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audio-converter: improve processing loop

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Process as many samples as we can from the input and return the number
of processed samples from the chain. This simplifies some code.
Fix the IN_WRITABLE handling, don't overwrite the flags.
This commit is contained in:
Wim Taymans 2016-01-12 11:37:17 +01:00
parent 20f6af651b
commit 8a8b12189e
1 changed files with 104 additions and 65 deletions
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169
gst-libs/gst/audio/audio-converter.c
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@ -103,8 +103,11 @@ struct _GstAudioConverter
GstAudioLayout current_layout; GstAudioLayout current_layout;
gint current_channels; gint current_channels;
gboolean in_writable;
gpointer *in_data; gpointer *in_data;
gsize in_samples;
gpointer *out_data; gpointer *out_data;
gsize out_samples;
/* unpack */ /* unpack */
gboolean in_default; gboolean in_default;
@ -135,14 +138,10 @@ struct _GstAudioConverter
gboolean passthrough; gboolean passthrough;
}; };
typedef gboolean (*AudioChainFunc) (AudioChain * chain, gsize num_samples, typedef gboolean (*AudioChainFunc) (AudioChain * chain, gpointer user_data);
gpointer user_data);
typedef gpointer *(*AudioChainAllocFunc) (AudioChain * chain, gsize num_samples, typedef gpointer *(*AudioChainAllocFunc) (AudioChain * chain, gsize num_samples,
gpointer user_data); gpointer user_data);
static gpointer *get_output_samples (AudioChain * chain, gsize num_samples,
gpointer user_data);
struct _AudioChain struct _AudioChain
{ {
AudioChain *prev; AudioChain *prev;
@ -165,6 +164,7 @@ struct _AudioChain
gsize tmpsize; gsize tmpsize;
gpointer *samples; gpointer *samples;
gsize num_samples;
}; };
static AudioChain * static AudioChain *
@ -214,15 +214,26 @@ audio_chain_alloc_samples (AudioChain * chain, gsize num_samples)
return chain->alloc_func (chain, num_samples, chain->alloc_data); return chain->alloc_func (chain, num_samples, chain->alloc_data);
} }
static void
audio_chain_set_samples (AudioChain * chain, gpointer * samples,
gsize num_samples)
{
GST_LOG ("set samples %p %" G_GSIZE_FORMAT, samples, num_samples);
chain->samples = samples;
chain->num_samples = num_samples;
}
static gpointer * static gpointer *
audio_chain_get_samples (AudioChain * chain, gsize num_samples) audio_chain_get_samples (AudioChain * chain, gsize * avail)
{ {
gpointer *res; gpointer *res;
while (!chain->samples) while (!chain->samples)
chain->make_func (chain, num_samples, chain->make_func_data); chain->make_func (chain, chain->make_func_data);
res = chain->samples; res = chain->samples;
*avail = chain->num_samples;
chain->samples = NULL; chain->samples = NULL;
return res; return res;
@ -299,6 +310,18 @@ gst_audio_converter_update_config (GstAudioConverter * convert,
{ {
g_return_val_if_fail (convert != NULL, FALSE); g_return_val_if_fail (convert != NULL, FALSE);
g_return_val_if_fail (config != NULL, FALSE); g_return_val_if_fail (config != NULL, FALSE);
g_return_val_if_fail ((in_rate == 0 && out_rate == 0) ||
convert->flags & GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE, FALSE);
GST_LOG ("new rate %d -> %d", in_rate, out_rate);
if (in_rate <= 0)
in_rate = convert->in.rate;
if (out_rate <= 0)
out_rate = convert->out.rate;
convert->in.rate = in_rate;
convert->out.rate = out_rate;
gst_structure_foreach (config, copy_config, convert); gst_structure_foreach (config, copy_config, convert);
gst_structure_free (config); gst_structure_free (config);
@ -331,22 +354,64 @@ gst_audio_converter_get_config (GstAudioConverter * convert,
return convert->config; return convert->config;
} }
static gboolean static gpointer *
do_unpack (AudioChain * chain, gsize num_samples, gpointer user_data) get_output_samples (AudioChain * chain, gsize num_samples, gpointer user_data)
{ {
GstAudioConverter *convert = user_data; GstAudioConverter *convert = user_data;
GST_LOG ("output samples %p %" G_GSIZE_FORMAT, convert->out_data,
num_samples);
return convert->out_data;
}
static gpointer *
get_temp_samples (AudioChain * chain, gsize num_samples, gpointer user_data)
{
gsize needed;
/* first part contains the pointers, second part the data */
needed = (num_samples * chain->stride + sizeof (gpointer)) * chain->blocks;
if (needed > chain->tmpsize) {
gint i;
guint8 *s;
GST_DEBUG ("alloc samples %" G_GSIZE_FORMAT, needed);
chain->tmp = g_realloc (chain->tmp, needed);
chain->tmpsize = needed;
/* jump to the data */
s = (guint8 *) & chain->tmp[chain->blocks];
/* set up the pointers */
for (i = 0; i < chain->blocks; i++)
chain->tmp[i] = s + (i * num_samples * chain->stride);
}
GST_LOG ("temp samples %p %" G_GSIZE_FORMAT, chain->tmp, num_samples);
return chain->tmp;
}
static gboolean
do_unpack (AudioChain * chain, gpointer user_data)
{
GstAudioConverter *convert = user_data;
gsize num_samples;
gpointer *tmp; gpointer *tmp;
gboolean src_writable; gboolean in_writable;
src_writable = (convert->flags & GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE); in_writable = convert->in_writable;
num_samples = convert->in_samples;
if (!chain->allow_ip || !src_writable || !convert->in_default) { if (!chain->allow_ip || !in_writable || !convert->in_default) {
gint i; gint i;
if (src_writable && chain->allow_ip) if (in_writable && chain->allow_ip)
tmp = convert->in_data; tmp = convert->in_data;
else else
tmp = audio_chain_alloc_samples (chain, num_samples); tmp = audio_chain_alloc_samples (chain, num_samples);
GST_LOG ("unpack %p, %p, %" G_GSIZE_FORMAT, tmp, convert->in_data, GST_LOG ("unpack %p, %p, %" G_GSIZE_FORMAT, tmp, convert->in_data,
num_samples); num_samples);
@ -359,79 +424,83 @@ do_unpack (AudioChain * chain, gsize num_samples, gpointer user_data)
tmp = convert->in_data; tmp = convert->in_data;
GST_LOG ("get in samples %p", tmp); GST_LOG ("get in samples %p", tmp);
} }
chain->samples = tmp; audio_chain_set_samples (chain, tmp, num_samples);
return TRUE; return TRUE;
} }
static gboolean static gboolean
do_convert_in (AudioChain * chain, gsize num_samples, gpointer user_data) do_convert_in (AudioChain * chain, gpointer user_data)
{ {
GstAudioConverter *convert = user_data; GstAudioConverter *convert = user_data;
gsize num_samples;
gpointer *in, *out; gpointer *in, *out;
gint i; gint i;
in = audio_chain_get_samples (chain->prev, num_samples); in = audio_chain_get_samples (chain->prev, &num_samples);
out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples)); out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples));
GST_LOG ("convert in %p, %p, %" G_GSIZE_FORMAT, in, out, num_samples); GST_LOG ("convert in %p, %p, %" G_GSIZE_FORMAT, in, out, num_samples);
for (i = 0; i < chain->blocks; i++) for (i = 0; i < chain->blocks; i++)
convert->convert_in (out[i], in[i], num_samples * chain->inc); convert->convert_in (out[i], in[i], num_samples * chain->inc);
chain->samples = out; audio_chain_set_samples (chain, out, num_samples);
return TRUE; return TRUE;
} }
static gboolean static gboolean
do_mix (AudioChain * chain, gsize num_samples, gpointer user_data) do_mix (AudioChain * chain, gpointer user_data)
{ {
GstAudioConverter *convert = user_data; GstAudioConverter *convert = user_data;
gsize num_samples;
gpointer *in, *out; gpointer *in, *out;
in = audio_chain_get_samples (chain->prev, num_samples); in = audio_chain_get_samples (chain->prev, &num_samples);
out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples)); out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples));
GST_LOG ("mix %p, %p, %" G_GSIZE_FORMAT, in, out, num_samples); GST_LOG ("mix %p, %p, %" G_GSIZE_FORMAT, in, out, num_samples);
gst_audio_channel_mixer_samples (convert->mix, in, out, num_samples); gst_audio_channel_mixer_samples (convert->mix, in, out, num_samples);
chain->samples = out; audio_chain_set_samples (chain, out, num_samples);
return TRUE; return TRUE;
} }
static gboolean static gboolean
do_convert_out (AudioChain * chain, gsize num_samples, gpointer user_data) do_convert_out (AudioChain * chain, gpointer user_data)
{ {
GstAudioConverter *convert = user_data; GstAudioConverter *convert = user_data;
gsize num_samples;
gpointer *in, *out; gpointer *in, *out;
gint i; gint i;
in = audio_chain_get_samples (chain->prev, num_samples); in = audio_chain_get_samples (chain->prev, &num_samples);
out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples)); out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples));
GST_LOG ("convert out %p, %p, %" G_GSIZE_FORMAT, in, out, num_samples); GST_LOG ("convert out %p, %p, %" G_GSIZE_FORMAT, in, out, num_samples);
for (i = 0; i < chain->blocks; i++) for (i = 0; i < chain->blocks; i++)
convert->convert_out (out[i], in[i], num_samples * chain->inc); convert->convert_out (out[i], in[i], num_samples * chain->inc);
chain->samples = out; audio_chain_set_samples (chain, out, num_samples);
return TRUE; return TRUE;
} }
static gboolean static gboolean
do_quantize (AudioChain * chain, gsize num_samples, gpointer user_data) do_quantize (AudioChain * chain, gpointer user_data)
{ {
GstAudioConverter *convert = user_data; GstAudioConverter *convert = user_data;
gsize num_samples;
gpointer *in, *out; gpointer *in, *out;
in = audio_chain_get_samples (chain->prev, num_samples); in = audio_chain_get_samples (chain->prev, &num_samples);
out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples)); out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples));
GST_LOG ("quantize %p, %p, %" G_GSIZE_FORMAT, in, out, num_samples); GST_LOG ("quantize %p, %p, %" G_GSIZE_FORMAT, in, out, num_samples);
gst_audio_quantize_samples (convert->quant, in, out, num_samples); gst_audio_quantize_samples (convert->quant, in, out, num_samples);
chain->samples = out; audio_chain_set_samples (chain, out, num_samples);
return TRUE; return TRUE;
} }
@ -612,41 +681,6 @@ chain_pack (GstAudioConverter * convert, AudioChain * prev)
return prev; return prev;
} }
static gpointer *
get_output_samples (AudioChain * chain, gsize samples, gpointer user_data)
{
GstAudioConverter *convert = user_data;
GST_LOG ("output samples %" G_GSIZE_FORMAT, samples);
return convert->out_data;
}
static gpointer *
get_temp_samples (AudioChain * chain, gsize num_samples, gpointer user_data)
{
gsize needed;
/* first part contains the pointers, second part the data */
needed = (num_samples * chain->stride + sizeof (gpointer)) * chain->blocks;
if (needed > chain->tmpsize) {
gint i;
guint8 *s;
GST_DEBUG ("alloc samples %" G_GSIZE_FORMAT, needed);
chain->tmp = g_realloc (chain->tmp, needed);
chain->tmpsize = needed;
/* jump to the data */
s = (guint8 *) & chain->tmp[chain->blocks];
/* set up the pointers */
for (i = 0; i < chain->blocks; i++)
chain->tmp[i] = s + (i * num_samples * chain->stride);
}
return chain->tmp;
}
static void static void
setup_allocators (GstAudioConverter * convert) setup_allocators (GstAudioConverter * convert)
{ {
@ -668,6 +702,7 @@ setup_allocators (GstAudioConverter * convert)
chain->alloc_func = alloc_func; chain->alloc_func = alloc_func;
chain->alloc_data = convert; chain->alloc_data = convert;
chain->allow_ip = allow_ip && chain->allow_ip; chain->allow_ip = allow_ip && chain->allow_ip;
GST_LOG ("chain %p: %d %d", chain, allow_ip, chain->allow_ip);
if (!chain->pass_alloc) { if (!chain->pass_alloc) {
/* can't pass allocator, make new temp line allocator */ /* can't pass allocator, make new temp line allocator */
@ -712,6 +747,7 @@ gst_audio_converter_new (GstAudioConverterFlags flags, GstAudioInfo * in_info,
convert = g_slice_new0 (GstAudioConverter); convert = g_slice_new0 (GstAudioConverter);
convert->flags = flags;
convert->in = *in_info; convert->in = *in_info;
convert->out = *out_info; convert->out = *out_info;
@ -871,6 +907,7 @@ gst_audio_converter_samples (GstAudioConverter * convert,
AudioChain *chain; AudioChain *chain;
gpointer *tmp; gpointer *tmp;
gint i; gint i;
gsize produced;
g_return_val_if_fail (convert != NULL, FALSE); g_return_val_if_fail (convert != NULL, FALSE);
g_return_val_if_fail (in != NULL, FALSE); g_return_val_if_fail (in != NULL, FALSE);
@ -898,19 +935,21 @@ gst_audio_converter_samples (GstAudioConverter * convert,
GST_LOG ("converting %" G_GSIZE_FORMAT, in_samples); GST_LOG ("converting %" G_GSIZE_FORMAT, in_samples);
convert->flags = flags; convert->in_writable = flags & GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE;
convert->in_data = in; convert->in_data = in;
convert->in_samples = in_samples;
convert->out_data = out; convert->out_data = out;
convert->out_samples = out_samples;
/* get samples to pack */ /* get samples to pack */
tmp = audio_chain_get_samples (chain, in_samples); tmp = audio_chain_get_samples (chain, &produced);
if (!convert->out_default) { if (!convert->out_default) {
GST_LOG ("pack %p, %p %" G_GSIZE_FORMAT, tmp, out, in_samples); GST_LOG ("pack %p, %p %" G_GSIZE_FORMAT, tmp, out, produced);
/* and pack if needed */ /* and pack if needed */
for (i = 0; i < chain->blocks; i++) for (i = 0; i < chain->blocks; i++)
convert->out.finfo->pack_func (convert->out.finfo, 0, tmp[i], out[i], convert->out.finfo->pack_func (convert->out.finfo, 0, tmp[i], out[i],
in_samples * chain->inc); produced * chain->inc);
} }
return TRUE; return TRUE;
} }