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-rw-r--r--src/audio_format.h14
-rw-r--r--src/audio_parser.c10
-rw-r--r--src/decoder/_flac_common.c5
-rw-r--r--src/decoder/audiofile_plugin.c8
-rw-r--r--src/decoder/faad_plugin.c6
-rw-r--r--src/decoder/ffmpeg_plugin.c9
-rw-r--r--src/decoder/mad_plugin.c10
-rw-r--r--src/decoder/mikmod_plugin.c4
-rw-r--r--src/decoder/modplug_plugin.c4
-rw-r--r--src/decoder/mp4ff_plugin.c6
-rw-r--r--src/decoder/mpcdec_plugin.c4
-rw-r--r--src/decoder/sidplay_plugin.cxx4
-rw-r--r--src/decoder/sndfile_decoder_plugin.c4
-rw-r--r--src/decoder/vorbis_plugin.c3
-rw-r--r--src/decoder/wavpack_plugin.c6
-rw-r--r--src/filter/convert_filter_plugin.c1
-rw-r--r--src/output/alsa_plugin.c52
-rw-r--r--src/output_thread.c10
-rw-r--r--src/pcm_byteswap.c71
-rw-r--r--src/pcm_byteswap.h50
-rw-r--r--src/pcm_convert.c19
21 files changed, 240 insertions, 60 deletions
diff --git a/src/audio_format.h b/src/audio_format.h
index 64087d070..54514ff93 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 +35,16 @@ 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,
+ 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)
@@ -88,7 +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/audio_parser.c b/src/audio_parser.c
index 906b0f819..d29f5f449 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 713dfe9b2..7b3453854 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 f66d90dc1..b4959f6c2 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 d0537dd5b..1b8b2b784 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 03c46a732..f6003d2f3 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 c6b9d32d3..85f4506d2 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 065c34319..e7b7bfb03 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 31f0a47c2..6c375e6a0 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 cf9382904..d2c63f983 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 26349f93a..a684da104 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 c62e6b4b6..54ab746e2 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/sndfile_decoder_plugin.c b/src/decoder/sndfile_decoder_plugin.c
index 0c5d2f063..4cc64459f 100644
--- a/src/decoder/sndfile_decoder_plugin.c
+++ b/src/decoder/sndfile_decoder_plugin.c
@@ -124,12 +124,10 @@ sndfile_stream_decode(struct decoder *decoder, struct input_stream *is)
return;
}
- audio_format.sample_rate = info.samplerate;
/* for now, always read 32 bit samples. Later, we could lower
MPD's CPU usage by reading 16 bit samples with
sf_readf_short() on low-quality source files. */
- audio_format.bits = 32;
- audio_format.channels = info.channels;
+ audio_format_init(&audio_format, info.samplerate, 32, info.channels);
if (!audio_format_valid(&audio_format)) {
g_warning("invalid audio format");
diff --git a/src/decoder/vorbis_plugin.c b/src/decoder/vorbis_plugin.c
index d4f81e91f..bab1d57ec 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 821536fb5..f3d701144 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/src/filter/convert_filter_plugin.c b/src/filter/convert_filter_plugin.c
index f4d03ebef..b7f16de4f 100644
--- a/src/filter/convert_filter_plugin.c
+++ b/src/filter/convert_filter_plugin.c
@@ -149,6 +149,7 @@ convert_filter_set(struct filter *_filter,
assert(audio_format_valid(&filter->out_audio_format));
assert(out_audio_format != NULL);
assert(audio_format_valid(out_audio_format));
+ assert(filter->in_audio_format.reverse_endian == 0);
filter->out_audio_format = *out_audio_format;
}
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c
index 818c83ca2..f271668b1 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;
+ default: 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 +248,38 @@ configure_hw:
}
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams, bitformat);
+ if (err == -EINVAL &&
+ byteswap_bitformat(bitformat) != SND_PCM_FORMAT_UNKNOWN) {
+ err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
+ byteswap_bitformat(bitformat));
+ 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 +292,17 @@ 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,
diff --git a/src/output_thread.c b/src/output_thread.c
index 2592b2456..c7bd069b1 100644
--- a/src/output_thread.c
+++ b/src/output_thread.c
@@ -93,18 +93,20 @@ ao_open(struct audio_output *ao)
g_mutex_unlock(ao->mutex);
g_debug("opened plugin=%s name=\"%s\" "
- "audio_format=%u:%u:%u",
+ "audio_format=%u:%u:%u:%u",
ao->plugin->name, ao->name,
ao->out_audio_format.sample_rate,
ao->out_audio_format.bits,
- ao->out_audio_format.channels);
+ ao->out_audio_format.channels,
+ ao->out_audio_format.reverse_endian);
if (!audio_format_equals(&ao->in_audio_format,
&ao->out_audio_format))
- g_debug("converting from %u:%u:%u",
+ g_debug("converting from %u:%u:%u:%u",
ao->in_audio_format.sample_rate,
ao->in_audio_format.bits,
- ao->in_audio_format.channels);
+ ao->in_audio_format.channels,
+ ao->in_audio_format.reverse_endian);
}
static void
diff --git a/src/pcm_byteswap.c b/src/pcm_byteswap.c
new file mode 100644
index 000000000..6bdec1f24
--- /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 000000000..e1196d9b2
--- /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 ebb4adff5..2d72628b2 100644
--- a/src/pcm_convert.c
+++ b/src/pcm_convert.c
@@ -20,6 +20,7 @@
#include "pcm_convert.h"
#include "pcm_channels.h"
#include "pcm_format.h"
+#include "pcm_byteswap.h"
#include "audio_format.h"
#include <assert.h>
@@ -83,6 +84,12 @@ 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 +127,12 @@ 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 +170,12 @@ 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;
}
generated by cgit v1.2.3 (git 2.25.1) at 2024-12-18 09:20:10 +0000