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authorLuiz Augusto von Dentz <luiz.dentz@openbossa.org>2008-01-24 14:25:29 +0000
committerLuiz Augusto von Dentz <luiz.dentz@openbossa.org>2008-01-24 14:25:29 +0000
commit449c9728f515a3b1263f6961c608328b3ad582a9 (patch)
treece5f7affe29df1c19fa165ecf850fd181c1c586d
parent6f15a199e037d789188b3b89e77ac78acee7812f (diff)
Fix gtreamer payloader sending fragmented frames.
-rw-r--r--audio/gstrtpsbcpay.c16
-rw-r--r--audio/gstsbcutil.c21
2 files changed, 28 insertions, 9 deletions
diff --git a/audio/gstrtpsbcpay.c b/audio/gstrtpsbcpay.c
index fdb42d18..25bf70eb 100644
--- a/audio/gstrtpsbcpay.c
+++ b/audio/gstrtpsbcpay.c
@@ -162,6 +162,8 @@ static GstFlowReturn gst_rtp_sbc_pay_flush_buffers(GstRtpSBCPay *sbcpay)
GstBuffer* outbuf;
guint8 *payload_data;
guint8 *data;
+ guint frame_count;
+ guint payload_length;
struct rtp_payload *payload;
if (sbcpay->frame_length == 0) {
@@ -176,25 +178,27 @@ static GstFlowReturn gst_rtp_sbc_pay_flush_buffers(GstRtpSBCPay *sbcpay)
0, 0);
max_payload = MIN(max_payload, available);
+ frame_count = max_payload / sbcpay->frame_length;
+ payload_length = frame_count * sbcpay->frame_length;
- outbuf = gst_rtp_buffer_new_allocate(max_payload +
+ outbuf = gst_rtp_buffer_new_allocate(payload_length +
RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
gst_rtp_buffer_set_payload_type(outbuf,
GST_BASE_RTP_PAYLOAD_PT(sbcpay));
- data = gst_adapter_take(sbcpay->adapter, max_payload);
payload_data = gst_rtp_buffer_get_payload(outbuf);
-
payload = (struct rtp_payload*) payload_data;
memset(payload, 0, sizeof(struct rtp_payload));
- payload->frame_count = max_payload / sbcpay->frame_length;
+ payload->frame_count = frame_count;
- memcpy(payload_data + RTP_SBC_PAYLOAD_HEADER_SIZE, data, max_payload);
+ data = gst_adapter_take(sbcpay->adapter, payload_length);
+ memcpy(payload_data + RTP_SBC_PAYLOAD_HEADER_SIZE, data,
+ payload_length);
g_free(data);
GST_BUFFER_TIMESTAMP(outbuf) = sbcpay->timestamp;
- GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes", max_payload);
+ GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes", payload_length);
return gst_basertppayload_push(GST_BASE_RTP_PAYLOAD(sbcpay), outbuf);
}
diff --git a/audio/gstsbcutil.c b/audio/gstsbcutil.c
index 04b7217a..78024f7e 100644
--- a/audio/gstsbcutil.c
+++ b/audio/gstsbcutil.c
@@ -100,8 +100,8 @@ const gchar *gst_sbc_get_mode_from_list(const GValue *list)
for (i = 0; i < size; i++) {
value = gst_value_list_get_value(list, i);
aux = g_value_get_string(value);
- if (strcmp("stereo", aux) == 0) {
- return "stereo";
+ if (strcmp("joint", aux) == 0) {
+ return "joint";
}
}
return g_value_get_string(gst_value_list_get_value(list, size-1));
@@ -135,6 +135,19 @@ gint gst_sbc_get_mode_int(const gchar *mode)
return -1;
}
+/* FIXME add dual when sbc_t supports it */
+gboolean gst_sbc_get_mode_int_for_sbc_t(const gchar *mode)
+{
+ if (g_ascii_strcasecmp(mode, "joint") == 0)
+ return TRUE;
+ else if (g_ascii_strcasecmp(mode, "stereo") == 0)
+ return FALSE;
+ else if (g_ascii_strcasecmp(mode, "mono") == 0)
+ return FALSE;
+ else
+ return -1;
+}
+
const gchar *gst_sbc_get_mode_string(int joint)
{
switch (joint) {
@@ -183,6 +196,8 @@ const gchar *gst_sbc_get_mode_string_from_sbc_t(int channels, int joint)
return "joint";
else if (channels == 2 && joint == 0)
return "stereo";
+ else if (channels == 1 && joint == 0)
+ return "mono";
else
return NULL;
}
@@ -434,7 +449,7 @@ gboolean gst_sbc_util_fill_sbc_params(sbc_t *sbc, GstCaps *caps)
sbc->blocks = blocks;
sbc->subbands = subbands;
sbc->bitpool = bitpool;
- sbc->joint = gst_sbc_get_mode_int(mode);
+ sbc->joint = gst_sbc_get_mode_int_for_sbc_t(mode);
sbc->allocation = gst_sbc_get_allocation_mode_int(allocation);
return TRUE;