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gst-libs/gst/audio/: Various small cleanups.
Original commit message from CVS: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_class_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time), (gst_baseaudiosink_set_property), (build_linear_format), (debug_spec_caps), (debug_spec_buffer), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_is_acquired), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_stop), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample), (wait_segment), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_advance), (gst_ringbuffer_clear): Various small cleanups. * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain), (gst_audio_convert_change_state): * gst/subparse/gstsubparse.c: (gst_subparse_chain): No need to take the locks anymore.
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5 changed files with 38 additions and 13 deletions
34
ChangeLog
34
ChangeLog
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@ -1,3 +1,37 @@
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2005-05-25 Wim Taymans <wim@fluendo.com>
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* gst-libs/gst/audio/gstaudiosink.c:
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(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
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(audioringbuffer_thread_func), (gst_audioringbuffer_init),
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(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
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(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
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(gst_audioringbuffer_delay), (gst_audiosink_class_init),
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(gst_audiosink_create_ringbuffer):
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
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(gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
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(gst_baseaudiosink_set_property), (build_linear_format),
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(debug_spec_caps), (debug_spec_buffer),
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(gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
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(gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
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(gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
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(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
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* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
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(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
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(gst_ringbuffer_release), (gst_ringbuffer_is_acquired),
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(gst_ringbuffer_play), (gst_ringbuffer_pause),
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(gst_ringbuffer_stop), (gst_ringbuffer_delay),
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(gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample),
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(wait_segment), (gst_ringbuffer_commit),
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(gst_ringbuffer_prepare_read), (gst_ringbuffer_advance),
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(gst_ringbuffer_clear):
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Various small cleanups.
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* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
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(gst_audio_convert_change_state):
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* gst/subparse/gstsubparse.c: (gst_subparse_chain):
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No need to take the locks anymore.
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2005-05-25 Wim Taymans <wim@fluendo.com>
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* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
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@ -294,6 +294,7 @@ gst_audioringbuffer_release (GstRingBuffer * buf)
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/* free the buffer */
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gst_buffer_unref (buf->data);
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buf->data = NULL;
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if (csink->close)
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result = csink->close (sink);
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@ -457,6 +457,7 @@ gst_baseaudiosink_event (GstBaseSink * bsink, GstEvent * event)
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sample = 0;
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have_value:
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GST_DEBUG ("discont now at %lld", sample);
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gst_ringbuffer_set_sample (sink->ringbuffer, sample);
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break;
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}
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@ -483,7 +484,8 @@ gst_baseaudiosink_render (GstBaseSink * bsink, GstBuffer * buf)
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offset = GST_BUFFER_OFFSET (buf);
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GST_DEBUG ("in offset %llu, time %lld", offset, GST_BUFFER_TIMESTAMP (buf));
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GST_DEBUG ("in offset %llu, time %" GST_TIME_FORMAT, offset,
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
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if (!gst_ringbuffer_is_acquired (sink->ringbuffer))
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goto wrong_state;
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@ -244,8 +244,6 @@ gst_audio_convert_chain (GstPad * pad, GstBuffer * buf)
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* - convert rate and channels
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* - convert back to output format
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*/
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GST_STREAM_LOCK (pad);
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if (!GST_RPAD_CAPS (this->sink)) {
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goto not_negotiated;
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} else if (!GST_RPAD_CAPS (this->src)) {
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@ -262,8 +260,6 @@ gst_audio_convert_chain (GstPad * pad, GstBuffer * buf)
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ret = gst_pad_push (this->src, buf);
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GST_STREAM_UNLOCK (pad);
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return ret;
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not_negotiated:
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@ -271,7 +267,6 @@ not_negotiated:
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GST_ELEMENT_ERROR (this, CORE, NEGOTIATION, (NULL),
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("Pad not negotiated before chain function was called"));
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gst_buffer_unref (buf);
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GST_STREAM_UNLOCK (pad);
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return GST_FLOW_NOT_NEGOTIATED;
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}
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no_format:
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@ -279,7 +274,6 @@ no_format:
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GST_ELEMENT_ERROR (this, CORE, NEGOTIATION, (NULL),
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("Could not negotiate format"));
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gst_buffer_unref (buf);
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GST_STREAM_UNLOCK (pad);
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return GST_FLOW_ERROR;
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}
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}
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@ -648,12 +642,10 @@ gst_audio_convert_change_state (GstElement * element)
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switch (transition) {
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case GST_STATE_PAUSED_TO_READY:
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GST_STREAM_LOCK (this->sink);
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this->convert_internal = NULL;
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gst_audio_convert_unset_matrix (this);
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gst_caps_replace (&GST_RPAD_CAPS (this->sink), NULL);
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gst_caps_replace (&GST_RPAD_CAPS (this->src), NULL);
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GST_STREAM_UNLOCK (this->sink);
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break;
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default:
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break;
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@ -618,12 +618,8 @@ gst_subparse_chain (GstPad * sinkpad, GstBuffer * buf)
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GST_DEBUG ("gst_subparse_chain");
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self = GST_SUBPARSE (GST_OBJECT_PARENT (sinkpad));
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GST_STREAM_LOCK (sinkpad);
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ret = handle_buffer (self, buf);
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GST_STREAM_UNLOCK (sinkpad);
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return ret;
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}
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