Commit c962e657 authored by Wim Taymans's avatar Wim Taymans

gst/audioresample/: Fix audioresample, seek torture, new segments, reverse...

gst/audioresample/: Fix audioresample, seek torture, new segments, reverse negotiation etc.. work fine.

Original commit message from CVS:
* gst/audioresample/buffer.c: (audioresample_buffer_queue_flush):
* gst/audioresample/buffer.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c: (resample_input_flush),
(resample_input_pushthrough), (resample_input_eos),
(resample_get_output_size_for_input),
(resample_get_input_size_for_output), (resample_get_output_size),
(resample_get_output_data):
* gst/audioresample/resample.h:
* gst/audioresample/resample_ref.c: (resample_scale_ref):
Fix audioresample, seek torture, new segments, reverse negotiation
etc.. work fine.
parent 130b68ae
......@@ -237,3 +237,17 @@ audioresample_buffer_queue_peek (AudioresampleBufferQueue * queue, int length)
return newbuffer;
}
void
audioresample_buffer_queue_flush (AudioresampleBufferQueue * queue)
{
GList *g;
for (g = g_list_first (queue->buffers); g; g = g_list_next (g)) {
audioresample_buffer_unref ((AudioresampleBuffer *) g->data);
}
g_list_free (queue->buffers);
queue->buffers = NULL;
queue->depth = 0;
queue->offset = 0;
}
......@@ -28,21 +28,24 @@ struct _AudioresampleBufferQueue
int offset;
};
AudioresampleBuffer *audioresample_buffer_new (void);
AudioresampleBuffer *audioresample_buffer_new_and_alloc (int size);
AudioresampleBuffer *audioresample_buffer_new_with_data (void *data, int size);
AudioresampleBuffer *audioresample_buffer_new_subbuffer (AudioresampleBuffer * buffer, int offset,
int length);
void audioresample_buffer_ref (AudioresampleBuffer * buffer);
void audioresample_buffer_unref (AudioresampleBuffer * buffer);
AudioresampleBufferQueue *audioresample_buffer_queue_new (void);
void audioresample_buffer_queue_free (AudioresampleBufferQueue * queue);
int audioresample_buffer_queue_get_depth (AudioresampleBufferQueue * queue);
int audioresample_buffer_queue_get_offset (AudioresampleBufferQueue * queue);
void audioresample_buffer_queue_push (AudioresampleBufferQueue * queue,
AudioresampleBuffer * buffer);
AudioresampleBuffer *audioresample_buffer_queue_pull (AudioresampleBufferQueue * queue, int len);
AudioresampleBuffer *audioresample_buffer_queue_peek (AudioresampleBufferQueue * queue, int len);
AudioresampleBuffer * audioresample_buffer_new (void);
AudioresampleBuffer * audioresample_buffer_new_and_alloc (int size);
AudioresampleBuffer * audioresample_buffer_new_with_data (void *data, int size);
AudioresampleBuffer * audioresample_buffer_new_subbuffer (AudioresampleBuffer * buffer,
int offset,
int length);
void audioresample_buffer_ref (AudioresampleBuffer * buffer);
void audioresample_buffer_unref (AudioresampleBuffer * buffer);
AudioresampleBufferQueue *
audioresample_buffer_queue_new (void);
void audioresample_buffer_queue_free (AudioresampleBufferQueue * queue);
int audioresample_buffer_queue_get_depth (AudioresampleBufferQueue * queue);
int audioresample_buffer_queue_get_offset (AudioresampleBufferQueue * queue);
void audioresample_buffer_queue_push (AudioresampleBufferQueue * queue,
AudioresampleBuffer * buffer);
AudioresampleBuffer * audioresample_buffer_queue_pull (AudioresampleBufferQueue * queue, int len);
AudioresampleBuffer * audioresample_buffer_queue_peek (AudioresampleBufferQueue * queue, int len);
void audioresample_buffer_queue_flush (AudioresampleBufferQueue * queue);
#endif
......@@ -48,6 +48,8 @@ enum
LAST_SIGNAL
};
#define DEFAULT_FILTERLEN 16
enum
{
ARG_0,
......@@ -97,8 +99,12 @@ GST_STATIC_CAPS ( \
GstCaps * outcaps, guint * outsize);
gboolean audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps);
static GstFlowReturn audioresample_pushthrough (GstAudioresample *
audioresample);
static GstFlowReturn audioresample_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
static gboolean audioresample_event (GstBaseTransform * base,
GstEvent * event);
/*static guint gst_audioresample_signals[LAST_SIGNAL] = { 0 }; */
......@@ -133,7 +139,8 @@ static void gst_audioresample_class_init (GstAudioresampleClass * klass)
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FILTERLEN,
g_param_spec_int ("filter_length", "filter_length", "filter_length",
0, G_MAXINT, 16, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
0, G_MAXINT, DEFAULT_FILTERLEN,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
GST_DEBUG_FUNCPTR (audioresample_transform_size);
......@@ -145,19 +152,32 @@ static void gst_audioresample_class_init (GstAudioresampleClass * klass)
GST_DEBUG_FUNCPTR (audioresample_set_caps);
GST_BASE_TRANSFORM_CLASS (klass)->transform =
GST_DEBUG_FUNCPTR (audioresample_transform);
GST_BASE_TRANSFORM_CLASS (klass)->event =
GST_DEBUG_FUNCPTR (audioresample_event);
GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
}
static void gst_audioresample_init (GstAudioresample * audioresample,
static void
gst_audioresample_init (GstAudioresample * audioresample,
GstAudioresampleClass * klass)
{
ResampleState *r;
GstBaseTransform *trans;
trans = GST_BASE_TRANSFORM (audioresample);
/* buffer alloc passthrough is too impossible. FIXME, it
* is trivial in the passtrough case. */
gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
r = resample_new ();
audioresample->resample = r;
audioresample->ts_offset = -1;
audioresample->offset = -1;
audioresample->next_ts = -1;
resample_set_filter_length (r, 64);
resample_set_filter_length (r, DEFAULT_FILTERLEN);
resample_set_format (r, RESAMPLE_FORMAT_S16);
}
......@@ -197,16 +217,14 @@ gboolean
GstCaps *audioresample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps)
{
GstCaps *temp, *res;
const GstCaps *templcaps;
GstCaps *res;
GstStructure *structure;
temp = gst_caps_copy (caps);
structure = gst_caps_get_structure (temp, 0);
gst_structure_remove_field (structure, "rate");
templcaps = gst_pad_get_pad_template_caps (base->srcpad);
res = gst_caps_intersect (templcaps, temp);
gst_caps_unref (temp);
/* transform caps gives one single caps so we can just replace
* the rate property with our range. */
res = gst_caps_copy (caps);
structure = gst_caps_get_structure (res, 0);
gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
return res;
}
......@@ -286,6 +304,7 @@ gboolean
GST_DEBUG_OBJECT (audioresample,
"caps are not the set caps, creating state");
state = resample_new ();
resample_set_filter_length (state, audioresample->filter_length);
resample_set_state_from_caps (state, sinkcaps, srccaps, NULL, NULL, NULL);
}
......@@ -293,12 +312,9 @@ gboolean
/* asked to convert size of an incoming buffer */
*othersize = resample_get_output_size_for_input (state, size);
} else {
/* take a best guess, this is called cheating */
*othersize = floor (size * state->i_rate / state->o_rate);
*othersize -= *othersize % state->sample_size;
/* asked to convert size of an outgoing buffer */
*othersize = resample_get_input_size_for_output (state, size);
}
*othersize += state->sample_size;
g_assert (*othersize % state->sample_size == 0);
/* we make room for one extra sample, given that the resampling filter
......@@ -346,35 +362,50 @@ gboolean
return TRUE;
}
static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event)
{
GstAudioresample *audioresample;
audioresample = GST_AUDIORESAMPLE (base);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
break;
case GST_EVENT_FLUSH_STOP:
resample_input_flush (audioresample->resample);
audioresample->ts_offset = -1;
audioresample->next_ts = -1;
audioresample->offset = -1;
break;
case GST_EVENT_NEWSEGMENT:
resample_input_pushthrough (audioresample->resample);
audioresample_pushthrough (audioresample);
audioresample->ts_offset = -1;
audioresample->next_ts = -1;
audioresample->offset = -1;
break;
case GST_EVENT_EOS:
resample_input_eos (audioresample->resample);
audioresample_pushthrough (audioresample);
break;
default:
break;
}
parent_class->event (base, event);
return TRUE;
}
static GstFlowReturn
audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
audioresample_do_output (GstAudioresample * audioresample,
GstBuffer * outbuf)
{
/* FIXME: this-> */
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
ResampleState *r;
guchar *data;
gulong size;
int outsize;
int outsamples;
/* FIXME: move to _inplace */
#if 0
if (audioresample->passthru) {
gst_pad_push (audioresample->srcpad, GST_DATA (buf));
return;
}
#endif
ResampleState *r;
r = audioresample->resample;
data = GST_BUFFER_DATA (inbuf);
size = GST_BUFFER_SIZE (inbuf);
GST_DEBUG_OBJECT (audioresample, "got buffer of %ld bytes", size);
resample_add_input_data (r, data, size, NULL, NULL);
outsize = resample_get_output_size (r);
GST_DEBUG_OBJECT (audioresample, "audioresample can give me %d bytes",
outsize);
......@@ -399,18 +430,27 @@ static GstFlowReturn
outsize, outsamples);
GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
GST_BUFFER_TIMESTAMP (outbuf) = base->segment.start +
audioresample->offset * GST_SECOND / audioresample->o_rate;
audioresample->offset += outsamples;
GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;
/* we calculate DURATION as the difference between "next" timestamp
* and current timestamp so we ensure a contiguous stream, instead of
* having rounding errors. */
GST_BUFFER_DURATION (outbuf) = base->segment.start +
audioresample->offset * GST_SECOND / audioresample->o_rate -
GST_BUFFER_TIMESTAMP (outbuf);
GST_BUFFER_TIMESTAMP (outbuf) = audioresample->next_ts;
if (audioresample->ts_offset != -1) {
audioresample->offset += outsamples;
audioresample->ts_offset += outsamples;
audioresample->next_ts =
gst_util_uint64_scale_int (audioresample->ts_offset, GST_SECOND,
audioresample->o_rate);
GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;
/* we calculate DURATION as the difference between "next" timestamp
* and current timestamp so we ensure a contiguous stream, instead of
* having rounding errors. */
GST_BUFFER_DURATION (outbuf) = audioresample->next_ts -
GST_BUFFER_TIMESTAMP (outbuf);
} else {
/* no valid offset know, we can still sortof calculate the duration though */
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (outsamples, GST_SECOND,
audioresample->o_rate);
}
/* check for possible mem corruption */
if (outsize > GST_BUFFER_SIZE (outbuf)) {
......@@ -429,10 +469,87 @@ static GstFlowReturn
"audioresample's written outsize %d too far from outbuffer's size %d",
outsize, GST_BUFFER_SIZE (outbuf));
}
GST_BUFFER_SIZE (outbuf) = outsize;
return GST_FLOW_OK;
}
static GstFlowReturn
audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstAudioresample *audioresample;
ResampleState *r;
guchar *data;
gulong size;
GstClockTime timestamp;
audioresample = GST_AUDIORESAMPLE (base);
r = audioresample->resample;
data = GST_BUFFER_DATA (inbuf);
size = GST_BUFFER_SIZE (inbuf);
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
GST_DEBUG_OBJECT (audioresample, "got buffer of %ld bytes", size);
if (audioresample->ts_offset == -1) {
/* if we don't know the initial offset yet, calculate it based on the
* input timestamp. */
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
GstClockTime stime;
/* offset used to calculate the timestamps. We use the sample offset for this
* to make it more accurate. We want the first buffer to have the same timestamp
* as the incomming timestamp. */
audioresample->next_ts = timestamp;
audioresample->ts_offset =
gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND);
/* offset used to set as the buffer offset, this offset is always relative
* to the stream time, note that timestamp is not... */
stime = (timestamp - base->segment.start) + base->segment.time;
audioresample->offset =
gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);
}
}
/* need to memdup, resample takes ownership. */
resample_add_input_data (r, g_memdup (data, size), size, NULL, NULL);
return audioresample_do_output (audioresample, outbuf);
}
/* push remaining data in the buffers out */
static GstFlowReturn
audioresample_pushthrough (GstAudioresample * audioresample)
{
int outsize;
ResampleState *r;
GstBuffer *outbuf;
GstFlowReturn res = GST_FLOW_OK;
GstBaseTransform *trans;
r = audioresample->resample;
outsize = resample_get_output_size (r);
if (outsize == 0)
goto done;
outbuf = gst_buffer_new_and_alloc (outsize);
res = audioresample_do_output (audioresample, outbuf);
if (res != GST_FLOW_OK)
goto done;
trans = GST_BASE_TRANSFORM (audioresample);
res = gst_pad_push (trans->srcpad, outbuf);
done:
return res;
}
static void
gst_audioresample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
......
......@@ -53,6 +53,8 @@ struct _GstAudioresample {
gboolean passthru;
guint64 offset;
guint64 ts_offset;
GstClockTime next_ts;
int channels;
int i_rate;
......
......@@ -121,19 +121,50 @@ resample_add_input_data (ResampleState * r, void *data, int size,
}
void
resample_input_eos (ResampleState * r)
resample_input_flush (ResampleState * r)
{
RESAMPLE_DEBUG ("flush");
audioresample_buffer_queue_flush (r->queue);
r->buffer_filled = 0;
r->need_reinit = 1;
}
void
resample_input_pushthrough (ResampleState * r)
{
AudioresampleBuffer *buffer;
int sample_size;
int filter_bytes;
int buffer_filled;
if (r->sample_size == 0)
return;
filter_bytes = r->filter_length * r->sample_size;
buffer_filled = r->buffer_filled;
sample_size = r->n_channels * resample_format_size (r->format);
RESAMPLE_DEBUG ("pushthrough filter_bytes %d, filled %d",
filter_bytes, buffer_filled);
buffer = audioresample_buffer_new_and_alloc (sample_size *
(r->filter_length / 2));
/* if we have no pending samples, we don't need to do anything. */
if (buffer_filled <= 0)
return;
/* send filter_length/2 number of samples so we can get to the
* last queued samples */
buffer = audioresample_buffer_new_and_alloc (filter_bytes / 2);
memset (buffer->data, 0, buffer->length);
RESAMPLE_DEBUG ("pushthrough", buffer->length);
audioresample_buffer_queue_push (r->queue, buffer);
}
void
resample_input_eos (ResampleState * r)
{
RESAMPLE_DEBUG ("EOS");
resample_input_pushthrough (r);
r->eos = 1;
}
......@@ -142,22 +173,61 @@ resample_get_output_size_for_input (ResampleState * r, int size)
{
int outsize;
double outd;
int avail;
int filter_bytes;
int buffer_filled;
if (r->sample_size == 0)
return 0;
filter_bytes = r->filter_length * r->sample_size;
buffer_filled = filter_bytes / 2 - r->buffer_filled / 2;
g_return_val_if_fail (r->sample_size != 0, 0);
avail =
audioresample_buffer_queue_get_depth (r->queue) + size - buffer_filled;
RESAMPLE_DEBUG ("avail %d, o_rate %f, i_rate %f, filter_bytes %d, filled %d",
avail, r->o_rate, r->i_rate, filter_bytes, buffer_filled);
if (avail <= 0)
return 0;
outd = (double) avail *r->o_rate / r->i_rate;
RESAMPLE_DEBUG ("size %d, o_rate %f, i_rate %f", size, r->o_rate, r->i_rate);
outd = (double) size / r->i_rate * r->o_rate;
outsize = (int) floor (outd);
/* round off for sample size */
return outsize - (outsize % r->sample_size);
outsize -= outsize % r->sample_size;
return outsize;
}
int
resample_get_input_size_for_output (ResampleState * r, int size)
{
int outsize;
double outd;
int avail;
if (r->sample_size == 0)
return 0;
avail = size;
RESAMPLE_DEBUG ("size %d, o_rate %f, i_rate %f", avail, r->o_rate, r->i_rate);
outd = (double) avail *r->i_rate / r->o_rate;
outsize = (int) ceil (outd);
/* round off for sample size */
outsize -= outsize % r->sample_size;
return outsize;
}
int
resample_get_output_size (ResampleState * r)
{
return resample_get_output_size_for_input (r,
audioresample_buffer_queue_get_depth (r->queue));
return resample_get_output_size_for_input (r, 0);
}
int
......@@ -166,6 +236,9 @@ resample_get_output_data (ResampleState * r, void *data, int size)
r->o_buf = data;
r->o_size = size;
if (size == 0)
return 0;
switch (r->method) {
case 0:
resample_scale_ref (r);
......
......@@ -67,6 +67,7 @@ struct _ResampleState {
void *buffer;
int buffer_len;
int buffer_filled;
double i_start;
double o_start;
......@@ -98,8 +99,12 @@ void resample_free (ResampleState *state);
void resample_add_input_data (ResampleState * r, void *data, int size,
ResampleCallback free_func, void *closure);
void resample_input_eos (ResampleState *r);
void resample_input_flush (ResampleState *r);
void resample_input_pushthrough (ResampleState *r);
int resample_get_output_size_for_input (ResampleState * r, int size);
int resample_get_input_size_for_output (ResampleState * r, int size);
int resample_get_output_size (ResampleState *r);
int resample_get_output_data (ResampleState *r, void *data, int size);
......
......@@ -63,6 +63,7 @@ resample_scale_ref (ResampleState * r)
r->buffer_len = r->sample_size * r->filter_length;
r->buffer = malloc (r->buffer_len);
memset (r->buffer, 0, r->buffer_len);
r->buffer_filled = 0;
r->i_inc = r->o_rate / r->i_rate;
r->o_inc = r->i_rate / r->o_rate;
......@@ -127,6 +128,8 @@ resample_scale_ref (ResampleState * r)
memcpy (r->buffer + r->buffer_len - r->sample_size, buffer->data,
r->sample_size);
r->buffer_filled = MIN (r->buffer_filled + r->sample_size, r->buffer_len);
audioresample_buffer_unref (buffer);
}
......
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