cmus: backport a patch to build against ffmpeg-2.1.
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2 changed files with 253 additions and 2 deletions
srcpkgs/cmus
251
srcpkgs/cmus/patches/ffmpeg-2.1.patch
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251
srcpkgs/cmus/patches/ffmpeg-2.1.patch
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commit fb939e5238c128ae1837430cbf838b945d4c99a9
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Author: Stone Kang <stonekyx@gmail.com>
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Date: Sun Jul 28 19:42:16 2013 +0800
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Update ffmpeg.c to work with newer versions of FFmpeg
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Since ffmpeg 2.0, the macro AVCODEC_MAX_AUDIO_FRAME_SIZE is deprecated
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and removed. This commit simply defines the macro if it's not defined
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previously, using the same value it used to be in old versions of
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libavcodec.
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Actually, as my branch has been using swr_convert, where caller specifies
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the size of output buffer, even if the current size of 192000 proved to
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be not enough in the future, we can simply let swr_convert deal with the
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insufficient size by a small modification.
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Currently the parameters to swr_convert() still assumes that the size of
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192000 samples is always enough for the output buffer.
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Signed-off-by: Gregory Petrosyan <gregory.petrosyan@gmail.com>
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diff --git a/Makefile b/Makefile
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index 826ac51..5e3f491 100644
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--- Makefile
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+++ Makefile
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@@ -18,6 +18,9 @@ include scripts/lib.mk
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CFLAGS += -D_FILE_OFFSET_BITS=64
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+FFMPEG_CFLAGS += $(shell pkg-config --cflags libswresample)
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+FFMPEG_LIBS += $(shell pkg-config --libs libswresample)
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+
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CMUS_LIBS = $(PTHREAD_LIBS) $(NCURSES_LIBS) $(ICONV_LIBS) $(DL_LIBS) $(DISCID_LIBS) $(CUE_LIBS) -lm $(COMPAT_LIBS)
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input.o main.o ui_curses.o pulse.lo: .version
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diff --git a/ffmpeg.c b/ffmpeg.c
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index 93c1831..e4da4df 100644
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--- ffmpeg.c
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+++ ffmpeg.c
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@@ -30,10 +30,16 @@
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#include <ffmpeg/avcodec.h>
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#include <ffmpeg/avformat.h>
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#include <ffmpeg/avio.h>
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+#include <ffmpeg/swresample.h>
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+#include <ffmpeg/opt.h>
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+#include <ffmpeg/audioconvert.h>
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#else
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#include <libavcodec/avcodec.h>
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#include <libavformat/avformat.h>
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#include <libavformat/avio.h>
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+#include <libswresample/swresample.h>
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+#include <libavutil/opt.h>
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+#include <libavutil/audioconvert.h>
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#ifndef AVUTIL_MATHEMATICS_H
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#include <libavutil/mathematics.h>
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#endif
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@@ -63,6 +69,10 @@
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#define AVDictionaryEntry AVMetadataTag
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#endif
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+#ifndef AVCODEC_MAX_AUDIO_FRAME_SIZE
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+#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000
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+#endif
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+
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struct ffmpeg_input {
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AVPacket pkt;
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int curr_pkt_size;
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@@ -83,6 +93,7 @@ struct ffmpeg_private {
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AVCodecContext *codec_context;
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AVFormatContext *input_context;
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AVCodec *codec;
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+ SwrContext *swr;
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int stream_index;
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struct ffmpeg_input *input;
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@@ -182,6 +193,7 @@ static int ffmpeg_open(struct input_plugin_data *ip_data)
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AVCodec *codec;
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AVCodecContext *cc = NULL;
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AVFormatContext *ic = NULL;
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+ SwrContext *swr = NULL;
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ffmpeg_init();
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@@ -241,20 +253,16 @@ static int ffmpeg_open(struct input_plugin_data *ip_data)
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break;
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}
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-#if (LIBAVCODEC_VERSION_INT > ((51<<16)+(64<<8)+0))
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- if (cc->sample_fmt == AV_SAMPLE_FMT_FLT || cc->sample_fmt == AV_SAMPLE_FMT_DBL) {
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-#else
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- if (cc->sample_fmt == AV_SAMPLE_FMT_FLT) {
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-#endif
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- err = -IP_ERROR_SAMPLE_FORMAT;
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- break;
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- }
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/* We assume below that no more errors follow. */
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} while (0);
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if (err < 0) {
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/* Clean up. cc is never opened at this point. (See above assumption.) */
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+#if (LIBAVCODEC_VERSION_INT < ((53<<16)+(25<<8)+0))
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av_close_input_file(ic);
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+#else
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+ avformat_close_input(&ic);
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+#endif
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return err;
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}
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@@ -266,26 +274,43 @@ static int ffmpeg_open(struct input_plugin_data *ip_data)
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priv->input = ffmpeg_input_create();
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if (priv->input == NULL) {
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avcodec_close(cc);
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+#if (LIBAVCODEC_VERSION_INT < ((53<<16)+(25<<8)+0))
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av_close_input_file(ic);
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+#else
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+ avformat_close_input(&ic);
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+#endif
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free(priv);
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return -IP_ERROR_INTERNAL;
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}
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priv->output = ffmpeg_output_create();
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+ /* Prepare for resampling. */
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+ swr = swr_alloc();
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+ av_opt_set_int(swr, "in_channel_layout", av_get_default_channel_layout(cc->channels), 0);
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+ av_opt_set_int(swr, "out_channel_layout", av_get_default_channel_layout(cc->channels), 0);
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+ av_opt_set_int(swr, "in_sample_rate", cc->sample_rate, 0);
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+ av_opt_set_int(swr, "out_sample_rate", cc->sample_rate, 0);
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+ av_opt_set_sample_fmt(swr, "in_sample_fmt", cc->sample_fmt, 0);
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+ priv->swr = swr;
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+
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ip_data->private = priv;
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ip_data->sf = sf_rate(cc->sample_rate) | sf_channels(cc->channels);
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switch (cc->sample_fmt) {
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case AV_SAMPLE_FMT_U8:
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ip_data->sf |= sf_bits(8) | sf_signed(0);
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+ av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_U8, 0);
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break;
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case AV_SAMPLE_FMT_S32:
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ip_data->sf |= sf_bits(32) | sf_signed(1);
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+ av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S32, 0);
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break;
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/* AV_SAMPLE_FMT_S16 */
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default:
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ip_data->sf |= sf_bits(16) | sf_signed(1);
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+ av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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break;
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}
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+ swr_init(swr);
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#ifdef WORDS_BIGENDIAN
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ip_data->sf |= sf_bigendian(1);
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#endif
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@@ -301,7 +326,12 @@ static int ffmpeg_close(struct input_plugin_data *ip_data)
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struct ffmpeg_private *priv = ip_data->private;
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avcodec_close(priv->codec_context);
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+#if (LIBAVCODEC_VERSION_INT < ((53<<16)+(25<<8)+0))
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av_close_input_file(priv->input_context);
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+#else
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+ avformat_close_input(&priv->input_context);
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+#endif
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+ swr_free(&priv->swr);
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ffmpeg_input_free(priv->input);
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ffmpeg_output_free(priv->output);
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free(priv);
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@@ -314,8 +344,12 @@ static int ffmpeg_close(struct input_plugin_data *ip_data)
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* It returns < 0 on error. 0 on EOF.
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*/
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static int ffmpeg_fill_buffer(AVFormatContext *ic, AVCodecContext *cc, struct ffmpeg_input *input,
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- struct ffmpeg_output *output)
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+ struct ffmpeg_output *output, SwrContext *swr)
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{
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+#if (LIBAVCODEC_VERSION_INT >= ((53<<16) + (25<<8) + 0))
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+ AVFrame *frame = avcodec_alloc_frame();
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+ int got_frame;
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+#endif
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while (1) {
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/* frame_size specifies the size of output->buffer for
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* avcodec_decode_audio2. */
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@@ -326,6 +360,9 @@ static int ffmpeg_fill_buffer(AVFormatContext *ic, AVCodecContext *cc, struct ff
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av_free_packet(&input->pkt);
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if (av_read_frame(ic, &input->pkt) < 0) {
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/* Force EOF once we can read no longer. */
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+#if (LIBAVCODEC_VERSION_INT >= ((53<<16) + (25<<8) + 0))
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+ avcodec_free_frame(&frame);
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+#endif
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return 0;
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}
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input->curr_pkt_size = input->pkt.size;
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@@ -345,7 +382,7 @@ static int ffmpeg_fill_buffer(AVFormatContext *ic, AVCodecContext *cc, struct ff
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#elif (LIBAVCODEC_VERSION_INT <= ((52<<16) + (25<<8) + 0))
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len = avcodec_decode_audio2(cc, (int16_t *) output->buffer, &frame_size,
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input->curr_pkt_buf, input->curr_pkt_size);
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-#else
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+#elif (LIBAVCODEC_VERSION_INT < ((53<<16) + (25<<8) + 0))
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{
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AVPacket avpkt;
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av_init_packet(&avpkt);
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@@ -354,6 +391,14 @@ static int ffmpeg_fill_buffer(AVFormatContext *ic, AVCodecContext *cc, struct ff
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len = avcodec_decode_audio3(cc, (int16_t *) output->buffer, &frame_size, &avpkt);
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av_free_packet(&avpkt);
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}
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+#else
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+ {
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+ AVPacket avpkt;
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+ av_new_packet(&avpkt, input->curr_pkt_size);
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+ memcpy(avpkt.data, input->curr_pkt_buf, input->curr_pkt_size);
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+ len = avcodec_decode_audio4(cc, frame, &got_frame, &avpkt);
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+ av_free_packet(&avpkt);
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+ }
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#endif
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if (len < 0) {
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/* this is often reached when seeking, not sure why */
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@@ -362,11 +407,27 @@ static int ffmpeg_fill_buffer(AVFormatContext *ic, AVCodecContext *cc, struct ff
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}
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input->curr_pkt_size -= len;
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input->curr_pkt_buf += len;
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+#if (LIBAVCODEC_VERSION_INT < ((53<<16) + (25<<8) + 0))
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if (frame_size > 0) {
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output->buffer_pos = output->buffer;
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output->buffer_used_len = frame_size;
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return frame_size;
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}
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+#else
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+ if (got_frame) {
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+ int res = swr_convert(swr,
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+ &output->buffer,
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+ frame->nb_samples,
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+ (const uint8_t **)frame->extended_data,
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+ frame->nb_samples);
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+ if (res < 0)
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+ res = 0;
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+ output->buffer_pos = output->buffer;
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+ output->buffer_used_len = res * cc->channels * sizeof(int16_t);
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+ avcodec_free_frame(&frame);
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+ return output->buffer_used_len;
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+ }
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+#endif
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}
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/* This should never get here. */
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return -IP_ERROR_INTERNAL;
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@@ -380,7 +441,8 @@ static int ffmpeg_read(struct input_plugin_data *ip_data, char *buffer, int coun
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int out_size;
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if (output->buffer_used_len == 0) {
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- rc = ffmpeg_fill_buffer(priv->input_context, priv->codec_context, priv->input, priv->output);
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+ rc = ffmpeg_fill_buffer(priv->input_context, priv->codec_context,
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+ priv->input, priv->output, priv->swr);
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if (rc <= 0) {
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return rc;
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}
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@ -1,7 +1,7 @@
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# Template file for 'cmus'
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pkgname=cmus
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version=2.5.0
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revision=3
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revision=4
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wrksrc="cmus-v${version}"
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short_desc="Small, fast and powerful console music player"
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maintainer="Juan RP <xtraeme@gmail.com>"
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@ -13,7 +13,7 @@ checksum=2021fab64088898d62d1439525ec345e219f58d91a9e933dc85e84eda21c8ddf
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hostmakedepends="pkg-config"
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makedepends="faad2-devel libflac-devel libao-devel libmad-devel
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libmodplug-devel libmp4v2-devel>=2.0.0 libmpcdec-devel pulseaudio-devel
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opusfile-devel wavpack-devel alsa-lib-devel libvorbis-devel ffmpeg-devel
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opusfile-devel wavpack-devel alsa-lib-devel libvorbis-devel ffmpeg-devel>=2.1
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ncurses-devel"
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do_configure() {
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