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rtpsource: refactor bitrate estimation
Don't reuse the same variable we need for stats for the bitrate estimation because we're updating it. Refactor the bitrate estimation code so that both sender and receivers use the same code path.
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commit
c971d1a9ab
2 changed files with 39 additions and 55 deletions
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@ -886,6 +886,40 @@ init_seq (RTPSource * src, guint16 seq)
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GST_DEBUG ("base_seq %d", seq);
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}
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static void
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do_bitrate_estimation (RTPSource * src, GstClockTime running_time,
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guint64 * bytes_handled)
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{
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guint64 elapsed;
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if (src->prev_rtime) {
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elapsed = running_time - src->prev_rtime;
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if (elapsed > (G_GINT64_CONSTANT (1) << 31)) {
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guint64 rate;
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rate =
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gst_util_uint64_scale (*bytes_handled, elapsed,
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(G_GINT64_CONSTANT (1) << 29));
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GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT
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", rate %" G_GUINT64_FORMAT, elapsed, *bytes_handled, rate);
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if (src->bitrate == 0)
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src->bitrate = rate;
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else
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src->bitrate = ((src->bitrate * 3) + rate) / 4;
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src->prev_rtime = running_time;
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*bytes_handled = 0;
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}
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} else {
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GST_LOG ("Reset bitrate measurement");
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src->prev_rtime = running_time;
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src->bitrate = 0;
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}
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}
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/**
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* rtp_source_process_rtp:
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* @src: an #RTPSource
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@ -903,7 +937,6 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
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guint16 seqnr, udelta;
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RTPSourceStats *stats;
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guint16 expected;
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guint64 elapsed;
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g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
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g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
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@ -981,37 +1014,13 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
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src->stats.octets_received += arrival->payload_len;
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src->stats.bytes_received += arrival->bytes;
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src->stats.packets_received++;
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/* for the bitrate estimation */
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src->bytes_received += arrival->bytes;
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/* the source that sent the packet must be a sender */
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src->is_sender = TRUE;
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src->validated = TRUE;
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if (src->prev_rtime) {
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elapsed = arrival->running_time - src->prev_rtime;
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if (elapsed > (G_GINT64_CONSTANT (1) << 31)) {
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guint64 rate;
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rate =
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gst_util_uint64_scale (src->stats.bytes_received, elapsed,
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(G_GINT64_CONSTANT (1) << 29));
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GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT
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", rate %" G_GUINT64_FORMAT, elapsed, src->stats.bytes_received,
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rate);
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if (src->bitrate == 0)
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src->bitrate = rate;
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else
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src->bitrate = ((src->bitrate * 3) + rate) / 4;
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src->prev_rtime = arrival->running_time;
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src->stats.bytes_received = 0;
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}
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} else {
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GST_LOG ("Reset bitrate measurement");
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src->prev_rtime = arrival->running_time;
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src->bitrate = 0;
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}
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do_bitrate_estimation (src, arrival->running_time, &src->bytes_received);
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GST_LOG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
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seqnr, src->stats.packets_received, src->stats.octets_received);
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@ -1094,7 +1103,6 @@ rtp_source_send_rtp (RTPSource * src, gpointer data, gboolean is_list,
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guint32 rtptime;
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guint64 ext_rtptime;
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guint64 rt_diff, rtp_diff;
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guint64 elapsed;
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GstBufferList *list = NULL;
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GstBuffer *buffer = NULL;
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guint packets;
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@ -1133,32 +1141,7 @@ rtp_source_send_rtp (RTPSource * src, gpointer data, gboolean is_list,
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src->stats.octets_sent += len;
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src->bytes_sent += len;
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if (src->prev_rtime) {
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elapsed = running_time - src->prev_rtime;
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if (elapsed > (G_GINT64_CONSTANT (1) << 31)) {
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guint64 rate;
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rate =
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gst_util_uint64_scale (src->bytes_sent, elapsed,
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(G_GINT64_CONSTANT (1) << 29));
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GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT
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", rate %" G_GUINT64_FORMAT, elapsed, src->bytes_sent, rate);
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if (src->bitrate == 0)
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src->bitrate = rate;
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else
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src->bitrate = ((src->bitrate * 3) + rate) / 4;
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src->prev_rtime = running_time;
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src->bytes_sent = 0;
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}
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} else {
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GST_LOG ("Reset bitrate measurement");
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src->prev_rtime = running_time;
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src->bitrate = 0;
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}
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do_bitrate_estimation (src, running_time, &src->bytes_sent);
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if (is_list) {
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rtptime = gst_rtp_buffer_list_get_timestamp (list);
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@ -143,6 +143,7 @@ struct _RTPSource {
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guint64 bitrate;
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GstClockTime prev_rtime;
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guint64 bytes_sent;
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guint64 bytes_received;
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GQueue *packets;
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