mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-02 16:52:42 +00:00
Revert "audioresample: Add GAP flag support"
This reverts commit 129af0d8e6.
This shouldn't be committed at all, it isn't ready and apparently
was in the wrong branch locally.
This commit is contained in:
Sebastian Dröge
parent
7ac1938f0f
commit
35c76b3409
5 changed files with 54 additions and 175 deletions
|
|
@ -223,7 +223,6 @@ gst_audio_resample_init (GstAudioResample * resample,
|
|||
|
||||
resample->quality = SPEEX_RESAMPLER_QUALITY_DEFAULT;
|
||||
|
||||
gst_base_transform_set_gap_aware (trans, TRUE);
|
||||
gst_pad_set_query_function (trans->srcpad, gst_audio_resample_query);
|
||||
gst_pad_set_query_type_function (trans->srcpad,
|
||||
gst_audio_resample_query_type);
|
||||
|
|
@ -237,8 +236,6 @@ gst_audio_resample_start (GstBaseTransform * base)
|
|||
|
||||
resample->need_discont = TRUE;
|
||||
|
||||
resample->count_gap = 0;
|
||||
resample->count_nongap = 0;
|
||||
resample->t0 = GST_CLOCK_TIME_NONE;
|
||||
resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
|
||||
resample->out_offset0 = GST_BUFFER_OFFSET_NONE;
|
||||
|
|
@ -388,6 +385,8 @@ gst_audio_resample_init_state (GstAudioResample * resample, gint width,
|
|||
return NULL;
|
||||
}
|
||||
|
||||
funcs->skip_zeros (ret);
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
|
|
@ -781,48 +780,18 @@ gst_audio_resample_workspace_realloc (guint8 ** workspace, guint * size,
|
|||
return *workspace;
|
||||
}
|
||||
|
||||
/* Push history_len zeros into the filter, but discard the output. */
|
||||
static void
|
||||
gst_audio_resample_dump_drain (GstAudioResample * resample, guint history_len)
|
||||
{
|
||||
gint outsize;
|
||||
guint in_len, in_processed;
|
||||
guint out_len, out_processed;
|
||||
guint num, den;
|
||||
void *buf;
|
||||
|
||||
g_assert (resample->state != NULL);
|
||||
|
||||
resample->funcs->get_ratio (resample->state, &num, &den);
|
||||
|
||||
in_len = in_processed = history_len;
|
||||
out_processed = out_len =
|
||||
gst_util_uint64_scale_int_ceil (history_len, den, num);
|
||||
outsize = out_len * resample->channels * (resample->funcs->width / 8);
|
||||
|
||||
if (out_len == 0)
|
||||
return;
|
||||
|
||||
buf = g_malloc (outsize);
|
||||
resample->funcs->process (resample->state, NULL, &in_processed, buf,
|
||||
&out_processed);
|
||||
g_free (buf);
|
||||
|
||||
g_assert (in_len == in_processed);
|
||||
}
|
||||
|
||||
static void
|
||||
gst_audio_resample_push_drain (GstAudioResample * resample, guint history_len)
|
||||
gst_audio_resample_push_drain (GstAudioResample * resample)
|
||||
{
|
||||
GstBuffer *outbuf;
|
||||
GstFlowReturn res;
|
||||
gint outsize;
|
||||
guint in_len, in_processed;
|
||||
guint out_len, out_processed;
|
||||
guint history_len, out_len, out_processed;
|
||||
gint err;
|
||||
guint num, den;
|
||||
|
||||
g_assert (resample->state != NULL);
|
||||
if (!resample->state)
|
||||
return;
|
||||
|
||||
/* Don't drain samples if we were reset. */
|
||||
if (!GST_CLOCK_TIME_IS_VALID (resample->t0))
|
||||
|
|
@ -830,14 +799,11 @@ gst_audio_resample_push_drain (GstAudioResample * resample, guint history_len)
|
|||
|
||||
resample->funcs->get_ratio (resample->state, &num, &den);
|
||||
|
||||
in_len = in_processed = history_len;
|
||||
history_len = resample->funcs->get_input_latency (resample->state);
|
||||
out_len = out_processed =
|
||||
gst_util_uint64_scale_int_ceil (history_len, den, num);
|
||||
outsize = out_len * resample->channels * (resample->width / 8);
|
||||
|
||||
if (out_len == 0)
|
||||
return;
|
||||
|
||||
res =
|
||||
gst_pad_alloc_buffer_and_set_caps (GST_BASE_TRANSFORM_SRC_PAD (resample),
|
||||
GST_BUFFER_OFFSET_NONE, outsize,
|
||||
|
|
@ -858,7 +824,7 @@ gst_audio_resample_push_drain (GstAudioResample * resample, guint history_len)
|
|||
}
|
||||
|
||||
/* process */
|
||||
err = resample->funcs->process (resample->state, NULL, &in_processed,
|
||||
err = resample->funcs->process (resample->state, NULL, &history_len,
|
||||
resample->tmp_out, &out_processed);
|
||||
|
||||
/* convert output format */
|
||||
|
|
@ -866,7 +832,7 @@ gst_audio_resample_push_drain (GstAudioResample * resample, guint history_len)
|
|||
GST_BUFFER_DATA (outbuf), out_processed, TRUE);
|
||||
} else {
|
||||
/* don't need to convert data format; process */
|
||||
err = resample->funcs->process (resample->state, NULL, &in_processed,
|
||||
err = resample->funcs->process (resample->state, NULL, &history_len,
|
||||
GST_BUFFER_DATA (outbuf), &out_processed);
|
||||
}
|
||||
|
||||
|
|
@ -881,6 +847,12 @@ gst_audio_resample_push_drain (GstAudioResample * resample, guint history_len)
|
|||
return;
|
||||
}
|
||||
|
||||
if (G_UNLIKELY (out_processed == 0)) {
|
||||
GST_WARNING_OBJECT (resample, "Failed to get drain, dropping buffer");
|
||||
gst_buffer_unref (outbuf);
|
||||
return;
|
||||
}
|
||||
|
||||
if (GST_CLOCK_TIME_IS_VALID (resample->t0)) {
|
||||
GST_BUFFER_OFFSET (outbuf) = resample->next_out_offset;
|
||||
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET (outbuf) + out_processed;
|
||||
|
|
@ -892,7 +864,7 @@ gst_audio_resample_push_drain (GstAudioResample * resample, guint history_len)
|
|||
resample->out_offset0, GST_SECOND, resample->outrate) -
|
||||
GST_BUFFER_TIMESTAMP (outbuf);
|
||||
resample->next_out_offset += out_processed;
|
||||
resample->next_in_offset += history_len;
|
||||
resample->next_in_offset += 0;
|
||||
} else {
|
||||
GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
|
||||
GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
|
||||
|
|
@ -900,12 +872,6 @@ gst_audio_resample_push_drain (GstAudioResample * resample, guint history_len)
|
|||
GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
|
||||
}
|
||||
|
||||
if (G_UNLIKELY (out_processed == 0 && in_len * den > num)) {
|
||||
GST_WARNING_OBJECT (resample, "Failed to get drain, dropping buffer");
|
||||
gst_buffer_unref (outbuf);
|
||||
return;
|
||||
}
|
||||
|
||||
GST_BUFFER_SIZE (outbuf) =
|
||||
out_processed * resample->channels * (resample->width / 8);
|
||||
|
||||
|
|
@ -934,11 +900,6 @@ gst_audio_resample_event (GstBaseTransform * base, GstEvent * event)
|
|||
switch (GST_EVENT_TYPE (event)) {
|
||||
case GST_EVENT_FLUSH_STOP:
|
||||
gst_audio_resample_reset_state (resample);
|
||||
if (resample->state)
|
||||
resample->count_gap = resample->funcs->get_filt_len (resample->state);
|
||||
else
|
||||
resample->count_gap = 0;
|
||||
resample->count_nongap = 0;
|
||||
resample->t0 = GST_CLOCK_TIME_NONE;
|
||||
resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
|
||||
resample->out_offset0 = GST_BUFFER_OFFSET_NONE;
|
||||
|
|
@ -947,14 +908,8 @@ gst_audio_resample_event (GstBaseTransform * base, GstEvent * event)
|
|||
resample->need_discont = TRUE;
|
||||
break;
|
||||
case GST_EVENT_NEWSEGMENT:
|
||||
if (resample->state)
|
||||
gst_audio_resample_push_drain (resample, resample->count_nongap);
|
||||
gst_audio_resample_push_drain (resample);
|
||||
gst_audio_resample_reset_state (resample);
|
||||
if (resample->state)
|
||||
resample->count_gap = resample->funcs->get_filt_len (resample->state);
|
||||
else
|
||||
resample->count_gap = 0;
|
||||
resample->count_nongap = 0;
|
||||
resample->t0 = GST_CLOCK_TIME_NONE;
|
||||
resample->in_offset0 = GST_BUFFER_OFFSET_NONE;
|
||||
resample->out_offset0 = GST_BUFFER_OFFSET_NONE;
|
||||
|
|
@ -963,8 +918,7 @@ gst_audio_resample_event (GstBaseTransform * base, GstEvent * event)
|
|||
resample->need_discont = TRUE;
|
||||
break;
|
||||
case GST_EVENT_EOS:
|
||||
if (resample->state)
|
||||
gst_audio_resample_push_drain (resample, resample->count_nongap);
|
||||
gst_audio_resample_push_drain (resample);
|
||||
gst_audio_resample_reset_state (resample);
|
||||
break;
|
||||
default:
|
||||
|
|
@ -979,9 +933,6 @@ gst_audio_resample_check_discont (GstAudioResample * resample, GstBuffer * buf)
|
|||
{
|
||||
guint64 offset;
|
||||
guint64 delta;
|
||||
guint filt_len = resample->funcs->get_filt_len (resample->state);
|
||||
guint64 delay =
|
||||
gst_util_uint64_scale_round (filt_len, GST_SECOND, 2 * resample->inrate);
|
||||
|
||||
/* is the incoming buffer a discontinuity? */
|
||||
if (G_UNLIKELY (GST_BUFFER_IS_DISCONT (buf)))
|
||||
|
|
@ -998,7 +949,7 @@ gst_audio_resample_check_discont (GstAudioResample * resample, GstBuffer * buf)
|
|||
offset =
|
||||
resample->in_offset0 +
|
||||
gst_util_uint64_scale_int_round (GST_BUFFER_TIMESTAMP (buf) -
|
||||
resample->t0 - delay, resample->inrate, GST_SECOND);
|
||||
resample->t0, resample->inrate, GST_SECOND);
|
||||
|
||||
/* many elements generate imperfect streams due to rounding errors, so we
|
||||
* permit a small error (up to one sample) without triggering a filter
|
||||
|
|
@ -1021,7 +972,7 @@ gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
|
|||
{
|
||||
guint32 in_len, in_processed;
|
||||
guint32 out_len, out_processed;
|
||||
guint filt_len = resample->funcs->get_filt_len (resample->state);
|
||||
gint err;
|
||||
|
||||
in_len = GST_BUFFER_SIZE (inbuf) / resample->channels;
|
||||
out_len = GST_BUFFER_SIZE (outbuf) / resample->channels;
|
||||
|
|
@ -1032,93 +983,47 @@ gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
|
|||
in_processed = in_len;
|
||||
out_processed = out_len;
|
||||
|
||||
if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
|
||||
resample->count_nongap = 0;
|
||||
if (resample->count_gap < filt_len) {
|
||||
guint zeros_to_push;
|
||||
if (in_len >= filt_len - resample->count_gap)
|
||||
zeros_to_push = filt_len - resample->count_gap;
|
||||
else
|
||||
zeros_to_push = in_len;
|
||||
|
||||
gst_audio_resample_push_drain (resample, zeros_to_push);
|
||||
in_len -= zeros_to_push;
|
||||
resample->count_gap += zeros_to_push;
|
||||
}
|
||||
|
||||
{
|
||||
guint num, den;
|
||||
resample->funcs->get_ratio (resample->state, &num, &den);
|
||||
out_processed =
|
||||
gst_util_uint64_scale_int_ceil (resample->next_in_offset -
|
||||
resample->in_offset0 + in_len, den,
|
||||
num) - resample->next_out_offset + resample->out_offset0;
|
||||
|
||||
memset (GST_BUFFER_DATA (outbuf), 0, GST_BUFFER_SIZE (outbuf));
|
||||
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
|
||||
resample->count_gap += in_len;
|
||||
in_processed = in_len;
|
||||
}
|
||||
} else { /* not a gap */
|
||||
|
||||
gint err;
|
||||
|
||||
if (resample->count_gap > filt_len) {
|
||||
/* push in enough zeros to restore the filter to the right offset */
|
||||
guint num, den;
|
||||
resample->funcs->get_ratio (resample->state, &num, &den);
|
||||
gst_audio_resample_dump_drain (resample,
|
||||
(resample->count_gap - filt_len) % num);
|
||||
}
|
||||
resample->count_gap = 0;
|
||||
if (resample->count_nongap < filt_len) {
|
||||
resample->count_nongap += in_len;
|
||||
if (resample->count_nongap > filt_len)
|
||||
resample->count_nongap = filt_len;
|
||||
}
|
||||
|
||||
if (resample->funcs->width != resample->width) {
|
||||
/* need to convert data format for processing; ensure we have enough
|
||||
* workspace available */
|
||||
if (!gst_audio_resample_workspace_realloc (&resample->tmp_in,
|
||||
&resample->tmp_in_size, in_len * resample->channels *
|
||||
(resample->funcs->width / 8)) ||
|
||||
!gst_audio_resample_workspace_realloc (&resample->tmp_out,
|
||||
&resample->tmp_out_size, out_len * resample->channels *
|
||||
(resample->funcs->width / 8))) {
|
||||
GST_ERROR_OBJECT (resample, "failed to allocate workspace");
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
|
||||
/* convert input */
|
||||
gst_audio_resample_convert_buffer (resample, GST_BUFFER_DATA (inbuf),
|
||||
resample->tmp_in, in_len, FALSE);
|
||||
|
||||
/* process */
|
||||
err = resample->funcs->process (resample->state,
|
||||
resample->tmp_in, &in_processed, resample->tmp_out, &out_processed);
|
||||
|
||||
/* convert output */
|
||||
gst_audio_resample_convert_buffer (resample, resample->tmp_out,
|
||||
GST_BUFFER_DATA (outbuf), out_processed, TRUE);
|
||||
} else {
|
||||
/* no format conversion required; process */
|
||||
err = resample->funcs->process (resample->state,
|
||||
GST_BUFFER_DATA (inbuf), &in_processed,
|
||||
GST_BUFFER_DATA (outbuf), &out_processed);
|
||||
}
|
||||
|
||||
if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) {
|
||||
GST_ERROR_OBJECT (resample, "Failed to convert data: %s",
|
||||
resample->funcs->strerror (err));
|
||||
if (resample->funcs->width != resample->width) {
|
||||
/* need to convert data format for processing; ensure we have enough
|
||||
* workspace available */
|
||||
if (!gst_audio_resample_workspace_realloc (&resample->tmp_in,
|
||||
&resample->tmp_in_size, in_len * resample->channels *
|
||||
(resample->funcs->width / 8)) ||
|
||||
!gst_audio_resample_workspace_realloc (&resample->tmp_out,
|
||||
&resample->tmp_out_size, out_len * resample->channels *
|
||||
(resample->funcs->width / 8))) {
|
||||
GST_ERROR_OBJECT (resample, "failed to allocate workspace");
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
|
||||
/* convert input */
|
||||
gst_audio_resample_convert_buffer (resample, GST_BUFFER_DATA (inbuf),
|
||||
resample->tmp_in, in_len, FALSE);
|
||||
|
||||
/* process */
|
||||
err = resample->funcs->process (resample->state,
|
||||
resample->tmp_in, &in_processed, resample->tmp_out, &out_processed);
|
||||
|
||||
/* convert output */
|
||||
gst_audio_resample_convert_buffer (resample, resample->tmp_out,
|
||||
GST_BUFFER_DATA (outbuf), out_processed, TRUE);
|
||||
} else {
|
||||
/* no format conversion required; process */
|
||||
err = resample->funcs->process (resample->state,
|
||||
GST_BUFFER_DATA (inbuf), &in_processed,
|
||||
GST_BUFFER_DATA (outbuf), &out_processed);
|
||||
}
|
||||
|
||||
/* If we wrote more than allocated something is really wrong now and we
|
||||
* should better abort immediately */
|
||||
g_assert (out_len >= out_processed);
|
||||
|
||||
if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) {
|
||||
GST_ERROR_OBJECT (resample, "Failed to convert data: %s",
|
||||
resample->funcs->strerror (err));
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
|
||||
if (G_UNLIKELY (in_len != in_processed)) {
|
||||
GST_WARNING_OBJECT (resample, "converted %d of %d input samples",
|
||||
in_processed, in_len);
|
||||
|
|
@ -1203,11 +1108,7 @@ gst_audio_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
|
|||
/* resync the timestamp and offset counters if possible */
|
||||
if (GST_BUFFER_TIMESTAMP_IS_VALID (inbuf) &&
|
||||
GST_BUFFER_OFFSET_IS_VALID (inbuf)) {
|
||||
guint filt_len = resample->funcs->get_filt_len (resample->state);
|
||||
guint64 delay = gst_util_uint64_scale_round (filt_len, GST_SECOND,
|
||||
2 * resample->inrate);
|
||||
resample->count_gap = resample->funcs->get_filt_len (resample->state);
|
||||
resample->t0 = GST_BUFFER_TIMESTAMP (inbuf) - delay;
|
||||
resample->t0 = GST_BUFFER_TIMESTAMP (inbuf);
|
||||
resample->in_offset0 = GST_BUFFER_OFFSET (inbuf);
|
||||
resample->out_offset0 =
|
||||
gst_util_uint64_scale_int_round (resample->in_offset0,
|
||||
|
|
|
|||
|
|
@ -64,9 +64,6 @@ struct _GstAudioResample {
|
|||
guint64 next_in_offset;
|
||||
guint64 next_out_offset;
|
||||
|
||||
guint count_gap;
|
||||
guint count_nongap;
|
||||
|
||||
gint channels;
|
||||
gint inrate;
|
||||
gint outrate;
|
||||
|
|
|
|||
|
|
@ -1310,12 +1310,6 @@ speex_resampler_get_output_latency (SpeexResamplerState * st)
|
|||
(st->num_rate >> 1)) / st->num_rate;
|
||||
}
|
||||
|
||||
EXPORT int
|
||||
speex_resampler_get_filt_len (SpeexResamplerState * st)
|
||||
{
|
||||
return st->filt_len;
|
||||
}
|
||||
|
||||
EXPORT int
|
||||
speex_resampler_skip_zeros (SpeexResamplerState * st)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -73,7 +73,6 @@
|
|||
#define speex_resampler_get_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_stride)
|
||||
#define speex_resampler_get_input_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_latency)
|
||||
#define speex_resampler_get_output_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_latency)
|
||||
#define speex_resampler_get_filt_len CAT_PREFIX(RANDOM_PREFIX,_resampler_get_filt_len)
|
||||
#define speex_resampler_skip_zeros CAT_PREFIX(RANDOM_PREFIX,_resampler_skip_zeros)
|
||||
#define speex_resampler_reset_mem CAT_PREFIX(RANDOM_PREFIX,_resampler_reset_mem)
|
||||
#define speex_resampler_strerror CAT_PREFIX(RANDOM_PREFIX,_resampler_strerror)
|
||||
|
|
@ -334,11 +333,6 @@ int speex_resampler_get_input_latency(SpeexResamplerState *st);
|
|||
*/
|
||||
int speex_resampler_get_output_latency(SpeexResamplerState *st);
|
||||
|
||||
/** Get the length of the filter in input samples.
|
||||
* @param st Resampler state
|
||||
*/
|
||||
int speex_resampler_get_filt_len(SpeexResamplerState *st);
|
||||
|
||||
/** Make sure that the first samples to go out of the resamplers don't have
|
||||
* leading zeros. This is only useful before starting to use a newly created
|
||||
* resampler. It is recommended to use that when resampling an audio file, as
|
||||
|
|
|
|||
|
|
@ -52,7 +52,6 @@ typedef struct {
|
|||
void (*get_ratio) (SpeexResamplerState * st,
|
||||
guint32 * ratio_num, guint32 * ratio_den);
|
||||
int (*get_input_latency) (SpeexResamplerState * st);
|
||||
int (*get_filt_len) (SpeexResamplerState * st);
|
||||
int (*set_quality) (SpeexResamplerState * st, gint quality);
|
||||
int (*reset_mem) (SpeexResamplerState * st);
|
||||
int (*skip_zeros) (SpeexResamplerState * st);
|
||||
|
|
@ -72,7 +71,6 @@ void resample_float_resampler_get_rate (SpeexResamplerState * st,
|
|||
void resample_float_resampler_get_ratio (SpeexResamplerState * st,
|
||||
guint32 * ratio_num, guint32 * ratio_den);
|
||||
int resample_float_resampler_get_input_latency (SpeexResamplerState * st);
|
||||
int resample_float_resampler_get_filt_len (SpeexResamplerState * st);
|
||||
int resample_float_resampler_set_quality (SpeexResamplerState * st, gint quality);
|
||||
int resample_float_resampler_reset_mem (SpeexResamplerState * st);
|
||||
int resample_float_resampler_skip_zeros (SpeexResamplerState * st);
|
||||
|
|
@ -87,7 +85,6 @@ static const SpeexResampleFuncs float_funcs =
|
|||
resample_float_resampler_get_rate,
|
||||
resample_float_resampler_get_ratio,
|
||||
resample_float_resampler_get_input_latency,
|
||||
resample_float_resampler_get_filt_len,
|
||||
resample_float_resampler_set_quality,
|
||||
resample_float_resampler_reset_mem,
|
||||
resample_float_resampler_skip_zeros,
|
||||
|
|
@ -107,7 +104,6 @@ void resample_double_resampler_get_rate (SpeexResamplerState * st,
|
|||
void resample_double_resampler_get_ratio (SpeexResamplerState * st,
|
||||
guint32 * ratio_num, guint32 * ratio_den);
|
||||
int resample_double_resampler_get_input_latency (SpeexResamplerState * st);
|
||||
int resample_double_resampler_get_filt_len (SpeexResamplerState * st);
|
||||
int resample_double_resampler_set_quality (SpeexResamplerState * st, gint quality);
|
||||
int resample_double_resampler_reset_mem (SpeexResamplerState * st);
|
||||
int resample_double_resampler_skip_zeros (SpeexResamplerState * st);
|
||||
|
|
@ -122,7 +118,6 @@ static const SpeexResampleFuncs double_funcs =
|
|||
resample_double_resampler_get_rate,
|
||||
resample_double_resampler_get_ratio,
|
||||
resample_double_resampler_get_input_latency,
|
||||
resample_double_resampler_get_filt_len,
|
||||
resample_double_resampler_set_quality,
|
||||
resample_double_resampler_reset_mem,
|
||||
resample_double_resampler_skip_zeros,
|
||||
|
|
@ -142,7 +137,6 @@ void resample_int_resampler_get_rate (SpeexResamplerState * st,
|
|||
void resample_int_resampler_get_ratio (SpeexResamplerState * st,
|
||||
guint32 * ratio_num, guint32 * ratio_den);
|
||||
int resample_int_resampler_get_input_latency (SpeexResamplerState * st);
|
||||
int resample_int_resampler_get_filt_len (SpeexResamplerState * st);
|
||||
int resample_int_resampler_set_quality (SpeexResamplerState * st, gint quality);
|
||||
int resample_int_resampler_reset_mem (SpeexResamplerState * st);
|
||||
int resample_int_resampler_skip_zeros (SpeexResamplerState * st);
|
||||
|
|
@ -157,7 +151,6 @@ static const SpeexResampleFuncs int_funcs =
|
|||
resample_int_resampler_get_rate,
|
||||
resample_int_resampler_get_ratio,
|
||||
resample_int_resampler_get_input_latency,
|
||||
resample_int_resampler_get_filt_len,
|
||||
resample_int_resampler_set_quality,
|
||||
resample_int_resampler_reset_mem,
|
||||
resample_int_resampler_skip_zeros,
|
||||
|
|
|
|||
Loading…
Reference in a new issue