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cmus: update to 2.6.0.

This commit is contained in:
Juan RP 2014-08-14 12:22:07 +02:00
parent 64286e44b2
commit 5e30580d80
3 changed files with 5 additions and 671 deletions
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414
srcpkgs/cmus/patches/cmus-2.5.0-opus-support.patch
View File

@ -1,414 +0,0 @@
--- Makefile 2012-11-07 09:46:17.000000000 +0100
+++ Makefile 2013-01-04 07:15:35.293069953 +0100
@@ -74,6 +74,7 @@ mikmod-objs := mikmod.lo
modplug-objs := modplug.lo
mpc-objs := mpc.lo
vorbis-objs := vorbis.lo
+opus-objs := opus.lo
wavpack-objs := wavpack.lo
wav-objs := wav.lo
mp4-objs := mp4.lo
@@ -88,6 +89,7 @@ ip-$(CONFIG_MIKMOD) += mikmod.so
ip-$(CONFIG_MODPLUG) += modplug.so
ip-$(CONFIG_MPC) += mpc.so
ip-$(CONFIG_VORBIS) += vorbis.so
+ip-$(CONFIG_OPUS) += opus.so
ip-$(CONFIG_WAVPACK) += wavpack.so
ip-$(CONFIG_WAV) += wav.so
ip-$(CONFIG_MP4) += mp4.so
@@ -102,6 +104,7 @@ $(mikmod-objs): CFLAGS += $(MIKMOD_CFLA
$(modplug-objs): CFLAGS += $(MODPLUG_CFLAGS)
$(mpc-objs): CFLAGS += $(MPC_CFLAGS)
$(vorbis-objs): CFLAGS += $(VORBIS_CFLAGS)
+$(opus-objs): CFLAGS += $(OPUS_CFLAGS)
$(wavpack-objs): CFLAGS += $(WAVPACK_CFLAGS)
$(mp4-objs): CFLAGS += $(MP4_CFLAGS)
$(aac-objs): CFLAGS += $(AAC_CFLAGS)
@@ -129,6 +132,9 @@ mpc.so: $(mpc-objs) $(libcmus-y)
vorbis.so: $(vorbis-objs) $(libcmus-y)
$(call cmd,ld_dl,$(VORBIS_LIBS))
+opus.so: $(opus-objs) $(libcmus-y)
+ $(call cmd,ld_dl,$(OPUS_LIBS))
+
wavpack.so: $(wavpack-objs) $(libcmus-y)
$(call cmd,ld_dl,$(WAVPACK_LIBS))
--- configure 2012-11-07 09:46:17.000000000 +0100
+++ configure 2013-01-04 07:15:35.293069953 +0100
@@ -228,6 +228,12 @@ check_vorbis()
fi
}
+check_opus()
+{
+ pkg_config OPUS "opusfile"
+ return $?
+}
+
check_wavpack()
{
pkg_config WAVPACK "wavpack" "" "-lwavpack"
@@ -425,6 +431,7 @@ Optional Features: y/n
CONFIG_MPC libmpcdec (Musepack .mpc, .mpp, .mp+) [auto]
CONFIG_VORBIS Ogg/Vorbis (.ogg, application/ogg, audio/x-ogg) [auto]
CONFIG_TREMOR Use Tremor as Ogg/Vorbis input plugin [n]
+ CONFIG_OPUS Opus (.opus) [auto]
CONFIG_WAV WAV [y]
CONFIG_WAVPACK WavPack (.wv, audio/x-wavpack) [auto]
CONFIG_MP4 MPEG-4 AAC (.mp4, .m4a, .m4b) [auto]
@@ -485,6 +492,7 @@ check check_mikmod CONFIG_MIKMOD
check check_modplug CONFIG_MODPLUG
check check_mpc CONFIG_MPC
check check_vorbis CONFIG_VORBIS
+check check_opus CONFIG_OPUS
check check_wavpack CONFIG_WAVPACK
check check_mp4 CONFIG_MP4
check check_aac CONFIG_AAC
@@ -534,7 +542,7 @@ config_header config/cue.h CONFIG_CUE
CFLAGS="${CFLAGS} -DHAVE_CONFIG"
makefile_vars bindir datadir libdir mandir exampledir
-makefile_vars CONFIG_CDIO CONFIG_FLAC CONFIG_MAD CONFIG_MIKMOD CONFIG_MODPLUG CONFIG_MPC CONFIG_VORBIS CONFIG_WAVPACK CONFIG_WAV CONFIG_MP4 CONFIG_AAC CONFIG_FFMPEG CONFIG_CUE
+makefile_vars CONFIG_CDIO CONFIG_FLAC CONFIG_MAD CONFIG_MIKMOD CONFIG_MODPLUG CONFIG_MPC CONFIG_VORBIS CONFIG_OPUS CONFIG_WAVPACK CONFIG_WAV CONFIG_MP4 CONFIG_AAC CONFIG_FFMPEG CONFIG_CUE
makefile_vars CONFIG_ROAR CONFIG_PULSE CONFIG_ALSA CONFIG_AO CONFIG_ARTS CONFIG_OSS CONFIG_SUN CONFIG_WAVEOUT
generate_config_mk
--- opus.c 1970-01-01 01:00:00.000000000 +0100
+++ opus.c 2013-01-04 07:15:35.493069945 +0100
@@ -0,0 +1,335 @@
+/*
+ * Copyright 2008-2012 Various Authors
+ * Copyright 2004-2005 Timo Hirvonen
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as
+ * published by the Free Software Foundation; either version 2 of the
+ * License, or (at your option) any later version.
+ *
+ * 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
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "ip.h"
+#include "xmalloc.h"
+#include "read_wrapper.h"
+#include "debug.h"
+#include "comment.h"
+
+#include <opusfile.h>
+
+#include <errno.h>
+#include <string.h>
+#include <sys/types.h>
+#include <unistd.h>
+
+#define SAMPLING_RATE 48000
+#define CHANNELS 2
+
+struct opus_private {
+ OggOpusFile *of;
+ int current_link;
+};
+
+static int read_func(void *datasource, unsigned char *ptr, int size)
+{
+ struct input_plugin_data *ip_data = datasource;
+ return read_wrapper(ip_data, ptr, size);
+}
+
+static int seek_func(void *datasource, opus_int64 offset, int whence)
+{
+ struct input_plugin_data *ip_data = datasource;
+ return lseek(ip_data->fd, offset, whence);
+}
+
+static int close_func(void *datasource)
+{
+ struct input_plugin_data *ip_data;
+ int rc;
+
+ ip_data = datasource;
+ rc = close(ip_data->fd);
+ ip_data->fd = -1;
+ return rc;
+}
+
+static opus_int64 tell_func(void *datasource)
+{
+ struct input_plugin_data *ip_data = datasource;
+ return lseek(ip_data->fd, 0, SEEK_CUR);
+}
+
+static OpusFileCallbacks callbacks = {
+ .read = read_func,
+ .seek = seek_func,
+ .tell = tell_func,
+ .close = close_func
+};
+
+static int opus_open(struct input_plugin_data *ip_data)
+{
+ struct opus_private *priv;
+ int rc;
+ void *source;
+
+ priv = xnew(struct opus_private, 1);
+ priv->current_link = -1;
+ priv->of = NULL;
+
+ source = op_fdopen(&callbacks, ip_data->fd, "r");
+ if (source == NULL) {
+ free(priv);
+ return -IP_ERROR_INTERNAL;
+ }
+
+ priv->of = op_open_callbacks(source, &callbacks, NULL, 0, &rc);
+ if (rc != 0) {
+ d_print("op_open_callbacks failed: %d:%s\n", rc, strerror(rc));
+ free(priv);
+ /* ogg is a container format, so it is likely to contain
+ * something else if it isn't opus */
+ return -IP_ERROR_UNSUPPORTED_FILE_TYPE;
+ }
+ ip_data->private = priv;
+
+ ip_data->sf = sf_rate(SAMPLING_RATE)
+ | sf_channels(CHANNELS)
+ | sf_bits(16)
+ | sf_signed(1);
+#ifdef WORDS_BIGENDIAN
+ ip_data->sf |= sf_bigendian(1);
+#endif
+ return 0;
+}
+
+static int opus_close(struct input_plugin_data *ip_data)
+{
+ struct opus_private *priv = ip_data->private;
+ /* this closes ip_data->fd! */
+ op_free(priv->of);
+ ip_data->fd = -1;
+ free(priv);
+ ip_data->private = NULL;
+ return 0;
+}
+
+/*
+ * -n
+ * indicates error
+ * 0
+ * indicates EOF
+ * n
+ * indicates actual number of bytes read
+ */
+static int opus_read(struct input_plugin_data *ip_data, char *buffer, int count)
+{
+ struct opus_private *priv;
+ int samples, current_link, rc;
+
+ priv = ip_data->private;
+
+ /* samples = number of samples read per channel */
+ samples = op_read_stereo(priv->of, (opus_int16*)buffer,
+ count / sizeof(opus_int16));
+ if (samples < 0) {
+ switch (samples) {
+ case OP_HOLE:
+ errno = EAGAIN;
+ rc = -1;
+ break;
+ case OP_EREAD:
+ errno = EINVAL;
+ rc = -1;
+ break;
+ case OP_EFAULT:
+ errno = EINVAL;
+ rc = -1;
+ break;
+ case OP_EIMPL:
+ rc = -IP_ERROR_FUNCTION_NOT_SUPPORTED;
+ break;
+ case OP_EINVAL:
+ errno = EINVAL;
+ rc = -1;
+ break;
+ case OP_ENOTFORMAT:
+ rc = -IP_ERROR_FILE_FORMAT;
+ break;
+ case OP_EBADHEADER:
+ rc = -IP_ERROR_FILE_FORMAT;
+ break;
+ case OP_EVERSION:
+ rc = -IP_ERROR_FILE_FORMAT;
+ break;
+ case OP_EBADPACKET:
+ errno = EINVAL;
+ rc = -1;
+ break;
+ case OP_EBADLINK:
+ errno = EINVAL;
+ rc = -1;
+ break;
+ case OP_EBADTIMESTAMP:
+ rc = -IP_ERROR_FILE_FORMAT;
+ break;
+ default:
+ d_print("error: %d\n", samples);
+ rc = -IP_ERROR_FILE_FORMAT;
+ }
+ } else if (samples == 0) {
+ /* EOF or buffer too small */
+ rc = 0;
+ } else {
+ current_link = op_current_link(priv->of);
+ if (current_link < 0) {
+ d_print("error: %d\n", current_link);
+ rc = -1;
+ } else {
+ if (ip_data->remote && current_link != priv->current_link) {
+ ip_data->metadata_changed = 1;
+ priv->current_link = current_link;
+ }
+
+ /* bytes = samples * channels * sample_size */
+ rc = samples * CHANNELS * sizeof(opus_int16);
+ }
+ }
+
+ return rc;
+}
+
+static int opus_seek(struct input_plugin_data *ip_data, double offset)
+{
+ struct opus_private *priv;
+ int rc;
+
+ priv = ip_data->private;
+
+ rc = op_pcm_seek(priv->of, offset * SAMPLING_RATE);
+ switch (rc) {
+ case OP_ENOSEEK:
+ return -IP_ERROR_FUNCTION_NOT_SUPPORTED;
+ case OP_EINVAL:
+ return -IP_ERROR_INTERNAL;
+ case OP_EREAD:
+ return -IP_ERROR_INTERNAL;
+ case OP_EFAULT:
+ return -IP_ERROR_INTERNAL;
+ case OP_EBADLINK:
+ return -IP_ERROR_INTERNAL;
+ }
+ return 0;
+}
+
+static int opus_read_comments(struct input_plugin_data *ip_data,
+ struct keyval **comments)
+{
+ GROWING_KEYVALS(c);
+ struct opus_private *priv;
+ const OpusTags *ot;
+ int i;
+
+ priv = ip_data->private;
+
+ ot = op_tags(priv->of, -1);
+ if (ot == NULL) {
+ d_print("ot == NULL\n");
+ *comments = keyvals_new(0);
+ return 0;
+ }
+
+ for (i = 0; i < ot->comments; i++) {
+ const char *str = ot->user_comments[i];
+ const char *eq = strchr(str, '=');
+ char *key;
+
+ if (!eq) {
+ d_print("invalid comment: '%s' ('=' expected)\n", str);
+ continue;
+ }
+
+ key = xstrndup(str, eq - str);
+ comments_add_const(&c, key, eq + 1);
+ free(key);
+ }
+ keyvals_terminate(&c);
+ *comments = c.keyvals;
+ return 0;
+}
+
+static int opus_duration(struct input_plugin_data *ip_data)
+{
+ struct opus_private *priv;
+ ogg_int64_t samples;
+
+ priv = ip_data->private;
+
+ samples = op_pcm_total(priv->of, -1);
+ if (samples < 0)
+ return -IP_ERROR_FUNCTION_NOT_SUPPORTED;
+
+ return samples / SAMPLING_RATE;
+}
+
+static long opus_bitrate(struct input_plugin_data *ip_data)
+{
+ struct opus_private *priv;
+ opus_int32 bitrate;
+
+ priv = ip_data->private;
+
+ bitrate = op_bitrate(priv->of, -1);
+ if (bitrate < 0)
+ return -IP_ERROR_FUNCTION_NOT_SUPPORTED;
+ else
+ return bitrate;
+}
+
+static long opus_current_bitrate(struct input_plugin_data *ip_data)
+{
+ struct opus_private *priv;
+ opus_int32 bitrate;
+
+ priv = ip_data->private;
+
+ bitrate = op_bitrate_instant(priv->of);
+ if (bitrate < 0)
+ return -IP_ERROR_FUNCTION_NOT_SUPPORTED;
+ else
+ return bitrate;
+}
+
+static char *opus_codec(struct input_plugin_data *ip_data)
+{
+ return xstrdup("opus");
+}
+
+static char *opus_codec_profile(struct input_plugin_data *ip_data)
+{
+ return NULL;
+}
+
+const struct input_plugin_ops ip_ops = {
+ .open = opus_open,
+ .close = opus_close,
+ .read = opus_read,
+ .seek = opus_seek,
+ .read_comments = opus_read_comments,
+ .duration = opus_duration,
+ .bitrate = opus_bitrate,
+ .bitrate_current = opus_current_bitrate,
+ .codec = opus_codec,
+ .codec_profile = opus_codec_profile
+};
+
+const int ip_priority = 50;
+const char * const ip_extensions[] = { "opus", NULL };
+const char * const ip_mime_types[] = { NULL };
+const char * const ip_options[] = { NULL };

251
srcpkgs/cmus/patches/ffmpeg-2.1.patch
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@ -1,251 +0,0 @@
commit fb939e5238c128ae1837430cbf838b945d4c99a9
Author: Stone Kang <stonekyx@gmail.com>
Date: Sun Jul 28 19:42:16 2013 +0800
Update ffmpeg.c to work with newer versions of FFmpeg
Since ffmpeg 2.0, the macro AVCODEC_MAX_AUDIO_FRAME_SIZE is deprecated
and removed. This commit simply defines the macro if it's not defined
previously, using the same value it used to be in old versions of
libavcodec.
Actually, as my branch has been using swr_convert, where caller specifies
the size of output buffer, even if the current size of 192000 proved to
be not enough in the future, we can simply let swr_convert deal with the
insufficient size by a small modification.
Currently the parameters to swr_convert() still assumes that the size of
192000 samples is always enough for the output buffer.
Signed-off-by: Gregory Petrosyan <gregory.petrosyan@gmail.com>
diff --git a/Makefile b/Makefile
index 826ac51..5e3f491 100644
--- Makefile
+++ Makefile
@@ -18,6 +18,9 @@ include scripts/lib.mk
CFLAGS += -D_FILE_OFFSET_BITS=64
+FFMPEG_CFLAGS += $(shell pkg-config --cflags libswresample)
+FFMPEG_LIBS += $(shell pkg-config --libs libswresample)
+
CMUS_LIBS = $(PTHREAD_LIBS) $(NCURSES_LIBS) $(ICONV_LIBS) $(DL_LIBS) $(DISCID_LIBS) $(CUE_LIBS) -lm $(COMPAT_LIBS)
input.o main.o ui_curses.o pulse.lo: .version
diff --git a/ffmpeg.c b/ffmpeg.c
index 93c1831..e4da4df 100644
--- ffmpeg.c
+++ ffmpeg.c
@@ -30,10 +30,16 @@
#include <ffmpeg/avcodec.h>
#include <ffmpeg/avformat.h>
#include <ffmpeg/avio.h>
+#include <ffmpeg/swresample.h>
+#include <ffmpeg/opt.h>
+#include <ffmpeg/audioconvert.h>
#else
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
+#include <libswresample/swresample.h>
+#include <libavutil/opt.h>
+#include <libavutil/audioconvert.h>
#ifndef AVUTIL_MATHEMATICS_H
#include <libavutil/mathematics.h>
#endif
@@ -63,6 +69,10 @@
#define AVDictionaryEntry AVMetadataTag
#endif
+#ifndef AVCODEC_MAX_AUDIO_FRAME_SIZE
+#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000
+#endif
+
struct ffmpeg_input {
AVPacket pkt;
int curr_pkt_size;
@@ -83,6 +93,7 @@ struct ffmpeg_private {
AVCodecContext *codec_context;
AVFormatContext *input_context;
AVCodec *codec;
+ SwrContext *swr;
int stream_index;
struct ffmpeg_input *input;
@@ -182,6 +193,7 @@ static int ffmpeg_open(struct input_plugin_data *ip_data)
AVCodec *codec;
AVCodecContext *cc = NULL;
AVFormatContext *ic = NULL;
+ SwrContext *swr = NULL;
ffmpeg_init();
@@ -241,20 +253,16 @@ static int ffmpeg_open(struct input_plugin_data *ip_data)
break;
}
-#if (LIBAVCODEC_VERSION_INT > ((51<<16)+(64<<8)+0))
- if (cc->sample_fmt == AV_SAMPLE_FMT_FLT || cc->sample_fmt == AV_SAMPLE_FMT_DBL) {
-#else
- if (cc->sample_fmt == AV_SAMPLE_FMT_FLT) {
-#endif
- err = -IP_ERROR_SAMPLE_FORMAT;
- break;
- }
/* We assume below that no more errors follow. */
} while (0);
if (err < 0) {
/* Clean up. cc is never opened at this point. (See above assumption.) */
+#if (LIBAVCODEC_VERSION_INT < ((53<<16)+(25<<8)+0))
av_close_input_file(ic);
+#else
+ avformat_close_input(&ic);
+#endif
return err;
}
@@ -266,26 +274,43 @@ static int ffmpeg_open(struct input_plugin_data *ip_data)
priv->input = ffmpeg_input_create();
if (priv->input == NULL) {
avcodec_close(cc);
+#if (LIBAVCODEC_VERSION_INT < ((53<<16)+(25<<8)+0))
av_close_input_file(ic);
+#else
+ avformat_close_input(&ic);
+#endif
free(priv);
return -IP_ERROR_INTERNAL;
}
priv->output = ffmpeg_output_create();
+ /* Prepare for resampling. */
+ swr = swr_alloc();
+ av_opt_set_int(swr, "in_channel_layout", av_get_default_channel_layout(cc->channels), 0);
+ av_opt_set_int(swr, "out_channel_layout", av_get_default_channel_layout(cc->channels), 0);
+ av_opt_set_int(swr, "in_sample_rate", cc->sample_rate, 0);
+ av_opt_set_int(swr, "out_sample_rate", cc->sample_rate, 0);
+ av_opt_set_sample_fmt(swr, "in_sample_fmt", cc->sample_fmt, 0);
+ priv->swr = swr;
+
ip_data->private = priv;
ip_data->sf = sf_rate(cc->sample_rate) | sf_channels(cc->channels);
switch (cc->sample_fmt) {
case AV_SAMPLE_FMT_U8:
ip_data->sf |= sf_bits(8) | sf_signed(0);
+ av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_U8, 0);
break;
case AV_SAMPLE_FMT_S32:
ip_data->sf |= sf_bits(32) | sf_signed(1);
+ av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S32, 0);
break;
/* AV_SAMPLE_FMT_S16 */
default:
ip_data->sf |= sf_bits(16) | sf_signed(1);
+ av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
break;
}
+ swr_init(swr);
#ifdef WORDS_BIGENDIAN
ip_data->sf |= sf_bigendian(1);
#endif
@@ -301,7 +326,12 @@ static int ffmpeg_close(struct input_plugin_data *ip_data)
struct ffmpeg_private *priv = ip_data->private;
avcodec_close(priv->codec_context);
+#if (LIBAVCODEC_VERSION_INT < ((53<<16)+(25<<8)+0))
av_close_input_file(priv->input_context);
+#else
+ avformat_close_input(&priv->input_context);
+#endif
+ swr_free(&priv->swr);
ffmpeg_input_free(priv->input);
ffmpeg_output_free(priv->output);
free(priv);
@@ -314,8 +344,12 @@ static int ffmpeg_close(struct input_plugin_data *ip_data)
* It returns < 0 on error. 0 on EOF.
*/
static int ffmpeg_fill_buffer(AVFormatContext *ic, AVCodecContext *cc, struct ffmpeg_input *input,
- struct ffmpeg_output *output)
+ struct ffmpeg_output *output, SwrContext *swr)
{
+#if (LIBAVCODEC_VERSION_INT >= ((53<<16) + (25<<8) + 0))
+ AVFrame *frame = avcodec_alloc_frame();
+ int got_frame;
+#endif
while (1) {
/* frame_size specifies the size of output->buffer for
* avcodec_decode_audio2. */
@@ -326,6 +360,9 @@ static int ffmpeg_fill_buffer(AVFormatContext *ic, AVCodecContext *cc, struct ff
av_free_packet(&input->pkt);
if (av_read_frame(ic, &input->pkt) < 0) {
/* Force EOF once we can read no longer. */
+#if (LIBAVCODEC_VERSION_INT >= ((53<<16) + (25<<8) + 0))
+ avcodec_free_frame(&frame);
+#endif
return 0;
}
input->curr_pkt_size = input->pkt.size;
@@ -345,7 +382,7 @@ static int ffmpeg_fill_buffer(AVFormatContext *ic, AVCodecContext *cc, struct ff
#elif (LIBAVCODEC_VERSION_INT <= ((52<<16) + (25<<8) + 0))
len = avcodec_decode_audio2(cc, (int16_t *) output->buffer, &frame_size,
input->curr_pkt_buf, input->curr_pkt_size);
-#else
+#elif (LIBAVCODEC_VERSION_INT < ((53<<16) + (25<<8) + 0))
{
AVPacket avpkt;
av_init_packet(&avpkt);
@@ -354,6 +391,14 @@ static int ffmpeg_fill_buffer(AVFormatContext *ic, AVCodecContext *cc, struct ff
len = avcodec_decode_audio3(cc, (int16_t *) output->buffer, &frame_size, &avpkt);
av_free_packet(&avpkt);
}
+#else
+ {
+ AVPacket avpkt;
+ av_new_packet(&avpkt, input->curr_pkt_size);
+ memcpy(avpkt.data, input->curr_pkt_buf, input->curr_pkt_size);
+ len = avcodec_decode_audio4(cc, frame, &got_frame, &avpkt);
+ av_free_packet(&avpkt);
+ }
#endif
if (len < 0) {
/* this is often reached when seeking, not sure why */
@@ -362,11 +407,27 @@ static int ffmpeg_fill_buffer(AVFormatContext *ic, AVCodecContext *cc, struct ff
}
input->curr_pkt_size -= len;
input->curr_pkt_buf += len;
+#if (LIBAVCODEC_VERSION_INT < ((53<<16) + (25<<8) + 0))
if (frame_size > 0) {
output->buffer_pos = output->buffer;
output->buffer_used_len = frame_size;
return frame_size;
}
+#else
+ if (got_frame) {
+ int res = swr_convert(swr,
+ &output->buffer,
+ frame->nb_samples,
+ (const uint8_t **)frame->extended_data,
+ frame->nb_samples);
+ if (res < 0)
+ res = 0;
+ output->buffer_pos = output->buffer;
+ output->buffer_used_len = res * cc->channels * sizeof(int16_t);
+ avcodec_free_frame(&frame);
+ return output->buffer_used_len;
+ }
+#endif
}
/* This should never get here. */
return -IP_ERROR_INTERNAL;
@@ -380,7 +441,8 @@ static int ffmpeg_read(struct input_plugin_data *ip_data, char *buffer, int coun
int out_size;
if (output->buffer_used_len == 0) {
- rc = ffmpeg_fill_buffer(priv->input_context, priv->codec_context, priv->input, priv->output);
+ rc = ffmpeg_fill_buffer(priv->input_context, priv->codec_context,
+ priv->input, priv->output, priv->swr);
if (rc <= 0) {
return rc;
}

11
srcpkgs/cmus/template
View File

@ -1,14 +1,13 @@
# Template file for 'cmus'
pkgname=cmus
version=2.5.0
revision=5
wrksrc="cmus-v${version}"
version=2.6.0
revision=1
short_desc="Small, fast and powerful console music player"
maintainer="Juan RP <xtraeme@gmail.com>"
license="GPL-2"
homepage="http://cmus.sourceforge.net"
distfiles="${SOURCEFORGE_SITE}/cmus/cmus-v${version}.tar.bz2"
checksum=2021fab64088898d62d1439525ec345e219f58d91a9e933dc85e84eda21c8ddf
homepage="http://cmus.github.io"
distfiles="https://github.com/cmus/cmus/archive/v${version}.tar.gz"
checksum=0c53c072432e026d682d1749bbe89777507b31107ca04afad21a9bf375208e8e
hostmakedepends="pkg-config"
makedepends="ncurses-devel faad2-devel libflac-devel libao-devel libmad-devel