diff options
Diffstat (limited to 'gst/audiobuffer/gstaudioringbuffer.c')
| -rw-r--r-- | gst/audiobuffer/gstaudioringbuffer.c | 1178 |
1 files changed, 0 insertions, 1178 deletions
diff --git a/gst/audiobuffer/gstaudioringbuffer.c b/gst/audiobuffer/gstaudioringbuffer.c deleted file mode 100644 index 9c43402df..000000000 --- a/gst/audiobuffer/gstaudioringbuffer.c +++ /dev/null @@ -1,1178 +0,0 @@ -/* 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_static_pad_template (gstelement_class, &srctemplate); - gst_element_class_add_static_pad_template (gstelement_class, &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) |