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Revert "audioresample: Add GAP flag support"

Browse source 
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 2010-09-15 11:28:29 +02:00
parent 7ac1938f0f
commit 35c76b3409
5 changed files with 54 additions and 175 deletions
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149
gst/audioresample/gstaudioresample.c
View file

@ -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;
@ -1032,51 +983,6 @@ 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 */
@ -1108,16 +1014,15 @@ gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
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 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 (in_len != in_processed)) {
GST_WARNING_OBJECT (resample, "converted %d of %d input samples",
@ -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,

3
gst/audioresample/gstaudioresample.h
View file

@ -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;

6
gst/audioresample/resample.c
View file

@ -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)
{

6
gst/audioresample/speex_resampler.h
View file

@ -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

7
gst/audioresample/speex_resampler_wrapper.h
View file

@ -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,