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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-06-27 10:20:42 +00:00
libs: audio: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
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11b47c4e29
commit
506c65aa27
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@ -33,9 +33,9 @@ gst_audio_buffer_unmap_internal (GstAudioBuffer * buffer, guint n_unmap)
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gst_buffer_unmap (buffer->buffer, &buffer->map_infos[i]);
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gst_buffer_unmap (buffer->buffer, &buffer->map_infos[i]);
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}
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}
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if (buffer->planes != buffer->priv_planes_arr)
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if (buffer->planes != buffer->priv_planes_arr)
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g_slice_free1 (buffer->n_planes * sizeof (gpointer), buffer->planes);
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g_free (buffer->planes);
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if (buffer->map_infos != buffer->priv_map_infos_arr)
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if (buffer->map_infos != buffer->priv_map_infos_arr)
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g_slice_free1 (buffer->n_planes * sizeof (GstMapInfo), buffer->map_infos);
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g_free (buffer->map_infos);
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}
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}
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/**
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/**
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@ -146,9 +146,8 @@ gst_audio_buffer_map (GstAudioBuffer * buffer, const GstAudioInfo * info,
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buffer->n_planes = GST_AUDIO_BUFFER_CHANNELS (buffer);
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buffer->n_planes = GST_AUDIO_BUFFER_CHANNELS (buffer);
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if (G_UNLIKELY (buffer->n_planes > 8)) {
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if (G_UNLIKELY (buffer->n_planes > 8)) {
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buffer->planes = g_slice_alloc (buffer->n_planes * sizeof (gpointer));
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buffer->planes = g_new (gpointer, buffer->n_planes);
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buffer->map_infos =
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buffer->map_infos = g_new (GstMapInfo, buffer->n_planes);
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g_slice_alloc (buffer->n_planes * sizeof (GstMapInfo));
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} else {
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} else {
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buffer->planes = buffer->priv_planes_arr;
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buffer->planes = buffer->priv_planes_arr;
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buffer->map_infos = buffer->priv_map_infos_arr;
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buffer->map_infos = buffer->priv_map_infos_arr;
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@ -95,7 +95,7 @@ gst_audio_channel_mixer_free (GstAudioChannelMixer * mix)
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g_free (mix->matrix_int);
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g_free (mix->matrix_int);
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mix->matrix_int = NULL;
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mix->matrix_int = NULL;
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g_slice_free (GstAudioChannelMixer, mix);
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g_free (mix);
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}
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}
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/*
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/*
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@ -836,7 +836,7 @@ gst_audio_channel_mixer_new_with_matrix (GstAudioChannelMixerFlags flags,
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g_return_val_if_fail (in_channels > 0 && in_channels < 64, NULL);
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g_return_val_if_fail (in_channels > 0 && in_channels < 64, NULL);
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g_return_val_if_fail (out_channels > 0 && out_channels < 64, NULL);
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g_return_val_if_fail (out_channels > 0 && out_channels < 64, NULL);
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mix = g_slice_new0 (GstAudioChannelMixer);
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mix = g_new0 (GstAudioChannelMixer, 1);
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mix->in_channels = in_channels;
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mix->in_channels = in_channels;
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mix->out_channels = out_channels;
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mix->out_channels = out_channels;
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@ -197,7 +197,7 @@ audio_chain_new (AudioChain * prev, GstAudioConverter * convert)
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{
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{
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AudioChain *chain;
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AudioChain *chain;
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chain = g_slice_new0 (AudioChain);
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chain = g_new0 (AudioChain, 1);
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chain->prev = prev;
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chain->prev = prev;
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if (convert->current_layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
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if (convert->current_layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
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@ -229,7 +229,7 @@ audio_chain_free (AudioChain * chain)
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if (chain->make_func_notify)
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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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chain->make_func_notify (chain->make_func_data);
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g_free (chain->tmp);
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g_free (chain->tmp);
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g_slice_free (AudioChain, chain);
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g_free (chain);
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}
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}
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static gpointer *
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static gpointer *
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@ -1347,7 +1347,7 @@ gst_audio_converter_new (GstAudioConverterFlags flags, GstAudioInfo * in_info,
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&& !opt_matrix)
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&& !opt_matrix)
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goto unpositioned;
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goto unpositioned;
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convert = g_slice_new0 (GstAudioConverter);
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convert = g_new0 (GstAudioConverter, 1);
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convert->flags = flags;
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convert->flags = flags;
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convert->in = *in_info;
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convert->in = *in_info;
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@ -1481,7 +1481,7 @@ gst_audio_converter_free (GstAudioConverter * convert)
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gst_structure_free (convert->config);
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gst_structure_free (convert->config);
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g_slice_free (GstAudioConverter, convert);
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g_free (convert);
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}
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}
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/**
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/**
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@ -61,7 +61,7 @@ ensure_debug_category (void)
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GstAudioInfo *
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GstAudioInfo *
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gst_audio_info_copy (const GstAudioInfo * info)
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gst_audio_info_copy (const GstAudioInfo * info)
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{
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{
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return g_slice_dup (GstAudioInfo, info);
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return g_memdup2 (info, sizeof (GstAudioInfo));
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}
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}
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/**
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/**
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@ -74,7 +74,7 @@ gst_audio_info_copy (const GstAudioInfo * info)
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void
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void
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gst_audio_info_free (GstAudioInfo * info)
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gst_audio_info_free (GstAudioInfo * info)
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{
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{
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g_slice_free (GstAudioInfo, info);
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g_free (info);
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}
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}
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G_DEFINE_BOXED_TYPE (GstAudioInfo, gst_audio_info,
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G_DEFINE_BOXED_TYPE (GstAudioInfo, gst_audio_info,
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@ -93,7 +93,7 @@ gst_audio_info_new (void)
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{
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{
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GstAudioInfo *info;
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GstAudioInfo *info;
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info = g_slice_new (GstAudioInfo);
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info = g_new (GstAudioInfo, 1);
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gst_audio_info_init (info);
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gst_audio_info_init (info);
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return info;
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return info;
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@ -446,7 +446,7 @@ gst_audio_quantize_new (GstAudioDitherMethod dither,
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g_return_val_if_fail (format == GST_AUDIO_FORMAT_S32, NULL);
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g_return_val_if_fail (format == GST_AUDIO_FORMAT_S32, NULL);
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g_return_val_if_fail (channels > 0, NULL);
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g_return_val_if_fail (channels > 0, NULL);
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quant = g_slice_new0 (GstAudioQuantize);
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quant = g_new0 (GstAudioQuantize, 1);
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quant->dither = dither;
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quant->dither = dither;
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quant->ns = ns;
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quant->ns = ns;
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quant->flags = flags;
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quant->flags = flags;
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@ -490,7 +490,7 @@ gst_audio_quantize_free (GstAudioQuantize * quant)
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g_free (quant->last_random);
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g_free (quant->last_random);
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g_free (quant->dither_buf);
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g_free (quant->dither_buf);
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g_slice_free (GstAudioQuantize, quant);
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g_free (quant);
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}
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}
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/**
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/**
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@ -1366,7 +1366,7 @@ gst_audio_resampler_new (GstAudioResamplerMethod method,
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audio_resampler_init ();
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audio_resampler_init ();
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resampler = g_slice_new0 (GstAudioResampler);
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resampler = g_new0 (GstAudioResampler, 1);
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resampler->method = method;
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resampler->method = method;
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resampler->flags = flags;
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resampler->flags = flags;
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resampler->format = format;
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resampler->format = format;
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@ -1634,7 +1634,7 @@ gst_audio_resampler_free (GstAudioResampler * resampler)
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g_free (resampler->sbuf);
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g_free (resampler->sbuf);
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if (resampler->options)
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if (resampler->options)
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gst_structure_free (resampler->options);
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gst_structure_free (resampler->options);
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g_slice_free (GstAudioResampler, resampler);
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g_free (resampler);
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}
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}
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/**
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/**
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@ -327,7 +327,7 @@ gst_audio_meta_free (GstMeta * meta, GstBuffer * buffer)
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GstAudioMeta *ameta = (GstAudioMeta *) meta;
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GstAudioMeta *ameta = (GstAudioMeta *) meta;
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if (ameta->offsets && ameta->offsets != ameta->priv_offsets_arr)
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if (ameta->offsets && ameta->offsets != ameta->priv_offsets_arr)
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g_slice_free1 (ameta->info.channels * sizeof (gsize), ameta->offsets);
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g_free (ameta->offsets);
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}
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}
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static gboolean
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static gboolean
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@ -413,7 +413,7 @@ gst_buffer_add_audio_meta (GstBuffer * buffer, const GstAudioInfo * info,
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#endif
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#endif
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if (G_UNLIKELY (info->channels > 8))
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if (G_UNLIKELY (info->channels > 8))
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meta->offsets = g_slice_alloc (info->channels * sizeof (gsize));
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meta->offsets = g_new (gsize, info->channels);
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else
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else
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meta->offsets = meta->priv_offsets_arr;
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meta->offsets = meta->priv_offsets_arr;
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@ -633,7 +633,7 @@ gst_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
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GST_INFO_OBJECT (buf, "Allocating an array for %d timestamps",
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GST_INFO_OBJECT (buf, "Allocating an array for %d timestamps",
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spec->segtotal);
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spec->segtotal);
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buf->timestamps = g_slice_alloc0 (sizeof (GstClockTime) * spec->segtotal);
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buf->timestamps = g_new0 (GstClockTime, spec->segtotal);
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/* initialize array with invalid timestamps */
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/* initialize array with invalid timestamps */
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for (i = 0; i < spec->segtotal; i++) {
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for (i = 0; i < spec->segtotal; i++) {
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buf->timestamps[i] = GST_CLOCK_TIME_NONE;
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buf->timestamps[i] = GST_CLOCK_TIME_NONE;
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@ -727,7 +727,7 @@ gst_audio_ring_buffer_release (GstAudioRingBuffer * buf)
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if (G_LIKELY (buf->timestamps)) {
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if (G_LIKELY (buf->timestamps)) {
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GST_INFO_OBJECT (buf, "Freeing timestamp buffer, %d entries",
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GST_INFO_OBJECT (buf, "Freeing timestamp buffer, %d entries",
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buf->spec.segtotal);
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buf->spec.segtotal);
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g_slice_free1 (sizeof (GstClockTime) * buf->spec.segtotal, buf->timestamps);
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g_free (buf->timestamps);
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buf->timestamps = NULL;
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buf->timestamps = NULL;
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}
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}
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