Author: adrian Update of /cvs/free/rpms/mpd/devel In directory se02.es.rpmfusion.net:/tmp/cvs-serv32137 Modified Files: .cvsignore mpd.spec sources Added Files: 6a071efa2794806ad5a2a62f0fcdee4b1843b41f.patch mpd.git-00503c9251141b427457c17a9677444bf29c3992.patch mpd.git-9e9d7b73d2165f197eeec12ee953add5f49746b7.patch mpd.git-f3a5b753ae053eb1a862343b0fd3d62973cacc18.patch Log Message: * Tue Aug 25 2009 Adrian Reber <adrian(a)lisas.de&gt; - 0.15.2 - updated to 0.15.2 - applied patches from David Woodhouse to fix "mpd fails to play to usb audio device" (#731) - fix description (#765) 6a071efa2794806ad5a2a62f0fcdee4b1843b41f.patch: --- NEW FILE 6a071efa2794806ad5a2a62f0fcdee4b1843b41f.patch --- From: Max Kellermann&lt;max(a)duempel.org&gt; Date: Wed, 22 Jul 2009 13:56:36 +0000 Subject: audio_format: initialize reverse_endian in audio_format_init() This line was missing in the reverse_endian patch, and led to undefined values and crashes in that attribute. --- diff --git a/src/audio_format.h b/src/audio_format.h index 54514ff..6373d1e 100644 --- a/src/audio_format.h +++ b/src/audio_format.h @@ -45,6 +45,7 @@ static inline void audio_format_init(struct audio_format *af, af->sample_rate = sample_rate; af->bits = bits; af->channels = channels; + af->reverse_endian = 0; } static inline bool audio_format_defined(const struct audio_format *af) -- cgit v0.8.1-8-g7b5c mpd.git-00503c9251141b427457c17a9677444bf29c3992.patch: --- NEW FILE mpd.git-00503c9251141b427457c17a9677444bf29c3992.patch --- From: David Woodhouse <David.Woodhouse(a)intel.com&gt; Date: Sun, 19 Jul 2009 16:43:08 +0100 Subject: [PATCH 1/1] Support wrong-endian ALSA output --- src/output/alsa_plugin.c | 52 ++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 50 insertions(+), 2 deletions(-) diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c index 818c83c..9f5c53b 100644 --- a/src/output/alsa_plugin.c +++ b/src/output/alsa_plugin.c @@ -183,6 +183,19 @@ get_bitformat(const struct audio_format *af) return SND_PCM_FORMAT_UNKNOWN; } +static snd_pcm_format_t +byteswap_bitformat(snd_pcm_format_t fmt) +{ + switch(fmt) { + case SND_PCM_FORMAT_S16_LE: return SND_PCM_FORMAT_S16_BE; + case SND_PCM_FORMAT_S24_LE: return SND_PCM_FORMAT_S24_BE; + case SND_PCM_FORMAT_S32_LE: return SND_PCM_FORMAT_S32_BE; + case SND_PCM_FORMAT_S16_BE: return SND_PCM_FORMAT_S16_LE; + case SND_PCM_FORMAT_S24_BE: return SND_PCM_FORMAT_S24_LE; + case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE; + } + return SND_PCM_FORMAT_UNKNOWN; +} /** * Set up the snd_pcm_t object which was opened by the caller. Set up * the configured settings and the audio format. @@ -208,7 +221,6 @@ alsa_setup(struct alsa_data *ad, struct audio_format *audio_format, configure_hw: /* configure HW params */ snd_pcm_hw_params_alloca(&hwparams); - cmd = "snd_pcm_hw_params_any"; err = snd_pcm_hw_params_any(ad->pcm, hwparams); if (err < 0) @@ -236,13 +248,37 @@ configure_hw: } err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, bitformat); + if (err == -EINVAL) { + snd_pcm_format_t fmt = byteswap_bitformat(bitformat); + err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, fmt); + if (err == 0) { + g_debug("ALSA device \"%s\": converting %u bit to reverse-endian\n", + alsa_device(ad), audio_format->bits); + audio_format->reverse_endian = 1; + } + } if (err == -EINVAL && (audio_format->bits == 24 || audio_format->bits == 16)) { /* fall back to 32 bit, let pcm_convert.c do the conversion */ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, SND_PCM_FORMAT_S32); - if (err == 0) + if (err == 0) { + g_debug("ALSA device \"%s\": converting %u bit to 32 bit\n", + alsa_device(ad), audio_format->bits); + audio_format->bits = 32; + } + } + if (err == -EINVAL && (audio_format->bits == 24 || + audio_format->bits == 16)) { + /* fall back to 32 bit, let pcm_convert.c do the conversion */ + err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, + byteswap_bitformat(SND_PCM_FORMAT_S32)); + if (err == 0) { + g_debug("ALSA device \"%s\": converting %u bit to 32 bit backward-endian\n", + alsa_device(ad), audio_format->bits); audio_format->bits = 32; + audio_format->reverse_endian = 1; + } } if (err == -EINVAL && audio_format->bits != 16) { @@ -255,6 +291,17 @@ configure_hw: audio_format->bits = 16; } } + if (err == -EINVAL && audio_format->bits != 16) { + /* fall back to 16 bit, let pcm_convert.c do the conversion */ + err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, + byteswap_bitformat(SND_PCM_FORMAT_S16)); + if (err == 0) { + g_debug("ALSA device \"%s\": converting %u bit to 16 bit backward-endian\n", + alsa_device(ad), audio_format->bits); + audio_format->bits = 16; + audio_format->reverse_endian = 1; + } + } if (err < 0) { g_set_error(error, alsa_output_quark(), err, @@ -470,6 +517,7 @@ static size_t alsa_play(void *data, const void *chunk, size_t size, GError **error) { struct alsa_data *ad = data; + int i; size /= ad->frame_size; -- 1.6.2.5 mpd.git-9e9d7b73d2165f197eeec12ee953add5f49746b7.patch: --- NEW FILE mpd.git-9e9d7b73d2165f197eeec12ee953add5f49746b7.patch --- From: David Woodhouse <David.Woodhouse(a)intel.com&gt; Date: Sun, 19 Jul 2009 16:24:43 +0100 Subject: [PATCH 1/1] Add audio_format_init() function --- src/audio_format.h | 9 +++++++++ src/audio_parser.c | 10 +++++++--- src/decoder/_flac_common.c | 5 ++--- src/decoder/audiofile_plugin.c | 8 +++----- src/decoder/faad_plugin.c | 6 +----- src/decoder/ffmpeg_plugin.c | 9 +++++---- src/decoder/mad_plugin.c | 10 ++-------- src/decoder/mikmod_plugin.c | 4 +--- src/decoder/modplug_plugin.c | 4 +--- src/decoder/mp4ff_plugin.c | 6 +----- src/decoder/mpcdec_plugin.c | 4 +--- src/decoder/sidplay_plugin.cxx | 4 +--- src/decoder/sndfile_decoder_plugin.c | 4 +--- src/decoder/vorbis_plugin.c | 3 +-- src/decoder/wavpack_plugin.c | 6 +++--- test/run_encoder.c | 8 +++----- test/run_filter.c | 8 +++----- test/run_output.c | 8 +++----- test/software_volume.c | 7 ++----- 19 files changed, 50 insertions(+), 73 deletions(-) diff --git a/src/audio_format.h b/src/audio_format.h index 64087d0..e325c1b 100644 --- a/src/audio_format.h +++ b/src/audio_format.h @@ -36,6 +36,15 @@ static inline void audio_format_clear(struct audio_format *af) af->channels = 0; } +static inline void audio_format_init(struct audio_format *af, + uint32_t sample_rate, + uint8_t bits, uint8_t channels) +{ + af->sample_rate = sample_rate; + af->bits = bits; + af->channels = channels; +} + static inline bool audio_format_defined(const struct audio_format *af) { return af->sample_rate != 0; diff --git a/src/audio_parser.c b/src/audio_parser.c index 906b0f8..d29f5f4 100644 --- a/src/audio_parser.c +++ b/src/audio_parser.c @@ -41,6 +41,8 @@ audio_format_parse(struct audio_format *dest, const char *src, GError **error) { char *endptr; unsigned long value; + uint32_t rate; + uint8_t bits, channels; audio_format_clear(dest); @@ -61,7 +63,7 @@ audio_format_parse(struct audio_format *dest, const char *src, GError **error) return false; } - dest->sample_rate = value; + rate = value; /* parse sample format */ @@ -81,7 +83,7 @@ audio_format_parse(struct audio_format *dest, const char *src, GError **error) return false; } - dest->bits = value; + bits = value; /* parse channel count */ @@ -93,7 +95,9 @@ audio_format_parse(struct audio_format *dest, const char *src, GError **error) return false; } - dest->channels = value; + channels = value; + + audio_format_init(dest, rate, bits, channels); return true; } diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c index 713dfe9..7b34538 100644 --- a/src/decoder/_flac_common.c +++ b/src/decoder/_flac_common.c @@ -195,9 +195,8 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block, switch (block->type) { case FLAC__METADATA_TYPE_STREAMINFO: - data->audio_format.bits = (int8_t)si->bits_per_sample; - data->audio_format.sample_rate = si->sample_rate; - data->audio_format.channels = (int8_t)si->channels; + audio_format_init(&data->audio_format, si->sample_rate, + si->bits_per_sample, si->channels); data->total_time = ((float)si->total_samples) / (si->sample_rate); break; case FLAC__METADATA_TYPE_VORBIS_COMMENT: diff --git a/src/decoder/audiofile_plugin.c b/src/decoder/audiofile_plugin.c index f66d90d..b4959f6 100644 --- a/src/decoder/audiofile_plugin.c +++ b/src/decoder/audiofile_plugin.c @@ -136,11 +136,9 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is) afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, bits); afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); - audio_format.bits = (uint8_t)bits; - audio_format.sample_rate = - (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK); - audio_format.channels = - (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK); + + audio_format_init(&audio_format, afGetRate(af_fp, AF_DEFAULT_TRACK), + bits, afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK)); if (!audio_format_valid(&audio_format)) { g_warning("Invalid audio format: %u:%u:%u\n", diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_plugin.c index d0537dd..1b8b2b7 100644 --- a/src/decoder/faad_plugin.c +++ b/src/decoder/faad_plugin.c @@ -262,11 +262,7 @@ faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer, decoder_buffer_consume(buffer, nbytes); - *audio_format = (struct audio_format){ - .bits = 16, - .channels = channels, - .sample_rate = sample_rate, - }; + audio_format_init(audio_format, sample_rate, 16, channels); return true; } diff --git a/src/decoder/ffmpeg_plugin.c b/src/decoder/ffmpeg_plugin.c index 03c46a7..f6003d2 100644 --- a/src/decoder/ffmpeg_plugin.c +++ b/src/decoder/ffmpeg_plugin.c @@ -267,6 +267,7 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx) struct audio_format audio_format; enum decoder_command cmd; int total_time; + uint8_t bits; total_time = 0; @@ -275,13 +276,13 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx) } #if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0) - audio_format.bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt); + bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt); #else /* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */ - audio_format.bits = (uint8_t) 16; + bits = (uint8_t) 16; #endif - audio_format.sample_rate = (unsigned int)codec_context->sample_rate; - audio_format.channels = codec_context->channels; + audio_format_init(&audio_format, codec_context->sample_rate, bits, + codec_context->channels); if (!audio_format_valid(&audio_format)) { g_warning("Invalid audio format: %u:%u:%u\n", diff --git a/src/decoder/mad_plugin.c b/src/decoder/mad_plugin.c index c6b9d32..85f4506 100644 --- a/src/decoder/mad_plugin.c +++ b/src/decoder/mad_plugin.c @@ -1148,13 +1148,6 @@ mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r) return ret != DECODE_BREAK; } -static void mp3_audio_format(struct mp3_data *data, struct audio_format *af) -{ - af->bits = 24; - af->sample_rate = (data->frame).header.samplerate; - af->channels = MAD_NCHANNELS(&(data->frame).header); -} - static void mp3_decode(struct decoder *decoder, struct input_stream *input_stream) { @@ -1170,7 +1163,8 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream) return; } - mp3_audio_format(&data, &audio_format); + audio_format_init(&audio_format, data.frame.header.samplerate, 24, + MAD_NCHANNELS(&data.frame.header)); decoder_initialized(decoder, &audio_format, data.input_stream->seekable, data.total_time); diff --git a/src/decoder/mikmod_plugin.c b/src/decoder/mikmod_plugin.c index 065c343..e7b7bfb 100644 --- a/src/decoder/mikmod_plugin.c +++ b/src/decoder/mikmod_plugin.c @@ -175,9 +175,7 @@ mod_decode(struct decoder *decoder, const char *path) return; } - audio_format.bits = 16; - audio_format.sample_rate = 44100; - audio_format.channels = 2; + audio_format_init(&audio_format, 44100, 16, 2); secPerByte = 1.0 / ((audio_format.bits * audio_format.channels / 8.0) * diff --git a/src/decoder/modplug_plugin.c b/src/decoder/modplug_plugin.c index 31f0a47..6c375e6 100644 --- a/src/decoder/modplug_plugin.c +++ b/src/decoder/modplug_plugin.c @@ -121,9 +121,7 @@ mod_decode(struct decoder *decoder, struct input_stream *is) return; } - audio_format.bits = 16; - audio_format.sample_rate = 44100; - audio_format.channels = 2; + audio_format_init(&audio_format, 44100, 16, 2); sec_perbyte = 1.0 / ((audio_format.bits * audio_format.channels / 8.0) * diff --git a/src/decoder/mp4ff_plugin.c b/src/decoder/mp4ff_plugin.c index cf93829..d2c63f9 100644 --- a/src/decoder/mp4ff_plugin.c +++ b/src/decoder/mp4ff_plugin.c @@ -131,11 +131,7 @@ mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format) } *track_r = track; - *audio_format = (struct audio_format){ - .bits = 16, - .channels = channels, - .sample_rate = sample_rate, - }; + audio_format_init(audio_format, sample_rate, 16, channels); if (!audio_format_valid(audio_format)) { g_warning("Invalid audio format: %u:%u:%u\n", diff --git a/src/decoder/mpcdec_plugin.c b/src/decoder/mpcdec_plugin.c index 26349f9..a684da1 100644 --- a/src/decoder/mpcdec_plugin.c +++ b/src/decoder/mpcdec_plugin.c @@ -193,9 +193,7 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is) mpc_demux_get_info(demux, &info); #endif - audio_format.bits = 24; - audio_format.channels = info.channels; - audio_format.sample_rate = info.sample_freq; + audio_format_init(&audio_format, info.sample_freq, 24, info.channels); if (!audio_format_valid(&audio_format)) { #ifndef MPC_IS_OLD_API diff --git a/src/decoder/sidplay_plugin.cxx b/src/decoder/sidplay_plugin.cxx index c62e6b4..54ab746 100644 --- a/src/decoder/sidplay_plugin.cxx +++ b/src/decoder/sidplay_plugin.cxx @@ -103,9 +103,7 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs) /* initialize the MPD decoder */ struct audio_format audio_format; - audio_format.sample_rate = 48000; - audio_format.bits = 16; - audio_format.channels = 2; + audio_format_init(&audio_format, 48000, 16, 2); decoder_initialized(decoder, &audio_format, false, -1); diff --git a/src/decoder/vorbis_plugin.c b/src/decoder/vorbis_plugin.c index d4f81e9..bab1d57 100644 --- a/src/decoder/vorbis_plugin.c +++ b/src/decoder/vorbis_plugin.c @@ -324,8 +324,7 @@ vorbis_stream_decode(struct decoder *decoder, vorbis_info *vi = ov_info(&vf, -1); struct replay_gain_info *new_rgi; - audio_format.channels = vi->channels; - audio_format.sample_rate = vi->rate; + audio_format_init(&audio_format, vi->rate, 16, vi->channels); if (!audio_format_valid(&audio_format)) { g_warning("Invalid audio format: %u:%u:%u\n", diff --git a/src/decoder/wavpack_plugin.c b/src/decoder/wavpack_plugin.c index 821536f..f3d7011 100644 --- a/src/decoder/wavpack_plugin.c +++ b/src/decoder/wavpack_plugin.c @@ -145,9 +145,9 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek, int bytes_per_sample, output_sample_size; int position; - audio_format.sample_rate = WavpackGetSampleRate(wpc); - audio_format.channels = WavpackGetReducedChannels(wpc); - audio_format.bits = WavpackGetBitsPerSample(wpc); + audio_format_init(&audio_format, WavpackGetSampleRate(wpc), + WavpackGetBitsPerSample(wpc), + WavpackGetReducedChannels(wpc)); /* round bitwidth to 8-bit units */ audio_format.bits = (audio_format.bits + 7) & (~7); diff --git a/test/run_encoder.c b/test/run_encoder.c index 8cb1c6d..a9b00e9 100644 --- a/test/run_encoder.c +++ b/test/run_encoder.c @@ -41,11 +41,7 @@ encoder_to_stdout(struct encoder *encoder) int main(int argc, char **argv) { GError *error = NULL; - struct audio_format audio_format = { - .sample_rate = 44100, - .bits = 16, - .channels = 2, - }; + struct audio_format audio_format; bool ret; const char *encoder_name; const struct encoder_plugin *plugin; @@ -66,6 +62,8 @@ int main(int argc, char **argv) else encoder_name = "vorbis"; + audio_format_init(&audio_format, 44100, 16, 2); + /* create the encoder */ plugin = encoder_plugin_get(encoder_name); diff --git a/test/run_output.c b/test/run_output.c index adf6e1d..a280f88 100644 --- a/test/run_output.c +++ b/test/run_output.c @@ -100,11 +100,7 @@ load_audio_output(struct audio_output *ao, const char *name) int main(int argc, char **argv) { struct audio_output ao; - struct audio_format audio_format = { - .sample_rate = 44100, - .bits = 16, - .channels = 2, - }; + struct audio_format audio_format; bool success; GError *error = NULL; char buffer[4096]; @@ -116,6 +112,8 @@ int main(int argc, char **argv) return 1; } + audio_format_init(&audio_format, 44100, 16, 2); + g_thread_init(NULL); /* read configuration file (mpd.conf) */ diff --git a/test/software_volume.c b/test/software_volume.c index 9a9fd56..9e8c8e7 100644 --- a/test/software_volume.c +++ b/test/software_volume.c @@ -35,11 +35,7 @@ int main(int argc, char **argv) { GError *error = NULL; - struct audio_format audio_format = { - .sample_rate = 48000, - .bits = 16, - .channels = 2, - }; + struct audio_format audio_format; bool ret; static char buffer[4096]; ssize_t nbytes; @@ -57,6 +53,7 @@ int main(int argc, char **argv) return 1; } } + audio_format_init(&audio_format, 48000, 16, 2); while ((nbytes = read(0, buffer, sizeof(buffer))) > 0) { pcm_volume(buffer, nbytes, &audio_format, PCM_VOLUME_1 / 2); -- 1.6.2.5 mpd.git-f3a5b753ae053eb1a862343b0fd3d62973cacc18.patch: --- NEW FILE mpd.git-f3a5b753ae053eb1a862343b0fd3d62973cacc18.patch --- From: David Woodhouse <David.Woodhouse(a)intel.com&gt; Date: Sun, 19 Jul 2009 16:42:19 +0100 Subject: [PATCH] Add reverse_endian field to struct audio_format and handle conversion --- Makefile.am | 2 + src/audio_format.h | 5 ++- src/filter/convert_filter_plugin.c | 1 + src/output_thread.c | 10 +++-- src/pcm_byteswap.c | 71 ++++++++++++++++++++++++++++++++++++ src/pcm_byteswap.h | 50 +++++++++++++++++++++++++ src/pcm_convert.c | 18 +++++++++ 7 files changed, 152 insertions(+), 5 deletions(-) create mode 100644 src/pcm_byteswap.c create mode 100644 src/pcm_byteswap.h diff --git a/Makefile.am b/Makefile.am index 5c53ca5..e53dfb1 100644 --- a/Makefile.am +++ b/Makefile.am @@ -114,6 +114,7 @@ mpd_headers = \ src/pcm_convert.h \ src/pcm_volume.h \ src/pcm_mix.h \ + src/pcm_byteswap.h \ src/pcm_channels.h \ src/pcm_format.h \ src/pcm_resample.h \ @@ -218,6 +219,7 @@ src_mpd_SOURCES = \ src/pcm_convert.c \ src/pcm_volume.c \ src/pcm_mix.c \ + src/pcm_byteswap.c \ src/pcm_channels.c \ src/pcm_format.c \ src/pcm_resample.c \ diff --git a/src/audio_format.h b/src/audio_format.h index e325c1b..54514ff 100644 --- a/src/audio_format.h +++ b/src/audio_format.h @@ -27,6 +27,7 @@ struct audio_format { uint32_t sample_rate; uint8_t bits; uint8_t channels; + uint8_t reverse_endian; }; static inline void audio_format_clear(struct audio_format *af) @@ -34,6 +35,7 @@ static inline void audio_format_clear(struct audio_format *af) af->sample_rate = 0; af->bits = 0; af->channels = 0; + af->reverse_endian = 0; } static inline void audio_format_init(struct audio_format *af, @@ -97,7 +99,8 @@ static inline bool audio_format_equals(const struct audio_format *a, { return a->sample_rate == b->sample_rate && a->bits == b->bits && - a->channels == b->channels; + a->channels == b->channels && + a->reverse_endian == b->reverse_endian; } /** diff --git a/src/pcm_byteswap.c b/src/pcm_byteswap.c new file mode 100644 index 0000000..6bdec1f --- /dev/null +++ b/src/pcm_byteswap.c @@ -0,0 +1,71 @@ +/* + * Copyright (C) 2003-2009 The Music Player Daemon Project + * http://www.musicpd.org + * + * 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, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#include "pcm_byteswap.h" +#include "pcm_buffer.h" + +#include <glib.h> + +#include <assert.h> + +#undef G_LOG_DOMAIN +#define G_LOG_DOMAIN "pcm" + +static inline uint16_t swab16(uint16_t x) +{ + return (x << 8) | (x >> 8); +} + +const int16_t *pcm_byteswap_16(struct pcm_buffer *buffer, + const int16_t *src, size_t len) +{ + unsigned i; + int16_t *buf = pcm_buffer_get(buffer, len); + + if (!buf) + return NULL; + + for (i = 0; i < len / 2; i++) + buf[i] = swab16(src[i]); + + return buf; +} + +static inline uint32_t swab32(uint32_t x) +{ + return (x << 24) | + ((x & 0xff00) << 8) | + ((x & 0xff0000) >> 8) | + (x >> 24); +} + +const int32_t *pcm_byteswap_32(struct pcm_buffer *buffer, + const int32_t *src, size_t len) +{ + unsigned i; + int32_t *buf = pcm_buffer_get(buffer, len); + + if (!buf) + return NULL; + + for (i = 0; i < len / 4; i++) + buf[i] = swab32(src[i]); + + return buf; +} diff --git a/src/pcm_byteswap.h b/src/pcm_byteswap.h new file mode 100644 index 0000000..e1196d9 --- /dev/null +++ b/src/pcm_byteswap.h @@ -0,0 +1,50 @@ +/* + * Copyright (C) 2003-2009 The Music Player Daemon Project + * http://www.musicpd.org + * + * 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, write to the Free Software Foundation, Inc., + * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. + */ + +#ifndef MPD_PCM_BYTESWAP_H +#define MPD_PCM_BYTESWAP_H + +#include <stdint.h> +#include <stddef.h> + +struct pcm_buffer; + +/** + * Changes the endianness of 16 bit PCM data. + * + * @param buffer the destination pcm_buffer object + * @param src the source PCM buffer + * @param src_size the number of bytes in #src + * @return the destination buffer + */ +const int16_t *pcm_byteswap_16(struct pcm_buffer *buffer, + const int16_t *src, size_t len); + +/** + * Changes the endianness of 32-bit (or 24-bit) PCM data. + * + * @param buffer the destination pcm_buffer object + * @param src the source PCM buffer + * @param src_size the number of bytes in #src + * @return the destination buffer + */ +const int32_t *pcm_byteswap_32(struct pcm_buffer *buffer, + const int32_t *src, size_t len); + +#endif diff --git a/src/pcm_convert.c b/src/pcm_convert.c index ebb4adf..a96c338 100644 --- a/src/pcm_convert.c +++ b/src/pcm_convert.c @@ -83,6 +83,12 @@ pcm_convert_16(struct pcm_convert_state *state, dest_format->sample_rate, &len); + if (dest_format->reverse_endian) { + buf = pcm_byteswap_16(&state->format_buffer, buf, len); + if (!buf) + g_error("pcm_byteswap_16() failed"); + } + *dest_size_r = len; return buf; } @@ -120,6 +126,12 @@ pcm_convert_24(struct pcm_convert_state *state, dest_format->sample_rate, &len); + if (dest_format->reverse_endian) { + buf = pcm_byteswap_32(&state->format_buffer, buf, len); + if (!buf) + g_error("pcm_byteswap_32() failed"); + } + *dest_size_r = len; return buf; } @@ -157,6 +169,12 @@ pcm_convert_32(struct pcm_convert_state *state, dest_format->sample_rate, &len); + if (dest_format->reverse_endian) { + buf = pcm_byteswap_32(&state->format_buffer, buf, len); + if (!buf) + g_error("pcm_byteswap_32() failed"); + } + *dest_size_r = len; return buf; } -- 1.6.2.5 Index: .cvsignore =================================================================== RCS file: /cvs/free/rpms/mpd/devel/.cvsignore,v retrieving revision 1.5 retrieving revision 1.6 diff -u -r1.5 -r1.6 --- .cvsignore 29 Jun 2009 12:13:10 -0000 1.5 +++ .cvsignore 25 Aug 2009 11:59:29 -0000 1.6 @@ -1 +1 @@ -mpd-0.15.tar.bz2 +mpd-0.15.2.tar.bz2 Index: mpd.spec =================================================================== RCS file: /cvs/free/rpms/mpd/devel/mpd.spec,v retrieving revision 1.10 retrieving revision 1.11 diff -u -r1.10 -r1.11 --- mpd.spec 29 Jun 2009 12:13:10 -0000 1.10 +++ mpd.spec 25 Aug 2009 11:59:29 -0000 1.11 @@ -1,13 +1,17 @@ Name: mpd -Version: 0.15 +Version: 0.15.2 Release: 1%{?dist} Summary: The Music Player Daemon License: GPLv2+ Group: Applications/Multimedia URL: http://mpd.wikia.com/ -Source: http://downloads.sourceforge.net/musicpd/mpd-0.15.tar.bz2 +Source: http://downloads.sourceforge.net/musicpd/mpd-0.15.2.tar.bz2 Source1: mpd.init Source2: 95-grant-audio-devices-to-mpd.fdi +Patch0: mpd.git-9e9d7b73d2165f197eeec12ee953add5f49746b7.patch +Patch1: mpd.git-f3a5b753ae053eb1a862343b0fd3d62973cacc18.patch +Patch2: mpd.git-00503c9251141b427457c17a9677444bf29c3992.patch +Patch3: 6a071efa2794806ad5a2a62f0fcdee4b1843b41f.patch BuildRoot: %{_tmppath}/%{name}-%{version}-%{release}-root-%(%{__id_u} -n) @@ -38,6 +42,7 @@ BuildRequires: bzip2-devel BuildRequires: zziplib-devel BuildRequires: sqlite-devel +BuildRequires: autoconf Requires(pre): shadow-utils Requires(post): chkconfig Requires(preun): chkconfig /sbin/service @@ -50,10 +55,15 @@ Vorbis, FLAC, Mod, AAC and wave files) and managing playlists. MPD is designed for integrating a computer into a stereo system that provides control for music playback over a local network. It is also makes a great desktop music player, -especially if your a console junkie, like frontend options, or restart X often. +especially if you are a console junkie, like frontend options or restart X often. %prep %setup -q +%patch0 -p1 +%patch1 -p1 +%patch2 -p1 +%patch3 -p1 +autoreconf --force --install %build %configure --enable-mikmod --enable-bzip2 --enable-zip @@ -149,6 +159,12 @@ %ghost %{_localstatedir}/lib/%{name}/mpdstate %changelog +* Tue Aug 25 2009 Adrian Reber <adrian(a)lisas.de&gt; - 0.15.2 +- updated to 0.15.2 +- applied patches from David Woodhouse to fix + "mpd fails to play to usb audio device" (#731) +- fix description (#765) + * Mon Jun 29 2009 Adrian Reber <adrian(a)lisas.de&gt; - 0.15-1 - updated to 0.15 - added "Conflicts: mpich2" (#593) Index: sources =================================================================== RCS file: /cvs/free/rpms/mpd/devel/sources,v retrieving revision 1.5 retrieving revision 1.6 diff -u -r1.5 -r1.6 --- sources 29 Jun 2009 12:13:10 -0000 1.5 +++ sources 25 Aug 2009 11:59:29 -0000 1.6 @@ -1 +1 @@ -2ed93a60bd703ba46d6794e12cfb5f1d mpd-0.15.tar.bz2 +869779fdf592ac45d067c7dec228cb51 mpd-0.15.2.tar.bz2