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-rw-r--r--src/filter/chain_filter_plugin.c39
-rw-r--r--src/filter/convert_filter_plugin.c5
-rw-r--r--src/filter/normalize_filter_plugin.c18
-rw-r--r--src/filter/null_filter_plugin.c3
-rw-r--r--src/filter/route_filter_plugin.c3
-rw-r--r--src/filter/volume_filter_plugin.c12
6 files changed, 47 insertions, 33 deletions
diff --git a/src/filter/chain_filter_plugin.c b/src/filter/chain_filter_plugin.c
index edfa70f35..56fa3a5e1 100644
--- a/src/filter/chain_filter_plugin.c
+++ b/src/filter/chain_filter_plugin.c
@@ -23,6 +23,7 @@
#include "filter_plugin.h"
#include "filter_internal.h"
#include "filter_registry.h"
+#include "audio_format.h"
#include <assert.h>
@@ -33,6 +34,12 @@ struct filter_chain {
GSList *children;
};
+static inline GQuark
+filter_quark(void)
+{
+ return g_quark_from_static_string("filter");
+}
+
static struct filter *
chain_filter_init(G_GNUC_UNUSED const struct config_param *param,
G_GNUC_UNUSED GError **error_r)
@@ -92,16 +99,42 @@ chain_close_until(struct filter_chain *chain, const struct filter *until)
}
static const struct audio_format *
-chain_filter_open(struct filter *_filter,
- const struct audio_format *audio_format,
+chain_open_child(struct filter *filter,
+ const struct audio_format *prev_audio_format,
+ GError **error_r)
+{
+ struct audio_format conv_audio_format = *prev_audio_format;
+ const struct audio_format *next_audio_format;
+
+ next_audio_format = filter_open(filter, &conv_audio_format, error_r);
+ if (next_audio_format == NULL)
+ return NULL;
+
+ if (!audio_format_equals(&conv_audio_format, prev_audio_format)) {
+ struct audio_format_string s;
+
+ filter_close(filter);
+ g_set_error(error_r, filter_quark(), 0,
+ "Audio format not supported by filter '%s': %s",
+ filter->plugin->name,
+ audio_format_to_string(prev_audio_format, &s));
+ return NULL;
+ }
+
+ return next_audio_format;
+}
+
+static const struct audio_format *
+chain_filter_open(struct filter *_filter, struct audio_format *in_audio_format,
GError **error_r)
{
struct filter_chain *chain = (struct filter_chain *)_filter;
+ const struct audio_format *audio_format = in_audio_format;
for (GSList *i = chain->children; i != NULL; i = g_slist_next(i)) {
struct filter *filter = i->data;
- audio_format = filter_open(filter, audio_format, error_r);
+ audio_format = chain_open_child(filter, audio_format, error_r);
if (audio_format == NULL) {
/* rollback, close all children */
chain_close_until(chain, filter);
diff --git a/src/filter/convert_filter_plugin.c b/src/filter/convert_filter_plugin.c
index 982ec7c4c..87070fa3a 100644
--- a/src/filter/convert_filter_plugin.c
+++ b/src/filter/convert_filter_plugin.c
@@ -71,8 +71,7 @@ convert_filter_finish(struct filter *filter)
}
static const struct audio_format *
-convert_filter_open(struct filter *_filter,
- const struct audio_format *audio_format,
+convert_filter_open(struct filter *_filter, struct audio_format *audio_format,
G_GNUC_UNUSED GError **error_r)
{
struct convert_filter *filter = (struct convert_filter *)_filter;
@@ -82,7 +81,7 @@ convert_filter_open(struct filter *_filter,
filter->in_audio_format = filter->out_audio_format = *audio_format;
pcm_convert_init(&filter->state);
- return audio_format;
+ return &filter->in_audio_format;
}
static void
diff --git a/src/filter/normalize_filter_plugin.c b/src/filter/normalize_filter_plugin.c
index 2694646ee..4a8bde51d 100644
--- a/src/filter/normalize_filter_plugin.c
+++ b/src/filter/normalize_filter_plugin.c
@@ -61,23 +61,13 @@ normalize_filter_finish(struct filter *filter)
static const struct audio_format *
normalize_filter_open(struct filter *_filter,
- const struct audio_format *audio_format,
- GError **error_r)
+ struct audio_format *audio_format,
+ G_GNUC_UNUSED GError **error_r)
{
struct normalize_filter *filter = (struct normalize_filter *)_filter;
- if (audio_format->format != SAMPLE_FORMAT_S16) {
- g_set_error(error_r, normalize_quark(), 0,
- "Unsupported audio format");
- return false;
- }
-
- if (audio_format->reverse_endian) {
- g_set_error(error_r, normalize_quark(), 0,
- "Normalize for reverse endian "
- "samples is not implemented");
- return false;
- }
+ audio_format->format = SAMPLE_FORMAT_S16;
+ audio_format->reverse_endian = false;
filter->compressor = Compressor_new(0);
diff --git a/src/filter/null_filter_plugin.c b/src/filter/null_filter_plugin.c
index 5671ba907..d677780dc 100644
--- a/src/filter/null_filter_plugin.c
+++ b/src/filter/null_filter_plugin.c
@@ -54,8 +54,7 @@ null_filter_finish(struct filter *_filter)
}
static const struct audio_format *
-null_filter_open(struct filter *_filter,
- const struct audio_format *audio_format,
+null_filter_open(struct filter *_filter, struct audio_format *audio_format,
G_GNUC_UNUSED GError **error_r)
{
struct null_filter *filter = (struct null_filter *)_filter;
diff --git a/src/filter/route_filter_plugin.c b/src/filter/route_filter_plugin.c
index 7fef2bc12..d9ca149c8 100644
--- a/src/filter/route_filter_plugin.c
+++ b/src/filter/route_filter_plugin.c
@@ -248,8 +248,7 @@ route_filter_finish(struct filter *_filter)
}
static const struct audio_format *
-route_filter_open(struct filter *_filter,
- const struct audio_format *audio_format,
+route_filter_open(struct filter *_filter, struct audio_format *audio_format,
G_GNUC_UNUSED GError **error_r)
{
struct route_filter *filter = (struct route_filter *)_filter;
diff --git a/src/filter/volume_filter_plugin.c b/src/filter/volume_filter_plugin.c
index 11549784f..82eed3acc 100644
--- a/src/filter/volume_filter_plugin.c
+++ b/src/filter/volume_filter_plugin.c
@@ -69,18 +69,12 @@ volume_filter_finish(struct filter *filter)
}
static const struct audio_format *
-volume_filter_open(struct filter *_filter,
- const struct audio_format *audio_format,
- GError **error_r)
+volume_filter_open(struct filter *_filter, struct audio_format *audio_format,
+ G_GNUC_UNUSED GError **error_r)
{
struct volume_filter *filter = (struct volume_filter *)_filter;
- if (audio_format->reverse_endian) {
- g_set_error(error_r, volume_quark(), 0,
- "Software volume for reverse endian "
- "samples is not implemented");
- return false;
- }
+ audio_format->reverse_endian = false;
filter->audio_format = *audio_format;
pcm_buffer_init(&filter->buffer);