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audiofx: Sync properties to the stream time

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This commit is contained in:
Sebastian Dröge 2010-03-08 19:28:47 +00:00
parent 4381b9296f
commit 79e720052a
8 changed files with 104 additions and 27 deletions
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18
gst/audiofx/audioamplify.c
View file

@ -479,11 +479,21 @@ static GstFlowReturn
gst_audio_amplify_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
guint num_samples;
GstClockTime timestamp, stream_time;
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (filter), stream_time);
num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (gst_base_transform_is_passthrough (base) ||
G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))

18
gst/audiofx/audiodynamic.c
View file

@ -695,11 +695,21 @@ static GstFlowReturn
gst_audio_dynamic_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
guint num_samples;
GstClockTime timestamp, stream_time;
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (filter), stream_time);
num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (gst_base_transform_is_passthrough (base) ||
G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))

18
gst/audiofx/audioecho.c
View file

@ -357,11 +357,21 @@ static GstFlowReturn
gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioEcho *self = GST_AUDIO_ECHO (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (self)->format.width / 8);
guint num_samples;
GstClockTime timestamp, stream_time;
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (self), GST_BUFFER_TIMESTAMP (buf));
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (self), stream_time);
num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (self)->format.width / 8);
if (self->buffer == NULL) {
guint width, rate, channels;

11
gst/audiofx/audiofxbasefirfilter.c
View file

@ -822,15 +822,24 @@ gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
guint generated_samples;
guint64 output_offset;
gint64 diff = 0;
GstClockTime stream_time;
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
if (!GST_CLOCK_TIME_IS_VALID (timestamp)
&& !GST_CLOCK_TIME_IS_VALID (self->start_ts)) {
GST_ERROR_OBJECT (self, "Invalid timestamp");
return GST_FLOW_ERROR;
}
gst_object_sync_values (G_OBJECT (self), timestamp);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (self), stream_time);
g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR);
g_return_val_if_fail (channels != 0, GST_FLOW_ERROR);

18
gst/audiofx/audiofxbaseiirfilter.c
View file

@ -355,11 +355,21 @@ gst_audio_fx_base_iir_filter_transform_ip (GstBaseTransform * base,
GstBuffer * buf)
{
GstAudioFXBaseIIRFilter *filter = GST_AUDIO_FX_BASE_IIR_FILTER (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
guint num_samples;
GstClockTime timestamp, stream_time;
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (filter), stream_time);
num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (gst_base_transform_is_passthrough (base))
return GST_FLOW_OK;

18
gst/audiofx/audioinvert.c
View file

@ -236,11 +236,21 @@ static GstFlowReturn
gst_audio_invert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioInvert *filter = GST_AUDIO_INVERT (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
guint num_samples;
GstClockTime timestamp, stream_time;
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (filter), stream_time);
num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (gst_base_transform_is_passthrough (base) ||
G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))

18
gst/audiofx/audiokaraoke.c
View file

@ -341,11 +341,21 @@ static GstFlowReturn
gst_audio_karaoke_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioKaraoke *filter = GST_AUDIO_KARAOKE (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
guint num_samples;
GstClockTime timestamp, stream_time;
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (filter), stream_time);
num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (gst_base_transform_is_passthrough (base) ||
G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))

12
gst/audiofx/audiopanorama.c
View file

@ -631,9 +631,17 @@ gst_audio_panorama_transform (GstBaseTransform * base, GstBuffer * inbuf,
{
GstAudioPanorama *filter = GST_AUDIO_PANORAMA (base);
guint num_samples = GST_BUFFER_SIZE (outbuf) / (2 * filter->width);
GstClockTime timestamp, stream_time;
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (outbuf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (outbuf));
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (filter), stream_time);
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);