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audiortppay: fix sample duration calculation

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This commit is contained in:
Wim Taymans 2009-09-02 13:12:28 +02:00
parent 299ab7be0e
commit 5808041f44
1 changed files with 10 additions and 9 deletions
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19
gst-libs/gst/rtp/gstbasertpaudiopayload.c
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@ -75,7 +75,7 @@ typedef gboolean (*GetLengthsFunc) (GstBaseRTPPayload * basepayload,
guint * min_payload_len, guint * max_payload_len, guint * align); guint * min_payload_len, guint * max_payload_len, guint * align);
/* function to convert bytes to a duration */ /* function to convert bytes to a duration */
typedef GstClockTime (*GetDurationFunc) (GstBaseRTPAudioPayload * payload, typedef GstClockTime (*GetDurationFunc) (GstBaseRTPAudioPayload * payload,
guint bytes); guint64 bytes);
struct _GstBaseRTPAudioPayloadPrivate struct _GstBaseRTPAudioPayloadPrivate
{ {
@ -104,10 +104,10 @@ static gboolean gst_base_rtp_audio_payload_get_sample_lengths (GstBaseRTPPayload
/* duration functions */ /* duration functions */
static GstClockTime static GstClockTime
gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload * payload, gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload * payload,
guint bytes); guint64 bytes);
static GstClockTime static GstClockTime
gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload * gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload *
payload, guint bytes); payload, guint64 bytes);
static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload
* payload, GstBuffer * buffer); * payload, GstBuffer * buffer);
@ -392,7 +392,7 @@ gst_base_rtp_audio_payload_get_frame_lengths (GstBaseRTPPayload *
static GstClockTime static GstClockTime
gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload * gst_base_rtp_audio_payload_get_frame_duration (GstBaseRTPAudioPayload *
payload, guint bytes) payload, guint64 bytes)
{ {
return gst_util_uint64_scale (bytes, payload->frame_duration * GST_MSECOND, return gst_util_uint64_scale (bytes, payload->frame_duration * GST_MSECOND,
payload->frame_size); payload->frame_size);
@ -431,7 +431,7 @@ gst_base_rtp_audio_payload_get_sample_lengths (GstBaseRTPPayload *
maxptime_octets); maxptime_octets);
/* min number of bytes based on a given ptime, has to be a multiple /* min number of bytes based on a given ptime, has to be a multiple
of sample rate */ * of sample rate */
minptime_octets = gst_util_uint64_scale (basepayload->min_ptime * 8, minptime_octets = gst_util_uint64_scale (basepayload->min_ptime * 8,
basepayload->clock_rate, payload->sample_size * GST_SECOND); basepayload->clock_rate, payload->sample_size * GST_SECOND);
@ -445,10 +445,10 @@ gst_base_rtp_audio_payload_get_sample_lengths (GstBaseRTPPayload *
static GstClockTime static GstClockTime
gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload * gst_base_rtp_audio_payload_get_sample_duration (GstBaseRTPAudioPayload *
payload, guint bytes) payload, guint64 bytes)
{ {
return gst_util_uint64_scale (bytes * 8 * GST_SECOND, return (bytes * 8 * GST_SECOND) /
GST_BASE_RTP_PAYLOAD (payload)->clock_rate, payload->sample_size); (GST_BASE_RTP_PAYLOAD (payload)->clock_rate * payload->sample_size);
} }
static GstFlowReturn static GstFlowReturn
@ -617,7 +617,8 @@ gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event)
switch (GST_EVENT_TYPE (event)) { switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS: case GST_EVENT_EOS:
/* FIXME. push remaining bytes */ /* FIXME. push remaining bytes? maybe not because it would violate the
* min-ptime. */
gst_adapter_clear (payload->priv->adapter); gst_adapter_clear (payload->priv->adapter);
break; break;
case GST_EVENT_FLUSH_STOP: case GST_EVENT_FLUSH_STOP: