| /* 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 <gst/gst.h> |
| #include <gst/base/gstbasesrc.h> |
| #include "gstinteraudiosrc.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_dispose (GObject * object); |
| static void gst_inter_audio_src_finalize (GObject * object); |
| |
| static GstCaps *gst_inter_audio_src_get_caps (GstBaseSrc * src); |
| static gboolean gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps); |
| static gboolean gst_inter_audio_src_negotiate (GstBaseSrc * src); |
| static gboolean gst_inter_audio_src_newsegment (GstBaseSrc * src); |
| 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 gboolean gst_inter_audio_src_is_seekable (GstBaseSrc * src); |
| static gboolean gst_inter_audio_src_unlock (GstBaseSrc * src); |
| static gboolean gst_inter_audio_src_event (GstBaseSrc * src, GstEvent * event); |
| static GstFlowReturn |
| gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size, |
| GstBuffer ** buf); |
| static gboolean gst_inter_audio_src_do_seek (GstBaseSrc * src, |
| GstSegment * segment); |
| static gboolean gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query); |
| static gboolean gst_inter_audio_src_check_get_range (GstBaseSrc * src); |
| static void gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps); |
| static gboolean gst_inter_audio_src_unlock_stop (GstBaseSrc * src); |
| static gboolean |
| gst_inter_audio_src_prepare_seek_segment (GstBaseSrc * src, GstEvent * seek, |
| GstSegment * segment); |
| |
| enum |
| { |
| PROP_0, |
| PROP_CHANNEL |
| }; |
| |
| /* pad templates */ |
| |
| static GstStaticPadTemplate gst_inter_audio_src_src_template = |
| GST_STATIC_PAD_TEMPLATE ("src", |
| GST_PAD_SRC, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/x-raw-int, " |
| "endianness = (int) BYTE_ORDER, " |
| "signed = (boolean) true, " |
| "width = (int) 16, " |
| "depth = (int) 16, " "rate = (int) 48000, " "channels = (int) 2") |
| ); |
| |
| |
| /* class initialization */ |
| |
| #define DEBUG_INIT(bla) \ |
| GST_DEBUG_CATEGORY_INIT (gst_inter_audio_src_debug_category, "interaudiosrc", 0, \ |
| "debug category for interaudiosrc element"); |
| |
| GST_BOILERPLATE_FULL (GstInterAudioSrc, gst_inter_audio_src, GstBaseSrc, |
| GST_TYPE_BASE_SRC, DEBUG_INIT); |
| |
| static void |
| gst_inter_audio_src_base_init (gpointer g_class) |
| { |
| GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); |
| |
| gst_element_class_add_pad_template (element_class, |
| gst_static_pad_template_get (&gst_inter_audio_src_src_template)); |
| |
| gst_element_class_set_details_simple (element_class, |
| "Internal audio source", |
| "Source/Audio", |
| "Virtual audio source for internal process communication", |
| "David Schleef <ds@schleef.org>"); |
| } |
| |
| static void |
| gst_inter_audio_src_class_init (GstInterAudioSrcClass * klass) |
| { |
| GObjectClass *gobject_class = G_OBJECT_CLASS (klass); |
| GstBaseSrcClass *base_src_class = GST_BASE_SRC_CLASS (klass); |
| |
| gobject_class->set_property = gst_inter_audio_src_set_property; |
| gobject_class->get_property = gst_inter_audio_src_get_property; |
| gobject_class->dispose = gst_inter_audio_src_dispose; |
| 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); |
| if (0) |
| base_src_class->negotiate = |
| GST_DEBUG_FUNCPTR (gst_inter_audio_src_negotiate); |
| base_src_class->newsegment = |
| GST_DEBUG_FUNCPTR (gst_inter_audio_src_newsegment); |
| 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); |
| if (0) |
| base_src_class->is_seekable = |
| GST_DEBUG_FUNCPTR (gst_inter_audio_src_is_seekable); |
| base_src_class->unlock = GST_DEBUG_FUNCPTR (gst_inter_audio_src_unlock); |
| base_src_class->event = GST_DEBUG_FUNCPTR (gst_inter_audio_src_event); |
| base_src_class->create = GST_DEBUG_FUNCPTR (gst_inter_audio_src_create); |
| if (0) |
| base_src_class->do_seek = GST_DEBUG_FUNCPTR (gst_inter_audio_src_do_seek); |
| base_src_class->query = GST_DEBUG_FUNCPTR (gst_inter_audio_src_query); |
| if (0) |
| base_src_class->check_get_range = |
| GST_DEBUG_FUNCPTR (gst_inter_audio_src_check_get_range); |
| base_src_class->fixate = GST_DEBUG_FUNCPTR (gst_inter_audio_src_fixate); |
| if (0) |
| base_src_class->unlock_stop = |
| GST_DEBUG_FUNCPTR (gst_inter_audio_src_unlock_stop); |
| if (0) |
| base_src_class->prepare_seek_segment = |
| GST_DEBUG_FUNCPTR (gst_inter_audio_src_prepare_seek_segment); |
| |
| #if 0 |
| 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", G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| #endif |
| } |
| |
| static void |
| gst_inter_audio_src_init (GstInterAudioSrc * interaudiosrc, |
| GstInterAudioSrcClass * interaudiosrc_class) |
| { |
| |
| 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"); |
| } |
| |
| 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; |
| 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; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec); |
| break; |
| } |
| } |
| |
| void |
| gst_inter_audio_src_dispose (GObject * object) |
| { |
| /* GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object); */ |
| |
| /* clean up as possible. may be called multiple times */ |
| |
| G_OBJECT_CLASS (parent_class)->dispose (object); |
| } |
| |
| void |
| gst_inter_audio_src_finalize (GObject * object) |
| { |
| /* GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object); */ |
| |
| /* clean up object here */ |
| |
| G_OBJECT_CLASS (parent_class)->finalize (object); |
| } |
| |
| |
| static GstCaps * |
| gst_inter_audio_src_get_caps (GstBaseSrc * src) |
| { |
| GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); |
| |
| GST_DEBUG_OBJECT (interaudiosrc, "get_caps"); |
| |
| return NULL; |
| } |
| |
| static gboolean |
| gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps) |
| { |
| GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); |
| const GstStructure *structure; |
| gboolean ret; |
| int sample_rate; |
| |
| GST_DEBUG_OBJECT (interaudiosrc, "set_caps"); |
| |
| structure = gst_caps_get_structure (caps, 0); |
| |
| ret = gst_structure_get_int (structure, "rate", &sample_rate); |
| if (ret) { |
| interaudiosrc->sample_rate = sample_rate; |
| } |
| |
| return ret; |
| } |
| |
| static gboolean |
| gst_inter_audio_src_negotiate (GstBaseSrc * src) |
| { |
| GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); |
| |
| GST_DEBUG_OBJECT (interaudiosrc, "negotiate"); |
| |
| return TRUE; |
| } |
| |
| static gboolean |
| gst_inter_audio_src_newsegment (GstBaseSrc * src) |
| { |
| GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); |
| |
| GST_DEBUG_OBJECT (interaudiosrc, "newsegment"); |
| |
| 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); |
| |
| 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 (interaudiosrc, "get_times"); |
| |
| /* for live sources, sync on the timestamp of the buffer */ |
| if (gst_base_src_is_live (src)) { |
| GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer); |
| |
| if (GST_CLOCK_TIME_IS_VALID (timestamp)) { |
| /* get duration to calculate end time */ |
| GstClockTime duration = GST_BUFFER_DURATION (buffer); |
| |
| if (GST_CLOCK_TIME_IS_VALID (duration)) { |
| *end = timestamp + duration; |
| } |
| *start = timestamp; |
| } |
| } else { |
| *start = -1; |
| *end = -1; |
| } |
| } |
| |
| static gboolean |
| gst_inter_audio_src_is_seekable (GstBaseSrc * src) |
| { |
| GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); |
| |
| GST_DEBUG_OBJECT (interaudiosrc, "is_seekable"); |
| |
| return FALSE; |
| } |
| |
| static gboolean |
| gst_inter_audio_src_unlock (GstBaseSrc * src) |
| { |
| GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); |
| |
| GST_DEBUG_OBJECT (interaudiosrc, "unlock"); |
| |
| return TRUE; |
| } |
| |
| static gboolean |
| gst_inter_audio_src_event (GstBaseSrc * src, GstEvent * event) |
| { |
| GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); |
| gboolean ret; |
| |
| GST_DEBUG_OBJECT (interaudiosrc, "event"); |
| |
| switch (GST_EVENT_TYPE (event)) { |
| default: |
| ret = GST_BASE_SRC_CLASS (parent_class)->event (src, event); |
| } |
| |
| return ret; |
| } |
| |
| #define SIZE 1600 |
| |
| static GstFlowReturn |
| gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size, |
| GstBuffer ** buf) |
| { |
| GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); |
| GstBuffer *buffer; |
| int n; |
| |
| GST_DEBUG_OBJECT (interaudiosrc, "create"); |
| |
| buffer = NULL; |
| |
| g_mutex_lock (interaudiosrc->surface->mutex); |
| n = gst_adapter_available (interaudiosrc->surface->audio_adapter) / 4; |
| if (n > SIZE * 2) { |
| GST_DEBUG ("flushing %d samples", SIZE / 2); |
| gst_adapter_flush (interaudiosrc->surface->audio_adapter, (SIZE / 2) * 4); |
| n -= (SIZE / 2); |
| } |
| if (n > SIZE) |
| n = SIZE; |
| if (n > 0) { |
| buffer = gst_adapter_take_buffer (interaudiosrc->surface->audio_adapter, |
| n * 4); |
| } |
| g_mutex_unlock (interaudiosrc->surface->mutex); |
| |
| if (n < SIZE) { |
| GstBuffer *newbuf = gst_buffer_new_and_alloc (SIZE * 4); |
| |
| GST_DEBUG ("creating %d samples of silence", SIZE - n); |
| memset (GST_BUFFER_DATA (newbuf) + n * 4, 0, SIZE * 4 - n * 4); |
| if (buffer) { |
| memcpy (GST_BUFFER_DATA (newbuf), GST_BUFFER_DATA (buffer), n * 4); |
| gst_buffer_unref (buffer); |
| } |
| buffer = newbuf; |
| } |
| n = SIZE; |
| |
| GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples; |
| GST_BUFFER_OFFSET_END (buffer) = interaudiosrc->n_samples + n; |
| GST_BUFFER_TIMESTAMP (buffer) = |
| gst_util_uint64_scale_int (interaudiosrc->n_samples, GST_SECOND, |
| interaudiosrc->sample_rate); |
| GST_DEBUG_OBJECT (interaudiosrc, "create ts %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer))); |
| GST_BUFFER_DURATION (buffer) = |
| gst_util_uint64_scale_int (interaudiosrc->n_samples + n, GST_SECOND, |
| interaudiosrc->sample_rate) - GST_BUFFER_TIMESTAMP (buffer); |
| GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples; |
| GST_BUFFER_OFFSET_END (buffer) = -1; |
| GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT); |
| if (interaudiosrc->n_samples == 0) { |
| GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT); |
| } |
| gst_buffer_set_caps (buffer, GST_PAD_CAPS (GST_BASE_SRC_PAD (interaudiosrc))); |
| interaudiosrc->n_samples += n; |
| |
| *buf = buffer; |
| |
| return GST_FLOW_OK; |
| } |
| |
| static gboolean |
| gst_inter_audio_src_do_seek (GstBaseSrc * src, GstSegment * segment) |
| { |
| GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); |
| |
| GST_DEBUG_OBJECT (interaudiosrc, "do_seek"); |
| |
| return FALSE; |
| } |
| |
| static gboolean |
| gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query) |
| { |
| GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); |
| gboolean ret; |
| |
| GST_DEBUG_OBJECT (interaudiosrc, "query"); |
| |
| switch (GST_QUERY_TYPE (query)) { |
| case GST_QUERY_LATENCY:{ |
| GstClockTime min_latency, max_latency; |
| |
| min_latency = 30 * gst_util_uint64_scale_int (GST_SECOND, SIZE, 48000); |
| |
| max_latency = min_latency; |
| |
| GST_ERROR_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 (parent_class)->query (src, query); |
| break; |
| } |
| |
| return ret; |
| } |
| |
| static gboolean |
| gst_inter_audio_src_check_get_range (GstBaseSrc * src) |
| { |
| GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); |
| |
| GST_DEBUG_OBJECT (interaudiosrc, "get_range"); |
| |
| return FALSE; |
| } |
| |
| static void |
| gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps) |
| { |
| GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); |
| GstStructure *structure; |
| |
| structure = gst_caps_get_structure (caps, 0); |
| |
| GST_DEBUG_OBJECT (interaudiosrc, "fixate"); |
| |
| gst_structure_fixate_field_nearest_int (structure, "channels", 2); |
| gst_structure_fixate_field_nearest_int (structure, "rate", 48000); |
| |
| } |
| |
| static gboolean |
| gst_inter_audio_src_unlock_stop (GstBaseSrc * src) |
| { |
| GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); |
| |
| GST_DEBUG_OBJECT (interaudiosrc, "stop"); |
| |
| return TRUE; |
| } |
| |
| static gboolean |
| gst_inter_audio_src_prepare_seek_segment (GstBaseSrc * src, GstEvent * seek, |
| GstSegment * segment) |
| { |
| GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src); |
| |
| GST_DEBUG_OBJECT (interaudiosrc, "seek_segment"); |
| |
| return FALSE; |
| } |