mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 12:11:13 +00:00
audio-converter: rework the main processing loop
Rework the main processing loop. We now create an audio processing chain from small core functions. This is very similar to how the video-converter core works and allows us to statically calculate an optimal allocation strategy for all possible combinations of operations. Make sure we support non-interleaved data everywhere. Add functions to calculate in and out frames and latency.
This commit is contained in:
Wim Taymans
parent
8bcf183c7f
commit
08734e7598
3 changed files with 575 additions and 173 deletions
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@ -74,8 +74,9 @@ ensure_debug_category (void)
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#define ensure_debug_category() /* NOOP */
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#endif /* GST_DISABLE_GST_DEBUG */
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typedef void (*AudioConvertFunc) (gpointer dst, const gpointer src, gint count);
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typedef struct _AudioChain AudioChain;
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typedef void (*AudioConvertFunc) (gpointer dst, const gpointer src, gint count);
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/* int/int int/float float/int float/float
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*
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* unpack S32 S32 F64 F64
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@ -84,6 +85,11 @@ typedef void (*AudioConvertFunc) (gpointer dst, const gpointer src, gint count);
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* convert F64->S32
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* quantize S32 S32
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* pack S32 F64 S32 F64
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*
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*
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* interleave
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* deinterleave
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* resample
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*/
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struct _GstAudioConverter
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{
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@ -92,26 +98,136 @@ struct _GstAudioConverter
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GstStructure *config;
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GstAudioConverterFlags flags;
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GstAudioFormat current_format;
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GstAudioLayout current_layout;
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gint current_channels;
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gpointer *in_data;
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gpointer *out_data;
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/* unpack */
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gboolean in_default;
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gboolean unpack_ip;
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AudioChain *unpack_chain;
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/* convert in */
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AudioConvertFunc convert_in;
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AudioChain *convert_in_chain;
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/* channel mix */
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gboolean mix_passthrough;
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GstAudioChannelMix *mix;
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AudioChain *mix_chain;
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/* convert out */
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AudioConvertFunc convert_out;
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AudioChain *convert_out_chain;
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/* quant */
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GstAudioQuantize *quant;
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AudioChain *quant_chain;
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/* pack */
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gboolean out_default;
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AudioChain *pack_chain;
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gboolean passthrough;
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gpointer tmpbuf;
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gpointer tmpbuf2;
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gint tmpbufsize;
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};
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typedef gboolean (*AudioChainFunc) (AudioChain * chain, gsize samples,
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gpointer user_data);
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typedef gpointer *(*AudioChainAllocFunc) (AudioChain * chain, gsize samples,
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gpointer user_data);
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static gpointer *get_output_samples (AudioChain * chain, gsize samples,
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gpointer user_data);
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struct _AudioChain
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{
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AudioChain *prev;
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AudioChainFunc make_func;
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gpointer make_func_data;
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GDestroyNotify make_func_notify;
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gint stride;
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gint inc;
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gint blocks;
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gboolean pass_alloc;
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gboolean allow_ip;
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AudioChainAllocFunc alloc_func;
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gpointer alloc_data;
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gpointer *tmp;
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gsize tmpsize;
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gpointer *samples;
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};
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static AudioChain *
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audio_chain_new (AudioChain * prev, GstAudioConverter * convert)
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{
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AudioChain *chain;
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const GstAudioFormatInfo *finfo;
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chain = g_slice_new0 (AudioChain);
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chain->prev = prev;
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if (convert->current_layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
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chain->inc = 1;
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chain->blocks = convert->current_channels;
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} else {
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chain->inc = convert->current_channels;
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chain->blocks = 1;
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}
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finfo = gst_audio_format_get_info (convert->current_format);
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chain->stride = (finfo->width * chain->inc) / 8;
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return chain;
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}
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static void
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audio_chain_set_make_func (AudioChain * chain,
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AudioChainFunc make_func, gpointer user_data, GDestroyNotify notify)
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{
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chain->make_func = make_func;
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chain->make_func_data = user_data;
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chain->make_func_notify = notify;
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}
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static void
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audio_chain_free (AudioChain * chain)
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{
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GST_LOG ("free chain %p", chain);
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if (chain->make_func_notify)
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chain->make_func_notify (chain->make_func_data);
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g_free (chain->tmp);
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g_slice_free (AudioChain, chain);
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}
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static gpointer *
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audio_chain_alloc_samples (AudioChain * chain, guint samples)
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{
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return chain->alloc_func (chain, samples, chain->alloc_data);
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}
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static gpointer *
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audio_chain_get_samples (AudioChain * chain, guint samples)
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{
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gpointer *res;
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while (!chain->samples)
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chain->make_func (chain, samples, chain->make_func_data);
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res = chain->samples;
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chain->samples = NULL;
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return res;
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}
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/*
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static guint
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get_opt_uint (GstAudioConverter * convert, const gchar * opt, guint def)
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@ -202,6 +318,351 @@ gst_audio_converter_get_config (GstAudioConverter * convert)
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return convert->config;
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}
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static gboolean
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do_unpack (AudioChain * chain, gsize samples, gpointer user_data)
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{
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GstAudioConverter *convert = user_data;
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gpointer *tmp;
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gboolean src_writable;
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src_writable = (convert->flags & GST_AUDIO_CONVERTER_FLAG_SOURCE_WRITABLE);
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if (!chain->allow_ip || !src_writable || !convert->in_default) {
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gint i;
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if (src_writable && chain->allow_ip)
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tmp = convert->in_data;
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else
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tmp = audio_chain_alloc_samples (chain, samples);
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GST_LOG ("unpack %p %p, %" G_GSIZE_FORMAT, tmp, convert->in_data, samples);
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for (i = 0; i < chain->blocks; i++) {
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convert->in.finfo->unpack_func (convert->in.finfo,
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GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE, tmp[i], convert->in_data[i],
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samples * chain->inc);
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}
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} else {
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tmp = convert->in_data;
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GST_LOG ("get in samples %p", tmp);
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}
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chain->samples = tmp;
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return TRUE;
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}
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static gboolean
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do_convert_in (AudioChain * chain, gsize samples, gpointer user_data)
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{
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GstAudioConverter *convert = user_data;
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gpointer *in, *out;
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gint i;
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in = audio_chain_get_samples (chain->prev, samples);
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out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, samples));
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GST_LOG ("convert in %p, %p %" G_GSIZE_FORMAT, in, out, samples);
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for (i = 0; i < chain->blocks; i++)
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convert->convert_in (out[i], in[i], samples * chain->inc);
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chain->samples = out;
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return TRUE;
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}
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static gboolean
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do_mix (AudioChain * chain, gsize samples, gpointer user_data)
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{
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GstAudioConverter *convert = user_data;
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gpointer *in, *out;
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in = audio_chain_get_samples (chain->prev, samples);
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out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, samples));
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GST_LOG ("mix %p %p,%" G_GSIZE_FORMAT, in, out, samples);
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gst_audio_channel_mix_samples (convert->mix, in, out, samples);
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chain->samples = out;
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return TRUE;
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}
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static gboolean
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do_convert_out (AudioChain * chain, gsize samples, gpointer user_data)
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{
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GstAudioConverter *convert = user_data;
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gpointer *in, *out;
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gint i;
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in = audio_chain_get_samples (chain->prev, samples);
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out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, samples));
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GST_LOG ("convert out %p, %p %" G_GSIZE_FORMAT, in, out, samples);
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for (i = 0; i < chain->blocks; i++)
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convert->convert_out (out[i], in[i], samples * chain->inc);
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chain->samples = out;
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return TRUE;
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}
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static gboolean
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do_quantize (AudioChain * chain, gsize samples, gpointer user_data)
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{
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GstAudioConverter *convert = user_data;
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gpointer *in, *out;
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in = audio_chain_get_samples (chain->prev, samples);
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out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, samples));
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GST_LOG ("quantize %p, %p %" G_GSIZE_FORMAT, in, out, samples);
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gst_audio_quantize_samples (convert->quant, in, out, samples);
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chain->samples = out;
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return TRUE;
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}
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static AudioChain *
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chain_unpack (GstAudioConverter * convert)
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{
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AudioChain *prev;
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GstAudioInfo *in = &convert->in;
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const GstAudioFormatInfo *fup;
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convert->current_format = in->finfo->unpack_format;
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convert->current_layout = in->layout;
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convert->current_channels = in->channels;
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convert->in_default = in->finfo->unpack_format == in->finfo->format;
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GST_INFO ("unpack format %s to %s",
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gst_audio_format_to_string (in->finfo->format),
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gst_audio_format_to_string (convert->current_format));
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fup = gst_audio_format_get_info (in->finfo->unpack_format);
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prev = convert->unpack_chain = audio_chain_new (NULL, convert);
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prev->allow_ip = fup->width <= in->finfo->width;
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prev->pass_alloc = FALSE;
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audio_chain_set_make_func (prev, do_unpack, convert, NULL);
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return prev;
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}
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static AudioChain *
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chain_convert_in (GstAudioConverter * convert, AudioChain * prev)
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{
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gboolean in_int, out_int;
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GstAudioInfo *in = &convert->in;
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GstAudioInfo *out = &convert->out;
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in_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (in->finfo);
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out_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (out->finfo);
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if (in_int && !out_int) {
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GST_INFO ("convert S32 to F64");
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convert->convert_in = (AudioConvertFunc) audio_orc_s32_to_double;
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convert->current_format = GST_AUDIO_FORMAT_F64;
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prev = convert->convert_in_chain = audio_chain_new (prev, convert);
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prev->allow_ip = FALSE;
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prev->pass_alloc = FALSE;
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audio_chain_set_make_func (prev, do_convert_in, convert, NULL);
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}
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return prev;
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}
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static AudioChain *
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chain_mix (GstAudioConverter * convert, AudioChain * prev)
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{
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GstAudioChannelMixFlags flags;
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GstAudioInfo *in = &convert->in;
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GstAudioInfo *out = &convert->out;
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GstAudioFormat format = convert->current_format;
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flags =
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GST_AUDIO_INFO_IS_UNPOSITIONED (in) ?
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GST_AUDIO_CHANNEL_MIX_FLAGS_UNPOSITIONED_IN : 0;
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flags |=
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GST_AUDIO_INFO_IS_UNPOSITIONED (out) ?
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GST_AUDIO_CHANNEL_MIX_FLAGS_UNPOSITIONED_OUT : 0;
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convert->current_channels = out->channels;
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convert->mix =
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gst_audio_channel_mix_new (flags, format, in->channels, in->position,
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out->channels, out->position);
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convert->mix_passthrough =
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gst_audio_channel_mix_is_passthrough (convert->mix);
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GST_INFO ("mix format %s, passthrough %d, in_channels %d, out_channels %d",
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gst_audio_format_to_string (format), convert->mix_passthrough,
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in->channels, out->channels);
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if (!convert->mix_passthrough) {
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prev = convert->mix_chain = audio_chain_new (prev, convert);
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/* we can only do in-place when in >= out, else we don't have enough
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* memory. */
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prev->allow_ip = in->channels >= out->channels;
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prev->pass_alloc = in->channels <= out->channels;
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audio_chain_set_make_func (prev, do_mix, convert, NULL);
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}
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return prev;
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}
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static AudioChain *
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chain_convert_out (GstAudioConverter * convert, AudioChain * prev)
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{
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gboolean in_int, out_int;
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GstAudioInfo *in = &convert->in;
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GstAudioInfo *out = &convert->out;
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in_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (in->finfo);
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out_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (out->finfo);
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if (!in_int && out_int) {
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convert->convert_out = (AudioConvertFunc) audio_orc_double_to_s32;
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convert->current_format = GST_AUDIO_FORMAT_S32;
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GST_INFO ("convert F64 to S32");
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prev = convert->convert_out_chain = audio_chain_new (prev, convert);
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prev->allow_ip = TRUE;
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prev->pass_alloc = FALSE;
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audio_chain_set_make_func (prev, do_convert_out, convert, NULL);
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}
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return prev;
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}
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static AudioChain *
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chain_quantize (GstAudioConverter * convert, AudioChain * prev)
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{
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GstAudioInfo *in = &convert->in;
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GstAudioInfo *out = &convert->out;
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gint in_depth, out_depth;
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gboolean in_int, out_int;
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GstAudioDitherMethod dither;
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GstAudioNoiseShapingMethod ns;
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dither = GET_OPT_DITHER_METHOD (convert);
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ns = GET_OPT_NOISE_SHAPING_METHOD (convert);
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in_depth = GST_AUDIO_FORMAT_INFO_DEPTH (in->finfo);
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out_depth = GST_AUDIO_FORMAT_INFO_DEPTH (out->finfo);
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GST_INFO ("depth in %d, out %d", in_depth, out_depth);
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in_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (in->finfo);
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out_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (out->finfo);
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/* Don't dither or apply noise shaping if target depth is bigger than 20 bits
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* as DA converters only can do a SNR up to 20 bits in reality.
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* Also don't dither or apply noise shaping if target depth is larger than
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* source depth. */
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if (out_depth > 20 || (in_int && out_depth >= in_depth)) {
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dither = GST_AUDIO_DITHER_NONE;
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ns = GST_AUDIO_NOISE_SHAPING_NONE;
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GST_INFO ("using no dither and noise shaping");
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} else {
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GST_INFO ("using dither %d and noise shaping %d", dither, ns);
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/* Use simple error feedback when output sample rate is smaller than
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* 32000 as the other methods might move the noise to audible ranges */
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if (ns > GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK && out->rate < 32000)
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ns = GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK;
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}
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/* we still want to run the quantization step when reducing bits to get
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* the rounding correct */
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if (out_int && out_depth < 32) {
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GST_INFO ("quantize to %d bits, dither %d, ns %d", out_depth, dither, ns);
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convert->quant =
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gst_audio_quantize_new (dither, ns, 0, convert->current_format,
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out->channels, 1U << (32 - out_depth));
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prev = convert->quant_chain = audio_chain_new (prev, convert);
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prev->allow_ip = TRUE;
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prev->pass_alloc = TRUE;
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audio_chain_set_make_func (prev, do_quantize, convert, NULL);
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}
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return prev;
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}
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static AudioChain *
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chain_pack (GstAudioConverter * convert, AudioChain * prev)
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{
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GstAudioInfo *out = &convert->out;
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GstAudioFormat format = convert->current_format;
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convert->current_format = out->finfo->format;
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g_assert (out->finfo->unpack_format == format);
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convert->out_default = format == out->finfo->format;
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GST_INFO ("pack format %s to %s", gst_audio_format_to_string (format),
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gst_audio_format_to_string (out->finfo->format));
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return prev;
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}
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static gpointer *
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get_output_samples (AudioChain * chain, gsize samples, gpointer user_data)
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{
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GstAudioConverter *convert = user_data;
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GST_LOG ("output samples %" G_GSIZE_FORMAT, samples);
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return convert->out_data;
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}
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static gpointer *
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get_temp_samples (AudioChain * chain, gsize samples, gpointer user_data)
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{
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gsize needed;
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/* first part contains the pointers, second part the data */
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needed = (samples * chain->stride + sizeof (gpointer)) * chain->blocks;
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if (needed > chain->tmpsize) {
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gint i;
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guint8 *s;
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GST_DEBUG ("alloc samples %" G_GSIZE_FORMAT, needed);
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chain->tmp = g_realloc (chain->tmp, needed);
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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 * samples * chain->stride);
|
||||
}
|
||||
return chain->tmp;
|
||||
}
|
||||
|
||||
static void
|
||||
setup_allocators (GstAudioConverter * convert)
|
||||
{
|
||||
AudioChain *chain;
|
||||
AudioChainAllocFunc alloc_func;
|
||||
gboolean allow_ip;
|
||||
|
||||
/* start with using dest if we can directly write into it */
|
||||
if (convert->out_default) {
|
||||
alloc_func = get_output_samples;
|
||||
allow_ip = FALSE;
|
||||
} else {
|
||||
alloc_func = get_temp_samples;
|
||||
allow_ip = TRUE;
|
||||
}
|
||||
/* now walk backwards, we try to write into the dest samples directly
|
||||
* and keep track if the source needs to be writable */
|
||||
for (chain = convert->pack_chain; chain; chain = chain->prev) {
|
||||
chain->alloc_func = alloc_func;
|
||||
chain->alloc_data = convert;
|
||||
chain->allow_ip = allow_ip && chain->allow_ip;
|
||||
|
||||
if (!chain->pass_alloc) {
|
||||
/* can't pass allocator, make new temp line allocator */
|
||||
alloc_func = get_temp_samples;
|
||||
allow_ip = TRUE;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* gst_audio_converter_new: (skip)
|
||||
* @in: a source #GstAudioInfo
|
||||
|
|
@ -221,12 +682,7 @@ gst_audio_converter_new (GstAudioInfo * in, GstAudioInfo * out,
|
|||
GstStructure * config)
|
||||
{
|
||||
GstAudioConverter *convert;
|
||||
gint in_depth, out_depth;
|
||||
GstAudioChannelMixFlags flags;
|
||||
gboolean in_int, out_int;
|
||||
GstAudioFormat format;
|
||||
GstAudioDitherMethod dither;
|
||||
GstAudioNoiseShapingMethod ns;
|
||||
AudioChain *prev;
|
||||
|
||||
g_return_val_if_fail (in != NULL, FALSE);
|
||||
g_return_val_if_fail (out != NULL, FALSE);
|
||||
|
|
@ -249,85 +705,20 @@ gst_audio_converter_new (GstAudioInfo * in, GstAudioInfo * out,
|
|||
if (config)
|
||||
gst_audio_converter_set_config (convert, config);
|
||||
|
||||
dither = GET_OPT_DITHER_METHOD (convert);
|
||||
ns = GET_OPT_NOISE_SHAPING_METHOD (convert);
|
||||
|
||||
GST_INFO ("unitsizes: %d -> %d", in->bpf, out->bpf);
|
||||
|
||||
in_depth = GST_AUDIO_FORMAT_INFO_DEPTH (in->finfo);
|
||||
out_depth = GST_AUDIO_FORMAT_INFO_DEPTH (out->finfo);
|
||||
|
||||
GST_INFO ("depth in %d, out %d", in_depth, out_depth);
|
||||
|
||||
in_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (in->finfo);
|
||||
out_int = GST_AUDIO_FORMAT_INFO_IS_INTEGER (out->finfo);
|
||||
|
||||
flags =
|
||||
GST_AUDIO_INFO_IS_UNPOSITIONED (in) ?
|
||||
GST_AUDIO_CHANNEL_MIX_FLAGS_UNPOSITIONED_IN : 0;
|
||||
flags |=
|
||||
GST_AUDIO_INFO_IS_UNPOSITIONED (out) ?
|
||||
GST_AUDIO_CHANNEL_MIX_FLAGS_UNPOSITIONED_OUT : 0;
|
||||
|
||||
|
||||
/* step 1, unpack */
|
||||
format = in->finfo->unpack_format;
|
||||
convert->in_default = in->finfo->unpack_format == in->finfo->format;
|
||||
GST_INFO ("unpack format %s to %s",
|
||||
gst_audio_format_to_string (in->finfo->format),
|
||||
gst_audio_format_to_string (format));
|
||||
|
||||
prev = chain_unpack (convert);
|
||||
/* step 2, optional convert from S32 to F64 for channel mix */
|
||||
if (in_int && !out_int) {
|
||||
GST_INFO ("convert S32 to F64");
|
||||
convert->convert_in = (AudioConvertFunc) audio_orc_s32_to_double;
|
||||
format = GST_AUDIO_FORMAT_F64;
|
||||
}
|
||||
|
||||
prev = chain_convert_in (convert, prev);
|
||||
/* step 3, channel mix */
|
||||
convert->mix =
|
||||
gst_audio_channel_mix_new (flags, format, in->channels, in->position,
|
||||
out->channels, out->position);
|
||||
convert->mix_passthrough =
|
||||
gst_audio_channel_mix_is_passthrough (convert->mix);
|
||||
GST_INFO ("mix format %s, passthrough %d, in_channels %d, out_channels %d",
|
||||
gst_audio_format_to_string (format), convert->mix_passthrough,
|
||||
in->channels, out->channels);
|
||||
|
||||
prev = chain_mix (convert, prev);
|
||||
/* step 4, optional convert for quantize */
|
||||
if (!in_int && out_int) {
|
||||
GST_INFO ("convert F64 to S32");
|
||||
convert->convert_out = (AudioConvertFunc) audio_orc_double_to_s32;
|
||||
format = GST_AUDIO_FORMAT_S32;
|
||||
}
|
||||
prev = chain_convert_out (convert, prev);
|
||||
/* step 5, optional quantize */
|
||||
/* Don't dither or apply noise shaping if target depth is bigger than 20 bits
|
||||
* as DA converters only can do a SNR up to 20 bits in reality.
|
||||
* Also don't dither or apply noise shaping if target depth is larger than
|
||||
* source depth. */
|
||||
if (out_depth > 20 || (in_int && out_depth >= in_depth)) {
|
||||
dither = GST_AUDIO_DITHER_NONE;
|
||||
ns = GST_AUDIO_NOISE_SHAPING_NONE;
|
||||
GST_INFO ("using no dither and noise shaping");
|
||||
} else {
|
||||
GST_INFO ("using dither %d and noise shaping %d", dither, ns);
|
||||
/* Use simple error feedback when output sample rate is smaller than
|
||||
* 32000 as the other methods might move the noise to audible ranges */
|
||||
if (ns > GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK && out->rate < 32000)
|
||||
ns = GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK;
|
||||
}
|
||||
/* we still want to run the quantization step when reducing bits to get
|
||||
* the rounding correct */
|
||||
if (out_int && out_depth < 32) {
|
||||
GST_INFO ("quantize to %d bits, dither %d, ns %d", out_depth, dither, ns);
|
||||
convert->quant = gst_audio_quantize_new (dither, ns, 0, format,
|
||||
out->channels, 1U << (32 - out_depth));
|
||||
}
|
||||
prev = chain_quantize (convert, prev);
|
||||
/* step 6, pack */
|
||||
g_assert (out->finfo->unpack_format == format);
|
||||
convert->out_default = format == out->finfo->format;
|
||||
GST_INFO ("pack format %s to %s", gst_audio_format_to_string (format),
|
||||
gst_audio_format_to_string (out->finfo->format));
|
||||
convert->pack_chain = chain_pack (convert, prev);
|
||||
|
||||
/* optimize */
|
||||
if (out->finfo->format == in->finfo->format && convert->mix_passthrough) {
|
||||
|
|
@ -335,6 +726,9 @@ gst_audio_converter_new (GstAudioInfo * in, GstAudioInfo * out,
|
|||
convert->passthrough = TRUE;
|
||||
}
|
||||
|
||||
setup_allocators (convert);
|
||||
|
||||
|
||||
return convert;
|
||||
|
||||
/* ERRORS */
|
||||
|
|
@ -356,6 +750,17 @@ gst_audio_converter_free (GstAudioConverter * convert)
|
|||
{
|
||||
g_return_if_fail (convert != NULL);
|
||||
|
||||
if (convert->unpack_chain)
|
||||
audio_chain_free (convert->unpack_chain);
|
||||
if (convert->convert_in_chain)
|
||||
audio_chain_free (convert->convert_in_chain);
|
||||
if (convert->mix_chain)
|
||||
audio_chain_free (convert->mix_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)
|
||||
|
|
@ -363,13 +768,61 @@ gst_audio_converter_free (GstAudioConverter * convert)
|
|||
gst_audio_info_init (&convert->in);
|
||||
gst_audio_info_init (&convert->out);
|
||||
|
||||
g_free (convert->tmpbuf);
|
||||
g_free (convert->tmpbuf2);
|
||||
gst_structure_free (convert->config);
|
||||
|
||||
g_slice_free (GstAudioConverter, convert);
|
||||
}
|
||||
|
||||
/**
|
||||
* gst_audio_converter_get_out_frames:
|
||||
* @convert: a #GstAudioConverter
|
||||
* @in_frames: number of input frames
|
||||
*
|
||||
* Calculate how many output frames can be produced when @in_frames input
|
||||
* frames are given to @convert.
|
||||
*
|
||||
* Returns: the number of output frames
|
||||
*/
|
||||
gsize
|
||||
gst_audio_converter_get_out_frames (GstAudioConverter * convert,
|
||||
gsize in_frames)
|
||||
{
|
||||
return in_frames;
|
||||
}
|
||||
|
||||
/**
|
||||
* gst_audio_converter_get_in_frames:
|
||||
* @convert: a #GstAudioConverter
|
||||
* @out_frames: number of output frames
|
||||
*
|
||||
* Calculate how many input frames are currently needed by @convert to produce
|
||||
* @out_frames of output frames.
|
||||
*
|
||||
* Returns: the number of input frames
|
||||
*/
|
||||
gsize
|
||||
gst_audio_converter_get_in_frames (GstAudioConverter * convert,
|
||||
gsize out_frames)
|
||||
{
|
||||
return out_frames;
|
||||
}
|
||||
|
||||
/**
|
||||
* gst_audio_converter_get_max_latency:
|
||||
* @convert: a #GstAudioConverter
|
||||
*
|
||||
* Get the maximum number of input frames that the converter would
|
||||
* need before producing output.
|
||||
*
|
||||
* Returns: the latency of @convert as expressed in the number of
|
||||
* frames.
|
||||
*/
|
||||
gsize
|
||||
gst_audio_converter_get_max_latency (GstAudioConverter * convert)
|
||||
{
|
||||
return 0;
|
||||
}
|
||||
|
||||
/**
|
||||
* gst_audio_converter_samples:
|
||||
* @convert: a #GstAudioConverter
|
||||
|
|
@ -402,8 +855,9 @@ gst_audio_converter_samples (GstAudioConverter * convert,
|
|||
gpointer out[], gsize out_samples, gsize * in_consumed,
|
||||
gsize * out_produced)
|
||||
{
|
||||
guint size;
|
||||
gpointer outbuf, tmpbuf, tmpbuf2, inp, outp;
|
||||
AudioChain *chain;
|
||||
gpointer *tmp;
|
||||
gint i;
|
||||
|
||||
g_return_val_if_fail (convert != NULL, FALSE);
|
||||
g_return_val_if_fail (in != NULL, FALSE);
|
||||
|
|
@ -419,93 +873,31 @@ gst_audio_converter_samples (GstAudioConverter * convert,
|
|||
return TRUE;
|
||||
}
|
||||
|
||||
inp = in[0];
|
||||
outp = out[0];
|
||||
chain = convert->pack_chain;
|
||||
|
||||
if (convert->passthrough) {
|
||||
memcpy (outp, inp, in_samples * convert->in.bpf);
|
||||
for (i = 0; i < chain->blocks; i++)
|
||||
memcpy (out[i], in[i], in_samples * chain->inc);
|
||||
*out_produced = in_samples;
|
||||
*in_consumed = in_samples;
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
size =
|
||||
sizeof (gdouble) * in_samples * MAX (convert->in.channels,
|
||||
convert->out.channels);
|
||||
convert->flags = flags;
|
||||
convert->in_data = in;
|
||||
convert->out_data = out;
|
||||
|
||||
if (size > convert->tmpbufsize) {
|
||||
convert->tmpbuf = g_realloc (convert->tmpbuf, size);
|
||||
convert->tmpbuf2 = g_realloc (convert->tmpbuf2, size);
|
||||
convert->tmpbufsize = size;
|
||||
}
|
||||
tmpbuf = convert->tmpbuf;
|
||||
tmpbuf2 = convert->tmpbuf2;
|
||||
/* get samples to pack */
|
||||
tmp = audio_chain_get_samples (chain, in_samples);
|
||||
|
||||
/* 1. unpack */
|
||||
if (!convert->in_default) {
|
||||
if (!convert->convert_in && convert->mix_passthrough
|
||||
&& !convert->convert_out && !convert->quant && convert->out_default)
|
||||
outbuf = outp;
|
||||
else
|
||||
outbuf = tmpbuf;
|
||||
|
||||
convert->in.finfo->unpack_func (convert->in.finfo,
|
||||
GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE, outbuf, inp,
|
||||
in_samples * convert->in.channels);
|
||||
inp = outbuf;
|
||||
}
|
||||
|
||||
/* 2. optionally convert for mixing */
|
||||
if (convert->convert_in) {
|
||||
if (convert->mix_passthrough && !convert->convert_out && !convert->quant
|
||||
&& convert->out_default)
|
||||
outbuf = outp;
|
||||
else if (inp == tmpbuf)
|
||||
outbuf = tmpbuf2;
|
||||
else
|
||||
outbuf = tmpbuf;
|
||||
|
||||
convert->convert_in (outbuf, inp, in_samples * convert->in.channels);
|
||||
inp = outbuf;
|
||||
}
|
||||
|
||||
/* step 3, channel mix if not passthrough */
|
||||
if (!convert->mix_passthrough) {
|
||||
if (!convert->convert_out && !convert->quant && convert->out_default)
|
||||
outbuf = outp;
|
||||
else
|
||||
outbuf = tmpbuf;
|
||||
|
||||
gst_audio_channel_mix_samples (convert->mix, &inp, &outbuf, in_samples);
|
||||
inp = outbuf;
|
||||
}
|
||||
/* step 4, optional convert F64 -> S32 for quantize */
|
||||
if (convert->convert_out) {
|
||||
if (!convert->quant && convert->out_default)
|
||||
outbuf = outp;
|
||||
else
|
||||
outbuf = tmpbuf;
|
||||
|
||||
convert->convert_out (outbuf, inp, in_samples * convert->out.channels);
|
||||
inp = outbuf;
|
||||
}
|
||||
|
||||
/* step 5, optional quantize */
|
||||
if (convert->quant) {
|
||||
if (convert->out_default)
|
||||
outbuf = outp;
|
||||
else
|
||||
outbuf = tmpbuf;
|
||||
|
||||
gst_audio_quantize_samples (convert->quant, &inp, &outbuf, in_samples);
|
||||
inp = outbuf;
|
||||
}
|
||||
|
||||
/* step 6, pack */
|
||||
if (!convert->out_default) {
|
||||
convert->out.finfo->pack_func (convert->out.finfo, 0, inp, outp,
|
||||
in_samples * convert->out.channels);
|
||||
GST_LOG ("pack %p, %p %" G_GSIZE_FORMAT, tmp, out, in_samples);
|
||||
/* and pack if needed */
|
||||
for (i = 0; i < chain->blocks; i++)
|
||||
convert->out.finfo->pack_func (convert->out.finfo, 0, tmp[i], out[i],
|
||||
in_samples * chain->inc);
|
||||
}
|
||||
|
||||
*out_produced = in_samples;
|
||||
*in_consumed = in_samples;
|
||||
|
||||
|
|
|
|||
|
|
@ -78,6 +78,13 @@ void gst_audio_converter_free (GstAudioConverter * con
|
|||
gboolean gst_audio_converter_set_config (GstAudioConverter * convert, GstStructure *config);
|
||||
const GstStructure * gst_audio_converter_get_config (GstAudioConverter * convert);
|
||||
|
||||
gsize gst_audio_converter_get_out_frames (GstAudioConverter *convert,
|
||||
gsize in_frames);
|
||||
gsize gst_audio_converter_get_in_frames (GstAudioConverter *convert,
|
||||
gsize out_frames);
|
||||
|
||||
gsize gst_audio_converter_get_max_latency (GstAudioConverter *convert);
|
||||
|
||||
|
||||
gboolean gst_audio_converter_samples (GstAudioConverter * convert,
|
||||
GstAudioConverterFlags flags,
|
||||
|
|
|
|||
|
|
@ -49,6 +49,9 @@ EXPORTS
|
|||
gst_audio_converter_flags_get_type
|
||||
gst_audio_converter_free
|
||||
gst_audio_converter_get_config
|
||||
gst_audio_converter_get_in_frames
|
||||
gst_audio_converter_get_max_latency
|
||||
gst_audio_converter_get_out_frames
|
||||
gst_audio_converter_new
|
||||
gst_audio_converter_samples
|
||||
gst_audio_converter_set_config
|
||||
|
|
|
|||
Loading…
Reference in a new issue