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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)
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