diff options
Diffstat (limited to 'media-sound/xmms2/files/xmms2-0.8-audio4-p2.patch')
| -rw-r--r-- | media-sound/xmms2/files/xmms2-0.8-audio4-p2.patch | 171 |
1 files changed, 171 insertions, 0 deletions
diff --git a/media-sound/xmms2/files/xmms2-0.8-audio4-p2.patch b/media-sound/xmms2/files/xmms2-0.8-audio4-p2.patch new file mode 100644 index 000000000000..46b5b1d3b926 --- /dev/null +++ b/media-sound/xmms2/files/xmms2-0.8-audio4-p2.patch @@ -0,0 +1,171 @@ +commit b614459dc1ea353d6c24b4a77c7f92a5577d5bc3 +Author: Uli Franke <cls@nebadje.org> +Date: Thu Jan 19 11:53:57 2012 +0100 + + BUG(2510): Add more bitrates/samplerates to AAC/ALAC. + +diff --git a/src/plugins/avcodec/avcodec.c b/src/plugins/avcodec/avcodec.c +index 1b4a659..b32de4d 100644 +--- a/src/plugins/avcodec/avcodec.c ++++ b/src/plugins/avcodec/avcodec.c +@@ -60,6 +60,7 @@ static gint xmms_avcodec_read (xmms_xform_t *xform, xmms_sample_t *buf, gint len + xmms_error_t *error); + static gint64 xmms_avcodec_seek (xmms_xform_t *xform, gint64 samples, + xmms_xform_seek_mode_t whence, xmms_error_t *err); ++static xmms_sample_format_t xmms_avcodec_translate_sample_format (enum AVSampleFormat av_sample_format); + + /* + * Plugin header +@@ -168,12 +169,12 @@ xmms_avcodec_init (xmms_xform_t *xform) + data->channels = ret; + } + +- /* bitrate required for WMA files */ ++ /* Required by WMA xform. */ + xmms_xform_auxdata_get_int (xform, + "bitrate", + &data->bitrate); + +- /* ALAC and MAC require bits per sample field to be 16 */ ++ /* Required by tta and apefile xforms. */ + xmms_xform_auxdata_get_int (xform, + "samplebits", + &data->samplebits); +@@ -238,12 +239,17 @@ xmms_avcodec_init (xmms_xform_t *xform) + + data->samplerate = data->codecctx->sample_rate; + data->channels = data->codecctx->channels; ++ data->sampleformat = xmms_avcodec_translate_sample_format (data->codecctx->sample_fmt); ++ if (data->sampleformat == XMMS_SAMPLE_FORMAT_UNKNOWN) { ++ avcodec_close (data->codecctx); ++ goto err; ++ } + + xmms_xform_outdata_type_add (xform, + XMMS_STREAM_TYPE_MIMETYPE, + "audio/pcm", + XMMS_STREAM_TYPE_FMT_FORMAT, +- XMMS_SAMPLE_FORMAT_S16, ++ data->sampleformat, + XMMS_STREAM_TYPE_FMT_CHANNELS, + data->channels, + XMMS_STREAM_TYPE_FMT_SAMPLERATE, +@@ -428,3 +434,23 @@ xmms_avcodec_seek (xmms_xform_t *xform, gint64 samples, xmms_xform_seek_mode_t w + + return ret; + } ++ ++static xmms_sample_format_t ++xmms_avcodec_translate_sample_format (enum AVSampleFormat av_sample_format) ++{ ++ switch (av_sample_format) { ++ case AV_SAMPLE_FMT_U8: ++ return XMMS_SAMPLE_FORMAT_U8; ++ case AV_SAMPLE_FMT_S16: ++ return XMMS_SAMPLE_FORMAT_S16; ++ case AV_SAMPLE_FMT_S32: ++ return XMMS_SAMPLE_FORMAT_S32; ++ case AV_SAMPLE_FMT_FLT: ++ return XMMS_SAMPLE_FORMAT_FLOAT; ++ case AV_SAMPLE_FMT_DBL: ++ return XMMS_SAMPLE_FORMAT_DOUBLE; ++ default: ++ XMMS_DBG ("AVSampleFormat (%i) not supported.", av_sample_format); ++ return XMMS_SAMPLE_FORMAT_UNKNOWN; ++ } ++} +diff --git a/src/plugins/mp4/mp4.c b/src/plugins/mp4/mp4.c +index 7c915c4..3ee9357 100644 +--- a/src/plugins/mp4/mp4.c ++++ b/src/plugins/mp4/mp4.c +@@ -186,9 +186,6 @@ xmms_mp4_init (xmms_xform_t *xform) + xmms_xform_auxdata_set_bin (xform, "decoder_config", tmpbuf, tmpbuflen); + g_free (tmpbuf); + +- /* This is only for ALAC to set 16-bit samples, ignored for AAC */ +- xmms_xform_auxdata_set_int (xform, "samplebits", 16); +- + xmms_mp4_get_mediainfo (xform); + + XMMS_DBG ("MP4 demuxer inited successfully!"); +@@ -288,7 +285,7 @@ xmms_mp4_get_mediainfo (xmms_xform_t *xform) + data = xmms_xform_private_data_get (xform); + g_return_if_fail (data); + +- if ((temp = mp4ff_get_sample_rate (data->mp4ff, data->track)) >= 0) { ++ if ((temp = mp4ff_get_sample_rate (data->mp4ff, data->track)) > 0) { + glong srate = temp; + + if ((temp = mp4ff_get_track_duration_use_offsets (data->mp4ff, +@@ -492,7 +489,7 @@ xmms_mp4_get_track (xmms_xform_t *xform, mp4ff_t *infile) + case 0x69: /* MPEG-2 audio */ + case 0x6B: /* MPEG-1 audio */ + continue; +- case 0xff: ++ case 0xff: /* ALAC */ + chans = mp4ff_get_channel_count (infile, i); + rate = mp4ff_get_sample_rate (infile, i); + if (chans <= 0 || rate <= 0) { +diff --git a/src/plugins/mp4/mp4ff/README.xmms2 b/src/plugins/mp4/mp4ff/README.xmms2 +index c2737c5..8021618 100644 +--- a/src/plugins/mp4/mp4ff/README.xmms2 ++++ b/src/plugins/mp4/mp4ff/README.xmms2 +@@ -12,3 +12,4 @@ Changes: + * Add value_length variable to tag type and use it when adding new item-value pairs, + necessary for cover art since it's binary data and can't be handled as a string + * Add support for Apple Lossless audio files ++ * Add a workaround for supporting higher samplerates. +diff --git a/src/plugins/mp4/mp4ff/mp4ff.c b/src/plugins/mp4/mp4ff/mp4ff.c +index ee7f7fb..b6f0a37 100644 +--- a/src/plugins/mp4/mp4ff/mp4ff.c ++++ b/src/plugins/mp4/mp4ff/mp4ff.c +@@ -32,6 +32,8 @@ + #include <string.h> + #include "mp4ffint.h" + ++static uint32_t mp4ff_normalize_flawed_sample_rate (uint16_t samplerate); ++ + mp4ff_t *mp4ff_open_read(mp4ff_callback_t *f) + { + mp4ff_t *ff = malloc(sizeof(mp4ff_t)); +@@ -304,12 +306,39 @@ int32_t mp4ff_num_samples(const mp4ff_t *f, const int32_t track) + return total; + } + ++static uint32_t ++mp4ff_normalize_flawed_sample_rate (uint16_t samplerate) ++{ ++ /* A list of common rates can be found at ++ * http://en.wikipedia.org/wiki/Sampling_rate */ ++ uint32_t rates[] = {8000, 11025, 16000, 22050, 32000, 44056, 44100, ++ 47250, 48000, 50000, 50400, 88200, 96000, 176400, ++ 192000, 352800, 384000, 0}; ++ uint32_t* rate; ++ ++ /* First check standard rates. */ ++ for (rate = rates; *rate; rate++) { ++ if (*rate == samplerate) { ++ return *rate; ++ } ++ } ++ ++ /* No standard rates matching - check if sample rate got truncated when ++ * added to MP4 container */ ++ for (rate = rates; *rate; rate++) { ++ if ((*rate & 0x0000FFFF) == samplerate) { ++ return *rate; ++ } ++ } + ++ /* Failed to find a standard rate - we give up returning the original rate */ ++ return samplerate; ++} + + + uint32_t mp4ff_get_sample_rate(const mp4ff_t *f, const int32_t track) + { +- return f->track[track]->sampleRate; ++ return mp4ff_normalize_flawed_sample_rate (f->track[track]->sampleRate); + } + + uint32_t mp4ff_get_channel_count(const mp4ff_t * f,const int32_t track) |