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audio-resampler: add reset function

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Add a function to reset the audio-resampler.
Use new function in audio-converter
Use the new functions in gstaudioresample and fixup drain functions.
This commit is contained in:
Wim Taymans 2016-01-26 16:42:16 +01:00
parent ea469ad9a8
commit ed747492ef
5 changed files with 49 additions and 44 deletions
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2
gst-libs/gst/audio/audio-converter.c
View file

@ -1088,6 +1088,8 @@ gst_audio_converter_get_max_latency (GstAudioConverter * convert)
void
gst_audio_converter_reset (GstAudioConverter * convert)
{
if (convert->resampler)
gst_audio_resampler_reset (convert->resampler);
if (convert->quant)
gst_audio_quantize_reset (convert->quant);
}

28
gst-libs/gst/audio/audio-resampler.c
View file

@ -936,6 +936,34 @@ get_sample_bufs (GstAudioResampler * resampler, gsize need)
return resampler->sbuf;
}
/**
* gst_audio_resampler_reset:
* @resampler: a #GstAudioResampler
*
* Reset @resampler to the state it was when it was first created, discarding
* all sample history.
*/
void
gst_audio_resampler_reset (GstAudioResampler * resampler)
{
g_return_if_fail (resampler != NULL);
if (resampler->samples) {
gsize bytes;
gint c, blocks, bpf;
bpf = resampler->bps * resampler->inc;
bytes = (resampler->n_taps / 2) * bpf;
blocks = resampler->blocks;
for (c = 0; c < blocks; c++)
memset (resampler->sbuf[c], 0, bytes);
}
/* half of the filter is filled with 0 */
resampler->samp_index = 0;
resampler->samples_avail = resampler->n_taps / 2 - 1;
}
/**
* gst_audio_resampler_update:
* @resampler: a #GstAudioResampler

1
gst-libs/gst/audio/audio-resampler.h
View file

@ -164,6 +164,7 @@ GstAudioResampler * gst_audio_resampler_new (GstAudioResamplerMetho
GstStructure *options);
void gst_audio_resampler_free (GstAudioResampler *resampler);
void gst_audio_resampler_reset (GstAudioResampler *resampler);
gboolean gst_audio_resampler_update (GstAudioResampler *resampler,
guint in_rate, guint out_rate,

61
gst/audioresample/gstaudioresample.c
View file

@ -429,6 +429,8 @@ update_failed:
static void
gst_audio_resample_reset_state (GstAudioResample * resample)
{
if (resample->converter)
gst_audio_converter_reset (resample->converter);
}
static gboolean
@ -507,32 +509,20 @@ invalid_outcaps:
static void
gst_audio_resample_dump_drain (GstAudioResample * resample, guint history_len)
{
#if 0
gint outsize;
guint in_len G_GNUC_UNUSED, in_processed;
guint out_len, out_processed;
guint num, den;
gpointer buf;
g_assert (resample->converter != 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);
gsize out_len, outsize;
gpointer out[1];
out_len =
gst_audio_converter_get_out_frames (resample->converter, history_len);
if (out_len == 0)
return;
buf = g_malloc (outsize);
resample->funcs->process (resample->state, NULL, &in_processed, buf,
&out_processed);
g_free (buf);
outsize = out_len * resample->out.bpf;
g_assert (in_len == in_processed);
#endif
out[0] = g_malloc (outsize);
gst_audio_converter_samples (resample->converter, 0, NULL, history_len,
out, out_len);
g_free (out[0]);
}
static void
@ -616,10 +606,6 @@ gst_audio_resample_sink_event (GstBaseTransform * base, GstEvent * event)
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_audio_resample_reset_state (resample);
#if 0
if (resample->converter)
resample->funcs->skip_zeros (resample->converter);
#endif
resample->num_gap_samples = 0;
resample->num_nongap_samples = 0;
resample->t0 = GST_CLOCK_TIME_NONE;
@ -630,18 +616,12 @@ gst_audio_resample_sink_event (GstBaseTransform * base, GstEvent * event)
resample->need_discont = TRUE;
break;
case GST_EVENT_SEGMENT:
#if 0
if (resample->converter) {
guint latency =
resample->funcs->get_input_latency (resample->converter);
gsize latency =
gst_audio_converter_get_max_latency (resample->converter);
gst_audio_resample_push_drain (resample, latency);
}
#endif
gst_audio_resample_reset_state (resample);
#if 0
if (resample->converter)
resample->funcs->skip_zeros (resample->converter);
#endif
resample->num_gap_samples = 0;
resample->num_nongap_samples = 0;
resample->t0 = GST_CLOCK_TIME_NONE;
@ -652,13 +632,11 @@ gst_audio_resample_sink_event (GstBaseTransform * base, GstEvent * event)
resample->need_discont = TRUE;
break;
case GST_EVENT_EOS:
#if 0
if (resample->converter) {
guint latency =
resample->funcs->get_input_latency (resample->converter);
gsize latency =
gst_audio_converter_get_max_latency (resample->converter);
gst_audio_resample_push_drain (resample, latency);
}
#endif
gst_audio_resample_reset_state (resample);
break;
default:
@ -859,9 +837,6 @@ gst_audio_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
/* handle discontinuity */
if (G_UNLIKELY (resample->need_discont)) {
#if 0
resample->funcs->skip_zeros (resample->state);
#endif
resample->num_gap_samples = 0;
resample->num_nongap_samples = 0;
/* reset */
@ -962,12 +937,10 @@ gst_audio_resample_query (GstPad * pad, GstObject * parent, GstQuery * query)
gint rate = resample->in.rate;
gint resampler_latency;
#if 0
if (resample->state)
if (resample->converter)
resampler_latency =
resample->funcs->get_input_latency (resample->state);
gst_audio_converter_get_max_latency (resample->converter);
else
#endif
resampler_latency = 0;
if (gst_base_transform_is_passthrough (trans))

1
win32/common/libgstaudio.def
View file

@ -164,6 +164,7 @@ EXPORTS
gst_audio_resampler_new
gst_audio_resampler_options_set_quality
gst_audio_resampler_resample
gst_audio_resampler_reset
gst_audio_resampler_update
gst_audio_ring_buffer_acquire
gst_audio_ring_buffer_activate