summaryrefslogtreecommitdiffstats
diff options
context:
space:
mode:
Diffstat
-rw-r--r--contrib/ffmpeg/module.defs4
-rw-r--r--contrib/libopus/module.defs11
-rw-r--r--contrib/libopus/module.rules2
-rw-r--r--gtk/configure.ac2
-rw-r--r--gtk/src/hb-backend.c15
-rw-r--r--libhb/common.c206
-rw-r--r--libhb/common.h13
-rw-r--r--libhb/encavcodecaudio.c4
-rw-r--r--libhb/module.defs4
-rw-r--r--libhb/muxavformat.c17
-rw-r--r--libhb/work.c5
-rw-r--r--make/include/main.defs1
-rw-r--r--test/module.defs2
13 files changed, 207 insertions, 79 deletions
diff --git a/contrib/ffmpeg/module.defs b/contrib/ffmpeg/module.defs
index dfd6ec9b1..655df0651 100644
--- a/contrib/ffmpeg/module.defs
+++ b/contrib/ffmpeg/module.defs
@@ -1,4 +1,4 @@
-__deps__ := YASM BZIP2 ZLIB FDKAAC LIBVPX LAME
+__deps__ := YASM BZIP2 ZLIB FDKAAC LIBVPX LAME LIBOPUS
ifeq (1-mingw,$(BUILD.cross)-$(BUILD.system))
__deps__ += PTHREADW32
endif
@@ -43,6 +43,8 @@ FFMPEG.CONFIGURE.extra = \
--enable-encoder=mpeg2video \
--enable-encoder=mpeg4 \
--enable-encoder=libmp3lame \
+ --enable-libopus \
+ --enable-encoder=libopus \
--enable-libvpx \
--enable-encoder=libvpx_vp8 \
--disable-decoder=libvpx_vp8 \
diff --git a/contrib/libopus/module.defs b/contrib/libopus/module.defs
new file mode 100644
index 000000000..7292f70bd
--- /dev/null
+++ b/contrib/libopus/module.defs
@@ -0,0 +1,11 @@
+$(eval $(call import.MODULE.defs,LIBOPUS,libopus))
+$(eval $(call import.CONTRIB.defs,LIBOPUS))
+
+LIBOPUS.FETCH.url = http://download.handbrake.fr/contrib/opus-1.1.3.tar.gz
+LIBOPUS.FETCH.url += http://downloads.xiph.org/releases/opus/opus-1.1.3.tar.gz
+LIBOPUS.FETCH.md5 = 32bbb6b557fe1b6066adc0ae1f08b629
+
+LIBOPUS.CONFIGURE.shared = --enable-shared=no
+LIBOPUS.CONFIGURE.extra = --disable-doc --disable-extra-programs
+
+LIBOPUS.CONFIGURE.bootstrap = rm -fr aclocal.m4 autom4te.cache configure; autoreconf -I m4 -fiv;
diff --git a/contrib/libopus/module.rules b/contrib/libopus/module.rules
new file mode 100644
index 000000000..3493eaf8f
--- /dev/null
+++ b/contrib/libopus/module.rules
@@ -0,0 +1,2 @@
+$(eval $(call import.MODULE.rules,LIBOPUS))
+$(eval $(call import.CONTRIB.rules,LIBOPUS))
diff --git a/gtk/configure.ac b/gtk/configure.ac
index 3b690b21f..4c75f4a90 100644
--- a/gtk/configure.ac
+++ b/gtk/configure.ac
@@ -164,7 +164,7 @@ PKG_CHECK_MODULES(GHB, [$GHB_PACKAGES])
GHB_CFLAGS="$HBINC $GHB_CFLAGS"
-HB_LIBS="-lhandbrake -lavresample -lavformat -lavcodec -lavfilter -lavutil -ldvdnav -ldvdread -lmp3lame -lvorbis -lvorbisenc -logg -lsamplerate -lx264 -lswscale -ltheoraenc -ltheoradec -lvpx -lz -lbz2 -lbluray -lass -lfontconfig -lfreetype -lxml2 -ljansson"
+HB_LIBS="-lhandbrake -lavresample -lavformat -lavcodec -lavfilter -lavutil -ldvdnav -ldvdread -lmp3lame -lvorbis -lvorbisenc -logg -lsamplerate -lx264 -lswscale -ltheoraenc -ltheoradec -lvpx -lz -lbz2 -lbluray -lass -lfontconfig -lfreetype -lxml2 -ljansson -lopus"
case $host in
*-*-mingw*)
diff --git a/gtk/src/hb-backend.c b/gtk/src/hb-backend.c
index 5feababa5..1df18e703 100644
--- a/gtk/src/hb-backend.c
+++ b/gtk/src/hb-backend.c
@@ -1011,20 +1011,7 @@ lookup_param_option(const hb_filter_param_t *param, const GhbValue *gval)
gint
ghb_find_closest_audio_samplerate(gint irate)
{
- gint result;
- const hb_rate_t *rate;
-
- result = 0;
- for (rate = hb_audio_samplerate_get_next(NULL); rate != NULL;
- rate = hb_audio_samplerate_get_next(rate))
- {
- if (irate <= rate->rate)
- {
- result = rate->rate;
- break;
- }
- }
- return result;
+ return hb_audio_samplerate_find_closest(irate, HB_ACODEC_INVALID);
}
const iso639_lang_t* ghb_iso639_lookup_by_int(int idx)
diff --git a/libhb/common.c b/libhb/common.c
index c75e4e564..50b596a98 100644
--- a/libhb/common.c
+++ b/libhb/common.c
@@ -31,6 +31,8 @@
#include <windows.h>
#endif
+static int mixdown_get_opus_coupled_stream_count(int mixdown);
+
/**********************************************************************
* Global variables
*********************************************************************/
@@ -63,6 +65,7 @@ enum
HB_GID_ACODEC_MP3_PASS,
HB_GID_ACODEC_TRUEHD_PASS,
HB_GID_ACODEC_VORBIS,
+ HB_GID_ACODEC_OPUS,
HB_GID_MUX_MKV,
HB_GID_MUX_MP4,
};
@@ -324,6 +327,7 @@ hb_encoder_internal_t hb_audio_encoders[] =
{ { "FLAC 16-bit", "flac16", "FLAC 16-bit (libavcodec)", HB_ACODEC_FFFLAC, HB_MUX_MASK_MKV, }, NULL, 1, HB_GID_ACODEC_FLAC, },
{ { "FLAC 24-bit", "flac24", "FLAC 24-bit (libavcodec)", HB_ACODEC_FFFLAC24, HB_MUX_MASK_MKV, }, NULL, 1, HB_GID_ACODEC_FLAC, },
{ { "FLAC Passthru", "copy:flac", "FLAC Passthru", HB_ACODEC_FLAC_PASS, HB_MUX_MASK_MKV, }, NULL, 1, HB_GID_ACODEC_FLAC_PASS, },
+ { { "Opus", "opus", "Opus (libopus)", HB_ACODEC_OPUS, HB_MUX_MASK_MKV, }, NULL, 1, HB_GID_ACODEC_OPUS, },
{ { "Auto Passthru", "copy", "Auto Passthru", HB_ACODEC_AUTO_PASS, HB_MUX_MASK_MP4|HB_MUX_MASK_MKV, }, NULL, 1, HB_GID_ACODEC_AUTO_PASS, },
};
int hb_audio_encoders_count = sizeof(hb_audio_encoders) / sizeof(hb_audio_encoders[0]);
@@ -361,6 +365,9 @@ static int hb_audio_encoder_is_enabled(int encoder)
case HB_ACODEC_FFFLAC24:
return avcodec_find_encoder(AV_CODEC_ID_FLAC) != NULL;
+ case HB_ACODEC_OPUS:
+ return avcodec_find_encoder(AV_CODEC_ID_OPUS) != NULL;
+
// the following encoders are always enabled
case HB_ACODEC_LAME:
case HB_ACODEC_VORBIS:
@@ -742,56 +749,56 @@ int hb_video_framerate_get_close(hb_rational_t *framerate, double thresh)
return result;
}
-int hb_audio_samplerate_get_best(uint32_t codec, int samplerate, int *sr_shift)
+int hb_audio_samplerate_is_supported(int samplerate, uint32_t codec)
{
- int best_samplerate;
- if (samplerate < 32000 && (codec == HB_ACODEC_AC3 ||
- codec == HB_ACODEC_FFEAC3 ||
- codec == HB_ACODEC_CA_HAAC))
- {
- // ca_haac can't do samplerates < 32 kHz
- // libav's E-AC-3 encoder can't do samplerates < 32 kHz
- // AC-3 < 32 kHz suffers from poor hardware compatibility
- best_samplerate = 32000;
- }
- else if (samplerate < 16000 && codec == HB_ACODEC_FDK_HAAC)
- {
- // fdk_haac can't do samplerates < 16 kHz
- best_samplerate = 16000;
- }
- else
+ switch (codec)
{
- best_samplerate = hb_audio_rates_first_item->rate;
- const hb_rate_t *audio_samplerate = NULL;
- while ((audio_samplerate = hb_audio_samplerate_get_next(audio_samplerate)) != NULL)
- {
- if (samplerate == audio_samplerate->rate)
- {
- // valid samplerate
- best_samplerate = audio_samplerate->rate;
- break;
- }
- if (samplerate > audio_samplerate->rate)
+ case HB_ACODEC_AC3:
+ case HB_ACODEC_FFEAC3:
+ case HB_ACODEC_CA_HAAC:
+ // ca_haac can't do samplerates < 32 kHz
+ // libav's E-AC-3 encoder can't do samplerates < 32 kHz
+ // AC-3 < 32 kHz suffers from poor hardware compatibility
+ if (samplerate < 32000)
+ return 0;
+ else
+ return 1;
+ case HB_ACODEC_FDK_HAAC:
+ // fdk_haac can't do samplerates < 16 kHz
+ if (samplerate < 16000)
+ return 0;
+ else
+ return 1;
+ case HB_ACODEC_OPUS:
+ switch (samplerate)
{
- // samplerates are sanitized downwards
- best_samplerate = audio_samplerate->rate;
+ // Opus only supports samplerates 8kHz, 12kHz, 16kHz,
+ // 24kHz, 48kHz
+ case 8000:
+ case 12000:
+ case 16000:
+ case 24000:
+ case 48000:
+ return 1;
+ default:
+ return 0;
}
- }
- }
- if (sr_shift != NULL)
- {
- /* sr_shift: 0 -> 48000, 44100, 32000 Hz
- * 1 -> 24000, 22050, 16000 Hz
- * 2 -> 12000, 11025, 8000 Hz
- *
- * also, since samplerates are sanitized downwards:
- *
- * (samplerate < 32000) implies (samplerate <= 24000)
- */
- *sr_shift = ((best_samplerate < 16000) ? 2 :
- (best_samplerate < 32000) ? 1 : 0);
+ default:
+ return 1;
}
- return best_samplerate;
+}
+
+int hb_audio_samplerate_get_sr_shift(int samplerate)
+{
+ /* sr_shift: 0 -> 48000, 44100, 32000 Hz
+ * 1 -> 24000, 22050, 16000 Hz
+ * 2 -> 12000, 11025, 8000 Hz
+ *
+ * also, since samplerates are sanitized downwards:
+ *
+ * (samplerate < 32000) implies (samplerate <= 24000)
+ */
+ return ((samplerate < 16000) ? 2 : (samplerate < 32000) ? 1 : 0);
}
int hb_audio_samplerate_get_from_name(const char *name)
@@ -813,7 +820,7 @@ int hb_audio_samplerate_get_from_name(const char *name)
if (i >= hb_audio_rates_first_item->rate &&
i <= hb_audio_rates_last_item ->rate)
{
- return hb_audio_samplerate_get_best(0, i, NULL);
+ return hb_audio_samplerate_find_closest(i, HB_ACODEC_INVALID);
}
fail:
@@ -848,6 +855,44 @@ const hb_rate_t* hb_audio_samplerate_get_next(const hb_rate_t *last)
return ((hb_rate_internal_t*)last)->next;
}
+const hb_rate_t* hb_audio_samplerate_get_next_for_codec(const hb_rate_t *last,
+ uint32_t codec)
+{
+ while ((last = hb_audio_samplerate_get_next(last)) != NULL)
+ if (hb_audio_samplerate_is_supported(last->rate, codec))
+ return last;
+
+ // None found or end of list
+ return NULL;
+}
+
+int hb_audio_samplerate_find_closest(int samplerate, uint32_t codec)
+{
+ const hb_rate_t * rate, * prev, * next;
+
+ rate = prev = next = hb_audio_samplerate_get_next_for_codec(NULL, codec);
+ while (rate != NULL && next->rate < samplerate)
+ {
+ rate = hb_audio_samplerate_get_next_for_codec(rate, codec);
+ if (rate != NULL)
+ {
+ prev = next;
+ next = rate;
+ }
+ }
+
+ int delta_prev = samplerate - prev->rate;
+ int delta_next = next->rate - samplerate;
+ if (delta_prev <= delta_next)
+ {
+ return prev->rate;
+ }
+ else
+ {
+ return next->rate;
+ }
+}
+
// Given an input bitrate, find closest match in the set of allowed bitrates
static int hb_audio_bitrate_find_closest(int bitrate)
{
@@ -894,11 +939,11 @@ int hb_audio_bitrate_get_default(uint32_t codec, int samplerate, int mixdown)
if ((codec & HB_ACODEC_PASS_FLAG) || !(codec & HB_ACODEC_MASK))
goto fail;
- int bitrate, nchannels, sr_shift;
+ int bitrate, nchannels, nlfe, sr_shift;
/* full-bandwidth channels, sr_shift */
- nchannels = (hb_mixdown_get_discrete_channel_count(mixdown) -
- hb_mixdown_get_low_freq_channel_count(mixdown));
- hb_audio_samplerate_get_best(codec, samplerate, &sr_shift);
+ nlfe = hb_mixdown_get_low_freq_channel_count(mixdown);
+ nchannels = hb_mixdown_get_discrete_channel_count(mixdown) - nlfe;
+ sr_shift = hb_audio_samplerate_get_sr_shift(samplerate);
switch (codec)
{
@@ -924,6 +969,14 @@ int hb_audio_bitrate_get_default(uint32_t codec, int samplerate, int mixdown)
bitrate = nchannels * 32;
break;
+ case HB_ACODEC_OPUS:
+ {
+ int coupled = mixdown_get_opus_coupled_stream_count(mixdown);
+ int uncoupled = nchannels + nlfe - 2 * coupled;
+
+ bitrate = coupled * 96 + uncoupled * 64;
+ } break;
+
default:
bitrate = nchannels * 80;
break;
@@ -1070,7 +1123,8 @@ void hb_audio_bitrate_get_limits(uint32_t codec, int samplerate, int mixdown,
/* samplerate, sr_shift */
int sr_shift;
- samplerate = hb_audio_samplerate_get_best(codec, samplerate, &sr_shift);
+ samplerate = hb_audio_samplerate_find_closest(samplerate, codec);
+ sr_shift = hb_audio_samplerate_get_sr_shift(samplerate);
/* LFE, full-bandwidth channels */
int lfe_count, nchannels;
@@ -1174,6 +1228,11 @@ void hb_audio_bitrate_get_limits(uint32_t codec, int samplerate, int mixdown,
( 50 * (samplerate >= 44100)));
break;
+ case HB_ACODEC_OPUS:
+ *low = (nchannels + lfe_count) * 6;
+ *high = (nchannels + lfe_count) * 256;
+ break;
+
// Bitrates don't apply to passthrough audio, but may apply if we
// fall back to an encoder when the source can't be passed through.
default:
@@ -1473,6 +1532,13 @@ void hb_audio_quality_get_limits(uint32_t codec, float *low, float *high,
*high = 127.;
break;
+ case HB_ACODEC_OPUS:
+ *direction = 0;
+ *granularity = 1.;
+ *low = 0.;
+ *high = 10.;
+ break;
+
default:
*direction = 0;
*granularity = 1.;
@@ -1513,6 +1579,9 @@ float hb_audio_quality_get_default(uint32_t codec)
case HB_ACODEC_CA_AAC:
return 91.;
+ case HB_ACODEC_OPUS:
+ return 10.;
+
default:
return HB_INVALID_AUDIO_QUALITY;
}
@@ -1658,6 +1727,34 @@ const hb_dither_t* hb_audio_dither_get_next(const hb_dither_t *last)
return ((hb_dither_internal_t*)last)->next;
}
+static int mixdown_get_opus_coupled_stream_count(int mixdown)
+{
+ switch (mixdown)
+ {
+ case HB_AMIXDOWN_7POINT1:
+ case HB_AMIXDOWN_6POINT1:
+ return 3;
+
+ case HB_AMIXDOWN_5POINT1:
+ return 2;
+
+ case HB_AMIXDOWN_MONO:
+ case HB_AMIXDOWN_LEFT:
+ case HB_AMIXDOWN_RIGHT:
+ return 0;
+
+ case HB_AMIXDOWN_NONE:
+ case HB_INVALID_AMIXDOWN:
+ case HB_AMIXDOWN_5_2_LFE:
+ // The 5F/2R/LFE configuration is currently not supported by Opus,
+ // so don't set coupled streams.
+ return 0;
+
+ default:
+ return 1;
+ }
+}
+
int hb_mixdown_is_supported(int mixdown, uint32_t codec, uint64_t layout)
{
return (hb_mixdown_has_codec_support(mixdown, codec) &&
@@ -1679,6 +1776,7 @@ int hb_mixdown_has_codec_support(int mixdown, uint32_t codec)
case HB_ACODEC_VORBIS:
case HB_ACODEC_FFFLAC:
case HB_ACODEC_FFFLAC24:
+ case HB_ACODEC_OPUS:
return (mixdown <= HB_AMIXDOWN_7POINT1);
case HB_ACODEC_LAME:
@@ -1831,6 +1929,7 @@ int hb_mixdown_get_default(uint32_t codec, uint64_t layout)
// the FLAC encoder defaults to the best mixdown up to 7.1
case HB_ACODEC_FFFLAC:
case HB_ACODEC_FFFLAC24:
+ case HB_ACODEC_OPUS:
mixdown = HB_AMIXDOWN_7POINT1;
break;
@@ -2289,9 +2388,8 @@ void hb_autopassthru_apply_settings(hb_job_t *job)
if (audio->config.out.samplerate <= 0)
audio->config.out.samplerate = audio->config.in.samplerate;
audio->config.out.samplerate =
- hb_audio_samplerate_get_best(audio->config.out.codec,
- audio->config.out.samplerate,
- NULL);
+ hb_audio_samplerate_find_closest(
+ audio->config.out.samplerate, audio->config.out.codec);
int quality_not_allowed =
hb_audio_quality_get_default(audio->config.out.codec)
== HB_INVALID_AUDIO_QUALITY;
diff --git a/libhb/common.h b/libhb/common.h
index 92cc1dadf..ae05f2726 100644
--- a/libhb/common.h
+++ b/libhb/common.h
@@ -341,10 +341,16 @@ const hb_rate_t* hb_video_framerate_get_next(const hb_rate_t *last);
int hb_video_framerate_get_close(hb_rational_t *framerate,
double thresh);
-int hb_audio_samplerate_get_best(uint32_t codec, int samplerate, int *sr_shift);
+int hb_audio_samplerate_is_supported(int samplerate,
+ uint32_t codec);
+int hb_audio_samplerate_find_closest(int samplerate,
+ uint32_t codec);
+int hb_audio_samplerate_get_sr_shift(int samplerate);
int hb_audio_samplerate_get_from_name(const char *name);
const char* hb_audio_samplerate_get_name(int samplerate);
const hb_rate_t* hb_audio_samplerate_get_next(const hb_rate_t *last);
+const hb_rate_t* hb_audio_samplerate_get_next_for_codec(const hb_rate_t *last,
+ uint32_t codec);
int hb_audio_bitrate_get_best(uint32_t codec, int bitrate, int samplerate, int mixdown);
int hb_audio_bitrate_get_default(uint32_t codec, int samplerate, int mixdown);
@@ -659,7 +665,7 @@ struct hb_job_s
/* Audio starts here */
/* Audio Codecs: Update win/CS/HandBrake.Interop/HandBrakeInterop/HbLib/NativeConstants.cs when changing these consts */
#define HB_ACODEC_INVALID 0x00000000
-#define HB_ACODEC_MASK 0x03FFFF00
+#define HB_ACODEC_MASK 0x07FFFF00
#define HB_ACODEC_LAME 0x00000200
#define HB_ACODEC_VORBIS 0x00000400
#define HB_ACODEC_AC3 0x00000800
@@ -677,7 +683,8 @@ struct hb_job_s
#define HB_ACODEC_FDK_HAAC 0x00800000
#define HB_ACODEC_FFEAC3 0x01000000
#define HB_ACODEC_FFTRUEHD 0x02000000
-#define HB_ACODEC_FF_MASK 0x03FF2800
+#define HB_ACODEC_OPUS 0x04000000
+#define HB_ACODEC_FF_MASK 0x07FF2800
#define HB_ACODEC_PASS_FLAG 0x40000000
#define HB_ACODEC_PASS_MASK (HB_ACODEC_AC3 | HB_ACODEC_DCA | HB_ACODEC_DCA_HD | HB_ACODEC_FFAAC | HB_ACODEC_FFEAC3 | HB_ACODEC_FFFLAC | HB_ACODEC_MP3 | HB_ACODEC_FFTRUEHD)
#define HB_ACODEC_AUTO_PASS (HB_ACODEC_PASS_FLAG | HB_ACODEC_PASS_MASK)
diff --git a/libhb/encavcodecaudio.c b/libhb/encavcodecaudio.c
index 544ee385a..73d44a25c 100644
--- a/libhb/encavcodecaudio.c
+++ b/libhb/encavcodecaudio.c
@@ -125,6 +125,10 @@ static int encavcodecaInit(hb_work_object_t *w, hb_job_t *job)
codec_name = "libmp3lame";
break;
+ case HB_ACODEC_OPUS:
+ codec_name = "libopus";
+ break;
+
default:
hb_error("encavcodecaInit: unsupported codec (0x%x)",
audio->config.out.codec);
diff --git a/libhb/module.defs b/libhb/module.defs
index ca80e56fa..8ea828bfa 100644
--- a/libhb/module.defs
+++ b/libhb/module.defs
@@ -1,7 +1,7 @@
__deps__ := A52DEC BZIP2 LIBVPX FFMPEG FONTCONFIG FREETYPE LAME LIBASS LIBDCA \
LIBDVDREAD LIBDVDNAV LIBICONV LIBSAMPLERATE LIBTHEORA LIBVORBIS LIBOGG \
LIBXML2 PTHREADW32 X264 X265 ZLIB LIBBLURAY FDKAAC LIBMFX LIBGNURX JANSSON \
- HARFBUZZ
+ HARFBUZZ LIBOPUS
$(eval $(call import.MODULE.defs,LIBHB,libhb,$(__deps__)))
$(eval $(call import.GCC,LIBHB))
@@ -124,7 +124,7 @@ LIBHB.lib = $(LIBHB.build/)hb.lib
LIBHB.dll.libs = $(foreach n, \
ass avcodec avformat avfilter avutil avresample dvdnav dvdread fontconfig \
freetype mp3lame samplerate swscale vpx theora vorbis vorbisenc ogg \
- x264 xml2 bluray jansson harfbuzz, \
+ x264 xml2 bluray jansson harfbuzz opus, \
$(CONTRIB.build/)lib/lib$(n).a )
ifeq (1,$(FEATURE.fdk_aac))
diff --git a/libhb/muxavformat.c b/libhb/muxavformat.c
index b7533d4dc..47762721c 100644
--- a/libhb/muxavformat.c
+++ b/libhb/muxavformat.c
@@ -485,6 +485,23 @@ static int avformatInit( hb_mux_object_t * m )
size += ogg_headers[jj]->bytes;
}
} break;
+ case HB_ACODEC_OPUS:
+ track->st->codec->codec_id = AV_CODEC_ID_OPUS;
+
+ if (audio->priv.config.extradata.length)
+ {
+ priv_size = audio->priv.config.extradata.length;
+ priv_data = av_malloc(priv_size + FF_INPUT_BUFFER_PADDING_SIZE);
+ if (priv_data == NULL)
+ {
+ hb_error("OPUS extradata: malloc failure");
+ goto error;
+ }
+ memcpy(priv_data,
+ audio->priv.config.extradata.bytes,
+ audio->priv.config.extradata.length);
+ }
+ break;
case HB_ACODEC_FFFLAC:
case HB_ACODEC_FFFLAC24:
track->st->codec->codec_id = AV_CODEC_ID_FLAC;
diff --git a/libhb/work.c b/libhb/work.c
index afb866394..a2290cc8b 100644
--- a/libhb/work.c
+++ b/libhb/work.c
@@ -967,9 +967,8 @@ static int sanitize_audio(hb_job_t *job)
audio->config.out.samplerate = audio->config.in.samplerate;
}
best_samplerate =
- hb_audio_samplerate_get_best(audio->config.out.codec,
- audio->config.out.samplerate,
- NULL);
+ hb_audio_samplerate_find_closest(audio->config.out.samplerate,
+ audio->config.out.codec);
if (best_samplerate != audio->config.out.samplerate)
{
hb_log("work: sanitizing track %d unsupported samplerate %d Hz to %s kHz",
diff --git a/make/include/main.defs b/make/include/main.defs
index b95227908..e7f3a0980 100644
--- a/make/include/main.defs
+++ b/make/include/main.defs
@@ -50,6 +50,7 @@ ifneq (,$(filter $(BUILD.system),darwin cygwin mingw))
MODULES += contrib/libass
MODULES += contrib/libogg
MODULES += contrib/libvorbis
+ MODULES += contrib/libopus
MODULES += contrib/libtheora
MODULES += contrib/libsamplerate
MODULES += contrib/lame
diff --git a/test/module.defs b/test/module.defs
index 904a2e0f5..3750d6efd 100644
--- a/test/module.defs
+++ b/test/module.defs
@@ -17,7 +17,7 @@ TEST.GCC.l = \
ass avresample avformat avcodec avfilter avutil mp3lame dvdnav \
dvdread fontconfig fribidi \
samplerate swscale vpx theoraenc theoradec vorbis vorbisenc ogg x264 \
- bluray freetype xml2 bz2 z jansson harfbuzz
+ bluray freetype xml2 bz2 z jansson harfbuzz opus
ifeq (1,$(FEATURE.qsv))
TEST.GCC.D += USE_QSV HAVE_THREADS=1
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-16 18:58:54 +0000