#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type())
#define GST_TYPE_JACK_CLIENT (gst_jack_client_get_type ())
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+#define GST_JACK_FORMAT_STR "F32_LE"
+#else
+#define GST_JACK_FORMAT_STR "F32_BE"
+#endif
+
GType gst_jack_client_get_type(void);
GType gst_jack_connect_get_type(void);
buf = GST_RING_BUFFER_CAST (arg);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
- channels = buf->spec.channels;
+ channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
/* get target buffers */
for (i = 0; i < channels; i++) {
GstJackRingBuffer *abuf;
const char **ports;
gint sample_rate, buffer_size;
- gint i, channels, res;
+ gint i, rate, bpf, channels, res;
jack_client_t *client;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
client = gst_jack_audio_client_get_client (sink->client);
+ rate = GST_AUDIO_INFO_RATE (&spec->info);
+
/* sample rate must be that of the server */
sample_rate = jack_get_sample_rate (client);
- if (sample_rate != spec->rate)
+ if (sample_rate != rate)
goto wrong_samplerate;
- channels = spec->channels;
+ channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
+ bpf = GST_AUDIO_INFO_BPF (&spec->info);
if (!gst_jack_audio_sink_allocate_channels (sink, channels))
goto out_of_ports;
* for all channels */
spec->segsize = buffer_size * sizeof (gfloat) * channels;
spec->latency_time = gst_util_uint64_scale (spec->segsize,
- (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
+ (GST_SECOND / GST_USECOND), rate * bpf);
/* segtotal based on buffer-time latency */
spec->segtotal = spec->buffer_time / spec->latency_time;
if (spec->segtotal < 2) {
abuf->sample_rate = sample_rate;
abuf->buffer_size = buffer_size;
- abuf->channels = spec->channels;
+ abuf->channels = channels;
return TRUE;
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Wrong samplerate, server is running at %d and we received %d",
- sample_rate, spec->rate));
+ sample_rate, rate));
return FALSE;
}
out_of_ports:
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-float, "
- "endianness = (int) BYTE_ORDER, "
- "width = (int) 32, "
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " GST_JACK_FORMAT_STR ", "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
GST_DEBUG_OBJECT (sink, "got %d-%d ports, samplerate: %d", min, max, rate);
if (!sink->caps) {
- sink->caps = gst_caps_new_simple ("audio/x-raw-float",
- "endianness", G_TYPE_INT, G_BYTE_ORDER,
- "width", G_TYPE_INT, 32,
+ sink->caps = gst_caps_new_simple ("audio/x-raw",
+ "format", G_TYPE_STRING, GST_JACK_FORMAT_STR,
"rate", G_TYPE_INT, rate,
"channels", GST_TYPE_INT_RANGE, min, max, NULL);
}
buf = GST_RING_BUFFER_CAST (arg);
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
- channels = buf->spec.channels;
+ channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
/* get input buffers */
for (i = 0; i < channels; i++)
GstJackRingBuffer *abuf;
const char **ports;
gint sample_rate, buffer_size;
- gint i, channels, res;
+ gint i, bpf, rate, channels, res;
jack_client_t *client;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
client = gst_jack_audio_client_get_client (src->client);
+ rate = GST_AUDIO_INFO_RATE (&spec->info);
+
/* sample rate must be that of the server */
sample_rate = jack_get_sample_rate (client);
- if (sample_rate != spec->rate)
+ if (sample_rate != rate)
goto wrong_samplerate;
- channels = spec->channels;
+ bpf = GST_AUDIO_INFO_BPF (&spec->info);
+ channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
if (!gst_jack_audio_src_allocate_channels (src, channels))
goto out_of_ports;
* for all channels */
spec->segsize = buffer_size * sizeof (gfloat) * channels;
spec->latency_time = gst_util_uint64_scale (spec->segsize,
- (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
+ (GST_SECOND / GST_USECOND), rate * bpf);
/* segtotal based on buffer-time latency */
spec->segtotal = spec->buffer_time / spec->latency_time;
if (spec->segtotal < 2) {
abuf->sample_rate = sample_rate;
abuf->buffer_size = buffer_size;
- abuf->channels = spec->channels;
+ abuf->channels = channels;
return TRUE;
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Wrong samplerate, server is running at %d and we received %d",
- sample_rate, spec->rate));
+ sample_rate, rate));
return FALSE;
}
out_of_ports:
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-float, "
- "endianness = (int) BYTE_ORDER, "
- "width = (int) 32, "
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " GST_JACK_FORMAT_STR ", "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
GST_DEBUG_OBJECT (src, "got %d-%d ports, samplerate: %d", min, max, rate);
if (!src->caps) {
- src->caps = gst_caps_new_simple ("audio/x-raw-float",
- "endianness", G_TYPE_INT, G_BYTE_ORDER,
- "width", G_TYPE_INT, 32,
+ src->caps = gst_caps_new_simple ("audio/x-raw",
+ "format", G_TYPE_STRING, GST_JACK_FORMAT_STR,
"rate", G_TYPE_INT, rate,
"channels", GST_TYPE_INT_RANGE, min, max, NULL);
}
G_STMT_START { \
guint8 *sb = s, *db = d; \
while (s <= se && d < de) { \
- memcpy (d, s, bps); \
- s += bps; \
+ memcpy (d, s, bpf); \
+ s += bpf; \
*accum += outr; \
if ((*accum << 1) >= inr) { \
*accum -= inr; \
- d += bps; \
+ d += bpf; \
} \
} \
- in_samples -= (s - sb)/bps; \
- out_samples -= (d - db)/bps; \
+ in_samples -= (s - sb)/bpf; \
+ out_samples -= (d - db)/bpf; \
GST_DEBUG ("fwd_up end %d/%d",*accum,*toprocess); \
} G_STMT_END
G_STMT_START { \
guint8 *sb = s, *db = d; \
while (s <= se && d < de) { \
- memcpy (d, s, bps); \
- d += bps; \
+ memcpy (d, s, bpf); \
+ d += bpf; \
*accum += inr; \
if ((*accum << 1) >= outr) { \
*accum -= outr; \
- s += bps; \
+ s += bpf; \
} \
} \
- in_samples -= (s - sb)/bps; \
- out_samples -= (d - db)/bps; \
+ in_samples -= (s - sb)/bpf; \
+ out_samples -= (d - db)/bpf; \
GST_DEBUG ("fwd_down end %d/%d",*accum,*toprocess); \
} G_STMT_END
G_STMT_START { \
guint8 *sb = se, *db = d; \
while (s <= se && d < de) { \
- memcpy (d, se, bps); \
- se -= bps; \
+ memcpy (d, se, bpf); \
+ se -= bpf; \
*accum += outr; \
while (d < de && (*accum << 1) >= inr) { \
*accum -= inr; \
- d += bps; \
+ d += bpf; \
} \
} \
- in_samples -= (sb - se)/bps; \
- out_samples -= (d - db)/bps; \
+ in_samples -= (sb - se)/bpf; \
+ out_samples -= (d - db)/bpf; \
GST_DEBUG ("rev_up end %d/%d",*accum,*toprocess); \
} G_STMT_END
G_STMT_START { \
guint8 *sb = se, *db = d; \
while (s <= se && d < de) { \
- memcpy (d, se, bps); \
- d += bps; \
+ memcpy (d, se, bpf); \
+ d += bpf; \
*accum += inr; \
while (s <= se && (*accum << 1) >= outr) { \
*accum -= outr; \
- se -= bps; \
+ se -= bpf; \
} \
} \
- in_samples -= (sb - se)/bps; \
- out_samples -= (d - db)/bps; \
+ in_samples -= (sb - se)/bpf; \
+ out_samples -= (d - db)/bpf; \
GST_DEBUG ("rev_down end %d/%d",*accum,*toprocess); \
} G_STMT_END
guint8 *data_end;
gboolean reverse;
gint *toprocess;
- gint inr, outr, bps;
+ gint inr, outr, bpf;
gint64 offset;
guint bufsize;
GST_DEBUG_OBJECT (psink, "entering commit");
pbuf->in_commit = TRUE;
- bps = buf->spec.bytes_per_sample;
+ bpf = GST_AUDIO_INFO_BPF (&buf->spec.info);
bufsize = buf->spec.segsize * buf->spec.segtotal;
/* our toy resampler for trick modes */
/* data_end points to the last sample we have to write, not past it. This is
* needed to properly handle reverse playback: it points to the last sample. */
- data_end = data + (bps * inr);
+ data_end = data + (bpf * inr);
#ifdef HAVE_PULSE_1_0
if (g_atomic_int_get (&psink->format_lost)) {
goto was_paused;
/* offset is in bytes */
- offset = *sample * bps;
+ offset = *sample * bpf;
while (*toprocess > 0) {
size_t avail;
GST_LOG_OBJECT (psink, "discontinuity, offset is %" G_GINT64_FORMAT ", "
"last offset was %" G_GINT64_FORMAT, offset, pbuf->m_lastoffset);
- towrite = out_samples * bps;
+ towrite = out_samples * bpf;
/* Only ever write segsize bytes at once. This will
* also limit the PA shm buffer to segsize
GST_LOG_OBJECT (psink,
"flushing %u samples at offset %" G_GINT64_FORMAT,
- (guint) pbuf->m_towrite / bps, pbuf->m_offset);
+ (guint) pbuf->m_towrite / bpf, pbuf->m_offset);
if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) {
if (pbuf->m_writable == (size_t) - 1)
goto writable_size_failed;
- pbuf->m_writable /= bps;
- pbuf->m_writable *= bps; /* handle only complete samples */
+ pbuf->m_writable /= bpf;
+ pbuf->m_writable *= bpf; /* handle only complete samples */
if (pbuf->m_writable >= towrite)
break;
if (pbuf->m_writable < towrite)
towrite = pbuf->m_writable;
- avail = towrite / bps;
+ avail = towrite / bpf;
GST_LOG_OBJECT (psink, "writing %u samples at offset %" G_GUINT64_FORMAT,
(guint) avail, offset);
pbuf->m_towrite += towrite;
pbuf->m_writable -= towrite;
- avail = towrite / bps;
+ avail = towrite / bpf;
}
/* flush the buffer if it's full */
if ((pbuf->m_data != NULL) && (pbuf->m_towrite > 0)
&& (pbuf->m_writable == 0)) {
GST_LOG_OBJECT (psink, "flushing %u samples at offset %" G_GINT64_FORMAT,
- (guint) pbuf->m_towrite / bps, pbuf->m_offset);
+ (guint) pbuf->m_towrite / bpf, pbuf->m_offset);
if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) {
}
*sample += avail;
- offset += avail * bps;
+ offset += avail * bpf;
pbuf->m_lastoffset = offset;
/* check if we need to uncork after writing the samples */
fake_done:
#endif
/* we consumed all samples here */
- data = data_end + bps;
+ data = data_end + bpf;
pbuf->in_commit = FALSE;
pa_threaded_mainloop_unlock (mainloop);
done:
- result = inr - ((data_end - data) / bps);
+ result = inr - ((data_end - data) / bpf);
GST_LOG_OBJECT (psink, "wrote %d samples", result);
return result;
/* flush the buffer if possible */
if (pbuf->stream && (pbuf->m_data != NULL) && (pbuf->m_towrite > 0)) {
#ifndef GST_DISABLE_GST_DEBUG
- gint bps;
+ gint bpf;
- bps = (GST_RING_BUFFER_CAST (pbuf))->spec.bytes_per_sample;
+ bpf = (GST_RING_BUFFER_CAST (pbuf))->spec.info.bpf;
GST_LOG_OBJECT (psink,
"flushing %u samples at offset %" G_GINT64_FORMAT,
- (guint) pbuf->m_towrite / bps, pbuf->m_offset);
+ (guint) pbuf->m_towrite / bpf, pbuf->m_offset);
#endif
if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
GstStateChange transition);
#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
-# define ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
+# define FORMATS "{ S16_LE, S16_BE, F32_LE, F32_BE, S32_LE, S32_BE, " \
+ "S24_3LE, S24_3BE, S24_LE, S24_BE, S8 }"
#else
-# define ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
+# define FORMATS "{ S16_BE, S16_LE, F32_BE, F32_LE, S32_BE, S32_LE, " \
+ "S24_3BE, S24_3LE, S24_BE, S24_LE, S8 }"
#endif
static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "endianness = (int) { " ENDIANNESS " }, "
- "signed = (boolean) TRUE, "
- "width = (int) 16, "
- "depth = (int) 16, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, 32 ];"
- "audio/x-raw-float, "
- "endianness = (int) { " ENDIANNESS " }, "
- "width = (int) 32, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, 32 ];"
- "audio/x-raw-int, "
- "endianness = (int) { " ENDIANNESS " }, "
- "signed = (boolean) TRUE, "
- "width = (int) 32, "
- "depth = (int) 32, "
- "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 32 ];"
- "audio/x-raw-int, "
- "endianness = (int) { " ENDIANNESS " }, "
- "signed = (boolean) TRUE, "
- "width = (int) 24, "
- "depth = (int) 24, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, 32 ];"
- "audio/x-raw-int, "
- "endianness = (int) { " ENDIANNESS " }, "
- "signed = (boolean) TRUE, "
- "width = (int) 32, "
- "depth = (int) 24, "
- "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 32 ];"
- "audio/x-raw-int, "
- "signed = (boolean) FALSE, "
- "width = (int) 8, "
- "depth = (int) 8, "
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " FORMATS ", "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 32 ];"
"audio/x-alaw, "
element, GstStateChange transition);
#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
-# define ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
+# define FORMATS "{ S16_LE, S16_BE, F32_LE, F32_BE, S32_LE, S32_BE, U8 }"
#else
-# define ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
+# define FORMATS "{ S16_BE, S16_LE, F32_BE, F32_LE, S32_BE, S32_LE, U8 }"
#endif
static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "endianness = (int) { " ENDIANNESS " }, "
- "signed = (boolean) TRUE, "
- "width = (int) 16, "
- "depth = (int) 16, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, 32 ];"
- "audio/x-raw-float, "
- "endianness = (int) { " ENDIANNESS " }, "
- "width = (int) 32, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, 32 ];"
- "audio/x-raw-int, "
- "endianness = (int) { " ENDIANNESS " }, "
- "signed = (boolean) TRUE, "
- "width = (int) 32, "
- "depth = (int) 32, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, 32 ];"
- "audio/x-raw-int, "
- "signed = (boolean) FALSE, "
- "width = (int) 8, "
- "depth = (int) 8, "
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " FORMATS ", "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 32 ];"
"audio/x-alaw, "
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_has_field (s, "channel-layout") ||
!gst_pulse_gst_to_channel_map (&channel_map, &spec)) {
- if (spec.channels == 1)
+ if (spec.info.channels == 1)
pa_channel_map_init_mono (&channel_map);
- else if (spec.channels == 2)
+ else if (spec.info.channels == 2)
pa_channel_map_init_stereo (&channel_map);
else
need_channel_layout = TRUE;
gst_pulse_fill_sample_spec (GstRingBufferSpec * spec, pa_sample_spec * ss)
{
- if (spec->format == GST_MU_LAW && spec->width == 8)
+ if (spec->type == GST_BUFTYPE_RAW) {
+ switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
+ case GST_AUDIO_FORMAT_U8:
+ ss->format = PA_SAMPLE_U8;
+ break;
+ case GST_AUDIO_FORMAT_S16_LE:
+ ss->format = PA_SAMPLE_S16LE;
+ break;
+ case GST_AUDIO_FORMAT_S16_BE:
+ ss->format = PA_SAMPLE_S16BE;
+ break;
+ case GST_AUDIO_FORMAT_F32_LE:
+ ss->format = PA_SAMPLE_FLOAT32LE;
+ break;
+ case GST_AUDIO_FORMAT_F32_BE:
+ ss->format = PA_SAMPLE_FLOAT32BE;
+ break;
+ case GST_AUDIO_FORMAT_S32_LE:
+ ss->format = PA_SAMPLE_S32LE;
+ break;
+ case GST_AUDIO_FORMAT_S32_BE:
+ ss->format = PA_SAMPLE_S32BE;
+ break;
+ case GST_AUDIO_FORMAT_S24_3LE:
+ ss->format = PA_SAMPLE_S24LE;
+ break;
+ case GST_AUDIO_FORMAT_S24_3BE:
+ ss->format = PA_SAMPLE_S24BE;
+ break;
+ case GST_AUDIO_FORMAT_S24_LE:
+ ss->format = PA_SAMPLE_S24_32LE;
+ break;
+ case GST_AUDIO_FORMAT_S24_BE:
+ ss->format = PA_SAMPLE_S24_32BE;
+ break;
+ default:
+ return FALSE;
+ }
+ } else if (spec->type == GST_BUFTYPE_MU_LAW) {
ss->format = PA_SAMPLE_ULAW;
- else if (spec->format == GST_A_LAW && spec->width == 8)
+ } else if (spec->type == GST_BUFTYPE_A_LAW) {
ss->format = PA_SAMPLE_ALAW;
- else if (spec->format == GST_U8 && spec->width == 8)
- ss->format = PA_SAMPLE_U8;
- else if (spec->format == GST_S16_LE && spec->width == 16)
- ss->format = PA_SAMPLE_S16LE;
- else if (spec->format == GST_S16_BE && spec->width == 16)
- ss->format = PA_SAMPLE_S16BE;
- else if (spec->format == GST_FLOAT32_LE && spec->width == 32)
- ss->format = PA_SAMPLE_FLOAT32LE;
- else if (spec->format == GST_FLOAT32_BE && spec->width == 32)
- ss->format = PA_SAMPLE_FLOAT32BE;
- else if (spec->format == GST_S32_LE && spec->width == 32)
- ss->format = PA_SAMPLE_S32LE;
- else if (spec->format == GST_S32_BE && spec->width == 32)
- ss->format = PA_SAMPLE_S32BE;
- else if (spec->format == GST_S24_3LE && spec->width == 24)
- ss->format = PA_SAMPLE_S24LE;
- else if (spec->format == GST_S24_3BE && spec->width == 24)
- ss->format = PA_SAMPLE_S24BE;
- else if (spec->format == GST_S24_LE && spec->width == 32)
- ss->format = PA_SAMPLE_S24_32LE;
- else if (spec->format == GST_S24_BE && spec->width == 32)
- ss->format = PA_SAMPLE_S24_32BE;
- else
+ } else
return FALSE;
- ss->channels = spec->channels;
- ss->rate = spec->rate;
+ ss->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
+ ss->rate = GST_AUDIO_INFO_RATE (&spec->info);
if (!pa_sample_spec_valid (ss))
return FALSE;
return NULL;
}
- for (i = 0; i < spec->channels; i++) {
+ for (i = 0; i < spec->info.channels; i++) {
if (pos[i] == GST_AUDIO_CHANNEL_POSITION_NONE) {
/* no valid mappings for these channels */
g_free (pos);
}
g_free (pos);
- map->channels = spec->channels;
+ map->channels = spec->info.channels;
if (!pa_channel_map_valid (map)) {
return NULL;
int i;
GstAudioChannelPosition *pos;
gboolean invalid = FALSE;
+ gint channels;
+
+ channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
- g_return_val_if_fail (map->channels == spec->channels, NULL);
+ g_return_val_if_fail (map->channels == channels, NULL);
- pos = g_new0 (GstAudioChannelPosition, spec->channels + 1);
+ pos = g_new0 (GstAudioChannelPosition, channels + 1);
- for (i = 0; i < spec->channels; i++) {
+ for (i = 0; i < channels; i++) {
if (map->map[i] == PA_CHANNEL_POSITION_INVALID) {
invalid = TRUE;
break;
}
}
- if (!invalid && !gst_audio_check_channel_positions (pos, spec->channels))
+ if (!invalid && !gst_audio_check_channel_positions (pos, channels))
invalid = TRUE;
if (invalid) {
- for (i = 0; i < spec->channels; i++)
+ for (i = 0; i < channels; i++)
pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
}
gst_object_unref (targetpad);
}
-static GstStaticCaps raw_caps =
- GST_STATIC_CAPS ("audio/x-raw-int; audio/x-raw-float");
+static GstStaticCaps raw_caps = GST_STATIC_CAPS ("audio/x-raw");
static void
gst_auto_audio_sink_init (GstAutoAudioSink * sink)
gst_object_unref (targetpad);
}
-static GstStaticCaps raw_caps =
- GST_STATIC_CAPS ("audio/x-raw-int; audio/x-raw-float");
+static GstStaticCaps raw_caps = GST_STATIC_CAPS ("audio/x-raw");
static void
gst_auto_audio_src_init (GstAutoAudioSrc * src)
projects / platform / upstream / gstreamer.git / commitdiff