mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-25 01:30:38 +00:00
avimux: fix indx duration for PCM audio
GstBuffers for PCM audio usually contains more than 1 sample, we need to get the total number of samples to set the indx duration.
This commit is contained in:
parent
8dd78015f1
commit
831b1e958a
2 changed files with 49 additions and 6 deletions
|
@ -342,6 +342,8 @@ gst_avi_mux_pad_reset (GstAviPad * avipad, gboolean free)
|
||||||
} else {
|
} else {
|
||||||
GstAviAudioPad *audpad = (GstAviAudioPad *) avipad;
|
GstAviAudioPad *audpad = (GstAviAudioPad *) avipad;
|
||||||
|
|
||||||
|
audpad->samples = 0;
|
||||||
|
|
||||||
avipad->hdr.type = GST_MAKE_FOURCC ('a', 'u', 'd', 's');
|
avipad->hdr.type = GST_MAKE_FOURCC ('a', 'u', 'd', 's');
|
||||||
if (audpad->auds_codec_data) {
|
if (audpad->auds_codec_data) {
|
||||||
gst_buffer_unref (audpad->auds_codec_data);
|
gst_buffer_unref (audpad->auds_codec_data);
|
||||||
|
@ -1467,8 +1469,8 @@ gst_avi_mux_riff_get_header (GstAviPad * avipad, guint32 video_frame_size)
|
||||||
|
|
||||||
/* write an odml index chunk in the movi list */
|
/* write an odml index chunk in the movi list */
|
||||||
static GstFlowReturn
|
static GstFlowReturn
|
||||||
gst_avi_mux_write_avix_index (GstAviMux * avimux, gchar * code,
|
gst_avi_mux_write_avix_index (GstAviMux * avimux, GstAviPad * avipad,
|
||||||
gchar * chunk, gst_avi_superindex_entry * super_index,
|
gchar * code, gchar * chunk, gst_avi_superindex_entry * super_index,
|
||||||
gint * super_index_count)
|
gint * super_index_count)
|
||||||
{
|
{
|
||||||
GstFlowReturn res;
|
GstFlowReturn res;
|
||||||
|
@ -1477,6 +1479,17 @@ gst_avi_mux_write_avix_index (GstAviMux * avimux, gchar * code,
|
||||||
gst_riff_index_entry *entry;
|
gst_riff_index_entry *entry;
|
||||||
gint i;
|
gint i;
|
||||||
guint32 size, entry_count;
|
guint32 size, entry_count;
|
||||||
|
gboolean is_pcm = FALSE;
|
||||||
|
guint32 pcm_samples = 0;
|
||||||
|
|
||||||
|
/* check if it is pcm */
|
||||||
|
if (avipad && !avipad->is_video) {
|
||||||
|
GstAviAudioPad *audiopad = (GstAviAudioPad *) avipad;
|
||||||
|
if (audiopad->auds.format == GST_RIFF_WAVE_FORMAT_PCM) {
|
||||||
|
pcm_samples = audiopad->samples;
|
||||||
|
is_pcm = TRUE;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
/* allocate the maximum possible */
|
/* allocate the maximum possible */
|
||||||
buffer = gst_buffer_new_and_alloc (32 + 8 * avimux->idx_index);
|
buffer = gst_buffer_new_and_alloc (32 + 8 * avimux->idx_index);
|
||||||
|
@ -1528,7 +1541,11 @@ gst_avi_mux_write_avix_index (GstAviMux * avimux, gchar * code,
|
||||||
i = *super_index_count;
|
i = *super_index_count;
|
||||||
super_index[i].offset = GUINT64_TO_LE (avimux->total_data);
|
super_index[i].offset = GUINT64_TO_LE (avimux->total_data);
|
||||||
super_index[i].size = GUINT32_TO_LE (size);
|
super_index[i].size = GUINT32_TO_LE (size);
|
||||||
|
if (is_pcm) {
|
||||||
|
super_index[i].duration = GUINT32_TO_LE (pcm_samples);
|
||||||
|
} else {
|
||||||
super_index[i].duration = GUINT32_TO_LE (entry_count);
|
super_index[i].duration = GUINT32_TO_LE (entry_count);
|
||||||
|
}
|
||||||
(*super_index_count)++;
|
(*super_index_count)++;
|
||||||
} else
|
} else
|
||||||
GST_WARNING_OBJECT (avimux, "No more room in superindex of stream %s",
|
GST_WARNING_OBJECT (avimux, "No more room in superindex of stream %s",
|
||||||
|
@ -1547,15 +1564,26 @@ gst_avi_mux_write_avix_index (GstAviMux * avimux, gchar * code,
|
||||||
/* some other usable functions (thankyou xawtv ;-) ) */
|
/* some other usable functions (thankyou xawtv ;-) ) */
|
||||||
|
|
||||||
static void
|
static void
|
||||||
gst_avi_mux_add_index (GstAviMux * avimux, gchar * code, guint32 flags,
|
gst_avi_mux_add_index (GstAviMux * avimux, GstAviPad * avipad, guint32 flags,
|
||||||
guint32 size)
|
guint32 size)
|
||||||
{
|
{
|
||||||
|
gchar *code = avipad->tag;
|
||||||
if (avimux->idx_index == avimux->idx_count) {
|
if (avimux->idx_index == avimux->idx_count) {
|
||||||
avimux->idx_count += 256;
|
avimux->idx_count += 256;
|
||||||
avimux->idx =
|
avimux->idx =
|
||||||
g_realloc (avimux->idx,
|
g_realloc (avimux->idx,
|
||||||
avimux->idx_count * sizeof (gst_riff_index_entry));
|
avimux->idx_count * sizeof (gst_riff_index_entry));
|
||||||
}
|
}
|
||||||
|
|
||||||
|
/* in case of pcm audio, we need to count the number of samples for
|
||||||
|
* putting in the indx entries */
|
||||||
|
if (!avipad->is_video) {
|
||||||
|
GstAviAudioPad *audiopad = (GstAviAudioPad *) avipad;
|
||||||
|
if (audiopad->auds.format == GST_RIFF_WAVE_FORMAT_PCM) {
|
||||||
|
audiopad->samples += size / audiopad->auds.blockalign;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
memcpy (&(avimux->idx[avimux->idx_index].id), code, 4);
|
memcpy (&(avimux->idx[avimux->idx_index].id), code, 4);
|
||||||
avimux->idx[avimux->idx_index].flags = GUINT32_TO_LE (flags);
|
avimux->idx[avimux->idx_index].flags = GUINT32_TO_LE (flags);
|
||||||
avimux->idx[avimux->idx_index].offset = GUINT32_TO_LE (avimux->idx_offset);
|
avimux->idx[avimux->idx_index].offset = GUINT32_TO_LE (avimux->idx_offset);
|
||||||
|
@ -1615,7 +1643,7 @@ gst_avi_mux_bigfile (GstAviMux * avimux, gboolean last)
|
||||||
|
|
||||||
node = node->next;
|
node = node->next;
|
||||||
|
|
||||||
res = gst_avi_mux_write_avix_index (avimux, avipad->tag,
|
res = gst_avi_mux_write_avix_index (avimux, avipad, avipad->tag,
|
||||||
avipad->idx_tag, avipad->idx, &avipad->idx_index);
|
avipad->idx_tag, avipad->idx, &avipad->idx_index);
|
||||||
if (res != GST_FLOW_OK)
|
if (res != GST_FLOW_OK)
|
||||||
return res;
|
return res;
|
||||||
|
@ -1657,6 +1685,15 @@ gst_avi_mux_bigfile (GstAviMux * avimux, gboolean last)
|
||||||
avimux->numx_frames = 0;
|
avimux->numx_frames = 0;
|
||||||
avimux->datax_size = 4; /* movi tag */
|
avimux->datax_size = 4; /* movi tag */
|
||||||
avimux->idx_index = 0;
|
avimux->idx_index = 0;
|
||||||
|
node = avimux->sinkpads;
|
||||||
|
while (node) {
|
||||||
|
GstAviPad *avipad = (GstAviPad *) node->data;
|
||||||
|
node = node->next;
|
||||||
|
if (!avipad->is_video) {
|
||||||
|
GstAviAudioPad *audiopad = (GstAviAudioPad *) avipad;
|
||||||
|
audiopad->samples = 0;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
header = gst_avi_mux_riff_get_avix_header (0);
|
header = gst_avi_mux_riff_get_avix_header (0);
|
||||||
avimux->total_data += GST_BUFFER_SIZE (header);
|
avimux->total_data += GST_BUFFER_SIZE (header);
|
||||||
|
@ -1946,7 +1983,7 @@ gst_avi_mux_do_buffer (GstAviMux * avimux, GstAviPad * avipad)
|
||||||
audpad->audio_time += GST_BUFFER_DURATION (data);
|
audpad->audio_time += GST_BUFFER_DURATION (data);
|
||||||
}
|
}
|
||||||
|
|
||||||
gst_avi_mux_add_index (avimux, avipad->tag, flags, GST_BUFFER_SIZE (data));
|
gst_avi_mux_add_index (avimux, avipad, flags, GST_BUFFER_SIZE (data));
|
||||||
|
|
||||||
/* prepare buffers for sending */
|
/* prepare buffers for sending */
|
||||||
gst_buffer_set_caps (header, GST_PAD_CAPS (avimux->srcpad));
|
gst_buffer_set_caps (header, GST_PAD_CAPS (avimux->srcpad));
|
||||||
|
|
|
@ -115,6 +115,12 @@ typedef struct _GstAviAudioPad {
|
||||||
/* audio info for bps calculation */
|
/* audio info for bps calculation */
|
||||||
guint32 audio_size;
|
guint32 audio_size;
|
||||||
guint64 audio_time;
|
guint64 audio_time;
|
||||||
|
|
||||||
|
/* counts the number of samples to put in indx chunk
|
||||||
|
* useful for raw audio where usually there are more than
|
||||||
|
* 1 sample in each GstBuffer */
|
||||||
|
gint samples;
|
||||||
|
|
||||||
/* extra data */
|
/* extra data */
|
||||||
GstBuffer *auds_codec_data;
|
GstBuffer *auds_codec_data;
|
||||||
} GstAviAudioPad;
|
} GstAviAudioPad;
|
||||||
|
|
Loading…
Reference in a new issue