diff options
Diffstat (limited to 'gst-libs/gst/audio/gstbaseaudiosrc.c')
| -rw-r--r-- | gst-libs/gst/audio/gstbaseaudiosrc.c | 1125 |
1 files changed, 0 insertions, 1125 deletions
diff --git a/gst-libs/gst/audio/gstbaseaudiosrc.c b/gst-libs/gst/audio/gstbaseaudiosrc.c deleted file mode 100644 index 02ce8ec7..00000000 --- a/gst-libs/gst/audio/gstbaseaudiosrc.c +++ /dev/null @@ -1,1125 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> - * 2005 Wim Taymans <wim@fluendo.com> - * - * gstbaseaudiosrc.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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -/** - * SECTION:gstbaseaudiosrc - * @short_description: Base class for audio sources - * @see_also: #GstAudioSrc, #GstRingBuffer. - * - * This is the base class for audio sources. Subclasses need to implement the - * ::create_ringbuffer vmethod. This base class will then take care of - * reading samples from the ringbuffer, synchronisation and flushing. - * - * Last reviewed on 2006-09-27 (0.10.12) - */ - -#ifdef HAVE_CONFIG_H -# include "config.h" -#endif - -#include <string.h> - -#include "gstbaseaudiosrc.h" - -#include "gst/gst-i18n-plugin.h" - -GST_DEBUG_CATEGORY_STATIC (gst_base_audio_src_debug); -#define GST_CAT_DEFAULT gst_base_audio_src_debug - -GType -gst_base_audio_src_slave_method_get_type (void) -{ - static GType slave_method_type = 0; - static const GEnumValue slave_method[] = { - {GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE, "Resampling slaving", "resample"}, - {GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP, "Re-timestamp", "re-timestamp"}, - {GST_BASE_AUDIO_SRC_SLAVE_SKEW, "Skew", "skew"}, - {GST_BASE_AUDIO_SRC_SLAVE_NONE, "No slaving", "none"}, - {0, NULL, NULL}, - }; - - if (!slave_method_type) { - slave_method_type = - g_enum_register_static ("GstBaseAudioSrcSlaveMethod", slave_method); - } - return slave_method_type; -} - -#define GST_BASE_AUDIO_SRC_GET_PRIVATE(obj) \ - (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_SRC, GstBaseAudioSrcPrivate)) - -struct _GstBaseAudioSrcPrivate -{ - gboolean provide_clock; - - /* the clock slaving algorithm in use */ - GstBaseAudioSrcSlaveMethod slave_method; -}; - -/* BaseAudioSrc signals and args */ -enum -{ - /* FILL ME */ - LAST_SIGNAL -}; - -#define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND) -#define DEFAULT_LATENCY_TIME ((10 * GST_MSECOND) / GST_USECOND) -#define DEFAULT_ACTUAL_BUFFER_TIME -1 -#define DEFAULT_ACTUAL_LATENCY_TIME -1 -#define DEFAULT_PROVIDE_CLOCK TRUE -#define DEFAULT_SLAVE_METHOD GST_BASE_AUDIO_SRC_SLAVE_SKEW - -enum -{ - PROP_0, - PROP_BUFFER_TIME, - PROP_LATENCY_TIME, - PROP_ACTUAL_BUFFER_TIME, - PROP_ACTUAL_LATENCY_TIME, - PROP_PROVIDE_CLOCK, - PROP_SLAVE_METHOD, - PROP_LAST -}; - -static void -_do_init (GType type) -{ - GST_DEBUG_CATEGORY_INIT (gst_base_audio_src_debug, "baseaudiosrc", 0, - "baseaudiosrc 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 */ -} - -GST_BOILERPLATE_FULL (GstBaseAudioSrc, gst_base_audio_src, GstPushSrc, - GST_TYPE_PUSH_SRC, _do_init); - -static void gst_base_audio_src_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_base_audio_src_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); -static void gst_base_audio_src_dispose (GObject * object); - -static GstStateChangeReturn gst_base_audio_src_change_state (GstElement * - element, GstStateChange transition); - -static GstClock *gst_base_audio_src_provide_clock (GstElement * elem); -static GstClockTime gst_base_audio_src_get_time (GstClock * clock, - GstBaseAudioSrc * src); - -static GstFlowReturn gst_base_audio_src_create (GstBaseSrc * bsrc, - guint64 offset, guint length, GstBuffer ** buf); -static gboolean gst_base_audio_src_check_get_range (GstBaseSrc * bsrc); - -static gboolean gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event); -static void gst_base_audio_src_get_times (GstBaseSrc * bsrc, - GstBuffer * buffer, GstClockTime * start, GstClockTime * end); -static gboolean gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps); -static gboolean gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query); -static void gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps); - -/* static guint gst_base_audio_src_signals[LAST_SIGNAL] = { 0 }; */ - -static void -gst_base_audio_src_base_init (gpointer g_class) -{ -} - -static void -gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass) -{ - GObjectClass *gobject_class; - GstElementClass *gstelement_class; - GstBaseSrcClass *gstbasesrc_class; - - gobject_class = (GObjectClass *) klass; - gstelement_class = (GstElementClass *) klass; - gstbasesrc_class = (GstBaseSrcClass *) klass; - - g_type_class_add_private (klass, sizeof (GstBaseAudioSrcPrivate)); - - gobject_class->set_property = gst_base_audio_src_set_property; - gobject_class->get_property = gst_base_audio_src_get_property; - gobject_class->dispose = gst_base_audio_src_dispose; - - g_object_class_install_property (gobject_class, PROP_BUFFER_TIME, - g_param_spec_int64 ("buffer-time", "Buffer Time", - "Size of audio buffer in microseconds", 1, - G_MAXINT64, DEFAULT_BUFFER_TIME, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - g_object_class_install_property (gobject_class, PROP_LATENCY_TIME, - g_param_spec_int64 ("latency-time", "Latency Time", - "Audio latency in microseconds", 1, - G_MAXINT64, DEFAULT_LATENCY_TIME, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - /** - * GstBaseAudioSrc:actual-buffer-time: - * - * Actual configured size of audio buffer in microseconds. - * - * Since: 0.10.20 - **/ - g_object_class_install_property (gobject_class, PROP_ACTUAL_BUFFER_TIME, - g_param_spec_int64 ("actual-buffer-time", "Actual Buffer Time", - "Actual configured size of audio buffer in microseconds", - DEFAULT_ACTUAL_BUFFER_TIME, G_MAXINT64, DEFAULT_ACTUAL_BUFFER_TIME, - G_PARAM_READABLE)); - - /** - * GstBaseAudioSrc:actual-latency-time: - * - * Actual configured audio latency in microseconds. - * - * Since: 0.10.20 - **/ - g_object_class_install_property (gobject_class, PROP_ACTUAL_LATENCY_TIME, - g_param_spec_int64 ("actual-latency-time", "Actual Latency Time", - "Actual configured audio latency in microseconds", - DEFAULT_ACTUAL_LATENCY_TIME, G_MAXINT64, DEFAULT_ACTUAL_LATENCY_TIME, - G_PARAM_READABLE)); - - g_object_class_install_property (gobject_class, PROP_PROVIDE_CLOCK, - g_param_spec_boolean ("provide-clock", "Provide Clock", - "Provide a clock to be used as the global pipeline clock", - DEFAULT_PROVIDE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - g_object_class_install_property (gobject_class, PROP_SLAVE_METHOD, - g_param_spec_enum ("slave-method", "Slave Method", - "Algorithm to use to match the rate of the masterclock", - GST_TYPE_BASE_AUDIO_SRC_SLAVE_METHOD, DEFAULT_SLAVE_METHOD, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - gstelement_class->change_state = - GST_DEBUG_FUNCPTR (gst_base_audio_src_change_state); - gstelement_class->provide_clock = - GST_DEBUG_FUNCPTR (gst_base_audio_src_provide_clock); - - gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_src_setcaps); - gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_src_event); - gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_src_query); - gstbasesrc_class->get_times = - GST_DEBUG_FUNCPTR (gst_base_audio_src_get_times); - gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_base_audio_src_create); - gstbasesrc_class->check_get_range = - GST_DEBUG_FUNCPTR (gst_base_audio_src_check_get_range); - gstbasesrc_class->fixate = GST_DEBUG_FUNCPTR (gst_base_audio_src_fixate); - - /* ref class from a thread-safe context to work around missing bit of - * thread-safety in GObject */ - g_type_class_ref (GST_TYPE_AUDIO_CLOCK); - g_type_class_ref (GST_TYPE_RING_BUFFER); -} - -static void -gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc, - GstBaseAudioSrcClass * g_class) -{ - baseaudiosrc->priv = GST_BASE_AUDIO_SRC_GET_PRIVATE (baseaudiosrc); - - baseaudiosrc->buffer_time = DEFAULT_BUFFER_TIME; - baseaudiosrc->latency_time = DEFAULT_LATENCY_TIME; - baseaudiosrc->priv->provide_clock = DEFAULT_PROVIDE_CLOCK; - baseaudiosrc->priv->slave_method = DEFAULT_SLAVE_METHOD; - /* reset blocksize we use latency time to calculate a more useful - * value based on negotiated format. */ - GST_BASE_SRC (baseaudiosrc)->blocksize = 0; - - baseaudiosrc->clock = gst_audio_clock_new ("GstAudioSrcClock", - (GstAudioClockGetTimeFunc) gst_base_audio_src_get_time, baseaudiosrc); - - /* we are always a live source */ - gst_base_src_set_live (GST_BASE_SRC (baseaudiosrc), TRUE); - /* we operate in time */ - gst_base_src_set_format (GST_BASE_SRC (baseaudiosrc), GST_FORMAT_TIME); -} - -static void -gst_base_audio_src_dispose (GObject * object) -{ - GstBaseAudioSrc *src; - - src = GST_BASE_AUDIO_SRC (object); - - GST_OBJECT_LOCK (src); - if (src->clock) - gst_object_unref (src->clock); - src->clock = NULL; - - if (src->ringbuffer) { - gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer)); - src->ringbuffer = NULL; - } - GST_OBJECT_UNLOCK (src); - - G_OBJECT_CLASS (parent_class)->dispose (object); -} - -static GstClock * -gst_base_audio_src_provide_clock (GstElement * elem) -{ - GstBaseAudioSrc *src; - GstClock *clock; - - src = GST_BASE_AUDIO_SRC (elem); - - /* we have no ringbuffer (must be NULL state) */ - if (src->ringbuffer == NULL) - goto wrong_state; - - if (!gst_ring_buffer_is_acquired (src->ringbuffer)) - goto wrong_state; - - GST_OBJECT_LOCK (src); - if (!src->priv->provide_clock) - goto clock_disabled; - - clock = GST_CLOCK_CAST (gst_object_ref (src->clock)); - GST_OBJECT_UNLOCK (src); - - return clock; - - /* ERRORS */ -wrong_state: - { - GST_DEBUG_OBJECT (src, "ringbuffer not acquired"); - return NULL; - } -clock_disabled: - { - GST_DEBUG_OBJECT (src, "clock provide disabled"); - GST_OBJECT_UNLOCK (src); - return NULL; - } -} - -static GstClockTime -gst_base_audio_src_get_time (GstClock * clock, GstBaseAudioSrc * src) -{ - guint64 raw, samples; - guint delay; - GstClockTime result; - - if (G_UNLIKELY (src->ringbuffer == NULL || src->ringbuffer->spec.rate == 0)) - return GST_CLOCK_TIME_NONE; - - raw = samples = gst_ring_buffer_samples_done (src->ringbuffer); - - /* the number of samples not yet processed, this is still queued in the - * device (not yet read for capture). */ - delay = gst_ring_buffer_delay (src->ringbuffer); - - samples += delay; - - result = gst_util_uint64_scale_int (samples, GST_SECOND, - src->ringbuffer->spec.rate); - - GST_DEBUG_OBJECT (src, - "processed samples: raw %" G_GUINT64_FORMAT ", delay %u, real %" - G_GUINT64_FORMAT ", time %" GST_TIME_FORMAT, raw, delay, samples, - GST_TIME_ARGS (result)); - - return result; -} - -static gboolean -gst_base_audio_src_check_get_range (GstBaseSrc * bsrc) -{ - /* we allow limited pull base operation of which the details - * will eventually exposed in an as of yet non-existing query. - * Basically pulling can be done on any number of bytes as long - * as the offset is -1 or sequentially increasing. */ - return TRUE; -} - -/** - * gst_base_audio_src_set_provide_clock: - * @src: a #GstBaseAudioSrc - * @provide: new state - * - * Controls whether @src will provide a clock or not. If @provide is %TRUE, - * gst_element_provide_clock() will return a clock that reflects the datarate - * of @src. If @provide is %FALSE, gst_element_provide_clock() will return NULL. - * - * Since: 0.10.16 - */ -void -gst_base_audio_src_set_provide_clock (GstBaseAudioSrc * src, gboolean provide) -{ - g_return_if_fail (GST_IS_BASE_AUDIO_SRC (src)); - - GST_OBJECT_LOCK (src); - src->priv->provide_clock = provide; - GST_OBJECT_UNLOCK (src); -} - -/** - * gst_base_audio_src_get_provide_clock: - * @src: a #GstBaseAudioSrc - * - * Queries whether @src will provide a clock or not. See also - * gst_base_audio_src_set_provide_clock. - * - * Returns: %TRUE if @src will provide a clock. - * - * Since: 0.10.16 - */ -gboolean -gst_base_audio_src_get_provide_clock (GstBaseAudioSrc * src) -{ - gboolean result; - - g_return_val_if_fail (GST_IS_BASE_AUDIO_SRC (src), FALSE); - - GST_OBJECT_LOCK (src); - result = src->priv->provide_clock; - GST_OBJECT_UNLOCK (src); - - return result; -} - -/** - * gst_base_audio_src_set_slave_method: - * @src: a #GstBaseAudioSrc - * @method: the new slave method - * - * Controls how clock slaving will be performed in @src. - * - * Since: 0.10.20 - */ -void -gst_base_audio_src_set_slave_method (GstBaseAudioSrc * src, - GstBaseAudioSrcSlaveMethod method) -{ - g_return_if_fail (GST_IS_BASE_AUDIO_SRC (src)); - - GST_OBJECT_LOCK (src); - src->priv->slave_method = method; - GST_OBJECT_UNLOCK (src); -} - -/** - * gst_base_audio_src_get_slave_method: - * @src: a #GstBaseAudioSrc - * - * Get the current slave method used by @src. - * - * Returns: The current slave method used by @src. - * - * Since: 0.10.20 - */ -GstBaseAudioSrcSlaveMethod -gst_base_audio_src_get_slave_method (GstBaseAudioSrc * src) -{ - GstBaseAudioSrcSlaveMethod result; - - g_return_val_if_fail (GST_IS_BASE_AUDIO_SRC (src), -1); - - GST_OBJECT_LOCK (src); - result = src->priv->slave_method; - GST_OBJECT_UNLOCK (src); - - return result; -} - -static void -gst_base_audio_src_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstBaseAudioSrc *src; - - src = GST_BASE_AUDIO_SRC (object); - - switch (prop_id) { - case PROP_BUFFER_TIME: - src->buffer_time = g_value_get_int64 (value); - break; - case PROP_LATENCY_TIME: - src->latency_time = g_value_get_int64 (value); - break; - case PROP_PROVIDE_CLOCK: - gst_base_audio_src_set_provide_clock (src, g_value_get_boolean (value)); - break; - case PROP_SLAVE_METHOD: - gst_base_audio_src_set_slave_method (src, g_value_get_enum (value)); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_base_audio_src_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstBaseAudioSrc *src; - - src = GST_BASE_AUDIO_SRC (object); - - switch (prop_id) { - case PROP_BUFFER_TIME: - g_value_set_int64 (value, src->buffer_time); - break; - case PROP_LATENCY_TIME: - g_value_set_int64 (value, src->latency_time); - break; - case PROP_ACTUAL_BUFFER_TIME: - GST_OBJECT_LOCK (src); - if (src->ringbuffer && src->ringbuffer->acquired) - g_value_set_int64 (value, src->ringbuffer->spec.buffer_time); - else - g_value_set_int64 (value, DEFAULT_ACTUAL_BUFFER_TIME); - GST_OBJECT_UNLOCK (src); - break; - case PROP_ACTUAL_LATENCY_TIME: - GST_OBJECT_LOCK (src); - if (src->ringbuffer && src->ringbuffer->acquired) - g_value_set_int64 (value, src->ringbuffer->spec.latency_time); - else - g_value_set_int64 (value, DEFAULT_ACTUAL_LATENCY_TIME); - GST_OBJECT_UNLOCK (src); - break; - case PROP_PROVIDE_CLOCK: - g_value_set_boolean (value, gst_base_audio_src_get_provide_clock (src)); - break; - case PROP_SLAVE_METHOD: - g_value_set_enum (value, gst_base_audio_src_get_slave_method (src)); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps) -{ - GstStructure *s; - gint width, depth; - - s = gst_caps_get_structure (caps, 0); - - /* fields for all formats */ - gst_structure_fixate_field_nearest_int (s, "rate", 44100); - gst_structure_fixate_field_nearest_int (s, "channels", 2); - gst_structure_fixate_field_nearest_int (s, "width", 16); - - /* fields for int */ - if (gst_structure_has_field (s, "depth")) { - gst_structure_get_int (s, "width", &width); - /* round width to nearest multiple of 8 for the depth */ - depth = GST_ROUND_UP_8 (width); - gst_structure_fixate_field_nearest_int (s, "depth", depth); - } - if (gst_structure_has_field (s, "signed")) - gst_structure_fixate_field_boolean (s, "signed", TRUE); - if (gst_structure_has_field (s, "endianness")) - gst_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER); -} - -static gboolean -gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps) -{ - GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc); - GstRingBufferSpec *spec; - - spec = &src->ringbuffer->spec; - - spec->buffer_time = src->buffer_time; - spec->latency_time = src->latency_time; - - if (!gst_ring_buffer_parse_caps (spec, caps)) - goto parse_error; - - /* calculate suggested segsize and segtotal */ - spec->segsize = - spec->rate * spec->bytes_per_sample * spec->latency_time / GST_MSECOND; - spec->segtotal = spec->buffer_time / spec->latency_time; - - GST_DEBUG ("release old ringbuffer"); - - gst_ring_buffer_release (src->ringbuffer); - - gst_ring_buffer_debug_spec_buff (spec); - - GST_DEBUG ("acquire new ringbuffer"); - - if (!gst_ring_buffer_acquire (src->ringbuffer, spec)) - goto acquire_error; - - /* calculate actual latency and buffer times */ - spec->latency_time = - spec->segsize * GST_MSECOND / (spec->rate * spec->bytes_per_sample); - spec->buffer_time = - spec->segtotal * spec->segsize * GST_MSECOND / (spec->rate * - spec->bytes_per_sample); - - gst_ring_buffer_debug_spec_buff (spec); - - g_object_notify (G_OBJECT (src), "actual-buffer-time"); - g_object_notify (G_OBJECT (src), "actual-latency-time"); - - return TRUE; - - /* ERRORS */ -parse_error: - { - GST_DEBUG ("could not parse caps"); - return FALSE; - } -acquire_error: - { - GST_DEBUG ("could not acquire ringbuffer"); - return FALSE; - } -} - -static void -gst_base_audio_src_get_times (GstBaseSrc * bsrc, GstBuffer * buffer, - GstClockTime * start, GstClockTime * end) -{ - /* no need to sync to a clock here, we schedule the samples based - * on our own clock for the moment. */ - *start = GST_CLOCK_TIME_NONE; - *end = GST_CLOCK_TIME_NONE; -} - -static gboolean -gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query) -{ - GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc); - gboolean res = FALSE; - - switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_LATENCY: - { - GstClockTime min_latency, max_latency; - GstRingBufferSpec *spec; - - if (G_UNLIKELY (src->ringbuffer == NULL - || src->ringbuffer->spec.rate == 0)) - goto done; - - spec = &src->ringbuffer->spec; - - /* we have at least 1 segment of latency */ - min_latency = - gst_util_uint64_scale_int (spec->segsize, GST_SECOND, - spec->rate * spec->bytes_per_sample); - /* we cannot delay more than the buffersize else we lose data */ - max_latency = - gst_util_uint64_scale_int (spec->segtotal * spec->segsize, GST_SECOND, - spec->rate * spec->bytes_per_sample); - - GST_DEBUG_OBJECT (src, - "report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT, - GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency)); - - /* we are always live, the min latency is 1 segment and the max latency is - * the complete buffer of segments. */ - gst_query_set_latency (query, TRUE, min_latency, max_latency); - - res = TRUE; - break; - } - default: - res = GST_BASE_SRC_CLASS (parent_class)->query (bsrc, query); - break; - } -done: - return res; -} - -static gboolean -gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event) -{ - GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc); - gboolean res; - - res = TRUE; - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_FLUSH_START: - GST_DEBUG_OBJECT (bsrc, "flush-start"); - gst_ring_buffer_pause (src->ringbuffer); - gst_ring_buffer_clear_all (src->ringbuffer); - break; - case GST_EVENT_FLUSH_STOP: - GST_DEBUG_OBJECT (bsrc, "flush-stop"); - /* always resync on sample after a flush */ - src->next_sample = -1; - gst_ring_buffer_clear_all (src->ringbuffer); - break; - case GST_EVENT_SEEK: - GST_DEBUG_OBJECT (bsrc, "refuse to seek"); - res = FALSE; - break; - default: - GST_DEBUG_OBJECT (bsrc, "dropping event %p", event); - break; - } - return res; -} - -/* get the next offset in the ringbuffer for reading samples. - * If the next sample is too far away, this function will position itself to the - * next most recent sample, creating discontinuity */ -static guint64 -gst_base_audio_src_get_offset (GstBaseAudioSrc * src) -{ - guint64 sample; - gint readseg, segdone, segtotal, sps; - gint diff; - - /* assume we can append to the previous sample */ - sample = src->next_sample; - - sps = src->ringbuffer->samples_per_seg; - segtotal = src->ringbuffer->spec.segtotal; - - /* get the currently processed segment */ - segdone = g_atomic_int_get (&src->ringbuffer->segdone) - - src->ringbuffer->segbase; - - if (sample != -1) { - GST_DEBUG_OBJECT (src, "at segment %d and sample %" G_GUINT64_FORMAT, - segdone, sample); - /* figure out the segment and the offset inside the segment where - * the sample should be read from. */ - readseg = sample / sps; - - /* see how far away it is from the read segment, normally segdone (where new - * data is written in the ringbuffer) is bigger than readseg (where we are - * reading). */ - diff = segdone - readseg; - if (diff >= segtotal) { - GST_DEBUG_OBJECT (src, "dropped, align to segment %d", segdone); - /* sample would be dropped, position to next playable position */ - sample = ((guint64) (segdone)) * sps; - } - } else { - /* no previous sample, go to the current position */ - GST_DEBUG_OBJECT (src, "first sample, align to current %d", segdone); - sample = ((guint64) (segdone)) * sps; - readseg = segdone; - } - - GST_DEBUG_OBJECT (src, - "reading from %d, we are at %d, sample %" G_GUINT64_FORMAT, readseg, - segdone, sample); - - return sample; -} - -static GstFlowReturn -gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length, - GstBuffer ** outbuf) -{ - GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc); - GstBuffer *buf; - guchar *data; - guint samples, total_samples; - guint64 sample; - gint bps; - GstRingBuffer *ringbuffer; - GstRingBufferSpec *spec; - guint read; - GstClockTime timestamp, duration; - GstClock *clock; - - ringbuffer = src->ringbuffer; - spec = &ringbuffer->spec; - - if (G_UNLIKELY (!gst_ring_buffer_is_acquired (ringbuffer))) - goto wrong_state; - - bps = spec->bytes_per_sample; - - if ((length == 0 && bsrc->blocksize == 0) || length == -1) - /* no length given, use the default segment size */ - length = spec->segsize; - else - /* make sure we round down to an integral number of samples */ - length -= length % bps; - - /* figure out the offset in the ringbuffer */ - if (G_UNLIKELY (offset != -1)) { - sample = offset / bps; - /* if a specific offset was given it must be the next sequential - * offset we expect or we fail for now. */ - if (src->next_sample != -1 && sample != src->next_sample) - goto wrong_offset; - } else { - /* calculate the sequentially next sample we need to read. This can jump and - * create a DISCONT. */ - sample = gst_base_audio_src_get_offset (src); - } - - GST_DEBUG_OBJECT (src, "reading from sample %" G_GUINT64_FORMAT, sample); - - /* get the number of samples to read */ - total_samples = samples = length / bps; - - /* FIXME, using a bufferpool would be nice here */ - buf = gst_buffer_new_and_alloc (length); - data = GST_BUFFER_DATA (buf); - - do { - read = gst_ring_buffer_read (ringbuffer, sample, data, samples); - GST_DEBUG_OBJECT (src, "read %u of %u", read, samples); - /* if we read all, we're done */ - if (read == samples) - break; - - /* else something interrupted us and we wait for playing again. */ - GST_DEBUG_OBJECT (src, "wait playing"); - if (gst_base_src_wait_playing (bsrc) != GST_FLOW_OK) - goto stopped; - - GST_DEBUG_OBJECT (src, "continue playing"); - - /* read next samples */ - sample += read; - samples -= read; - data += read * bps; - } while (TRUE); - - /* mark discontinuity if needed */ - if (G_UNLIKELY (sample != src->next_sample) && src->next_sample != -1) { - GST_WARNING_OBJECT (src, - "create DISCONT of %" G_GUINT64_FORMAT " samples at sample %" - G_GUINT64_FORMAT, sample - src->next_sample, sample); - GST_ELEMENT_WARNING (src, CORE, CLOCK, - (_("Can't record audio fast enough")), - ("Dropped %" G_GUINT64_FORMAT " samples. This is most likely because " - "downstream can't keep up and is consuming samples too slowly.", - sample - src->next_sample)); - GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); - } - - src->next_sample = sample + samples; - - /* get the normal timestamp to get the duration. */ - timestamp = gst_util_uint64_scale_int (sample, GST_SECOND, spec->rate); - duration = gst_util_uint64_scale_int (src->next_sample, GST_SECOND, - spec->rate) - timestamp; - - GST_OBJECT_LOCK (src); - if (!(clock = GST_ELEMENT_CLOCK (src))) - goto no_sync; - - if (clock != src->clock) { - /* we are slaved, check how to handle this */ - switch (src->priv->slave_method) { - case GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE: - /* not implemented, use skew algorithm. This algorithm should - * work on the readout pointer and produces more or less samples based - * on the clock drift */ - case GST_BASE_AUDIO_SRC_SLAVE_SKEW: - { - GstClockTime running_time; - GstClockTime base_time; - GstClockTime current_time; - guint64 running_time_sample; - gint running_time_segment; - gint last_read_segment; - gint segment_skew; - gint sps; - gint segments_written; - gint last_written_segment; - - /* get the amount of segments written from the device by now */ - segments_written = g_atomic_int_get (&ringbuffer->segdone); - - /* subtract the base to segments_written to get the number of the - last written segment in the ringbuffer (one segment written = segment 0) */ - last_written_segment = segments_written - ringbuffer->segbase - 1; - - /* samples per segment */ - sps = ringbuffer->samples_per_seg; - - /* get the current time */ - current_time = gst_clock_get_time (clock); - - /* get the basetime */ - base_time = GST_ELEMENT_CAST (src)->base_time; - - /* get the running_time */ - running_time = current_time - base_time; - - /* the running_time converted to a sample (relative to the ringbuffer) */ - running_time_sample = - gst_util_uint64_scale_int (running_time, spec->rate, GST_SECOND); - - /* the segmentnr corrensponding to running_time, round down */ - running_time_segment = running_time_sample / sps; - - /* the segment currently read from the ringbuffer */ - last_read_segment = sample / sps; - - /* the skew we have between running_time and the ringbuffertime (last written to) */ - segment_skew = running_time_segment - last_written_segment; - - GST_DEBUG_OBJECT (bsrc, - "\n running_time = %" - GST_TIME_FORMAT - "\n timestamp = %" - GST_TIME_FORMAT - "\n running_time_segment = %d" - "\n last_written_segment = %d" - "\n segment_skew (running time segment - last_written_segment) = %d" - "\n last_read_segment = %d", - GST_TIME_ARGS (running_time), GST_TIME_ARGS (timestamp), - running_time_segment, last_written_segment, segment_skew, - last_read_segment); - - /* Resync the ringbuffer if: - * - * 1. We are more than the length of the ringbuffer behind. - * The length of the ringbuffer then gets to dictate - * the threshold for what is concidered "too late" - * - * 2. If this is our first buffer. - * We know that we should catch up to running_time - * the first time we are ran. - */ - if ((segment_skew >= ringbuffer->spec.segtotal) || - (last_read_segment == 0)) { - gint new_read_segment; - gint segment_diff; - guint64 new_sample; - - /* the difference between running_time and the last written segment */ - segment_diff = running_time_segment - last_written_segment; - - /* advance the ringbuffer */ - gst_ring_buffer_advance (ringbuffer, segment_diff); - - /* we move the new read segment to the last known written segment */ - new_read_segment = - g_atomic_int_get (&ringbuffer->segdone) - ringbuffer->segbase; - - /* we calculate the new sample value */ - new_sample = ((guint64) new_read_segment) * sps; - - /* and get the relative time to this -> our new timestamp */ - timestamp = - gst_util_uint64_scale_int (new_sample, GST_SECOND, spec->rate); - - /* we update the next sample accordingly */ - src->next_sample = new_sample + samples; - - GST_DEBUG_OBJECT (bsrc, - "Timeshifted the ringbuffer with %d segments: " - "Updating the timestamp to %" GST_TIME_FORMAT ", " - "and src->next_sample to %" G_GUINT64_FORMAT, segment_diff, - GST_TIME_ARGS (timestamp), src->next_sample); - } - break; - } - case GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP: - { - GstClockTime base_time, latency; - - /* We are slaved to another clock, take running time of the pipeline clock and - * timestamp against it. Somebody else in the pipeline should figure out the - * clock drift. We keep the duration we calculated above. */ - timestamp = gst_clock_get_time (clock); - base_time = GST_ELEMENT_CAST (src)->base_time; - - if (GST_CLOCK_DIFF (timestamp, base_time) < 0) - timestamp -= base_time; - else - timestamp = 0; - - /* subtract latency */ - latency = - gst_util_uint64_scale_int (total_samples, GST_SECOND, spec->rate); - if (timestamp > latency) - timestamp -= latency; - else - timestamp = 0; - } - case GST_BASE_AUDIO_SRC_SLAVE_NONE: - break; - } - } else { - GstClockTime base_time; - - /* to get the timestamp against the clock we also need to add our offset */ - timestamp = gst_audio_clock_adjust (clock, timestamp); - - /* we are not slaved, subtract base_time */ - base_time = GST_ELEMENT_CAST (src)->base_time; - - if (GST_CLOCK_DIFF (timestamp, base_time) < 0) { - timestamp -= base_time; - GST_LOG_OBJECT (src, - "buffer timestamp %" GST_TIME_FORMAT " (base_time %" GST_TIME_FORMAT - ")", GST_TIME_ARGS (timestamp), GST_TIME_ARGS (base_time)); - } else { - GST_LOG_OBJECT (src, - "buffer timestamp 0, ts %" GST_TIME_FORMAT " <= base_time %" - GST_TIME_FORMAT, GST_TIME_ARGS (timestamp), - GST_TIME_ARGS (base_time)); - timestamp = 0; - } - } - -no_sync: - GST_OBJECT_UNLOCK (src); - - GST_BUFFER_TIMESTAMP (buf) = timestamp; - GST_BUFFER_DURATION (buf) = duration; - GST_BUFFER_OFFSET (buf) = sample; - GST_BUFFER_OFFSET_END (buf) = sample + samples; - - *outbuf = buf; - - return GST_FLOW_OK; - - /* ERRORS */ -wrong_state: - { - GST_DEBUG_OBJECT (src, "ringbuffer in wrong state"); - return GST_FLOW_WRONG_STATE; - } -wrong_offset: - { - GST_ELEMENT_ERROR (src, RESOURCE, SEEK, - (NULL), ("resource can only be operated on sequentially but offset %" - G_GUINT64_FORMAT " was given", offset)); - return GST_FLOW_ERROR; - } -stopped: - { - gst_buffer_unref (buf); - GST_DEBUG_OBJECT (src, "ringbuffer stopped"); - return GST_FLOW_WRONG_STATE; - } -} - -/** - * gst_base_audio_src_create_ringbuffer: - * @src: a #GstBaseAudioSrc. - * - * Create and return the #GstRingBuffer for @src. This function will call the - * ::create_ringbuffer vmethod and will set @src as the parent of the returned - * buffer (see gst_object_set_parent()). - * - * Returns: The new ringbuffer of @src. - */ -GstRingBuffer * -gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc * src) -{ - GstBaseAudioSrcClass *bclass; - GstRingBuffer *buffer = NULL; - - bclass = GST_BASE_AUDIO_SRC_GET_CLASS (src); - if (bclass->create_ringbuffer) - buffer = bclass->create_ringbuffer (src); - - if (G_LIKELY (buffer)) - gst_object_set_parent (GST_OBJECT_CAST (buffer), GST_OBJECT_CAST (src)); - - return buffer; -} - -static GstStateChangeReturn -gst_base_audio_src_change_state (GstElement * element, - GstStateChange transition) -{ - GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; - GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element); - - switch (transition) { - case GST_STATE_CHANGE_NULL_TO_READY: - GST_DEBUG_OBJECT (src, "NULL->READY"); - GST_OBJECT_LOCK (src); - if (src->ringbuffer == NULL) { - gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0); - src->ringbuffer = gst_base_audio_src_create_ringbuffer (src); - } - GST_OBJECT_UNLOCK (src); - if (!gst_ring_buffer_open_device (src->ringbuffer)) - goto open_failed; - break; - case GST_STATE_CHANGE_READY_TO_PAUSED: - GST_DEBUG_OBJECT (src, "READY->PAUSED"); - src->next_sample = -1; - gst_ring_buffer_set_flushing (src->ringbuffer, FALSE); - gst_ring_buffer_may_start (src->ringbuffer, FALSE); - break; - case GST_STATE_CHANGE_PAUSED_TO_PLAYING: - GST_DEBUG_OBJECT (src, "PAUSED->PLAYING"); - gst_ring_buffer_may_start (src->ringbuffer, TRUE); - break; - case GST_STATE_CHANGE_PLAYING_TO_PAUSED: - GST_DEBUG_OBJECT (src, "PLAYING->PAUSED"); - gst_ring_buffer_may_start (src->ringbuffer, FALSE); - gst_ring_buffer_pause (src->ringbuffer); - break; - case GST_STATE_CHANGE_PAUSED_TO_READY: - GST_DEBUG_OBJECT (src, "PAUSED->READY"); - gst_ring_buffer_set_flushing (src->ringbuffer, TRUE); - break; - default: - break; - } - - ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - switch (transition) { - case GST_STATE_CHANGE_PAUSED_TO_READY: - GST_DEBUG_OBJECT (src, "PAUSED->READY"); - gst_ring_buffer_release (src->ringbuffer); - break; - case GST_STATE_CHANGE_READY_TO_NULL: - GST_DEBUG_OBJECT (src, "READY->NULL"); - gst_ring_buffer_close_device (src->ringbuffer); - GST_OBJECT_LOCK (src); - gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer)); - src->ringbuffer = NULL; - GST_OBJECT_UNLOCK (src); - break; - default: - break; - } - - return ret; - - /* ERRORS */ -open_failed: - { - /* subclass must post a meaningfull error message */ - GST_DEBUG_OBJECT (src, "open failed"); - return GST_STATE_CHANGE_FAILURE; - } - -} |