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-rw-r--r--contrib/libsamplerate/module.defs18
-rw-r--r--contrib/libsamplerate/module.rules2
-rw-r--r--gtk/configure.ac2
-rw-r--r--libhb/audio_resample.c7
-rw-r--r--libhb/module.defs2
-rw-r--r--libhb/sync.c104
-rw-r--r--make/include/main.defs1
-rw-r--r--test/module.defs2
8 files changed, 50 insertions, 88 deletions
diff --git a/contrib/libsamplerate/module.defs b/contrib/libsamplerate/module.defs
deleted file mode 100644
index f5d6ac01a..000000000
--- a/contrib/libsamplerate/module.defs
+++ /dev/null
@@ -1,18 +0,0 @@
-$(eval $(call import.MODULE.defs,LIBSAMPLERATE,libsamplerate))
-$(eval $(call import.CONTRIB.defs,LIBSAMPLERATE))
-
-# libsamplerate 0.1.9-49-ga2eb281
-LIBSAMPLERATE.FETCH.url = https://download.handbrake.fr/handbrake/contrib/libsamplerate-a2eb2814150a4ada0b49ea3cc5e51170572fc606.tar.gz
-LIBSAMPLERATE.FETCH.url += https://github.com/erikd/libsamplerate/archive/a2eb2814150a4ada0b49ea3cc5e51170572fc606.tar.gz
-LIBSAMPLERATE.FETCH.sha256 = b8228e8bdbd754e95146745f85e37a25fb43d626cca7f785f3fd8e0a1c1f5f22
-LIBSAMPLERATE.EXTRACT.basename = libsamplerate-a2eb2814150a4ada0b49ea3cc5e51170572fc606.tar.gz
-LIBSAMPLERATE.EXTRACT.tarbase = libsamplerate-a2eb2814150a4ada0b49ea3cc5e51170572fc606
-
-LIBSAMPLERATE.CONFIGURE.bootstrap = rm -fr aclocal.m4 autom4te.cache configure; autoreconf -fiv;
-
-# Disable to avoid Carbon.h dependency on OSX
-LIBSAMPLERATE.CONFIGURE.extra = --disable-sndfile
-
-ifeq (1-msys,$(HOST.cross)-$(BUILD.system))
- LIBSAMPLERATE.CONFIGURE.args.build = --build=$(BUILD.machine)-unknown-linux-gnu
-endif
diff --git a/contrib/libsamplerate/module.rules b/contrib/libsamplerate/module.rules
deleted file mode 100644
index ef633856c..000000000
--- a/contrib/libsamplerate/module.rules
+++ /dev/null
@@ -1,2 +0,0 @@
-$(eval $(call import.MODULE.rules,LIBSAMPLERATE))
-$(eval $(call import.CONTRIB.rules,LIBSAMPLERATE))
diff --git a/gtk/configure.ac b/gtk/configure.ac
index e282b1d56..dc3a7dc26 100644
--- a/gtk/configure.ac
+++ b/gtk/configure.ac
@@ -192,7 +192,7 @@ PKG_CHECK_MODULES(GHB, [$GHB_PACKAGES])
GHB_CFLAGS="$HBINC $GHB_CFLAGS"
-HB_LIBS="-lhandbrake -lavformat -lavfilter -lavcodec -lavutil -lswresample -lpostproc -ldvdnav -ldvdread -lmp3lame -lvorbis -lvorbisenc -logg -lsamplerate -lx264 -lswscale -ltheoraenc -ltheoradec -lvpx -lz -lbz2 -lbluray -lass -lfontconfig -lfreetype -lxml2 -ljansson -lopus -lspeex -llzma"
+HB_LIBS="-lhandbrake -lavformat -lavfilter -lavcodec -lavutil -lswresample -lpostproc -ldvdnav -ldvdread -lmp3lame -lvorbis -lvorbisenc -logg -lx264 -lswscale -ltheoraenc -ltheoradec -lvpx -lz -lbz2 -lbluray -lass -lfontconfig -lfreetype -lxml2 -ljansson -lopus -lspeex -llzma"
if test "x$use_fdk_aac" = "xyes" ; then
HB_LIBS="$HB_LIBS -lfdk-aac"
diff --git a/libhb/audio_resample.c b/libhb/audio_resample.c
index 165368395..d415b574a 100644
--- a/libhb/audio_resample.c
+++ b/libhb/audio_resample.c
@@ -274,8 +274,8 @@ hb_buffer_t* hb_audio_resample(hb_audio_resample_t *resample,
if (resample->resample_needed)
{
- out_samples = (nsamples + 1) * resample->out.sample_rate /
- resample->in.sample_rate;
+ out_samples = nsamples * resample->out.sample_rate /
+ resample->in.sample_rate + 1;
out_size = av_samples_get_buffer_size(NULL, resample->out.channels,
out_samples,
resample->out.sample_fmt, 0);
@@ -286,7 +286,9 @@ hb_buffer_t* hb_audio_resample(hb_audio_resample_t *resample,
if (out_samples <= 0)
{
if (out_samples < 0)
+ {
hb_log("hb_audio_resample: swr_convert() failed");
+ }
// don't send empty buffers downstream (EOF)
hb_buffer_close(&out);
return NULL;
@@ -322,6 +324,7 @@ hb_buffer_t* hb_audio_resample(hb_audio_resample_t *resample,
}
out->size = out_samples * sample_size;
}
+ out->s.duration = 90000. * out_samples / resample->out.sample_rate;
return out;
}
diff --git a/libhb/module.defs b/libhb/module.defs
index c7321fffc..e76358f39 100644
--- a/libhb/module.defs
+++ b/libhb/module.defs
@@ -102,7 +102,7 @@ LIBHB.lib = $(LIBHB.build/)hb.lib
LIBHB.dll.libs = $(foreach n, \
ass avformat avfilter avcodec avutil swresample postproc dvdnav dvdread \
- freetype mp3lame samplerate swscale vpx theora vorbis vorbisenc ogg \
+ freetype mp3lame swscale vpx theora vorbis vorbisenc ogg \
x264 xml2 bluray jansson harfbuzz opus speex, \
$(CONTRIB.build/)lib/lib$(n).a )
diff --git a/libhb/sync.c b/libhb/sync.c
index 56edd2ba4..5ff7f1e17 100644
--- a/libhb/sync.c
+++ b/libhb/sync.c
@@ -10,7 +10,7 @@
#include "hb.h"
#include "hbffmpeg.h"
#include <stdio.h>
-#include "samplerate.h"
+#include "audio_resample.h"
#define SYNC_MAX_VIDEO_QUEUE_LEN 40
#define SYNC_MIN_VIDEO_QUEUE_LEN 20
@@ -121,16 +121,12 @@ typedef struct
// Audio stream context
struct
{
- hb_audio_t * audio;
+ hb_audio_t * audio;
// Audio filter settings
// Samplerate conversion
- struct
- {
- SRC_STATE * ctx;
- SRC_DATA pkt;
- } src;
- double gain_factor;
+ hb_audio_resample_t * resample;
+ double gain_factor;
} audio;
// Subtitle stream context
@@ -2245,13 +2241,24 @@ static int InitAudio( sync_common_t * common, int index )
if (!(audio->config.out.codec & HB_ACODEC_PASS_FLAG) &&
audio->config.in.samplerate != audio->config.out.samplerate)
{
- /* Initialize libsamplerate */
- int error;
- pv->stream->audio.src.ctx = src_new(SRC_SINC_MEDIUM_QUALITY,
- hb_mixdown_get_discrete_channel_count(audio->config.out.mixdown),
- &error);
- if (pv->stream->audio.src.ctx == NULL) goto fail;
- pv->stream->audio.src.pkt.end_of_input = 0;
+ /* Initialize samplerate conversion */
+ pv->stream->audio.resample =
+ hb_audio_resample_init(AV_SAMPLE_FMT_FLT,
+ audio->config.out.samplerate,
+ audio->config.out.mixdown,
+ audio->config.out.normalize_mix_level);
+ if (pv->stream->audio.resample == NULL)
+ {
+ hb_error("sync: audio 0x%x resample init failed", audio->id);
+ goto fail;
+ }
+ hb_audio_resample_set_sample_rate(pv->stream->audio.resample,
+ audio->config.in.samplerate);
+ if (hb_audio_resample_update(pv->stream->audio.resample))
+ {
+ hb_error("sync: audio 0x%x resample update failed", audio->id);
+ goto fail;
+ }
}
pv->stream->audio.gain_factor = pow(10, audio->config.out.gain / 20);
@@ -2265,9 +2272,9 @@ fail:
{
if (pv->stream != NULL)
{
- if (pv->stream->audio.src.ctx)
+ if (pv->stream->audio.resample)
{
- src_delete(pv->stream->audio.src.ctx);
+ hb_audio_resample_free(pv->stream->audio.resample);
}
hb_list_close(&pv->stream->delta_list);
hb_list_close(&pv->stream->in_queue);
@@ -3047,9 +3054,9 @@ static void syncAudioClose( hb_work_object_t * w )
}
// Free samplerate conversion context
- if (pv->stream->audio.src.ctx)
+ if (pv->stream->audio.resample)
{
- src_delete(pv->stream->audio.src.ctx);
+ hb_audio_resample_free(pv->stream->audio.resample);
}
sync_delta_t * delta;
@@ -3136,54 +3143,27 @@ static hb_buffer_t * FilterAudioFrame( sync_stream_t * stream,
// Audio is not passthru. Check if we need to modify the audio
// in any way.
- if (stream->audio.src.ctx != NULL)
+ if (stream->audio.resample != NULL)
{
/* do sample rate conversion */
- int count_in, count_out;
- hb_buffer_t * buf_raw = buf;
- int sample_size = hb_mixdown_get_discrete_channel_count(
- audio->config.out.mixdown ) * sizeof( float );
-
- count_in = buf_raw->size / sample_size;
- /*
- * When using stupid rates like 44.1 there will always be some
- * truncation error. E.g., a 1536 sample AC3 frame will turn into a
- * 1536*44.1/48.0 = 1411.2 sample frame. If we just truncate the .2
- * the error will build up over time and eventually the audio will
- * substantially lag the video. libsamplerate will keep track of the
- * fractional sample & give it to us when appropriate if we give it
- * an extra sample of space in the output buffer.
- */
- count_out = (buf->s.duration * audio->config.out.samplerate) /
- 90000 + 1;
-
- stream->audio.src.pkt.input_frames = count_in;
- stream->audio.src.pkt.output_frames = count_out;
- stream->audio.src.pkt.src_ratio =
- (double)audio->config.out.samplerate /
- audio->config.in.samplerate;
-
- buf = hb_buffer_init( count_out * sample_size );
- buf->s = buf_raw->s;
- stream->audio.src.pkt.data_in = (float *) buf_raw->data;
- stream->audio.src.pkt.data_out = (float *) buf->data;
- if (src_process(stream->audio.src.ctx, &stream->audio.src.pkt))
- {
- /* XXX If this happens, we're screwed */
- hb_error("sync: audio 0x%x src_process failed", audio->id);
- }
- hb_buffer_close(&buf_raw);
-
- if (stream->audio.src.pkt.output_frames_gen <= 0)
+ hb_buffer_t * out;
+ int nsamples, sample_size;
+
+ sample_size = hb_mixdown_get_discrete_channel_count(
+ audio->config.out.mixdown ) * sizeof( float );
+
+ nsamples = buf->size / sample_size;
+ out = hb_audio_resample(stream->audio.resample,
+ (const uint8_t **)&buf->data,
+ nsamples);
+ hb_buffer_close(&buf);
+ if (out == NULL)
{
- hb_buffer_close(&buf);
return NULL;
}
- buf->size = stream->audio.src.pkt.output_frames_gen * sample_size;
- buf->s.duration = 90000. * stream->audio.src.pkt.output_frames_gen /
- audio->config.out.samplerate;
- buf->s.start = stream->next_pts;
- buf->s.stop = stream->next_pts + buf->s.duration;
+ out->s.start = stream->next_pts;
+ out->s.stop = stream->next_pts + out->s.duration;
+ buf = out;
}
if (audio->config.out.gain > 0.0)
{
diff --git a/make/include/main.defs b/make/include/main.defs
index 380362d6e..b11d00d28 100644
--- a/make/include/main.defs
+++ b/make/include/main.defs
@@ -24,7 +24,6 @@ ifneq (,$(filter $(HOST.system),darwin cygwin mingw))
MODULES += contrib/libopus
MODULES += contrib/libspeex
MODULES += contrib/libtheora
- MODULES += contrib/libsamplerate
MODULES += contrib/lame
MODULES += contrib/x264
MODULES += contrib/jansson
diff --git a/test/module.defs b/test/module.defs
index db2ec169a..22000f73c 100644
--- a/test/module.defs
+++ b/test/module.defs
@@ -16,7 +16,7 @@ TEST.libs = $(LIBHB.a)
TEST.GCC.l = \
ass avformat avfilter avcodec avutil swresample postproc mp3lame dvdnav \
dvdread fribidi \
- samplerate swscale vpx theoraenc theoradec vorbis vorbisenc ogg x264 \
+ swscale vpx theoraenc theoradec vorbis vorbisenc ogg x264 \
bluray freetype xml2 bz2 z jansson harfbuzz opus speex lzma
ifeq (,$(filter $(HOST.system),darwin cygwin mingw))
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-16 15:31:46 +0000