mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-06-07 07:58:51 +00:00
Code Issues 1 Projects Releases Wiki Activity

audio-convert: simplify the chain free process

Browse source 
It is not needed to store a pointer to every single chain element to free it.
Instead walk the channel list backwards and free the chain elements one by one.

Rename GstAudioConverter->chain_pack to chain_end.

https://bugzilla.gnome.org/show_bug.cgi?id=773073
This commit is contained in:
Petr Kulhavy 2016-10-21 14:30:31 +02:00 committed by Sebastian Dröge
parent b2b8e77566
commit 640c54d8f8
1 changed files with 20 additions and 29 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

49
gst-libs/gst/audio/audio-converter.c
View file

@ -116,32 +116,26 @@ struct _GstAudioConverter
/* unpack */ /* unpack */
gboolean in_default; gboolean in_default;
gboolean unpack_ip; gboolean unpack_ip;
AudioChain *unpack_chain;
/* convert in */ /* convert in */
AudioConvertFunc convert_in; AudioConvertFunc convert_in;
AudioChain *convert_in_chain;
/* channel mix */ /* channel mix */
gboolean mix_passthrough; gboolean mix_passthrough;
GstAudioChannelMixer *mix; GstAudioChannelMixer *mix;
AudioChain *mix_chain;
/* resample */ /* resample */
GstAudioResampler *resampler; GstAudioResampler *resampler;
AudioChain *resample_chain;
/* convert out */ /* convert out */
AudioConvertFunc convert_out; AudioConvertFunc convert_out;
AudioChain *convert_out_chain;
/* quant */ /* quant */
GstAudioQuantize *quant; GstAudioQuantize *quant;
AudioChain *quant_chain;
/* pack */ /* pack */
gboolean out_default; gboolean out_default;
AudioChain *pack_chain; AudioChain *chain_end; /* NULL for empty chain or points to the last element in the chain */
AudioConvertSamplesFunc convert; AudioConvertSamplesFunc convert;
}; };
@ -598,7 +592,7 @@ chain_unpack (GstAudioConverter * convert)
gst_audio_format_to_string (in->finfo->format), gst_audio_format_to_string (in->finfo->format),
gst_audio_format_to_string (convert->current_format)); gst_audio_format_to_string (convert->current_format));
prev = convert->unpack_chain = audio_chain_new (NULL, convert); prev = audio_chain_new (NULL, convert);
prev->allow_ip = prev->finfo->width <= in->finfo->width; prev->allow_ip = prev->finfo->width <= in->finfo->width;
prev->pass_alloc = FALSE; prev->pass_alloc = FALSE;
audio_chain_set_make_func (prev, do_unpack, convert, NULL); audio_chain_set_make_func (prev, do_unpack, convert, NULL);
@ -621,7 +615,7 @@ chain_convert_in (GstAudioConverter * convert, AudioChain * prev)
convert->convert_in = (AudioConvertFunc) audio_orc_s32_to_double; convert->convert_in = (AudioConvertFunc) audio_orc_s32_to_double;
convert->current_format = GST_AUDIO_FORMAT_F64; convert->current_format = GST_AUDIO_FORMAT_F64;
prev = convert->convert_in_chain = audio_chain_new (prev, convert); prev = audio_chain_new (prev, convert);
prev->allow_ip = FALSE; prev->allow_ip = FALSE;
prev->pass_alloc = FALSE; prev->pass_alloc = FALSE;
audio_chain_set_make_func (prev, do_convert_in, convert, NULL); audio_chain_set_make_func (prev, do_convert_in, convert, NULL);
@ -656,7 +650,7 @@ chain_mix (GstAudioConverter * convert, AudioChain * prev)
in->channels, out->channels); in->channels, out->channels);
if (!convert->mix_passthrough) { if (!convert->mix_passthrough) {
prev = convert->mix_chain = audio_chain_new (prev, convert); prev = audio_chain_new (prev, convert);
prev->allow_ip = FALSE; prev->allow_ip = FALSE;
prev->pass_alloc = FALSE; prev->pass_alloc = FALSE;
audio_chain_set_make_func (prev, do_mix, convert, NULL); audio_chain_set_make_func (prev, do_mix, convert, NULL);
@ -692,7 +686,7 @@ chain_resample (GstAudioConverter * convert, AudioChain * prev)
gst_audio_resampler_new (method, flags, format, channels, in->rate, gst_audio_resampler_new (method, flags, format, channels, in->rate,
out->rate, convert->config); out->rate, convert->config);
prev = convert->resample_chain = audio_chain_new (prev, convert); prev = audio_chain_new (prev, convert);
prev->allow_ip = FALSE; prev->allow_ip = FALSE;
prev->pass_alloc = FALSE; prev->pass_alloc = FALSE;
audio_chain_set_make_func (prev, do_resample, convert, NULL); audio_chain_set_make_func (prev, do_resample, convert, NULL);
@ -715,7 +709,7 @@ chain_convert_out (GstAudioConverter * convert, AudioChain * prev)
convert->current_format = GST_AUDIO_FORMAT_S32; convert->current_format = GST_AUDIO_FORMAT_S32;
GST_INFO ("convert F64 to S32"); GST_INFO ("convert F64 to S32");
prev = convert->convert_out_chain = audio_chain_new (prev, convert); prev = audio_chain_new (prev, convert);
prev->allow_ip = TRUE; prev->allow_ip = TRUE;
prev->pass_alloc = FALSE; prev->pass_alloc = FALSE;
audio_chain_set_make_func (prev, do_convert_out, convert, NULL); audio_chain_set_make_func (prev, do_convert_out, convert, NULL);
@ -769,7 +763,7 @@ chain_quantize (GstAudioConverter * convert, AudioChain * prev)
gst_audio_quantize_new (dither, ns, 0, convert->current_format, gst_audio_quantize_new (dither, ns, 0, convert->current_format,
out->channels, 1U << (32 - out_depth)); out->channels, 1U << (32 - out_depth));
prev = convert->quant_chain = audio_chain_new (prev, convert); prev = audio_chain_new (prev, convert);
prev->allow_ip = TRUE; prev->allow_ip = TRUE;
prev->pass_alloc = TRUE; prev->pass_alloc = TRUE;
audio_chain_set_make_func (prev, do_quantize, convert, NULL); audio_chain_set_make_func (prev, do_quantize, convert, NULL);
@ -809,7 +803,7 @@ setup_allocators (GstAudioConverter * convert)
} }
/* now walk backwards, we try to write into the dest samples directly /* now walk backwards, we try to write into the dest samples directly
* and keep track if the source needs to be writable */ * and keep track if the source needs to be writable */
for (chain = convert->pack_chain; chain; chain = chain->prev) { for (chain = convert->chain_end; chain; chain = chain->prev) {
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;
@ -832,7 +826,7 @@ converter_passthrough (GstAudioConverter * convert,
AudioChain *chain; AudioChain *chain;
gsize samples; gsize samples;
chain = convert->pack_chain; chain = convert->chain_end;
samples = in_frames * chain->inc; samples = in_frames * chain->inc;
@ -863,7 +857,7 @@ converter_generic (GstAudioConverter * convert,
gint i; gint i;
gsize produced; gsize produced;
chain = convert->pack_chain; chain = convert->chain_end;
convert->in_writable = flags & GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE; convert->in_writable = flags & GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE;
convert->in_data = in; convert->in_data = in;
@ -953,7 +947,7 @@ gst_audio_converter_new (GstAudioConverterFlags flags, GstAudioInfo * in_info,
/* step 6, optional quantize */ /* step 6, optional quantize */
prev = chain_quantize (convert, prev); prev = chain_quantize (convert, prev);
/* step 7, pack */ /* step 7, pack */
convert->pack_chain = chain_pack (convert, prev); convert->chain_end = chain_pack (convert, prev);
convert->convert = converter_generic; convert->convert = converter_generic;
@ -993,20 +987,17 @@ unpositioned:
void void
gst_audio_converter_free (GstAudioConverter * convert) gst_audio_converter_free (GstAudioConverter * convert)
{ {
AudioChain *chain;
g_return_if_fail (convert != NULL); g_return_if_fail (convert != NULL);
if (convert->unpack_chain) /* walk the chain backwards and free all elements */
audio_chain_free (convert->unpack_chain); for (chain = convert->chain_end; chain;) {
if (convert->convert_in_chain) AudioChain *prev = chain->prev;
audio_chain_free (convert->convert_in_chain); audio_chain_free (chain);
if (convert->mix_chain) chain = prev;
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) if (convert->quant)
gst_audio_quantize_free (convert->quant); gst_audio_quantize_free (convert->quant);
if (convert->mix) if (convert->mix)