Author: sagitter Update of /cvs/free/rpms/moc/F-19 In directory old02.ovh.rpmfusion.lan:/tmp/cvs-serv17395/F-19 Modified Files: moc.spec Added Files: moc+warnings.patch moc-r2506:2526-samplesize.patch Log Message: moc+warnings.patch: audio.c | 2 +- decoder_plugins/ffmpeg/ffmpeg.c | 4 ++++ 2 files changed, 5 insertions(+), 1 deletion(-) --- NEW FILE moc+warnings.patch --- diff --git a/audio.c b/audio.c index 5ae0e1e..863238f 100644 --- a/audio.c +++ b/audio.c @@ -883,7 +883,7 @@ static void find_working_driver (lists_t_strs *drivers, struct hw_funcs *funcs) { int ix; - memset (funcs, 0, sizeof(funcs)); + memset (funcs, 0, sizeof(*funcs)); for (ix = 0; ix < lists_strs_size (drivers); ix += 1) { const char *name; diff --git a/decoder_plugins/ffmpeg/ffmpeg.c b/decoder_plugins/ffmpeg/ffmpeg.c index a85fd82..e8f0b91 100644 --- a/decoder_plugins/ffmpeg/ffmpeg.c +++ b/decoder_plugins/ffmpeg/ffmpeg.c @@ -301,6 +301,10 @@ static int locking_cb (void **mutex, enum AVLockOp op) free (*mutex); *mutex = NULL; break; + default: + /* Catch any FFmpeg/LibAV API changes. */ + assert (0); + result = 0; } return result; moc-r2506:2526-samplesize.patch: THANKS | 7 + TODO | 2 audio.c | 2 audio_conversion.c | 8 - configure.in | 6 - decoder_plugins/ffmpeg/ffmpeg.c | 190 +++++++++++++++++++++++++++++++-------- decoder_plugins/ffmpeg/ffmpeg.m4 | 19 +++ decoder_plugins/flac/flac.c | 14 +- decoder_plugins/mp3/mp3.c | 22 ++-- decoder_plugins/vorbis/vorbis.c | 25 ++--- io.c | 10 +- menu.c | 9 + mocp.1 | 5 + utf8.c | 63 ++++++------ 14 files changed, 270 insertions(+), 112 deletions(-) --- NEW FILE moc-r2506:2526-samplesize.patch --- Index: THANKS =================================================================== --- moc-2.5.0-beta1/THANKS (revision 2506) +++ moc-2.5.0-beta1/THANKS (revision 2526) @@ -10,6 +10,9 @@ * Fixed miscellaneous coding errors. * Assisted with testing. +Gregory Maxwell: + * Provided patch for off_t in io_* functions. + Morten Grunnet Buhl: * Provided Yellow/Red theme. @@ -44,6 +47,7 @@ * Headed the effort to port MOC to OpenWRT. * Provided signficant information on DTS, AAC and other formats. * Also contributed much time in the refinement of feature ideas. + * Provided 24-bit format conversion bug fixes. Juho Hämäläinen: * Added -Q (--format) FORMAT_STRING option to display file information. @@ -202,6 +206,9 @@ * Fixed segfault when trying to play a file using FFmpeg. * Migrated to newer FFmpeg API. +"firejox" + * Fixed screen upsets due to UTF-8 character handing. + "fluxid": * Fixed incorrect setting for themes red channel value. Index: mocp.1 =================================================================== --- moc-2.5.0-beta1/mocp.1 (revision 2506) +++ moc-2.5.0-beta1/mocp.1 (revision 2526) @@ -69,6 +69,11 @@ set in the configuration file as \fBStartInMusicDir\fP. .LP .TP +\fB\-q\fP, \fB\-\-enqueue\fP +Add files given after command line options to the queue. Don't start the +interface. +.LP +.TP \fB\-a\fP, \fB\-\-append\fP Append files, directories (recursively) and playlists given after command line options to the playlist. Don't start the interface. Index: menu.c =================================================================== --- moc-2.5.0-beta1/menu.c (revision 2506) +++ moc-2.5.0-beta1/menu.c (revision 2526) @@ -90,9 +90,14 @@ getyx (menu->win, y, x); if (title_width <= title_space || mi->align == MENU_ALIGN_LEFT) xwaddnstr (menu->win, mi->title, title_space); - else - xwaddstr (menu->win, mi->title + title_width - title_space); + else { + char *ptr; + ptr = xstrtail (mi->title, title_space); + xwaddstr (menu->win, ptr); + free (ptr); + } + /* Fill the remainder of the title field with spaces. */ if (mi == menu->selected) { getyx (menu->win, y, ix); Index: configure.in =================================================================== --- moc-2.5.0-beta1/configure.in (revision 2506) +++ moc-2.5.0-beta1/configure.in (revision 2526) @@ -142,7 +142,7 @@ PKG_PROG_PKG_CONFIG([0.20]) -if test "x$PKG_CONFIG" == "x" +if test "x$PKG_CONFIG" = "x" then AC_MSG_WARN([No pkg-config utility found or it's too old, I will have trouble finding installed libraries.]) fi @@ -293,7 +293,6 @@ dnl optional functions AC_CHECK_FUNCS([strcasestr strerror_r syslog]) -AC_CHECK_FUNCS([getrlimit pthread_attr_getstacksize]) AX_CHECK_UNAME_SYSCALL dnl MIME magic @@ -322,6 +321,7 @@ CC="$PTHREAD_CC" CFLAGS="$PTHREAD_CFLAGS $CFLAGS" EXTRA_LIBS="$EXTRA_LIBS $PTHREAD_LIBS" +AC_CHECK_FUNCS([getrlimit pthread_attr_getstacksize]) dnl __FUNCTION__ AC_TRY_COMPILE(,[printf(__FUNCTION__);], [AC_DEFINE([HAVE__FUNCTION__], 1, @@ -342,7 +342,7 @@ fi dnl popt -AC_SEARCH_LIBS([poptGetContext], [popt], , [POPT_MISSING="yes"]) +AC_CHECK_LIB([popt], [poptGetContext], [true], [POPT_MISSING="yes"]) dnl getopt AC_CHECK_FUNC(getopt_long, Index: utf8.c =================================================================== --- moc-2.5.0-beta1/utf8.c (revision 2506) +++ moc-2.5.0-beta1/utf8.c (revision 2526) @@ -199,47 +199,50 @@ int xwaddnstr (WINDOW *win, const char *str, const int n) { - int res; + int res, width, inv_char; + wchar_t *ucs; + char *mstr, *lstr; + size_t size, num_chars; assert (n > 0); assert (str != NULL); - if (using_utf8) { + mstr = iconv_str (iconv_desc, str); - /* This nasty hack is because we need to count n in chars, but - * [w]addnstr() takes arguments in bytes (in UTF-8 a char can be - * longer than 1 byte). There are also problems with [w]addnwstr() - * (screen garbled). I have no better idea. */ + size = xmbstowcs (NULL, mstr, -1, NULL) + 1; + ucs = (wchar_t *)xmalloc (sizeof(wchar_t) * size); + xmbstowcs (ucs, mstr, size, &inv_char); + width = wcswidth (ucs, WIDTH_MAX); - wchar_t *ucs; - size_t size; - size_t utf_num_chars; - int inv_char; - - size = xmbstowcs (NULL, str, -1, NULL) + 1; - ucs = (wchar_t *)xmalloc (sizeof(wchar_t) * size); - xmbstowcs (ucs, str, size, &inv_char); - if ((size_t)n < size - 1) - ucs[n] = L'\0'; - utf_num_chars = wcstombs (NULL, ucs, 0); - if (inv_char) { - char *utf8 = (char *)xmalloc (utf_num_chars + 1); - - wcstombs (utf8, ucs, utf_num_chars + 1); - res = waddstr (win, utf8); - free (utf8); + if (width == -1) { + size_t clidx; + for (clidx = 0; clidx < size - 1; clidx++) { + if (wcwidth (ucs[clidx]) == -1) + ucs[clidx] = L'?'; } - else - res = waddnstr (win, str, utf_num_chars); - free (ucs); + width = wcswidth (ucs, WIDTH_MAX); + inv_char = 1; } - else { - char *lstr = iconv_str (iconv_desc, str); - res = waddnstr (win, lstr, n); - free (lstr); + if (width > n) { + while (width > n) + width -= wcwidth (ucs[--size]); + ucs[size] = L'\0'; } + num_chars = wcstombs (NULL, ucs, 0); + lstr = (char *)xmalloc (num_chars + 1); + + if (inv_char) + wcstombs (lstr, ucs, num_chars + 1); + else + snprintf (lstr, num_chars + 1, "%s", mstr); + + res = waddstr (win, lstr); + + free (ucs); + free (lstr); + free (mstr); return res; } Index: audio_conversion.c =================================================================== --- moc-2.5.0-beta1/audio_conversion.c (revision 2506) +++ moc-2.5.0-beta1/audio_conversion.c (revision 2526) @@ -189,7 +189,7 @@ size_t i; /* maximum and minimum values of 32-bit samples */ - const unsigned int U32_MAX = (1 << 24); + const unsigned int U32_MAX = (1 << 24) - 1; const int S32_MAX = (1 << 23) - 1; const int S32_MIN = -(1 << 23); @@ -201,7 +201,7 @@ float f = in[i] * S32_MAX; if (f >= S32_MAX) - *out_val = U32_MAX; + *out_val = U32_MAX << 8; else if (f <= S32_MIN) *out_val = 0; else { @@ -231,9 +231,9 @@ float f = in[i] * S32_MAX; if (f >= S32_MAX) - *out_val = S32_MAX; + *out_val = S32_MAX << 8; else if (f <= S32_MIN) - *out_val = S32_MIN; + *out_val = S32_MIN << 8; else { #ifdef HAVE_LRINTF *out_val = lrintf(f) << 8; Index: io.c =================================================================== --- moc-2.5.0-beta1/io.c (revision 2506) +++ moc-2.5.0-beta1/io.c (revision 2526) @@ -141,20 +141,20 @@ } #ifdef HAVE_MMAP -static off_t io_seek_mmap (struct io_stream *s, const long where) +static off_t io_seek_mmap (struct io_stream *s, const off_t where) { - assert (RANGE(0, where, (long)s->size)); + assert (RANGE(0, where, (off_t)s->size)); return (s->mem_pos = where); } #endif -static off_t io_seek_fd (struct io_stream *s, const int where) +static off_t io_seek_fd (struct io_stream *s, const off_t where) { return lseek (s->fd, where, SEEK_SET); } -static off_t io_seek_buffered (struct io_stream *s, const long where) +static off_t io_seek_buffered (struct io_stream *s, const off_t where) { off_t res = -1; @@ -180,7 +180,7 @@ return res; } -static off_t io_seek_unbuffered (struct io_stream *s, const long where) +static off_t io_seek_unbuffered (struct io_stream *s, const off_t where) { off_t res = -1; Index: decoder_plugins/mp3/mp3.c =================================================================== --- moc-2.5.0-beta1/decoder_plugins/mp3/mp3.c (revision 2506) +++ moc-2.5.0-beta1/decoder_plugins/mp3/mp3.c (revision 2526) @@ -696,17 +696,19 @@ { char *ext; + strcpy (buf, "MPx"); + ext = ext_pos (file); - if (!strcasecmp (ext, "mp3")) - strcpy (buf, "MP3"); - else if (!strcasecmp (ext, "mp2")) - strcpy (buf, "MP2"); - else if (!strcasecmp (ext, "mp1")) - strcpy (buf, "MP1"); - else if (!strcasecmp (ext, "mpga")) - strcpy (buf, "MPG"); - else - strcpy (buf, "MPx"); + if (ext) { + if (!strcasecmp (ext, "mp3")) + strcpy (buf, "MP3"); + else if (!strcasecmp (ext, "mp2")) + strcpy (buf, "MP2"); + else if (!strcasecmp (ext, "mp1")) + strcpy (buf, "MP1"); + else if (!strcasecmp (ext, "mpga")) + strcpy (buf, "MPG"); + } } static int mp3_our_format_ext (const char *ext) Index: decoder_plugins/ffmpeg/ffmpeg.m4 =================================================================== --- moc-2.5.0-beta1/decoder_plugins/ffmpeg/ffmpeg.m4 (revision 2506) +++ moc-2.5.0-beta1/decoder_plugins/ffmpeg/ffmpeg.m4 (revision 2526) @@ -88,6 +88,25 @@ [#include <libavcodec/avcodec.h>]) AC_CHECK_DECLS([AV_CODEC_ID_OPUS], , , [#include <libavcodec/avcodec.h>]) + AC_SEARCH_LIBS(avcodec_free_frame, avcodec, + [AC_DEFINE([HAVE_AVCODEC_FREE_FRAME], 1, + [Define to 1 if you have the `avcodec_free_frame' function.])]) + AC_CHECK_DECLS([CODEC_ID_PCM_S8_PLANAR], , , + [#include <libavcodec/avcodec.h>]) + AC_CHECK_DECLS([AV_SAMPLE_FMT_U8P], , , + [#include <libavcodec/avcodec.h>]) + AC_CHECK_DECLS([AV_SAMPLE_FMT_S16P], , , + [#include <libavcodec/avcodec.h>]) + AC_CHECK_DECLS([AV_SAMPLE_FMT_S32P], , , + [#include <libavcodec/avcodec.h>]) + AC_CHECK_DECLS([AV_SAMPLE_FMT_FLTP], , , + [#include <libavcodec/avcodec.h>]) + AC_SEARCH_LIBS(av_get_sample_fmt_name, avutil, + [AC_DEFINE([HAVE_AV_GET_SAMPLE_FMT_NAME], 1, + [Define to 1 if you have the `av_get_sample_fmt_name' function.])]) + AC_SEARCH_LIBS(av_lockmgr_register, avcodec, + [AC_DEFINE([HAVE_LOCKMGR_REGISTER], 1, + [Define to 1 if you have the `av_lockmgr_register' function.])]) CPPFLAGS="$save_CPPFLAGS" CFLAGS="$save_CFLAGS" LIBS="$save_LIBS" Index: decoder_plugins/ffmpeg/ffmpeg.c =================================================================== --- moc-2.5.0-beta1/decoder_plugins/ffmpeg/ffmpeg.c (revision 2506) +++ moc-2.5.0-beta1/decoder_plugins/ffmpeg/ffmpeg.c (revision 2526) @@ -99,6 +99,7 @@ bool okay; /* was this stream successfully opened? */ struct decoder_error error; long fmt; + int sample_width; int bitrate; /* in bits per second */ int avg_bitrate; /* in bits per second */ #if SEEK_IN_DECODER @@ -106,6 +107,7 @@ int seek_sec; /* second to which to seek */ #endif bool seek_broken; /* FFmpeg seeking is broken */ + bool timing_broken; /* FFmpeg trashes duration and bit_rate */ #if SEEK_IN_DECODER && defined(DEBUG) pthread_t thread_id; #endif @@ -277,8 +279,72 @@ } } +/* Handle FFmpeg's locking requirements. */ +#ifdef HAVE_LOCKMGR_REGISTER +static int locking_cb (void **mutex, enum AVLockOp op) +{ + int result; + + switch (op) { + case AV_LOCK_CREATE: + *mutex = xmalloc (sizeof (pthread_mutex_t)); + result = pthread_mutex_init (*mutex, NULL); + break; + case AV_LOCK_OBTAIN: + result = pthread_mutex_lock (*mutex); + break; + case AV_LOCK_RELEASE: + result = pthread_mutex_unlock (*mutex); + break; + case AV_LOCK_DESTROY: + result = pthread_mutex_destroy (*mutex); + free (*mutex); + *mutex = NULL; + break; + } + + return result; +} +#endif + +/* Here we attempt to determine if FFmpeg/LibAV has trashed the 'duration' + * and 'bit_rate' fields in AVFormatContext for large files. Determining + * whether or not they are likely to be valid is imprecise and will vary + * depending (at least) on: + * + * - The file's size, + * - The file's codec, + * - The number and size of tags, + * - The version of FFmpeg/LibAV, and + * - Whether it's FFmpeg or LibAV. + * + * This function represents a best guess. +*/ +static bool is_timing_broken (AVFormatContext *ic) +{ + int64_t file_size; + + if (ic->duration < 0 || ic->bit_rate < 0) + return true; + +#ifdef HAVE_AVIO_SIZE + file_size = avio_size (ic->pb); +#else + file_size = ic->file_size; +#endif + + if (file_size < UINT32_MAX) + return false; + + return true; +} + static void ffmpeg_init () { +#ifdef HAVE_LOCKMGR_REGISTER + int rc; +#endif + #ifdef DEBUG av_log_set_level (AV_LOG_INFO); #else @@ -291,10 +357,20 @@ supported_extns = lists_strs_new (16); load_audio_extns (supported_extns); load_video_extns (supported_extns); + +#ifdef HAVE_LOCKMGR_REGISTER + rc = av_lockmgr_register (locking_cb); + if (rc < 0) + fatal ("Lock manager initialisation failed"); +#endif } static void ffmpeg_destroy () { +#ifdef HAVE_LOCKMGR_REGISTER + av_lockmgr_register (NULL); +#endif + av_log_set_level (AV_LOG_QUIET); ffmpeg_log_repeats (NULL); @@ -341,9 +417,9 @@ } #endif - if (tags_sel & TAGS_TIME) { + if (!is_timing_broken (ic) && tags_sel & TAGS_TIME) { info->time = -1; - if (ic->duration >= 0) + if (ic->duration != (int64_t)AV_NOPTS_VALUE && ic->duration >= 0) info->time = ic->duration / AV_TIME_BASE; } @@ -426,12 +502,6 @@ #endif - if (tags_sel & TAGS_TIME) { - info->time = -1; - if (ic->duration != (int64_t)AV_NOPTS_VALUE && ic->duration >= 0) - info->time = ic->duration / AV_TIME_BASE; - } - end: #ifdef HAVE_AVFORMAT_CLOSE_INPUT avformat_close_input (&ic); @@ -450,12 +520,16 @@ if (!strcmp (data->ic->iformat->name, "wav")) { switch (data->enc->codec_id) { case CODEC_ID_PCM_S8: +#if HAVE_DECL_CODEC_ID_PCM_S8_PLANAR + case CODEC_ID_PCM_S8_PLANAR: +#endif result = SFMT_S8; break; case CODEC_ID_PCM_U8: result = SFMT_U8; break; case CODEC_ID_PCM_S16LE: + case CODEC_ID_PCM_S16LE_PLANAR: case CODEC_ID_PCM_S16BE: result = SFMT_S16; break; @@ -490,15 +564,27 @@ switch (data->enc->sample_fmt) { case AV_SAMPLE_FMT_U8: +#if HAVE_DECL_AV_SAMPLE_FMT_U8P + case AV_SAMPLE_FMT_U8P: +#endif result = SFMT_U8; break; case AV_SAMPLE_FMT_S16: +#if HAVE_DECL_AV_SAMPLE_FMT_S16P + case AV_SAMPLE_FMT_S16P: +#endif result = SFMT_S16; break; case AV_SAMPLE_FMT_S32: +#if HAVE_DECL_AV_SAMPLE_FMT_S32P + case AV_SAMPLE_FMT_S32P: +#endif result = SFMT_S32; break; case AV_SAMPLE_FMT_FLT: +#if HAVE_DECL_AV_SAMPLE_FMT_FLTP + case AV_SAMPLE_FMT_FLTP: +#endif result = SFMT_FLOAT; break; default: @@ -597,6 +683,7 @@ data->stream = NULL; data->enc = NULL; data->codec = NULL; + data->sample_width = 0; data->bitrate = 0; data->avg_bitrate = 0; @@ -624,6 +711,7 @@ data->seek_sec = 0; #endif data->seek_broken = false; + data->timing_broken = false; decoder_error_init (&data->error); @@ -702,27 +790,38 @@ if (data->fmt == 0) data->fmt = fmt_from_sample_fmt (data); if (data->fmt == 0) { +#ifdef HAVE_AV_GET_SAMPLE_FMT_NAME decoder_error (&data->error, ERROR_FATAL, 0, + "Cannot get sample size from unknown sample format: %s", + av_get_sample_fmt_name (data->enc->sample_fmt)); +#else + decoder_error (&data->error, ERROR_FATAL, 0, "Unsupported sample size!"); +#endif + avcodec_close (data->enc); goto end; } + data->sample_width = sfmt_Bps (data->fmt); if (data->codec->capabilities & CODEC_CAP_DELAY) data->delay = true; data->seek_broken = is_seek_broken (data); + data->timing_broken = is_timing_broken (data->ic); data->okay = true; - if (data->ic->duration >= AV_TIME_BASE) { + if (!data->timing_broken && data->ic->duration >= AV_TIME_BASE) { #ifdef HAVE_AVIO_SIZE data->avg_bitrate = (int) (avio_size (data->ic->pb) / - (data->ic->duration / AV_TIME_BASE) * 8); + (data->ic->duration / AV_TIME_BASE) * 8); #else data->avg_bitrate = (int) (data->ic->file_size / - (data->ic->duration / AV_TIME_BASE) * 8); + (data->ic->duration / AV_TIME_BASE) * 8); #endif } - data->bitrate = data->ic->bit_rate; + if (!data->timing_broken && data->ic->bit_rate > 0) + data->bitrate = data->ic->bit_rate; + return data; end: @@ -943,12 +1042,14 @@ char *buf, int buf_len) { int filled = 0; + AVFrame *frame; + frame = avcodec_alloc_frame (); + do { int len, got_frame, is_planar, plane_size, data_size, copied; - AVFrame frame; - len = avcodec_decode_audio4 (data->enc, &frame, &got_frame, pkt); + len = avcodec_decode_audio4 (data->enc, frame, &got_frame, pkt); if (len < 0) { /* skip frame */ @@ -968,33 +1069,46 @@ is_planar = av_sample_fmt_is_planar (data->enc->sample_fmt); data_size = av_samples_get_buffer_size (&plane_size, - data->enc->channels, frame.nb_samples, + data->enc->channels, + frame->nb_samples, data->enc->sample_fmt, 1); if (data_size == 0) continue; - copied = copy_or_buffer (data, (char *)frame.extended_data[0], - plane_size, buf, buf_len); - buf += copied; - filled += copied; - buf_len -= copied; + if (is_planar && data->enc->channels > 1) { + int offset, ch; - if (is_planar && data->enc->channels > 1) { - int ch; + for (offset = 0; offset < plane_size; offset += data->sample_width) { + for (ch = 0; ch < data->enc->channels; ch += 1) { + copied = copy_or_buffer (data, + (char *)frame->extended_data[ch] + + offset, + data->sample_width, buf, buf_len); + buf += copied; + filled += copied; + buf_len -= copied; + } + } + } + else { + copied = copy_or_buffer (data, (char *)frame->extended_data[0], + plane_size, buf, buf_len); + buf += copied; + filled += copied; + buf_len -= copied; + } - for (ch = 1; ch < data->enc->channels; ch += 1) { - copied = copy_or_buffer (data, (char *)frame.extended_data[ch], - plane_size, buf, buf_len); - buf += copied; - filled += copied; - buf_len -= copied; - } - } - debug ("Copying %dB (%dB filled)", data_size, filled); } while (pkt->size > 0); + avcodec_get_frame_defaults (frame); +#ifdef HAVE_AVCODEC_FREE_FRAME + avcodec_free_frame (&frame); +#else + av_freep (&frame); +#endif + return filled; } #endif @@ -1122,9 +1236,10 @@ free_packet (pkt); } while (!bytes_produced && !data->eos); - data->bitrate = compute_bitrate (sound_params, bytes_used, - bytes_produced + data->remain_buf_len, - data->bitrate); + if (!data->timing_broken) + data->bitrate = compute_bitrate (sound_params, bytes_used, + bytes_produced + data->remain_buf_len, + data->bitrate); return bytes_produced; } @@ -1185,20 +1300,23 @@ { struct ffmpeg_data *data = (struct ffmpeg_data *)prv_data; - return data->bitrate / 1000; + return data->timing_broken ? -1 : data->bitrate / 1000; } static int ffmpeg_get_avg_bitrate (void *prv_data) { struct ffmpeg_data *data = (struct ffmpeg_data *)prv_data; - return data->avg_bitrate / 1000; + return data->timing_broken ? -1 : data->avg_bitrate / 1000; } static int ffmpeg_get_duration (void *prv_data) { struct ffmpeg_data *data = (struct ffmpeg_data *)prv_data; + if (data->timing_broken) + return -1; + if (!data->stream) return -1; Index: decoder_plugins/vorbis/vorbis.c =================================================================== --- moc-2.5.0-beta1/decoder_plugins/vorbis/vorbis.c (revision 2506) +++ moc-2.5.0-beta1/decoder_plugins/vorbis/vorbis.c (revision 2526) @@ -47,9 +47,9 @@ /* Tremor defines time as 64-bit integer milliseconds. */ #ifndef HAVE_TREMOR -static const double time_scaler = 1; +static const int64_t time_scaler = 1; #else -static const ogg_int64_t time_scaler = 1000; +static const int64_t time_scaler = 1000; #endif struct vorbis_data @@ -171,11 +171,11 @@ get_comment_tags (&vf, info); if (tags_sel & TAGS_TIME) { - int vorbis_time; + int64_t vorbis_time; - vorbis_time = ov_time_total (&vf, -1) / time_scaler; - if (vorbis_time >= 0) - info->time = vorbis_time; + vorbis_time = ov_time_total (&vf, -1); + if (vorbis_time >= 0) + info->time = vorbis_time / time_scaler; } ov_clear (&vf); @@ -244,12 +244,15 @@ io_close (data->stream); } else { + int64_t duration; + data->last_section = -1; data->avg_bitrate = ov_bitrate (&data->vf, -1) / 1000; data->bitrate = data->avg_bitrate; - data->duration = ov_time_total (&data->vf, -1) / time_scaler; - if (data->duration == OV_EINVAL) - data->duration = -1; + data->duration = -1; + duration = ov_time_total (&data->vf, -1); + if (duration >= 0) + data->duration = duration / time_scaler; data->ok = 1; get_comment_tags (&data->vf, data->tags); } @@ -280,9 +283,9 @@ static int vorbis_can_decode (struct io_stream *stream) { - char buf[34]; + char buf[35]; - if (io_peek (stream, buf, 34) == 34 && !memcmp (buf, "OggS", 4) + if (io_peek (stream, buf, 35) == 35 && !memcmp (buf, "OggS", 4) && !memcmp (buf + 28, "\01vorbis", 7)) return 1; Index: decoder_plugins/flac/flac.c =================================================================== --- moc-2.5.0-beta1/decoder_plugins/flac/flac.c (revision 2506) +++ moc-2.5.0-beta1/decoder_plugins/flac/flac.c (revision 2526) @@ -54,7 +54,7 @@ int abort; /* abort playing (due to an error) */ unsigned length; - unsigned total_samples; + FLAC__uint64 total_samples; FLAC__byte sample_buffer[SAMPLE_BUFFER_SIZE]; unsigned sample_buffer_fill; @@ -156,11 +156,8 @@ if (metadata->type == FLAC__METADATA_TYPE_STREAMINFO) { debug ("Got metadata info"); - data->total_samples = - (unsigned)(metadata->data.stream_info.total_samples - & 0xffffffff); - data->bits_per_sample = - metadata->data.stream_info.bits_per_sample; + data->total_samples = metadata->data.stream_info.total_samples; + data->bits_per_sample = metadata->data.stream_info.bits_per_sample; data->channels = metadata->data.stream_info.channels; data->sample_rate = metadata->data.stream_info.sample_rate; data->length = data->total_samples / data->sample_rate; @@ -536,9 +533,10 @@ if ((unsigned)sec > data->length) return -1; - target_sample = (FLAC__uint64)((sec/(double)data->length) * - (double)data->total_samples); + target_sample = (FLAC__uint64)(((double)sec / (double)data->length) * + (double)data->total_samples); + #ifdef LEGACY_FLAC if (FLAC__seekable_stream_decoder_seek_absolute(data->decoder, target_sample)) Index: audio.c =================================================================== --- moc-2.5.0-beta1/audio.c (revision 2506) +++ moc-2.5.0-beta1/audio.c (revision 2526) @@ -169,7 +169,7 @@ if (fmt1 & (SFMT_S16 | SFMT_U16) && fmt2 & (SFMT_S16 | SFMT_U16)) return 1; - if (fmt1 & (SFMT_S8 | SFMT_U8) + if (fmt1 & (SFMT_S32 | SFMT_U32) && fmt2 & (SFMT_S32 | SFMT_U32)) return 1; if (fmt1 & fmt2 & SFMT_FLOAT) Index: TODO =================================================================== --- moc-2.5.0-beta1/TODO (revision 2506) +++ moc-2.5.0-beta1/TODO (revision 2526) @@ -9,8 +9,6 @@ - Recognition of ID3 tags v1 is broken (example: small.mp3). [node/234] - Perhaps MOC can add support for the frame field_type to differentiate between ID3_FIELD_TYPE_LATIN1 and ID3_FIELD_TYPE_STRING. [node/234] - - Characters which are represented in more than one screen column cause - the text to overflow its field. [node/82, 99 and 788] - Some Cyrillic characters in file and directory names don't get displayed correctly. * Program crashes: Index: moc.spec =================================================================== RCS file: /cvs/free/rpms/moc/F-19/moc.spec,v retrieving revision 1.1 retrieving revision 1.2 diff -u -r1.1 -r1.2 --- moc.spec 8 Jun 2013 18:47:21 -0000 1.1 +++ moc.spec 18 Jun 2013 15:14:24 -0000 1.2 @@ -4,7 +4,7 @@ Name: moc Summary: Music on Console - Console audio player for Linux/UNIX Version: 2.5.0 -Release: 0.9.beta1%{?dist} +Release: 0.10.beta1%{?dist} License: GPLv2+ and GPLv3+ URL: http://www.moc.daper.net Source0: http://ftp.daper.net/pub/soft/moc/unstable/%{name}-%{version}-beta1.tar.bz2 @@ -12,6 +12,14 @@ ## This patch corrects all outdated FSF address Patch0: %{name}-r2506+fsf_addr.patch +## This patchset corrects "Unsupported sample size!" error +## See http://moc.daper.net/node/862 +Patch1: %{name}-r2506:2526-samplesize.patch + +## This patch corrects 'sizeof' argument bug; +## directly provided from upstream +Patch2: %{name}+warnings.patch + BuildRequires: pkgconfig(ncurses) BuildRequires: pkgconfig(alsa) BuildRequires: pkgconfig(jack) @@ -36,6 +44,9 @@ BuildRequires: libtool BuildRequires: librcc-devel +## Source code configuring tools +BuildRequires: autoconf, automake >= 1.13 + Requires: ffmpeg Requires: opus @@ -47,9 +58,19 @@ %prep %setup -q -n %{name}-%{version}-beta1 -%patch0 -p1 -b %{name}-r2506+fsf_addr.patch +%patch0 -p1 +%patch1 -p1 +%patch2 -p1 + +## This renaming is requested by Automake-1.13 +mv configure.in configure.ac %build + +## Latest patchset changes ffmpeg.m4 file +## Autoreconf is temporarily necessary +autoreconf -i --force + %configure --disable-static --with-rcc \ --with-oss --with-alsa --with-jack --with-aac --with-mp3 \ --with-musepack --with-vorbis --with-flac --with-wavpack \ @@ -76,6 +97,13 @@ %{_libdir}/%{name}/decoder_plugins %changelog +* Tue Jun 18 2013 Antonio Trande <sagitter(a)fedoraproject.org&gt; 2.5.0-0.10.beta1 +- Added patchset to fix "Unsupported sample size!" error + See http://moc.daper.net/node/862 for more details +- Added patch for 'sizeof' argument bug +- Added BR: Autoconf and Automake-1.13 (temporarily) +- 'configure.in' renaming + * Sat Jun 08 2013 Antonio Trande <sagitter(a)fedoraproject.org&gt; 2.5.0-0.9.beta1 - Removed some explicit Requires (curl, jack-audio-connection-kit, ncurses, speex)