(chain->allow_ip ? in : audio_chain_alloc_samples (chain, out_frames,
&out_frames));
- GST_LOG ("resample %p %p,%" G_GSIZE_FORMAT " %" G_GSIZE_FORMAT, in, out,
- in_frames, out_frames);
+ GST_LOG ("resample %p %p,%" G_GSIZE_FORMAT " %" G_GSIZE_FORMAT " %"
+ G_GSIZE_FORMAT, in, out, in_frames, out_frames, num_samples);
gst_audio_resampler_resample (convert->resampler, in, in_frames, out,
- out_frames, &produced, &consumed);
+ out_frames, &consumed, &produced);
audio_chain_set_samples (chain, out, produced);
/* optimize */
if (out_info->finfo->format == in_info->finfo->format
- && convert->mix_passthrough) {
- GST_INFO ("same formats and passthrough mixing -> passthrough");
+ && convert->mix_passthrough && convert->resampler == NULL) {
+ GST_INFO
+ ("same formats, no resampler and passthrough mixing -> passthrough");
convert->convert = converter_passthrough;
} else {
GST_INFO ("do full conversion");
audio_chain_free (convert->convert_in_chain);
if (convert->mix_chain)
audio_chain_free (convert->mix_chain);
+ if (convert->resample_chain)
+ audio_chain_free (convert->resample_chain);
if (convert->convert_out_chain)
audio_chain_free (convert->convert_out_chain);
if (convert->quant_chain)
audio_chain_free (convert->quant_chain);
-
if (convert->quant)
gst_audio_quantize_free (convert->quant);
if (convert->mix)
gst_audio_channel_mixer_free (convert->mix);
+ if (convert->resampler)
+ gst_audio_resampler_free (convert->resampler);
gst_audio_info_init (&convert->in);
gst_audio_info_init (&convert->out);
gst_audio_converter_get_out_frames (GstAudioConverter * convert,
gsize in_frames)
{
- return in_frames;
+ if (convert->resampler)
+ return gst_audio_resampler_get_out_frames (convert->resampler, in_frames);
+ else
+ return in_frames;
}
/**
gst_audio_converter_get_in_frames (GstAudioConverter * convert,
gsize out_frames)
{
- return out_frames;
+ if (convert->resampler)
+ return gst_audio_resampler_get_in_frames (convert->resampler, out_frames);
+ else
+ return out_frames;
}
/**
gsize
gst_audio_converter_get_max_latency (GstAudioConverter * convert)
{
- return 0;
+ if (convert->resampler)
+ return gst_audio_resampler_get_max_latency (convert->resampler);
+ else
+ return 0;
+}
+
+/**
+ * gst_audio_converter_update_rates:
+ * @convert: a #GstAudioConverter
+ * @in_rate: input rate
+ * @out_rate: output rate
+ * @options: resampler options
+ *
+ * Update the input and output rates, passing @options to the resampler.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_audio_converter_update_rates (GstAudioConverter * convert,
+ gint in_rate, gint out_rate, GstStructure * options)
+{
+ g_return_val_if_fail (convert != NULL, FALSE);
+ g_return_val_if_fail (in_rate > 0, FALSE);
+ g_return_val_if_fail (out_rate > 0, FALSE);
+
+ convert->in.rate = in_rate;
+ convert->out.rate = out_rate;
+
+ if (options)
+ gst_structure_free (options);
+
+ return TRUE;
+}
+
+/**
+ * gst_audio_converter_get_rates:
+ * @convert: a #GstAudioConverter
+ * @in_rate: input rate
+ * @out_rate: output rate
+ *
+ * Get the current input and output rates.
+ */
+void
+gst_audio_converter_get_rates (GstAudioConverter * convert,
+ gint * in_rate, gint * out_rate)
+{
+ if (in_rate)
+ *in_rate = convert->in.rate;
+ if (out_rate)
+ *out_rate = convert->out.rate;
}
/**
static inline gsize
calc_out (GstAudioResampler * resampler, gsize in)
{
- return ((in * resampler->out_rate -
+ gsize out;
+
+ out = ((in * resampler->out_rate -
resampler->samp_phase) / resampler->in_rate) + 1;
+ GST_LOG ("out %d = ((%d * %d - %d) / %d) + 1", (gint) out,
+ (gint) in, resampler->out_rate, resampler->samp_phase,
+ resampler->in_rate);
+ return out;
}
/**
need = resampler->n_taps + resampler->samp_index + resampler->skip;
avail = resampler->samples_avail + in_frames;
+ GST_LOG ("need %d = %d + %d + %d, avail %d = %d + %d", (gint) need,
+ resampler->n_taps, resampler->samp_index, resampler->skip,
+ (gint) avail, (gint) resampler->samples_avail, (gint) in_frames);
if (avail < need)
return 0;
* @in_frames: number of input frames
* @out: output samples
* @out_frames: maximum output frames
- * @consumed: number of frames consumed
- * @produced: number of frames produced
+ * @in_consumed: number of frames consumed
+ * @out_produced: number of frames produced
*
* Perform resampling on @in_frames frames in @in and write at most
* @out_frames of frames to @out.
void
gst_audio_resampler_resample (GstAudioResampler * resampler,
gpointer in[], gsize in_frames, gpointer out[], gsize out_frames,
- gsize * consumed, gsize * produced)
+ gsize * in_consumed, gsize * out_produced)
{
gsize samples_avail;
gsize out2, need;
if (resampler->skip >= in_frames) {
/* we need tp skip all input */
resampler->skip -= in_frames;
- *consumed = in_frames;
- *produced = 0;
+ *in_consumed = in_frames;
+ *out_produced = 0;
return;
}
/* skip the last samples by advancing the sample index */
need = resampler->n_taps + resampler->samp_index;
if (samples_avail < need) {
/* not enough samples to start */
- *consumed = in_frames;
- *produced = 0;
+ *in_consumed = in_frames;
+ *out_produced = 0;
return;
}
/* resample all channels */
resampler->resample (resampler, sbuf, samples_avail, out, out_frames,
- consumed, produced, TRUE);
+ in_consumed, out_produced, TRUE);
GST_LOG ("in %" G_GSIZE_FORMAT ", used %" G_GSIZE_FORMAT ", consumed %"
G_GSIZE_FORMAT ", produced %" G_GSIZE_FORMAT, in_frames, samples_avail,
- *consumed, *produced);
+ *in_consumed, *out_produced);
/* update pointers */
- if (*consumed > 0) {
- gssize left = samples_avail - *consumed;
+ if (*in_consumed > 0) {
+ gssize left = samples_avail - *in_consumed;
if (left > 0) {
/* we consumed part of our samples */
resampler->samples_avail = left;
resampler->skip = -left;
}
/* we always consume everything */
- *consumed = in_frames;
+ *in_consumed = in_frames;
}
}
PROP_SINC_FILTER_AUTO_THRESHOLD
};
-#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define SUPPORTED_CAPS \
- GST_AUDIO_CAPS_MAKE ("{ F32LE, F64LE, S32LE, S16LE }") \
+ GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) \
", layout = (string) { interleaved, non-interleaved }"
-#else
-#define SUPPORTED_CAPS \
- GST_AUDIO_CAPS_MAKE ("{ F32BE, F64BE, S32BE, S16BE }") \
- ", layout = (string) { interleaved, non-interleaved }"
-#endif
-
-/* If TRUE integer arithmetic resampling is faster and will be used if appropriate */
-#if defined AUDIORESAMPLE_FORMAT_INT
-static gboolean gst_audio_resample_use_int = TRUE;
-#elif defined AUDIORESAMPLE_FORMAT_FLOAT
-static gboolean gst_audio_resample_use_int = FALSE;
-#else
-static gboolean gst_audio_resample_use_int = FALSE;
-#endif
static GstStaticPadTemplate gst_audio_resample_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
{
GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
- if (resample->resamp) {
- gst_audio_resampler_free (resample->resamp);
- resample->resamp = NULL;
+ if (resample->converter) {
+ gst_audio_converter_free (resample->converter);
+ resample->converter = NULL;
}
return TRUE;
}
GstStructure *options;
options = gst_structure_new_empty ("resampler-options");
- gst_audio_resampler_options_set_quality (resample->method,
- resample->quality, in->rate, out->rate, options);
+ if (in != NULL && out != NULL)
+ gst_audio_resampler_options_set_quality (resample->method,
+ resample->quality, in->rate, out->rate, options);
gst_structure_set (options,
+ GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD, GST_TYPE_AUDIO_RESAMPLER_METHOD,
+ resample->method,
GST_AUDIO_RESAMPLER_OPT_FILTER_MODE, GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE,
resample->sinc_filter_mode, GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD,
G_TYPE_UINT, resample->sinc_filter_auto_threshold, NULL);
gsize old_latency = -1;
GstStructure *options;
- if (resample->resamp == NULL && in == NULL && out == NULL)
+ if (resample->converter == NULL && in == NULL && out == NULL)
return TRUE;
options = make_options (resample, in, out);
- if (resample->resamp)
- old_latency = gst_audio_resampler_get_max_latency (resample->resamp);
+ if (resample->converter)
+ old_latency = gst_audio_converter_get_max_latency (resample->converter);
/* if channels and layout changed, destroy existing resampler */
- if ((in->finfo != resample->in.finfo ||
+ if (in != NULL && (in->finfo != resample->in.finfo ||
in->channels != resample->in.channels ||
- in->layout != resample->in.layout) && resample->resamp) {
- gst_audio_resampler_free (resample->resamp);
- resample->resamp = NULL;
+ in->layout != resample->in.layout) && resample->converter) {
+ gst_audio_converter_free (resample->converter);
+ resample->converter = NULL;
}
- if (resample->resamp == NULL) {
- GstAudioResamplerFlags flags = 0;
-
- if (in->layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED)
- flags |= GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED;
-
- resample->resamp = gst_audio_resampler_new (resample->method,
- flags, in->finfo->format, in->channels, in->rate, out->rate, options);
- if (resample->resamp == NULL)
+ if (resample->converter == NULL) {
+ resample->converter = gst_audio_converter_new (0, in, out, options);
+ if (resample->converter == NULL)
goto resampler_failed;
- } else {
+ } else if (in && out) {
gboolean ret;
ret =
- gst_audio_resampler_update (resample->resamp, in->rate, out->rate,
- options);
+ gst_audio_converter_update_rates (resample->converter, in->rate,
+ out->rate, options);
if (!ret)
goto update_failed;
+ } else {
+ gst_structure_free (options);
}
if (old_latency != -1)
updated_latency =
- old_latency != gst_audio_resampler_get_max_latency (resample->resamp);
+ old_latency !=
+ gst_audio_converter_get_max_latency (resample->converter);
if (updated_latency)
gst_element_post_message (GST_ELEMENT (resample),
guint num, den;
gpointer buf;
- g_assert (resample->resamp != NULL);
+ g_assert (resample->converter != NULL);
resample->funcs->get_ratio (resample->state, &num, &den);
GstMapInfo map;
gpointer out[1];
- g_assert (resample->resamp != NULL);
+ g_assert (resample->converter != NULL);
/* Don't drain samples if we were reset. */
if (!GST_CLOCK_TIME_IS_VALID (resample->t0))
return;
- out_len = gst_audio_resampler_get_out_frames (resample->resamp, history_len);
+ out_len =
+ gst_audio_converter_get_out_frames (resample->converter, history_len);
if (out_len == 0)
return;
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
out[0] = map.data;
- gst_audio_resampler_resample (resample->resamp, NULL, history_len,
+ gst_audio_converter_samples (resample->converter, 0, NULL, history_len,
out, out_len, &in_processed, &out_processed);
/* If we wrote more than allocated something is really wrong now
case GST_EVENT_FLUSH_STOP:
gst_audio_resample_reset_state (resample);
#if 0
- if (resample->resamp)
- resample->funcs->skip_zeros (resample->resamp);
+ if (resample->converter)
+ resample->funcs->skip_zeros (resample->converter);
#endif
resample->num_gap_samples = 0;
resample->num_nongap_samples = 0;
break;
case GST_EVENT_SEGMENT:
#if 0
- if (resample->resamp) {
- guint latency = resample->funcs->get_input_latency (resample->resamp);
+ if (resample->converter) {
+ guint latency =
+ resample->funcs->get_input_latency (resample->converter);
gst_audio_resample_push_drain (resample, latency);
}
#endif
gst_audio_resample_reset_state (resample);
#if 0
- if (resample->resamp)
- resample->funcs->skip_zeros (resample->resamp);
+ if (resample->converter)
+ resample->funcs->skip_zeros (resample->converter);
#endif
resample->num_gap_samples = 0;
resample->num_nongap_samples = 0;
break;
case GST_EVENT_EOS:
#if 0
- if (resample->resamp) {
- guint latency = resample->funcs->get_input_latency (resample->resamp);
+ if (resample->converter) {
+ guint latency =
+ resample->funcs->get_input_latency (resample->converter);
gst_audio_resample_push_drain (resample, latency);
}
#endif
gsize outsize;
guint32 in_len, in_processed;
guint32 out_len, out_processed;
- guint filt_len = gst_audio_resampler_get_max_latency (resample->resamp) * 2;
+ guint filt_len =
+ gst_audio_converter_get_max_latency (resample->converter) * 2;
gst_buffer_map (inbuf, &in_map, GST_MAP_READ);
gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE);
gpointer in[1], out[1];
out_test =
- gst_audio_resampler_get_out_frames (resample->resamp, in_len);
+ gst_audio_converter_get_out_frames (resample->converter, in_len);
out_test = MIN (out_test, out_len);
in[0] = in_map.data;
out[0] = out_map.data;
- gst_audio_resampler_resample (resample->resamp, in, in_len,
+ gst_audio_converter_samples (resample->converter, 0, in, in_len,
out, out_len, &in_proc, &out_proc);
in_processed = in_proc;
}
}
-/* FIXME: should have a benchmark fallback for the case where orc is disabled */
-#if defined(AUDIORESAMPLE_FORMAT_AUTO) && !defined(DISABLE_ORC)
-
-#define BENCHMARK_SIZE 512
-
-static gboolean
-_benchmark_int_float (GstAudioResampler * st)
-{
- gint16 in[BENCHMARK_SIZE] = { 0, }, G_GNUC_UNUSED out[BENCHMARK_SIZE / 2];
- gfloat in_tmp[BENCHMARK_SIZE], out_tmp[BENCHMARK_SIZE / 2];
- gint i;
- guint32 inlen = BENCHMARK_SIZE, outlen = BENCHMARK_SIZE / 2;
- gpointer inp[1], outp[1];
- gsize produced, consumed;
-
- for (i = 0; i < BENCHMARK_SIZE; i++) {
- gfloat tmp = in[i];
- in_tmp[i] = tmp / G_MAXINT16;
- }
-
- inp[0] = in_tmp;
- outp[0] = out_tmp;
-
- gst_audio_resampler_resample (st,
- inp, inlen, outp, outlen, &produced, &consumed);
-
- if (outlen == 0) {
- GST_ERROR ("Failed to use float resampler");
- return FALSE;
- }
-
- for (i = 0; i < outlen; i++) {
- gfloat tmp = out_tmp[i];
- out[i] = CLAMP (tmp * G_MAXINT16 + 0.5, G_MININT16, G_MAXINT16);
- }
-
- return TRUE;
-}
-
-static gboolean
-_benchmark_int_int (GstAudioResampler * st)
-{
- gint16 in[BENCHMARK_SIZE] = { 0, }, out[BENCHMARK_SIZE / 2];
- guint32 inlen = BENCHMARK_SIZE, outlen = BENCHMARK_SIZE / 2;
- gpointer inp[1], outp[1];
- gsize produced, consumed;
-
- inp[0] = in;
- outp[0] = out;
-
- gst_audio_resampler_resample (st, inp, inlen, outp, outlen, &produced,
- &consumed);
-
- if (outlen == 0) {
- GST_ERROR ("Failed to use int resampler");
- return FALSE;
- }
-
- return TRUE;
-}
-
-static gboolean
-_benchmark_integer_resampling (void)
-{
- OrcProfile a, b;
- gdouble av, bv;
- GstAudioResampler *sta, *stb;
- int i;
-
- orc_profile_init (&a);
- orc_profile_init (&b);
-
- sta = gst_audio_resampler_new (GST_AUDIO_RESAMPLER_METHOD_KAISER,
- 0, GST_AUDIO_FORMAT_F32LE, 1, 48000, 24000, NULL);
- if (sta == NULL) {
- GST_ERROR ("Failed to create float resampler state");
- return FALSE;
- }
-
- stb = gst_audio_resampler_new (GST_AUDIO_RESAMPLER_METHOD_KAISER,
- 0, GST_AUDIO_FORMAT_S32LE, 1, 48000, 24000, NULL);
- if (stb == NULL) {
- gst_audio_resampler_free (sta);
- GST_ERROR ("Failed to create int resampler state");
- return FALSE;
- }
-
- /* Benchmark */
- for (i = 0; i < 10; i++) {
- orc_profile_start (&a);
- if (!_benchmark_int_float (sta))
- goto error;
- orc_profile_stop (&a);
- }
-
- /* Benchmark */
- for (i = 0; i < 10; i++) {
- orc_profile_start (&b);
- if (!_benchmark_int_int (stb))
- goto error;
- orc_profile_stop (&b);
- }
-
- /* Handle results */
- orc_profile_get_ave_std (&a, &av, NULL);
- orc_profile_get_ave_std (&b, &bv, NULL);
-
- /* Remember benchmark result in global variable */
- gst_audio_resample_use_int = (av > bv);
- gst_audio_resampler_free (sta);
- gst_audio_resampler_free (stb);
-
- if (av > bv)
- GST_INFO ("Using integer resampler if appropriate: %lf < %lf", bv, av);
- else
- GST_INFO ("Using float resampler for everything: %lf <= %lf", av, bv);
-
- return TRUE;
-
-error:
- gst_audio_resampler_free (sta);
- gst_audio_resampler_free (stb);
-
- return FALSE;
-}
-#endif /* defined(AUDIORESAMPLE_FORMAT_AUTO) && !defined(DISABLE_ORC) */
-
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (audio_resample_debug, "audioresample", 0,
"audio resampling element");
-#if defined(AUDIORESAMPLE_FORMAT_AUTO) && !defined(DISABLE_ORC)
- if (!_benchmark_integer_resampling ())
- return FALSE;
-#else
- GST_WARNING ("Orc disabled, can't benchmark int vs. float resampler");
- {
- GST_DEBUG_CATEGORY_GET (GST_CAT_PERFORMANCE, "GST_PERFORMANCE");
- GST_CAT_WARNING (GST_CAT_PERFORMANCE, "orc disabled, no benchmarking done");
- }
-#endif
-
if (!gst_element_register (plugin, "audioresample", GST_RANK_PRIMARY,
GST_TYPE_AUDIO_RESAMPLE)) {
return FALSE;
projects / platform / upstream / gstreamer.git / commitdiff