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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 12:11:13 +00:00
audio-converter: improve processing loop
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
parent
20f6af651b
commit
8a8b12189e
1 changed files with 104 additions and 65 deletions
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@ -103,8 +103,11 @@ struct _GstAudioConverter
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GstAudioLayout current_layout;
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gint current_channels;
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gboolean in_writable;
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gpointer *in_data;
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gsize in_samples;
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gpointer *out_data;
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gsize out_samples;
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/* unpack */
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gboolean in_default;
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@ -135,14 +138,10 @@ struct _GstAudioConverter
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gboolean passthrough;
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};
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typedef gboolean (*AudioChainFunc) (AudioChain * chain, gsize num_samples,
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gpointer user_data);
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typedef gboolean (*AudioChainFunc) (AudioChain * chain, gpointer user_data);
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typedef gpointer *(*AudioChainAllocFunc) (AudioChain * chain, gsize num_samples,
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gpointer user_data);
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static gpointer *get_output_samples (AudioChain * chain, gsize num_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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@ -165,6 +164,7 @@ struct _AudioChain
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gsize tmpsize;
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gpointer *samples;
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gsize num_samples;
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};
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static AudioChain *
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@ -214,15 +214,26 @@ audio_chain_alloc_samples (AudioChain * chain, gsize num_samples)
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return chain->alloc_func (chain, num_samples, chain->alloc_data);
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}
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static void
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audio_chain_set_samples (AudioChain * chain, gpointer * samples,
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gsize num_samples)
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{
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GST_LOG ("set samples %p %" G_GSIZE_FORMAT, samples, num_samples);
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chain->samples = samples;
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chain->num_samples = num_samples;
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}
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static gpointer *
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audio_chain_get_samples (AudioChain * chain, gsize num_samples)
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audio_chain_get_samples (AudioChain * chain, gsize * avail)
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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, num_samples, chain->make_func_data);
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chain->make_func (chain, chain->make_func_data);
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res = chain->samples;
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*avail = chain->num_samples;
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chain->samples = NULL;
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return res;
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@ -299,6 +310,18 @@ gst_audio_converter_update_config (GstAudioConverter * convert,
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{
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g_return_val_if_fail (convert != NULL, FALSE);
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g_return_val_if_fail (config != NULL, FALSE);
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g_return_val_if_fail ((in_rate == 0 && out_rate == 0) ||
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convert->flags & GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE, FALSE);
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GST_LOG ("new rate %d -> %d", in_rate, out_rate);
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if (in_rate <= 0)
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in_rate = convert->in.rate;
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if (out_rate <= 0)
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out_rate = convert->out.rate;
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convert->in.rate = in_rate;
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convert->out.rate = out_rate;
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gst_structure_foreach (config, copy_config, convert);
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gst_structure_free (config);
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@ -331,22 +354,64 @@ 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 num_samples, gpointer user_data)
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static gpointer *
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get_output_samples (AudioChain * chain, gsize num_samples, gpointer user_data)
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{
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GstAudioConverter *convert = user_data;
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GST_LOG ("output samples %p %" G_GSIZE_FORMAT, convert->out_data,
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num_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 num_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 = (num_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;
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/* jump to the data */
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s = (guint8 *) & chain->tmp[chain->blocks];
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/* set up the pointers */
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for (i = 0; i < chain->blocks; i++)
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chain->tmp[i] = s + (i * num_samples * chain->stride);
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}
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GST_LOG ("temp samples %p %" G_GSIZE_FORMAT, chain->tmp, num_samples);
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return chain->tmp;
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}
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static gboolean
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do_unpack (AudioChain * chain, gpointer user_data)
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{
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GstAudioConverter *convert = user_data;
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gsize num_samples;
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gpointer *tmp;
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gboolean src_writable;
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gboolean in_writable;
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src_writable = (convert->flags & GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE);
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in_writable = convert->in_writable;
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num_samples = convert->in_samples;
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if (!chain->allow_ip || !src_writable || !convert->in_default) {
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if (!chain->allow_ip || !in_writable || !convert->in_default) {
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gint i;
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if (src_writable && chain->allow_ip)
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if (in_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, num_samples);
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GST_LOG ("unpack %p, %p, %" G_GSIZE_FORMAT, tmp, convert->in_data,
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num_samples);
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@ -359,79 +424,83 @@ do_unpack (AudioChain * chain, gsize num_samples, gpointer user_data)
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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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audio_chain_set_samples (chain, tmp, num_samples);
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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 num_samples, gpointer user_data)
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do_convert_in (AudioChain * chain, gpointer user_data)
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{
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GstAudioConverter *convert = user_data;
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gsize num_samples;
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gpointer *in, *out;
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gint i;
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in = audio_chain_get_samples (chain->prev, num_samples);
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in = audio_chain_get_samples (chain->prev, &num_samples);
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out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples));
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GST_LOG ("convert in %p, %p, %" G_GSIZE_FORMAT, in, out, num_samples);
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for (i = 0; i < chain->blocks; i++)
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convert->convert_in (out[i], in[i], num_samples * chain->inc);
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chain->samples = out;
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audio_chain_set_samples (chain, out, num_samples);
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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 num_samples, gpointer user_data)
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do_mix (AudioChain * chain, gpointer user_data)
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{
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GstAudioConverter *convert = user_data;
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gsize num_samples;
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gpointer *in, *out;
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in = audio_chain_get_samples (chain->prev, num_samples);
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in = audio_chain_get_samples (chain->prev, &num_samples);
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out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples));
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GST_LOG ("mix %p, %p, %" G_GSIZE_FORMAT, in, out, num_samples);
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gst_audio_channel_mixer_samples (convert->mix, in, out, num_samples);
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chain->samples = out;
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audio_chain_set_samples (chain, out, num_samples);
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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 num_samples, gpointer user_data)
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do_convert_out (AudioChain * chain, gpointer user_data)
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{
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GstAudioConverter *convert = user_data;
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gsize num_samples;
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gpointer *in, *out;
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gint i;
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in = audio_chain_get_samples (chain->prev, num_samples);
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in = audio_chain_get_samples (chain->prev, &num_samples);
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out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples));
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GST_LOG ("convert out %p, %p, %" G_GSIZE_FORMAT, in, out, num_samples);
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for (i = 0; i < chain->blocks; i++)
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convert->convert_out (out[i], in[i], num_samples * chain->inc);
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chain->samples = out;
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audio_chain_set_samples (chain, out, num_samples);
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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 num_samples, gpointer user_data)
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do_quantize (AudioChain * chain, gpointer user_data)
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{
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GstAudioConverter *convert = user_data;
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gsize num_samples;
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gpointer *in, *out;
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in = audio_chain_get_samples (chain->prev, num_samples);
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in = audio_chain_get_samples (chain->prev, &num_samples);
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out = (chain->allow_ip ? in : audio_chain_alloc_samples (chain, num_samples));
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GST_LOG ("quantize %p, %p, %" G_GSIZE_FORMAT, in, out, num_samples);
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gst_audio_quantize_samples (convert->quant, in, out, num_samples);
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chain->samples = out;
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audio_chain_set_samples (chain, out, num_samples);
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return TRUE;
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}
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@ -612,41 +681,6 @@ chain_pack (GstAudioConverter * convert, AudioChain * prev)
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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 num_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 = (num_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;
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/* jump to the data */
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s = (guint8 *) & chain->tmp[chain->blocks];
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/* set up the pointers */
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for (i = 0; i < chain->blocks; i++)
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chain->tmp[i] = s + (i * num_samples * chain->stride);
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}
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return chain->tmp;
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}
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static void
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setup_allocators (GstAudioConverter * convert)
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{
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@ -668,6 +702,7 @@ setup_allocators (GstAudioConverter * convert)
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chain->alloc_func = alloc_func;
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chain->alloc_data = convert;
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chain->allow_ip = allow_ip && chain->allow_ip;
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GST_LOG ("chain %p: %d %d", chain, allow_ip, chain->allow_ip);
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if (!chain->pass_alloc) {
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/* can't pass allocator, make new temp line allocator */
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@ -712,6 +747,7 @@ gst_audio_converter_new (GstAudioConverterFlags flags, GstAudioInfo * in_info,
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convert = g_slice_new0 (GstAudioConverter);
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convert->flags = flags;
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convert->in = *in_info;
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convert->out = *out_info;
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@ -871,6 +907,7 @@ gst_audio_converter_samples (GstAudioConverter * convert,
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AudioChain *chain;
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gpointer *tmp;
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gint i;
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gsize produced;
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g_return_val_if_fail (convert != NULL, FALSE);
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g_return_val_if_fail (in != NULL, FALSE);
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@ -898,19 +935,21 @@ gst_audio_converter_samples (GstAudioConverter * convert,
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GST_LOG ("converting %" G_GSIZE_FORMAT, in_samples);
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convert->flags = flags;
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convert->in_writable = flags & GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE;
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convert->in_data = in;
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convert->in_samples = in_samples;
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convert->out_data = out;
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convert->out_samples = out_samples;
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/* get samples to pack */
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tmp = audio_chain_get_samples (chain, in_samples);
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tmp = audio_chain_get_samples (chain, &produced);
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if (!convert->out_default) {
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GST_LOG ("pack %p, %p %" G_GSIZE_FORMAT, tmp, out, in_samples);
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GST_LOG ("pack %p, %p %" G_GSIZE_FORMAT, tmp, out, produced);
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/* and pack if needed */
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for (i = 0; i < chain->blocks; i++)
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convert->out.finfo->pack_func (convert->out.finfo, 0, tmp[i], out[i],
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in_samples * chain->inc);
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produced * chain->inc);
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}
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return TRUE;
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}
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