb_get_work( WORK_DECAVCODECAI );
+ else
+ return hb_get_work( WORK_DECAVCODEC );
+ }
+ break;
}
return NULL;
}
@@ -351,7 +358,7 @@ void hb_display_job_info( hb_job_t * job )
hb_log( " + bitrate: %d kbps, samplerate: %d Hz", audio->config.in.bitrate / 1000, audio->config.in.samplerate );
}
- if( (audio->config.out.codec != HB_ACODEC_AC3_PASS) && (audio->config.out.codec != HB_ACODEC_DCA_PASS) )
+ if( !(audio->config.out.codec & HB_ACODEC_PASS_FLAG) )
{
for (j = 0; j < hb_audio_mixdowns_count; j++)
{
@@ -371,10 +378,12 @@ void hb_display_job_info( hb_job_t * job )
hb_log(" + dynamic range compression: %f", audio->config.out.dynamic_range_compression);
}
- if( (audio->config.out.codec == HB_ACODEC_AC3_PASS) || (audio->config.out.codec == HB_ACODEC_DCA_PASS) )
+ if( audio->config.out.codec & HB_ACODEC_PASS_FLAG )
{
- hb_log( " + %s passthrough", (audio->config.out.codec == HB_ACODEC_AC3_PASS) ?
- "AC3" : "DCA" );
+ hb_log( " + %s passthrough",
+ (audio->config.out.codec == HB_ACODEC_AC3_PASS) ? "AC3" :
+ (audio->config.out.codec == HB_ACODEC_DCA_PASS) ? "DTS" :
+ "DTS-HD");
}
else
{
@@ -416,7 +425,7 @@ static int check_ff_audio( hb_list_t *list_audio, hb_audio_t *ff_audio )
if ( audio == ff_audio )
break;
- if ( audio->config.in.codec == HB_ACODEC_FFMPEG &&
+ if ( ( audio->config.in.codec & HB_ACODEC_FF_MASK ) &&
audio->id == ff_audio->id )
{
hb_list_add( audio->priv.ff_audio_list, ff_audio );
@@ -547,8 +556,9 @@ static void do_job( hb_job_t * job )
for( i = 0; i < hb_list_count( title->list_audio ); )
{
audio = hb_list_item( title->list_audio, i );
- if( ( ( audio->config.out.codec == HB_ACODEC_AC3_PASS ) && ( audio->config.in.codec != HB_ACODEC_AC3 ) ) ||
- ( ( audio->config.out.codec == HB_ACODEC_DCA_PASS ) && ( audio->config.in.codec != HB_ACODEC_DCA ) ) )
+ if( ( audio->config.out.codec & HB_ACODEC_PASS_FLAG ) &&
+ !( audio->config.in.codec & audio->config.out.codec &
+ HB_ACODEC_PASS_MASK ) )
{
hb_log( "Passthru requested and input codec is not the same as output codec for track %d",
audio->config.out.track );
@@ -556,8 +566,7 @@ static void do_job( hb_job_t * job )
free( audio );
continue;
}
- if( audio->config.out.codec != HB_ACODEC_AC3_PASS &&
- audio->config.out.codec != HB_ACODEC_DCA_PASS &&
+ if( !(audio->config.out.codec & HB_ACODEC_PASS_FLAG) &&
audio->config.out.samplerate > 48000 )
{
hb_log( "Sample rate %d not supported. Down-sampling to 48kHz.",
@@ -712,7 +721,7 @@ static void do_job( hb_job_t * job )
audio->priv.fifo_out = hb_fifo_init( FIFO_LARGE, FIFO_LARGE_WAKE );
audio->priv.ff_audio_list = hb_list_init();
- if ( audio->config.in.codec == HB_ACODEC_FFMPEG )
+ if ( audio->config.in.codec & HB_ACODEC_FF_MASK )
{
if ( !check_ff_audio( title->list_audio, audio ) )
{
@@ -939,8 +948,7 @@ static void do_job( hb_job_t * job )
/*
* Audio Encoder Thread
*/
- if( audio->config.out.codec != HB_ACODEC_AC3_PASS &&
- audio->config.out.codec != HB_ACODEC_DCA_PASS )
+ if ( !(audio->config.out.codec & HB_ACODEC_PASS_FLAG ) )
{
/*
* Add the encoder thread if not doing AC-3 pass through