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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 12:11:13 +00:00
audio-converter: handle NULL input
Allow NULL as input to mean silence samples.
This commit is contained in:
parent
6050509b65
commit
2d971df593
1 changed files with 35 additions and 20 deletions
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@ -154,6 +154,7 @@ struct _AudioChain
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gpointer make_func_data;
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gpointer make_func_data;
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GDestroyNotify make_func_notify;
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GDestroyNotify make_func_notify;
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const GstAudioFormatInfo *finfo;
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gint stride;
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gint stride;
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gint inc;
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gint inc;
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gint blocks;
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gint blocks;
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@ -175,7 +176,6 @@ static AudioChain *
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audio_chain_new (AudioChain * prev, GstAudioConverter * convert)
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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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const GstAudioFormatInfo *finfo;
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chain = g_slice_new0 (AudioChain);
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chain = g_slice_new0 (AudioChain);
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chain->prev = prev;
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chain->prev = prev;
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@ -187,8 +187,8 @@ audio_chain_new (AudioChain * prev, GstAudioConverter * convert)
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chain->inc = convert->current_channels;
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chain->inc = convert->current_channels;
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chain->blocks = 1;
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chain->blocks = 1;
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}
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}
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finfo = gst_audio_format_get_info (convert->current_format);
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chain->finfo = gst_audio_format_get_info (convert->current_format);
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chain->stride = (finfo->width * chain->inc) / 8;
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chain->stride = (chain->finfo->width * chain->inc) / 8;
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return chain;
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return chain;
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}
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}
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@ -425,6 +425,7 @@ do_unpack (AudioChain * chain, gpointer user_data)
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GST_LOG ("unpack to tmp %p, %" G_GSIZE_FORMAT, tmp, num_samples);
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GST_LOG ("unpack to tmp %p, %" G_GSIZE_FORMAT, tmp, num_samples);
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}
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}
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if (convert->in_data) {
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for (i = 0; i < chain->blocks; i++) {
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for (i = 0; i < chain->blocks; i++) {
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GST_LOG ("unpack %p, %p, %" G_GSIZE_FORMAT, tmp[i], convert->in_data[i],
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GST_LOG ("unpack %p, %p, %" G_GSIZE_FORMAT, tmp[i], convert->in_data[i],
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num_samples);
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num_samples);
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@ -432,6 +433,12 @@ do_unpack (AudioChain * chain, gpointer user_data)
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GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE, tmp[i], convert->in_data[i],
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GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE, tmp[i], convert->in_data[i],
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num_samples * chain->inc);
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num_samples * chain->inc);
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}
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}
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} else {
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for (i = 0; i < chain->blocks; i++) {
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gst_audio_format_fill_silence (chain->finfo, tmp[i],
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num_samples * chain->inc);
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}
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}
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} else {
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} else {
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tmp = convert->in_data;
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tmp = convert->in_data;
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GST_LOG ("get in samples %p", tmp);
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GST_LOG ("get in samples %p", tmp);
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@ -531,7 +538,6 @@ chain_unpack (GstAudioConverter * convert)
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AudioChain *prev;
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AudioChain *prev;
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GstAudioInfo *in = &convert->in;
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GstAudioInfo *in = &convert->in;
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GstAudioInfo *out = &convert->out;
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GstAudioInfo *out = &convert->out;
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const GstAudioFormatInfo *fup;
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gboolean same_format;
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gboolean same_format;
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same_format = in->finfo->format == out->finfo->format;
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same_format = in->finfo->format == out->finfo->format;
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@ -552,10 +558,8 @@ chain_unpack (GstAudioConverter * convert)
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gst_audio_format_to_string (in->finfo->format),
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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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gst_audio_format_to_string (convert->current_format));
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fup = gst_audio_format_get_info (convert->current_format);
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prev = convert->unpack_chain = audio_chain_new (NULL, convert);
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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->allow_ip = prev->finfo->width <= in->finfo->width;
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prev->pass_alloc = FALSE;
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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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audio_chain_set_make_func (prev, do_unpack, convert, NULL);
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@ -747,18 +751,27 @@ converter_passthrough (GstAudioConverter * convert,
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gpointer out[], gsize out_frames)
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gpointer out[], gsize out_frames)
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{
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{
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gint i;
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gint i;
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gsize bytes;
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AudioChain *chain;
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AudioChain *chain;
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gsize samples;
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chain = convert->pack_chain;
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chain = convert->pack_chain;
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bytes = in_frames * chain->inc * (convert->in.bpf / convert->in.channels);
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samples = in_frames * chain->inc;
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GST_LOG ("passthrough: %" G_GSIZE_FORMAT " / %" G_GSIZE_FORMAT " bytes",
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GST_LOG ("passthrough: %" G_GSIZE_FORMAT " / %" G_GSIZE_FORMAT " samples",
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in_frames, bytes);
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in_frames, samples);
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if (in) {
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gsize bytes;
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bytes = samples * (convert->in.bpf / convert->in.channels);
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for (i = 0; i < chain->blocks; i++)
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for (i = 0; i < chain->blocks; i++)
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memcpy (out[i], in[i], bytes);
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memcpy (out[i], in[i], bytes);
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} else {
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for (i = 0; i < chain->blocks; i++)
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gst_audio_format_fill_silence (convert->in.finfo, out[i], samples);
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}
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return TRUE;
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return TRUE;
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}
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}
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@ -977,6 +990,9 @@ gst_audio_converter_get_max_latency (GstAudioConverter * convert)
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* If non-interleaved samples are used, @in and @out must point to an
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* If non-interleaved samples are used, @in and @out must point to an
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* array with pointers to memory blocks, one for each channel.
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* array with pointers to memory blocks, one for each channel.
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*
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*
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* @in may be %NULL, in which case @in_frames of silence samples are processed
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* by the converter.
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*
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* This function always produces @out_frames of output and consumes @in_frames of
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* This function always produces @out_frames of output and consumes @in_frames of
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* input. Use gst_audio_converter_get_out_frames() and
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* input. Use gst_audio_converter_get_out_frames() and
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* gst_audio_converter_get_in_frames() to make sure @in_frames and @out_frames
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* gst_audio_converter_get_in_frames() to make sure @in_frames and @out_frames
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@ -990,7 +1006,6 @@ gst_audio_converter_samples (GstAudioConverter * convert,
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gpointer out[], gsize out_frames)
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gpointer out[], gsize out_frames)
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{
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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 (convert != NULL, FALSE);
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g_return_val_if_fail (in != NULL, FALSE);
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g_return_val_if_fail (out != NULL, FALSE);
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g_return_val_if_fail (out != NULL, FALSE);
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in_frames = MIN (in_frames, out_frames);
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in_frames = MIN (in_frames, out_frames);
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