/* GStreamer * Copyright (C) 2011 David A. Schleef <ds@schleef.org> * * 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 Street, Suite 500, * Boston, MA 02110-1335, USA. */ /** * SECTION:element-gstinteraudiosrc * * The interaudiosrc element is an audio source element. It is used * in connection with a interaudiosink element in a different pipeline. * * <refsect2> * <title>Example launch line</title> * |[ * gst-launch -v interaudiosrc ! queue ! audiosink * ]| * * The interaudiosrc element cannot be used effectively with gst-launch, * as it requires a second pipeline in the application to send audio. * See the gstintertest.c example in the gst-plugins-bad source code for * more details. * </refsect2> */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstinteraudiosrc.h" #include <gst/gst.h> #include <gst/base/gstbasesrc.h> #include <gst/audio/audio.h> #include <string.h> GST_DEBUG_CATEGORY_STATIC (gst_inter_audio_src_debug_category); #define GST_CAT_DEFAULT gst_inter_audio_src_debug_category /* prototypes */ static void gst_inter_audio_src_set_property (GObject * object, guint property_id, const GValue * value, GParamSpec * pspec); static void gst_inter_audio_src_get_property (GObject * object, guint property_id, GValue * value, GParamSpec * pspec); static void gst_inter_audio_src_finalize (GObject * object); static GstCaps *gst_inter_audio_src_get_caps (GstBaseSrc * src, GstCaps * filter); static gboolean gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps); static gboolean gst_inter_audio_src_start (GstBaseSrc * src); static gboolean gst_inter_audio_src_stop (GstBaseSrc * src); static void gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer, GstClockTime * start, GstClockTime * end); static GstFlowReturn gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size, GstBuffer ** buf); static gboolean gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query); static GstCaps *gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps); enum { PROP_0, PROP_CHANNEL, PROP_BUFFER_TIME, PROP_LATENCY_TIME, PROP_PERIOD_TIME }; #define DEFAULT_CHANNEL ("default") /* pad templates */ static GstStaticPadTemplate gst_inter_audio_src_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) ", layout = (string) interleaved") ); /* class initialization */ #define parent_class gst_inter_audio_src_parent_class G_DEFINE_TYPE (GstInterAudioSrc, gst_inter_audio_src, GST_TYPE_BASE_SRC); static void gst_inter_audio_src_class_init (GstInterAudioSrcClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstBaseSrcClass *base_src_class = GST_BASE_SRC_CLASS (klass); GST_DEBUG_CATEGORY_INIT (gst_inter_audio_src_debug_category, "interaudiosrc", 0, "debug category for interaudiosrc element"); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_inter_audio_src_src_template)); gst_element_class_set_static_metadata (element_class, "Internal audio source", "Source/Audio", "Virtual audio source for internal process communication", "David Schleef <ds@schleef.org>"); gobject_class->set_property = gst_inter_audio_src_set_property; gobject_class->get_property = gst_inter_audio_src_get_property; gobject_class->finalize = gst_inter_audio_src_finalize; base_src_class->get_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_src_get_caps); base_src_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_src_set_caps); base_src_class->start = GST_DEBUG_FUNCPTR (gst_inter_audio_src_start); base_src_class->stop = GST_DEBUG_FUNCPTR (gst_inter_audio_src_stop); base_src_class->get_times = GST_DEBUG_FUNCPTR (gst_inter_audio_src_get_times); base_src_class->create = GST_DEBUG_FUNCPTR (gst_inter_audio_src_create); base_src_class->query = GST_DEBUG_FUNCPTR (gst_inter_audio_src_query); base_src_class->fixate = GST_DEBUG_FUNCPTR (gst_inter_audio_src_fixate); g_object_class_install_property (gobject_class, PROP_CHANNEL, g_param_spec_string ("channel", "Channel", "Channel name to match inter src and sink elements", DEFAULT_CHANNEL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_BUFFER_TIME, g_param_spec_uint64 ("buffer-time", "Buffer Time", "Size of audio buffer", 1, G_MAXUINT64, DEFAULT_AUDIO_BUFFER_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_LATENCY_TIME, g_param_spec_uint64 ("latency-time", "Latency Time", "Latency as reported by the source", 1, G_MAXUINT64, DEFAULT_AUDIO_LATENCY_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PERIOD_TIME, g_param_spec_uint64 ("period-time", "Period Time", "The minimum amount of data to read in each iteration", 1, G_MAXUINT64, DEFAULT_AUDIO_PERIOD_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); } static void gst_inter_audio_src_init (GstInterAudioSrc * interaudiosrc) { gst_base_src_set_format (GST_BASE_SRC (interaudiosrc), GST_FORMAT_TIME); gst_base_src_set_live (GST_BASE_SRC (interaudiosrc), TRUE); gst_base_src_set_blocksize (GST_BASE_SRC (interaudiosrc), -1); interaudiosrc->channel = g_strdup (DEFAULT_CHANNEL); interaudiosrc->buffer_time = DEFAULT_AUDIO_BUFFER_TIME; interaudiosrc->latency_time = DEFAULT_AUDIO_LATENCY_TIME; interaudiosrc->period_time = DEFAULT_AUDIO_PERIOD_TIME; } void gst_inter_audio_src_set_property (GObject * object, guint property_id, const GValue * value, GParamSpec * pspec) { GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object); switch (property_id) { case PROP_CHANNEL: g_free (interaudiosrc->channel); interaudiosrc->channel = g_value_dup_string (value); break; case PROP_BUFFER_TIME: interaudiosrc->buffer_time = g_value_get_uint64 (value); break; case PROP_LATENCY_TIME: interaudiosrc->latency_time = g_value_get_uint64 (value); break; case PROP_PERIOD_TIME: interaudiosrc->period_time = g_value_get_uint64 (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec); break; } } void gst_inter_audio_src_get_property (GObject * object, guint property_id, GValue * value, GParamSpec * pspec) { GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object); switch (property_id) { case PROP_CHANNEL: g_value_set_string (value, interaudiosrc->channel); break; case PROP_BUFFER_TIME: g_value_set_uint64 (value, interaudiosrc->buffer_time); break; case PROP_LATENCY_TIME: g_value_set_uint64 (value, interaudiosrc->latency_time); break; case PROP_PERIOD_TIME: g_value_set_uint64 (value, interaudiosrc->period_time); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec); break; } } void gst_inter_audio_src_finalize (GObject * object) { GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object); /* clean up object here */ g_free (interaudiosrc->channel); G_OBJECT_CLASS (gst_inter_audio_src_parent_class)->finalize (object); } static GstCaps * gst_inter_audio_src_get_caps (GstBaseSrc * src, GstCaps * filter) { GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); GstCaps *caps; GST_DEBUG_OBJECT (interaudiosrc, "get_caps"); if (!interaudiosrc->surface) return GST_BASE_SRC_CLASS (parent_class)->get_caps (src, filter); g_mutex_lock (&interaudiosrc->surface->mutex); if (interaudiosrc->surface->audio_info.finfo) { caps = gst_audio_info_to_caps (&interaudiosrc->surface->audio_info); if (filter) { GstCaps *tmp; tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (caps); caps = tmp; } } else { caps = NULL; } g_mutex_unlock (&interaudiosrc->surface->mutex); if (caps) return caps; else return GST_BASE_SRC_CLASS (parent_class)->get_caps (src, filter); } static gboolean gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps) { GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); GST_DEBUG_OBJECT (interaudiosrc, "set_caps"); if (!gst_audio_info_from_caps (&interaudiosrc->info, caps)) { GST_ERROR_OBJECT (src, "Failed to parse caps %" GST_PTR_FORMAT, caps); return FALSE; } return TRUE; } static gboolean gst_inter_audio_src_start (GstBaseSrc * src) { GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); GST_DEBUG_OBJECT (interaudiosrc, "start"); interaudiosrc->surface = gst_inter_surface_get (interaudiosrc->channel); interaudiosrc->timestamp_offset = 0; interaudiosrc->n_samples = 0; g_mutex_lock (&interaudiosrc->surface->mutex); interaudiosrc->surface->audio_buffer_time = interaudiosrc->buffer_time; interaudiosrc->surface->audio_latency_time = interaudiosrc->latency_time; interaudiosrc->surface->audio_period_time = interaudiosrc->period_time; g_mutex_unlock (&interaudiosrc->surface->mutex); return TRUE; } static gboolean gst_inter_audio_src_stop (GstBaseSrc * src) { GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); GST_DEBUG_OBJECT (interaudiosrc, "stop"); gst_inter_surface_unref (interaudiosrc->surface); interaudiosrc->surface = NULL; return TRUE; } static void gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer, GstClockTime * start, GstClockTime * end) { GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); GST_DEBUG_OBJECT (src, "get_times"); /* for live sources, sync on the timestamp of the buffer */ if (gst_base_src_is_live (src)) { if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) { *start = GST_BUFFER_TIMESTAMP (buffer); if (GST_BUFFER_DURATION_IS_VALID (buffer)) { *end = *start + GST_BUFFER_DURATION (buffer); } else { if (interaudiosrc->info.rate > 0) { *end = *start + gst_util_uint64_scale_int (gst_buffer_get_size (buffer), GST_SECOND, interaudiosrc->info.rate * interaudiosrc->info.bpf); } } } } } static GstFlowReturn gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size, GstBuffer ** buf) { GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); GstCaps *caps; GstBuffer *buffer; guint n, bpf; guint64 period_time; guint64 period_samples; GST_DEBUG_OBJECT (interaudiosrc, "create"); buffer = NULL; caps = NULL; g_mutex_lock (&interaudiosrc->surface->mutex); if (interaudiosrc->surface->audio_info.finfo) { if (!gst_audio_info_is_equal (&interaudiosrc->surface->audio_info, &interaudiosrc->info)) { caps = gst_audio_info_to_caps (&interaudiosrc->surface->audio_info); interaudiosrc->timestamp_offset += gst_util_uint64_scale (interaudiosrc->n_samples, GST_SECOND, interaudiosrc->info.rate); interaudiosrc->n_samples = 0; } } bpf = interaudiosrc->surface->audio_info.bpf; period_time = interaudiosrc->surface->audio_period_time; period_samples = gst_util_uint64_scale (period_time, interaudiosrc->info.rate, GST_SECOND); if (bpf > 0) n = gst_adapter_available (interaudiosrc->surface->audio_adapter) / bpf; else n = 0; if (n > period_samples) n = period_samples; if (n > 0) { buffer = gst_adapter_take_buffer (interaudiosrc->surface->audio_adapter, n * bpf); } else { buffer = gst_buffer_new (); GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_GAP); } g_mutex_unlock (&interaudiosrc->surface->mutex); if (caps) { gboolean ret = gst_base_src_set_caps (src, caps); gst_caps_unref (caps); if (!ret) { GST_ERROR_OBJECT (src, "Failed to set caps %" GST_PTR_FORMAT, caps); if (buffer) gst_buffer_unref (buffer); return GST_FLOW_NOT_NEGOTIATED; } } buffer = gst_buffer_make_writable (buffer); bpf = interaudiosrc->info.bpf; if (n < period_samples) { GstMapInfo map; GstMemory *mem; GST_DEBUG_OBJECT (interaudiosrc, "creating %" G_GUINT64_FORMAT " samples of silence", period_samples - n); mem = gst_allocator_alloc (NULL, (period_samples - n) * bpf, NULL); if (gst_memory_map (mem, &map, GST_MAP_WRITE)) { gst_audio_format_fill_silence (interaudiosrc->info.finfo, map.data, map.size); gst_memory_unmap (mem, &map); } gst_buffer_prepend_memory (buffer, mem); } n = period_samples; GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples; GST_BUFFER_OFFSET_END (buffer) = interaudiosrc->n_samples + n; GST_BUFFER_TIMESTAMP (buffer) = interaudiosrc->timestamp_offset + gst_util_uint64_scale (interaudiosrc->n_samples, GST_SECOND, interaudiosrc->info.rate); GST_DEBUG_OBJECT (interaudiosrc, "create ts %" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer))); GST_BUFFER_DURATION (buffer) = interaudiosrc->timestamp_offset + gst_util_uint64_scale (interaudiosrc->n_samples + n, GST_SECOND, interaudiosrc->info.rate) - GST_BUFFER_TIMESTAMP (buffer); GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT); if (interaudiosrc->n_samples == 0) { GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT); } interaudiosrc->n_samples += n; *buf = buffer; return GST_FLOW_OK; } static gboolean gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query) { GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); gboolean ret; GST_DEBUG_OBJECT (src, "query"); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_LATENCY:{ GstClockTime min_latency, max_latency; min_latency = interaudiosrc->latency_time; max_latency = interaudiosrc->buffer_time; GST_DEBUG_OBJECT (src, "report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT, GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency)); gst_query_set_latency (query, gst_base_src_is_live (src), min_latency, max_latency); ret = TRUE; break; } default: ret = GST_BASE_SRC_CLASS (gst_inter_audio_src_parent_class)->query (src, query); break; } return ret; } static GstCaps * gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps) { GstStructure *structure; GST_DEBUG_OBJECT (src, "fixate"); caps = gst_caps_make_writable (caps); caps = gst_caps_truncate (caps); structure = gst_caps_get_structure (caps, 0); gst_structure_fixate_field_string (structure, "format", GST_AUDIO_NE (S16)); gst_structure_fixate_field_nearest_int (structure, "channels", 2); gst_structure_fixate_field_nearest_int (structure, "rate", 48000); gst_structure_fixate_field_string (structure, "layout", "interleaved"); return caps; }