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Merge branch 'master' into 0.11

Browse source 
Conflicts:
	gst-libs/gst/rtsp/gstrtspconnection.c
	win32/common/libgstaudio.def
This commit is contained in:
Wim Taymans 2012-02-17 23:46:17 +01:00
commit 268d52fd33
3 changed files with 13 additions and 5 deletions
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8
docs/libs/gst-plugins-base-libs-sections.txt
View file

@ -182,17 +182,21 @@ gst_audio_decoder_finish_frame
gst_audio_decoder_get_audio_info
gst_audio_decoder_get_byte_time
gst_audio_decoder_get_delay
gst_audio_decoder_get_drainable
gst_audio_decoder_get_latency
gst_audio_decoder_get_max_errors
gst_audio_decoder_get_min_latency
gst_audio_decoder_get_needs_format
gst_audio_decoder_get_parse_state
gst_audio_decoder_get_plc
gst_audio_decoder_get_plc_aware
gst_audio_decoder_get_tolerance
gst_audio_decoder_set_byte_time
gst_audio_decoder_set_drainable
gst_audio_decoder_set_latency
gst_audio_decoder_set_max_errors
gst_audio_decoder_set_min_latency
gst_audio_decoder_set_needs_format
gst_audio_decoder_set_plc
gst_audio_decoder_set_plc_aware
gst_audio_decoder_set_tolerance
@ -219,9 +223,11 @@ GST_AUDIO_ENCODER_SRC_NAME
GST_AUDIO_ENCODER_SRC_PAD
gst_audio_encoder_finish_frame
gst_audio_encoder_get_audio_info
gst_audio_encoder_get_drainable
gst_audio_encoder_get_frame_max
gst_audio_encoder_get_frame_samples_min
gst_audio_encoder_get_frame_samples_max
gst_audio_encoder_get_hard_min
gst_audio_encoder_get_hard_resync
gst_audio_encoder_get_latency
gst_audio_encoder_get_lookahead
@ -229,9 +235,11 @@ gst_audio_encoder_get_mark_granule
gst_audio_encoder_get_perfect_timestamp
gst_audio_encoder_get_tolerance
gst_audio_encoder_proxy_getcaps
gst_audio_encoder_set_drainable
gst_audio_encoder_set_frame_max
gst_audio_encoder_set_frame_samples_min
gst_audio_encoder_set_frame_samples_max
gst_audio_encoder_set_hard_min
gst_audio_encoder_set_hard_resync
gst_audio_encoder_set_latency
gst_audio_encoder_set_lookahead

8
gst-libs/gst/audio/gstaudiodecoder.c
View file

@ -2502,7 +2502,7 @@ gst_audio_decoder_get_tolerance (GstAudioDecoder * dec)
/**
* gst_audio_decoder_set_drainable:
* @enc: a #GstAudioDecoder
* @dec: a #GstAudioDecoder
* @enabled: new state
*
* Configures decoder drain handling. If drainable, subclass might
@ -2526,7 +2526,7 @@ gst_audio_decoder_set_drainable (GstAudioDecoder * dec, gboolean enabled)
/**
* gst_audio_decoder_get_drainable:
* @enc: a #GstAudioDecoder
* @dec: a #GstAudioDecoder
*
* Queries decoder drain handling.
*
@ -2552,7 +2552,7 @@ gst_audio_decoder_get_drainable (GstAudioDecoder * dec)
/**
* gst_audio_decoder_set_needs_format:
* @enc: a #GstAudioDecoder
* @dec: a #GstAudioDecoder
* @enabled: new state
*
* Configures decoder format needs. If enabled, subclass needs to be
@ -2578,7 +2578,7 @@ gst_audio_decoder_set_needs_format (GstAudioDecoder * dec, gboolean enabled)
/**
* gst_audio_decoder_get_needs_format:
* @enc: a #GstAudioDecoder
* @dec: a #GstAudioDecoder
*
* Queries decoder required format handling.
*

2
gst-libs/gst/rtsp/gstrtspconnection.c
View file

@ -3320,7 +3320,7 @@ gst_rtsp_watch_write_data (GstRTSPWatch * watch, const guint8 * data,
g_mutex_lock (watch->mutex);
/* try to send the message synchronously first */
if (watch->messages->length == 0) {
if (watch->messages->length == 0 && watch->write_data == NULL) {
res =
write_bytes (watch->conn->write_socket, data, &off, size,
watch->conn->cancellable);