/* GStreamer * Copyright (C) 2008 Wim Taymans <wim.taymans@gmail.com> * * gstaudioringbuffer.c: * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:element-audioringbuffer * @short_description: Asynchronous audio ringbuffer. * * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include <string.h> #include <glib/gstdio.h> #include <gst/gst.h> #include <gst/gst-i18n-plugin.h> #include <gst/audio/gstringbuffer.h> static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS_ANY); static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS_ANY); GST_DEBUG_CATEGORY_STATIC (audioringbuffer_debug); #define GST_CAT_DEFAULT (audioringbuffer_debug) enum { LAST_SIGNAL }; #define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND) #define DEFAULT_SEGMENT_TIME ((10 * GST_MSECOND) / GST_USECOND) enum { PROP_0, PROP_BUFFER_TIME, PROP_SEGMENT_TIME }; #define GST_TYPE_AUDIO_RINGBUFFER \ (gst_audio_ringbuffer_get_type()) #define GST_AUDIO_RINGBUFFER(obj) \ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RINGBUFFER,GstAudioRingbuffer)) #define GST_AUDIO_RINGBUFFER_CLASS(klass) \ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RINGBUFFER,GstAudioRingbufferClass)) #define GST_IS_AUDIO_RINGBUFFER(obj) \ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RINGBUFFER)) #define GST_IS_AUDIO_RINGBUFFER_CLASS(klass) \ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RINGBUFFER)) #define GST_AUDIO_RINGBUFFER_CAST(obj) \ ((GstAudioRingbuffer *)(obj)) static GType gst_audio_ringbuffer_get_type (void); typedef struct _GstAudioRingbuffer GstAudioRingbuffer; typedef struct _GstAudioRingbufferClass GstAudioRingbufferClass; typedef struct _GstIntRingBuffer GstIntRingBuffer; typedef struct _GstIntRingBufferClass GstIntRingBufferClass; struct _GstAudioRingbuffer { GstElement element; /*< private > */ GstPad *sinkpad; GstPad *srcpad; gboolean pushing; gboolean pulling; /* segments to keep track of timestamps */ GstSegment sink_segment; GstSegment src_segment; /* flowreturn when srcpad is paused */ gboolean is_eos; gboolean flushing; gboolean waiting; GCond *cond; GstRingBuffer *buffer; GstClockTime buffer_time; GstClockTime segment_time; guint64 next_sample; guint64 last_align; }; struct _GstAudioRingbufferClass { GstElementClass parent_class; }; #define GST_TYPE_INT_RING_BUFFER (gst_int_ring_buffer_get_type()) #define GST_INT_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_INT_RING_BUFFER,GstIntRingBuffer)) #define GST_INT_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_INT_RING_BUFFER,GstIntRingBufferClass)) #define GST_INT_RING_BUFFER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_INT_RING_BUFFER, GstIntRingBufferClass)) #define GST_INT_RING_BUFFER_CAST(obj) ((GstIntRingBuffer *)obj) #define GST_IS_INT_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_INT_RING_BUFFER)) #define GST_IS_INT_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_INT_RING_BUFFER)) struct _GstIntRingBuffer { GstRingBuffer object; }; struct _GstIntRingBufferClass { GstRingBufferClass parent_class; }; GST_BOILERPLATE (GstIntRingBuffer, gst_int_ring_buffer, GstRingBuffer, GST_TYPE_RING_BUFFER); static gboolean gst_int_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) { spec->seglatency = spec->segtotal; buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize); memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data)); return TRUE; } static gboolean gst_int_ring_buffer_release (GstRingBuffer * buf) { gst_buffer_unref (buf->data); buf->data = NULL; return TRUE; } static gboolean gst_int_ring_buffer_start (GstRingBuffer * buf) { GstAudioRingbuffer *ringbuffer; ringbuffer = GST_AUDIO_RINGBUFFER (GST_OBJECT_PARENT (buf)); GST_OBJECT_LOCK (ringbuffer); if (G_UNLIKELY (ringbuffer->waiting)) { ringbuffer->waiting = FALSE; GST_DEBUG_OBJECT (ringbuffer, "start, sending signal"); g_cond_broadcast (ringbuffer->cond); } GST_OBJECT_UNLOCK (ringbuffer); return TRUE; } static void gst_int_ring_buffer_base_init (gpointer klass) { } static void gst_int_ring_buffer_class_init (GstIntRingBufferClass * klass) { GstRingBufferClass *gstringbuffer_class; gstringbuffer_class = (GstRingBufferClass *) klass; gstringbuffer_class->acquire = GST_DEBUG_FUNCPTR (gst_int_ring_buffer_acquire); gstringbuffer_class->release = GST_DEBUG_FUNCPTR (gst_int_ring_buffer_release); gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_int_ring_buffer_start); } static void gst_int_ring_buffer_init (GstIntRingBuffer * buff, GstIntRingBufferClass * g_class) { } static GstRingBuffer * gst_int_ring_buffer_new (void) { GstRingBuffer *res; res = g_object_new (GST_TYPE_INT_RING_BUFFER, NULL); return res; } /* can't use boilerplate as we need to register with Queue2 to avoid conflicts * with ringbuffer in core elements */ static void gst_audio_ringbuffer_class_init (GstAudioRingbufferClass * klass); static void gst_audio_ringbuffer_init (GstAudioRingbuffer * ringbuffer, GstAudioRingbufferClass * g_class); static GstElementClass *elem_parent_class; static GType gst_audio_ringbuffer_get_type (void) { static GType gst_audio_ringbuffer_type = 0; if (!gst_audio_ringbuffer_type) { static const GTypeInfo gst_audio_ringbuffer_info = { sizeof (GstAudioRingbufferClass), NULL, NULL, (GClassInitFunc) gst_audio_ringbuffer_class_init, NULL, NULL, sizeof (GstAudioRingbuffer), 0, (GInstanceInitFunc) gst_audio_ringbuffer_init, NULL }; gst_audio_ringbuffer_type = g_type_register_static (GST_TYPE_ELEMENT, "GstAudioRingbuffer", &gst_audio_ringbuffer_info, 0); } return gst_audio_ringbuffer_type; } static void gst_audio_ringbuffer_finalize (GObject * object); static void gst_audio_ringbuffer_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audio_ringbuffer_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstFlowReturn gst_audio_ringbuffer_chain (GstPad * pad, GstBuffer * buffer); static GstFlowReturn gst_audio_ringbuffer_bufferalloc (GstPad * pad, guint64 offset, guint size, GstCaps * caps, GstBuffer ** buf); static gboolean gst_audio_ringbuffer_handle_sink_event (GstPad * pad, GstEvent * event); static gboolean gst_audio_ringbuffer_handle_src_event (GstPad * pad, GstEvent * event); static gboolean gst_audio_ringbuffer_handle_src_query (GstPad * pad, GstQuery * query); static GstCaps *gst_audio_ringbuffer_getcaps (GstPad * pad); static gboolean gst_audio_ringbuffer_setcaps (GstPad * pad, GstCaps * caps); static GstFlowReturn gst_audio_ringbuffer_get_range (GstPad * pad, guint64 offset, guint length, GstBuffer ** buffer); static gboolean gst_audio_ringbuffer_src_checkgetrange_function (GstPad * pad); static gboolean gst_audio_ringbuffer_src_activate_pull (GstPad * pad, gboolean active); static gboolean gst_audio_ringbuffer_src_activate_push (GstPad * pad, gboolean active); static gboolean gst_audio_ringbuffer_sink_activate_push (GstPad * pad, gboolean active); static GstStateChangeReturn gst_audio_ringbuffer_change_state (GstElement * element, GstStateChange transition); /* static guint gst_audio_ringbuffer_signals[LAST_SIGNAL] = { 0 }; */ static void gst_audio_ringbuffer_class_init (GstAudioRingbufferClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); elem_parent_class = g_type_class_peek_parent (klass); gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_set_property); gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_get_property); g_object_class_install_property (gobject_class, PROP_BUFFER_TIME, g_param_spec_int64 ("buffer-time", "Buffer Time", "Size of audio buffer in nanoseconds", 1, G_MAXINT64, DEFAULT_BUFFER_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_SEGMENT_TIME, g_param_spec_int64 ("segment-time", "Segment Time", "Audio segment duration in nanoseconds", 1, G_MAXINT64, DEFAULT_SEGMENT_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&srctemplate)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&sinktemplate)); gst_element_class_set_static_metadata (gstelement_class, "AudioRingbuffer", "Generic", "Asynchronous Audio ringbuffer", "Wim Taymans <wim.taymans@gmail.com>"); /* set several parent class virtual functions */ gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_finalize); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_change_state); } static void gst_audio_ringbuffer_init (GstAudioRingbuffer * ringbuffer, GstAudioRingbufferClass * g_class) { ringbuffer->sinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink"); gst_pad_set_chain_function (ringbuffer->sinkpad, GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_chain)); gst_pad_set_activatepush_function (ringbuffer->sinkpad, GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_sink_activate_push)); gst_pad_set_event_function (ringbuffer->sinkpad, GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_handle_sink_event)); gst_pad_set_getcaps_function (ringbuffer->sinkpad, GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_getcaps)); gst_pad_set_setcaps_function (ringbuffer->sinkpad, GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_setcaps)); gst_pad_set_bufferalloc_function (ringbuffer->sinkpad, GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_bufferalloc)); gst_element_add_pad (GST_ELEMENT (ringbuffer), ringbuffer->sinkpad); ringbuffer->srcpad = gst_pad_new_from_static_template (&srctemplate, "src"); gst_pad_set_activatepull_function (ringbuffer->srcpad, GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_src_activate_pull)); gst_pad_set_activatepush_function (ringbuffer->srcpad, GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_src_activate_push)); gst_pad_set_getrange_function (ringbuffer->srcpad, GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_get_range)); gst_pad_set_checkgetrange_function (ringbuffer->srcpad, GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_src_checkgetrange_function)); gst_pad_set_getcaps_function (ringbuffer->srcpad, GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_getcaps)); gst_pad_set_event_function (ringbuffer->srcpad, GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_handle_src_event)); gst_pad_set_query_function (ringbuffer->srcpad, GST_DEBUG_FUNCPTR (gst_audio_ringbuffer_handle_src_query)); gst_element_add_pad (GST_ELEMENT (ringbuffer), ringbuffer->srcpad); gst_segment_init (&ringbuffer->sink_segment, GST_FORMAT_TIME); ringbuffer->cond = g_cond_new (); ringbuffer->is_eos = FALSE; ringbuffer->buffer_time = DEFAULT_BUFFER_TIME; ringbuffer->segment_time = DEFAULT_SEGMENT_TIME; GST_DEBUG_OBJECT (ringbuffer, "initialized ringbuffer's not_empty & not_full conditions"); } /* called only once, as opposed to dispose */ static void gst_audio_ringbuffer_finalize (GObject * object) { GstAudioRingbuffer *ringbuffer = GST_AUDIO_RINGBUFFER (object); GST_DEBUG_OBJECT (ringbuffer, "finalizing ringbuffer"); g_cond_free (ringbuffer->cond); G_OBJECT_CLASS (elem_parent_class)->finalize (object); } static GstCaps * gst_audio_ringbuffer_getcaps (GstPad * pad) { GstAudioRingbuffer *ringbuffer; GstPad *otherpad; GstCaps *result; ringbuffer = GST_AUDIO_RINGBUFFER (GST_PAD_PARENT (pad)); otherpad = (pad == ringbuffer->srcpad ? ringbuffer->sinkpad : ringbuffer->srcpad); result = gst_pad_peer_get_caps (otherpad); if (result == NULL) result = gst_caps_new_any (); return result; } static gboolean gst_audio_ringbuffer_setcaps (GstPad * pad, GstCaps * caps) { GstAudioRingbuffer *ringbuffer; GstRingBufferSpec *spec; ringbuffer = GST_AUDIO_RINGBUFFER (GST_PAD_PARENT (pad)); if (!ringbuffer->buffer) return FALSE; spec = &ringbuffer->buffer->spec; GST_DEBUG_OBJECT (ringbuffer, "release old ringbuffer"); /* release old ringbuffer */ gst_ring_buffer_activate (ringbuffer->buffer, FALSE); gst_ring_buffer_release (ringbuffer->buffer); GST_DEBUG_OBJECT (ringbuffer, "parse caps"); spec->buffer_time = ringbuffer->buffer_time; spec->latency_time = ringbuffer->segment_time; /* parse new caps */ if (!gst_ring_buffer_parse_caps (spec, caps)) goto parse_error; gst_ring_buffer_debug_spec_buff (spec); GST_DEBUG_OBJECT (ringbuffer, "acquire ringbuffer"); if (!gst_ring_buffer_acquire (ringbuffer->buffer, spec)) goto acquire_error; GST_DEBUG_OBJECT (ringbuffer, "activate ringbuffer"); gst_ring_buffer_activate (ringbuffer->buffer, TRUE); /* calculate actual latency and buffer times. * FIXME: In 0.11, store the latency_time internally in ns */ spec->latency_time = gst_util_uint64_scale (spec->segsize, (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample); spec->buffer_time = spec->segtotal * spec->latency_time; gst_ring_buffer_debug_spec_buff (spec); return TRUE; /* ERRORS */ parse_error: { GST_DEBUG_OBJECT (ringbuffer, "could not parse caps"); GST_ELEMENT_ERROR (ringbuffer, STREAM, FORMAT, (NULL), ("cannot parse audio format.")); return FALSE; } acquire_error: { GST_DEBUG_OBJECT (ringbuffer, "could not acquire ringbuffer"); return FALSE; } } static GstFlowReturn gst_audio_ringbuffer_bufferalloc (GstPad * pad, guint64 offset, guint size, GstCaps * caps, GstBuffer ** buf) { GstAudioRingbuffer *ringbuffer; GstFlowReturn result; ringbuffer = GST_AUDIO_RINGBUFFER (GST_PAD_PARENT (pad)); /* Forward to src pad, without setting caps on the src pad */ result = gst_pad_alloc_buffer (ringbuffer->srcpad, offset, size, caps, buf); return result; } static gboolean gst_audio_ringbuffer_handle_sink_event (GstPad * pad, GstEvent * event) { GstAudioRingbuffer *ringbuffer; gboolean forward; ringbuffer = GST_AUDIO_RINGBUFFER (GST_OBJECT_PARENT (pad)); forward = ringbuffer->pushing || ringbuffer->pulling; switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_START: { GST_LOG_OBJECT (ringbuffer, "received flush start event"); break; } case GST_EVENT_FLUSH_STOP: { ringbuffer->is_eos = FALSE; GST_LOG_OBJECT (ringbuffer, "received flush stop event"); break; } case GST_EVENT_NEWSEGMENT: { gboolean update; gdouble rate, arate; GstFormat format; gint64 start, stop, time; gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format, &start, &stop, &time); gst_segment_set_newsegment_full (&ringbuffer->sink_segment, update, rate, arate, format, start, stop, time); break; } case GST_EVENT_EOS: ringbuffer->is_eos = TRUE; break; default: break; } if (forward) { gst_pad_push_event (ringbuffer->srcpad, event); } else { if (event) gst_event_unref (event); } return TRUE; } #define DIFF_TOLERANCE 2 static GstFlowReturn gst_audio_ringbuffer_render (GstAudioRingbuffer * ringbuffer, GstBuffer * buf) { GstRingBuffer *rbuf; gint bps, accum; guint size; guint samples, written, out_samples; gint64 diff, align, ctime, cstop; guint8 *data; guint64 in_offset; GstClockTime time, stop, render_start, render_stop, sample_offset; gboolean align_next; rbuf = ringbuffer->buffer; /* can't do anything when we don't have the device */ if (G_UNLIKELY (!gst_ring_buffer_is_acquired (rbuf))) goto wrong_state; bps = rbuf->spec.bytes_per_sample; size = GST_BUFFER_SIZE (buf); if (G_UNLIKELY (size % bps) != 0) goto wrong_size; samples = size / bps; out_samples = samples; in_offset = GST_BUFFER_OFFSET (buf); time = GST_BUFFER_TIMESTAMP (buf); GST_DEBUG_OBJECT (ringbuffer, "time %" GST_TIME_FORMAT ", offset %llu, start %" GST_TIME_FORMAT ", samples %u", GST_TIME_ARGS (time), in_offset, GST_TIME_ARGS (ringbuffer->sink_segment.start), samples); data = GST_BUFFER_DATA (buf); stop = time + gst_util_uint64_scale_int (samples, GST_SECOND, rbuf->spec.rate); if (!gst_segment_clip (&ringbuffer->sink_segment, GST_FORMAT_TIME, time, stop, &ctime, &cstop)) goto out_of_segment; /* see if some clipping happened */ diff = ctime - time; if (diff > 0) { /* bring clipped time to samples */ diff = gst_util_uint64_scale_int (diff, rbuf->spec.rate, GST_SECOND); GST_DEBUG_OBJECT (ringbuffer, "clipping start to %" GST_TIME_FORMAT " %" G_GUINT64_FORMAT " samples", GST_TIME_ARGS (ctime), diff); samples -= diff; data += diff * bps; time = ctime; } diff = stop - cstop; if (diff > 0) { /* bring clipped time to samples */ diff = gst_util_uint64_scale_int (diff, rbuf->spec.rate, GST_SECOND); GST_DEBUG_OBJECT (ringbuffer, "clipping stop to %" GST_TIME_FORMAT " %" G_GUINT64_FORMAT " samples", GST_TIME_ARGS (cstop), diff); samples -= diff; stop = cstop; } /* bring buffer start and stop times to running time */ render_start = gst_segment_to_running_time (&ringbuffer->sink_segment, GST_FORMAT_TIME, time); render_stop = gst_segment_to_running_time (&ringbuffer->sink_segment, GST_FORMAT_TIME, stop); GST_DEBUG_OBJECT (ringbuffer, "running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT, GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop)); /* and bring the time to the rate corrected offset in the buffer */ render_start = gst_util_uint64_scale_int (render_start, rbuf->spec.rate, GST_SECOND); render_stop = gst_util_uint64_scale_int (render_stop, rbuf->spec.rate, GST_SECOND); /* positive playback rate, first sample is render_start, negative rate, first * sample is render_stop. When no rate conversion is active, render exactly * the amount of input samples to avoid aligning to rounding errors. */ if (ringbuffer->sink_segment.rate >= 0.0) { sample_offset = render_start; if (ringbuffer->sink_segment.rate == 1.0) render_stop = sample_offset + samples; } else { sample_offset = render_stop; if (ringbuffer->sink_segment.rate == -1.0) render_start = sample_offset + samples; } /* always resync after a discont */ if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) { GST_DEBUG_OBJECT (ringbuffer, "resync after discont"); goto no_align; } /* resync when we don't know what to align the sample with */ if (G_UNLIKELY (ringbuffer->next_sample == -1)) { GST_DEBUG_OBJECT (ringbuffer, "no align possible: no previous sample position known"); goto no_align; } /* now try to align the sample to the previous one, first see how big the * difference is. */ if (sample_offset >= ringbuffer->next_sample) diff = sample_offset - ringbuffer->next_sample; else diff = ringbuffer->next_sample - sample_offset; /* we tollerate half a second diff before we start resyncing. This * should be enough to compensate for various rounding errors in the timestamp * and sample offset position. We always resync if we got a discont anyway and * non-discont should be aligned by definition. */ if (G_LIKELY (diff < rbuf->spec.rate / DIFF_TOLERANCE)) { /* calc align with previous sample */ align = ringbuffer->next_sample - sample_offset; GST_DEBUG_OBJECT (ringbuffer, "align with prev sample, ABS (%" G_GINT64_FORMAT ") < %d", align, rbuf->spec.rate / DIFF_TOLERANCE); } else { /* bring sample diff to seconds for error message */ diff = gst_util_uint64_scale_int (diff, GST_SECOND, rbuf->spec.rate); /* timestamps drifted apart from previous samples too much, we need to * resync. We log this as an element warning. */ GST_ELEMENT_WARNING (ringbuffer, CORE, CLOCK, ("Compensating for audio synchronisation problems"), ("Unexpected discontinuity in audio timestamps of more " "than half a second (%" GST_TIME_FORMAT "), resyncing", GST_TIME_ARGS (diff))); align = 0; } ringbuffer->last_align = align; /* apply alignment */ render_start += align; render_stop += align; no_align: /* number of target samples is difference between start and stop */ out_samples = render_stop - render_start; /* we render the first or last sample first, depending on the rate */ if (ringbuffer->sink_segment.rate >= 0.0) sample_offset = render_start; else sample_offset = render_stop; GST_DEBUG_OBJECT (ringbuffer, "rendering at %" G_GUINT64_FORMAT " %d/%d", sample_offset, samples, out_samples); /* we need to accumulate over different runs for when we get interrupted */ accum = 0; align_next = TRUE; do { written = gst_ring_buffer_commit_full (rbuf, &sample_offset, data, samples, out_samples, &accum); GST_DEBUG_OBJECT (ringbuffer, "wrote %u of %u", written, samples); /* if we wrote all, we're done */ if (written == samples) break; GST_OBJECT_LOCK (ringbuffer); if (ringbuffer->flushing) goto flushing; GST_OBJECT_UNLOCK (ringbuffer); /* if we got interrupted, we cannot assume that the next sample should * be aligned to this one */ align_next = FALSE; samples -= written; data += written * bps; } while (TRUE); if (align_next) ringbuffer->next_sample = sample_offset; else ringbuffer->next_sample = -1; GST_DEBUG_OBJECT (ringbuffer, "next sample expected at %" G_GUINT64_FORMAT, ringbuffer->next_sample); if (GST_CLOCK_TIME_IS_VALID (stop) && stop >= ringbuffer->sink_segment.stop) { GST_DEBUG_OBJECT (ringbuffer, "start playback because we are at the end of segment"); gst_ring_buffer_start (rbuf); } return GST_FLOW_OK; /* SPECIAL cases */ out_of_segment: { GST_DEBUG_OBJECT (ringbuffer, "dropping sample out of segment time %" GST_TIME_FORMAT ", start %" GST_TIME_FORMAT, GST_TIME_ARGS (time), GST_TIME_ARGS (ringbuffer->sink_segment.start)); return GST_FLOW_OK; } /* ERRORS */ wrong_state: { GST_DEBUG_OBJECT (ringbuffer, "ringbuffer not negotiated"); GST_ELEMENT_ERROR (ringbuffer, STREAM, FORMAT, (NULL), ("ringbuffer not negotiated.")); return GST_FLOW_NOT_NEGOTIATED; } wrong_size: { GST_DEBUG_OBJECT (ringbuffer, "wrong size"); GST_ELEMENT_ERROR (ringbuffer, STREAM, WRONG_TYPE, (NULL), ("ringbuffer received buffer of wrong size.")); return GST_FLOW_ERROR; } flushing: { GST_DEBUG_OBJECT (ringbuffer, "ringbuffer is flushing"); GST_OBJECT_UNLOCK (ringbuffer); return GST_FLOW_FLUSHING; } } static GstFlowReturn gst_audio_ringbuffer_chain (GstPad * pad, GstBuffer * buffer) { GstFlowReturn res; GstAudioRingbuffer *ringbuffer; ringbuffer = GST_AUDIO_RINGBUFFER (GST_OBJECT_PARENT (pad)); if (ringbuffer->pushing) { GST_DEBUG_OBJECT (ringbuffer, "proxy pushing buffer"); res = gst_pad_push (ringbuffer->srcpad, buffer); } else { GST_DEBUG_OBJECT (ringbuffer, "render buffer in ringbuffer"); res = gst_audio_ringbuffer_render (ringbuffer, buffer); } return res; } static gboolean gst_audio_ringbuffer_handle_src_event (GstPad * pad, GstEvent * event) { gboolean res = TRUE; GstAudioRingbuffer *ringbuffer = GST_AUDIO_RINGBUFFER (GST_PAD_PARENT (pad)); /* just forward upstream */ res = gst_pad_push_event (ringbuffer->sinkpad, event); return res; } static gboolean gst_audio_ringbuffer_handle_src_query (GstPad * pad, GstQuery * query) { GstAudioRingbuffer *ringbuffer; ringbuffer = GST_AUDIO_RINGBUFFER (GST_PAD_PARENT (pad)); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_POSITION: break; case GST_QUERY_DURATION: break; case GST_QUERY_BUFFERING: break; default: break; } return TRUE; } static GstFlowReturn gst_audio_ringbuffer_get_range (GstPad * pad, guint64 offset, guint length, GstBuffer ** buffer) { GstAudioRingbuffer *ringbuffer; GstRingBuffer *rbuf; GstFlowReturn ret; ringbuffer = GST_AUDIO_RINGBUFFER_CAST (gst_pad_get_parent (pad)); rbuf = ringbuffer->buffer; if (ringbuffer->pulling) { GST_DEBUG_OBJECT (ringbuffer, "proxy pulling range"); ret = gst_pad_pull_range (ringbuffer->sinkpad, offset, length, buffer); } else { guint8 *data; guint len; guint64 sample; gint bps, segsize, segtotal, sps; gint sampleslen, segdone; gint readseg, sampleoff; guint8 *dest; GST_DEBUG_OBJECT (ringbuffer, "pulling data at %" G_GUINT64_FORMAT ", length %u", offset, length); if (offset != ringbuffer->src_segment.last_stop) { GST_DEBUG_OBJECT (ringbuffer, "expected offset %" G_GINT64_FORMAT, ringbuffer->src_segment.last_stop); } /* first wait till we have something in the ringbuffer and it * is running */ GST_OBJECT_LOCK (ringbuffer); if (ringbuffer->flushing) goto flushing; while (ringbuffer->waiting) { GST_DEBUG_OBJECT (ringbuffer, "waiting for unlock"); g_cond_wait (ringbuffer->cond, GST_OBJECT_GET_LOCK (ringbuffer)); GST_DEBUG_OBJECT (ringbuffer, "unlocked"); if (ringbuffer->flushing) goto flushing; } GST_OBJECT_UNLOCK (ringbuffer); bps = rbuf->spec.bytes_per_sample; if (G_UNLIKELY (length % bps) != 0) goto wrong_size; segsize = rbuf->spec.segsize; segtotal = rbuf->spec.segtotal; sps = rbuf->samples_per_seg; dest = GST_BUFFER_DATA (rbuf->data); sample = offset / bps; len = length / bps; *buffer = gst_buffer_new_and_alloc (length); data = GST_BUFFER_DATA (*buffer); while (len) { gint diff; /* figure out the segment and the offset inside the segment where * the sample should be read from. */ readseg = sample / sps; sampleoff = (sample % sps); segdone = g_atomic_int_get (&rbuf->segdone) - rbuf->segbase; diff = readseg - segdone; /* we can read now */ readseg = readseg % segtotal; sampleslen = MIN (sps - sampleoff, len); GST_DEBUG_OBJECT (ringbuffer, "read @%p seg %d, off %d, sampleslen %d, diff %d", dest + readseg * segsize, readseg, sampleoff, sampleslen, diff); memcpy (data, dest + (readseg * segsize) + (sampleoff * bps), (sampleslen * bps)); if (diff > 0) gst_ring_buffer_advance (rbuf, diff); len -= sampleslen; sample += sampleslen; data += sampleslen * bps; } ringbuffer->src_segment.last_stop += length; ret = GST_FLOW_OK; } gst_object_unref (ringbuffer); return ret; /* ERRORS */ flushing: { GST_DEBUG_OBJECT (ringbuffer, "we are flushing"); GST_OBJECT_UNLOCK (ringbuffer); gst_object_unref (ringbuffer); return GST_FLOW_FLUSHING; } wrong_size: { GST_DEBUG_OBJECT (ringbuffer, "wrong size"); GST_ELEMENT_ERROR (ringbuffer, STREAM, WRONG_TYPE, (NULL), ("asked to pull buffer of wrong size.")); return GST_FLOW_ERROR; } } static gboolean gst_audio_ringbuffer_src_checkgetrange_function (GstPad * pad) { gboolean ret; /* we can always operate in pull mode */ ret = TRUE; return ret; } /* sink currently only operates in push mode */ static gboolean gst_audio_ringbuffer_sink_activate_push (GstPad * pad, gboolean active) { gboolean result = TRUE; GstAudioRingbuffer *ringbuffer; ringbuffer = GST_AUDIO_RINGBUFFER (gst_pad_get_parent (pad)); if (active) { GST_DEBUG_OBJECT (ringbuffer, "activating push mode"); ringbuffer->is_eos = FALSE; ringbuffer->pulling = FALSE; } else { /* unblock chain function */ GST_DEBUG_OBJECT (ringbuffer, "deactivating push mode"); ringbuffer->pulling = FALSE; } gst_object_unref (ringbuffer); return result; } /* src operating in push mode, we will proxy the push from upstream, basically * acting as a passthrough element. */ static gboolean gst_audio_ringbuffer_src_activate_push (GstPad * pad, gboolean active) { gboolean result = FALSE; GstAudioRingbuffer *ringbuffer; ringbuffer = GST_AUDIO_RINGBUFFER (gst_pad_get_parent (pad)); if (active) { GST_DEBUG_OBJECT (ringbuffer, "activating push mode"); ringbuffer->is_eos = FALSE; ringbuffer->pushing = TRUE; ringbuffer->pulling = FALSE; result = TRUE; } else { GST_DEBUG_OBJECT (ringbuffer, "deactivating push mode"); ringbuffer->pushing = FALSE; ringbuffer->pulling = FALSE; result = TRUE; } gst_object_unref (ringbuffer); return result; } /* pull mode, downstream will call our getrange function */ static gboolean gst_audio_ringbuffer_src_activate_pull (GstPad * pad, gboolean active) { gboolean result; GstAudioRingbuffer *ringbuffer; ringbuffer = GST_AUDIO_RINGBUFFER (gst_pad_get_parent (pad)); if (active) { GST_DEBUG_OBJECT (ringbuffer, "activating pull mode"); /* try to activate upstream in pull mode as well. If it fails, no problems, * we'll be activated in push mode. Remember that we are pulling-through */ ringbuffer->pulling = gst_pad_activate_pull (ringbuffer->sinkpad, active); ringbuffer->is_eos = FALSE; ringbuffer->waiting = TRUE; ringbuffer->flushing = FALSE; gst_segment_init (&ringbuffer->src_segment, GST_FORMAT_BYTES); result = TRUE; } else { GST_DEBUG_OBJECT (ringbuffer, "deactivating pull mode"); if (ringbuffer->pulling) gst_pad_activate_pull (ringbuffer->sinkpad, active); ringbuffer->pulling = FALSE; ringbuffer->waiting = FALSE; ringbuffer->flushing = TRUE; result = TRUE; } gst_object_unref (ringbuffer); return result; } static GstStateChangeReturn gst_audio_ringbuffer_change_state (GstElement * element, GstStateChange transition) { GstAudioRingbuffer *ringbuffer; GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; ringbuffer = GST_AUDIO_RINGBUFFER (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: if (ringbuffer->buffer == NULL) { ringbuffer->buffer = gst_int_ring_buffer_new (); gst_object_set_parent (GST_OBJECT (ringbuffer->buffer), GST_OBJECT (ringbuffer)); gst_ring_buffer_open_device (ringbuffer->buffer); } break; case GST_STATE_CHANGE_READY_TO_PAUSED: ringbuffer->next_sample = -1; ringbuffer->last_align = -1; gst_ring_buffer_set_flushing (ringbuffer->buffer, FALSE); gst_ring_buffer_may_start (ringbuffer->buffer, TRUE); break; case GST_STATE_CHANGE_PAUSED_TO_READY: GST_OBJECT_LOCK (ringbuffer); ringbuffer->flushing = TRUE; ringbuffer->waiting = FALSE; g_cond_broadcast (ringbuffer->cond); GST_OBJECT_UNLOCK (ringbuffer); gst_ring_buffer_set_flushing (ringbuffer->buffer, TRUE); gst_ring_buffer_may_start (ringbuffer->buffer, FALSE); break; default: break; } ret = GST_ELEMENT_CLASS (elem_parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: gst_ring_buffer_activate (ringbuffer->buffer, FALSE); gst_ring_buffer_release (ringbuffer->buffer); break; case GST_STATE_CHANGE_READY_TO_NULL: if (ringbuffer->buffer != NULL) { gst_ring_buffer_close_device (ringbuffer->buffer); gst_object_unparent (GST_OBJECT (ringbuffer->buffer)); ringbuffer->buffer = NULL; } break; default: break; } return ret; } static void gst_audio_ringbuffer_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioRingbuffer *ringbuffer; ringbuffer = GST_AUDIO_RINGBUFFER (object); switch (prop_id) { case PROP_BUFFER_TIME: ringbuffer->buffer_time = g_value_get_int64 (value); break; case PROP_SEGMENT_TIME: ringbuffer->segment_time = g_value_get_int64 (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audio_ringbuffer_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioRingbuffer *ringbuffer; ringbuffer = GST_AUDIO_RINGBUFFER (object); switch (prop_id) { case PROP_BUFFER_TIME: g_value_set_int64 (value, ringbuffer->buffer_time); break; case PROP_SEGMENT_TIME: g_value_set_int64 (value, ringbuffer->segment_time); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static gboolean plugin_init (GstPlugin * plugin) { GST_DEBUG_CATEGORY_INIT (audioringbuffer_debug, "audioringbuffer", 0, "Audio ringbuffer element"); #ifdef ENABLE_NLS GST_DEBUG ("binding text domain %s to locale dir %s", GETTEXT_PACKAGE, LOCALEDIR); bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR); bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8"); #endif /* ENABLE_NLS */ return gst_element_register (plugin, "audioringbuffer", GST_RANK_NONE, GST_TYPE_AUDIO_RINGBUFFER); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, audioringbuffer, "An audio ringbuffer", plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)