/* GStreamer * Copyright (C) 2011 David Schleef * Copyright (C) 2014 Sebastian Dröge * * 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-decklinkaudiosink * @short_description: Outputs Audio to a BlackMagic DeckLink Device * @see_also: decklinkvideosink * * Playout Video and Audio to a BlackMagic DeckLink Device. Can only be used * in conjunction with decklinkvideosink. * * ## Sample pipeline * |[ * gst-launch-1.0 \ * videotestsrc ! decklinkvideosink device-number=0 mode=1080p25 \ * audiotestsrc ! decklinkaudiosink device-number=0 * ]| * Playout a 1080p25 test-video with a test-audio signal to the SDI-Out of Card 0. * Devices are numbered starting with 0. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstdecklinkaudiosink.h" #include "gstdecklinkvideosink.h" #include GST_DEBUG_CATEGORY_STATIC (gst_decklink_audio_sink_debug); #define GST_CAT_DEFAULT gst_decklink_audio_sink_debug #define DEFAULT_DEVICE_NUMBER (0) #define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND) #define DEFAULT_DISCONT_WAIT (1 * GST_SECOND) // Microseconds for audiobasesink compatibility... #define DEFAULT_BUFFER_TIME (50 * GST_MSECOND / 1000) #define DEFAULT_PERSISTENT_ID (-1) enum { PROP_0, PROP_DEVICE_NUMBER, PROP_HW_SERIAL_NUMBER, PROP_ALIGNMENT_THRESHOLD, PROP_DISCONT_WAIT, PROP_BUFFER_TIME, PROP_PERSISTENT_ID }; static void gst_decklink_audio_sink_set_property (GObject * object, guint property_id, const GValue * value, GParamSpec * pspec); static void gst_decklink_audio_sink_get_property (GObject * object, guint property_id, GValue * value, GParamSpec * pspec); static void gst_decklink_audio_sink_finalize (GObject * object); static GstStateChangeReturn gst_decklink_audio_sink_change_state (GstElement * element, GstStateChange transition); static GstClock *gst_decklink_audio_sink_provide_clock (GstElement * element); static GstCaps *gst_decklink_audio_sink_get_caps (GstBaseSink * bsink, GstCaps * filter); static gboolean gst_decklink_audio_sink_set_caps (GstBaseSink * bsink, GstCaps * caps); static GstFlowReturn gst_decklink_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer); static gboolean gst_decklink_audio_sink_open (GstBaseSink * bsink); static gboolean gst_decklink_audio_sink_close (GstBaseSink * bsink); static gboolean gst_decklink_audio_sink_stop (GstDecklinkAudioSink * self); static gboolean gst_decklink_audio_sink_unlock_stop (GstBaseSink * bsink); static void gst_decklink_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer, GstClockTime * start, GstClockTime * end); static gboolean gst_decklink_audio_sink_query (GstBaseSink * bsink, GstQuery * query); static gboolean gst_decklink_audio_sink_event (GstBaseSink * bsink, GstEvent * event); static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, format={S16LE,S32LE}, channels={2, 8, 16}, rate=48000, " "layout=interleaved") ); #define parent_class gst_decklink_audio_sink_parent_class G_DEFINE_TYPE (GstDecklinkAudioSink, gst_decklink_audio_sink, GST_TYPE_BASE_SINK); GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (decklinkaudiosink, "decklinkaudiosink", GST_RANK_NONE, GST_TYPE_DECKLINK_AUDIO_SINK, decklink_element_init (plugin)); static void gst_decklink_audio_sink_class_init (GstDecklinkAudioSinkClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstBaseSinkClass *basesink_class = GST_BASE_SINK_CLASS (klass); gobject_class->set_property = gst_decklink_audio_sink_set_property; gobject_class->get_property = gst_decklink_audio_sink_get_property; gobject_class->finalize = gst_decklink_audio_sink_finalize; element_class->change_state = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_change_state); element_class->provide_clock = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_provide_clock); basesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_get_caps); basesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_set_caps); basesink_class->render = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_render); // FIXME: These are misnamed in basesink! basesink_class->start = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_open); basesink_class->stop = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_close); basesink_class->unlock_stop = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_unlock_stop); basesink_class->get_times = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_get_times); basesink_class->query = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_query); basesink_class->event = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_event); g_object_class_install_property (gobject_class, PROP_DEVICE_NUMBER, g_param_spec_int ("device-number", "Device number", "Output device instance to use", 0, G_MAXINT, DEFAULT_DEVICE_NUMBER, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT))); /** * GstDecklinkAudioSink:persistent-id * * Decklink device to use. Higher priority than "device-number". * BMDDeckLinkPersistentID is a device specific, 32-bit unique identifier. * It is stable even when the device is plugged in a different connector, * across reboots, and when plugged into different computers. * * Since: 1.22 */ g_object_class_install_property (gobject_class, PROP_PERSISTENT_ID, g_param_spec_int64 ("persistent-id", "Persistent id", "Output device instance to use. Higher priority than \"device-number\".", DEFAULT_PERSISTENT_ID, G_MAXINT64, DEFAULT_PERSISTENT_ID, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT))); g_object_class_install_property (gobject_class, PROP_HW_SERIAL_NUMBER, g_param_spec_string ("hw-serial-number", "Hardware serial number", "The serial number (hardware ID) of the Decklink card", NULL, (GParamFlags) (G_PARAM_READABLE | G_PARAM_STATIC_STRINGS))); g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD, g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold", "Timestamp alignment threshold in nanoseconds", 0, G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_MUTABLE_READY))); g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT, g_param_spec_uint64 ("discont-wait", "Discont Wait", "Window of time in nanoseconds to wait before " "creating a discontinuity", 0, G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_MUTABLE_READY))); g_object_class_install_property (gobject_class, PROP_BUFFER_TIME, g_param_spec_uint64 ("buffer-time", "Buffer Time", "Size of audio buffer in microseconds, this is the minimum latency that the sink reports", 0, G_MAXUINT64, DEFAULT_BUFFER_TIME, (GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_MUTABLE_READY))); gst_element_class_add_static_pad_template (element_class, &sink_template); gst_element_class_set_static_metadata (element_class, "Decklink Audio Sink", "Audio/Sink/Hardware", "Decklink Sink", "David Schleef , " "Sebastian Dröge "); GST_DEBUG_CATEGORY_INIT (gst_decklink_audio_sink_debug, "decklinkaudiosink", 0, "debug category for decklinkaudiosink element"); } static void gst_decklink_audio_sink_init (GstDecklinkAudioSink * self) { self->device_number = DEFAULT_DEVICE_NUMBER; self->stream_align = gst_audio_stream_align_new (48000, DEFAULT_ALIGNMENT_THRESHOLD, DEFAULT_DISCONT_WAIT); self->buffer_time = DEFAULT_BUFFER_TIME * 1000; self->persistent_id = DEFAULT_PERSISTENT_ID; gst_base_sink_set_max_lateness (GST_BASE_SINK_CAST (self), 20 * GST_MSECOND); } void gst_decklink_audio_sink_set_property (GObject * object, guint property_id, const GValue * value, GParamSpec * pspec) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (object); switch (property_id) { case PROP_DEVICE_NUMBER: self->device_number = g_value_get_int (value); break; case PROP_ALIGNMENT_THRESHOLD: GST_OBJECT_LOCK (self); gst_audio_stream_align_set_alignment_threshold (self->stream_align, g_value_get_uint64 (value)); GST_OBJECT_UNLOCK (self); break; case PROP_DISCONT_WAIT: GST_OBJECT_LOCK (self); gst_audio_stream_align_set_discont_wait (self->stream_align, g_value_get_uint64 (value)); GST_OBJECT_UNLOCK (self); break; case PROP_BUFFER_TIME: GST_OBJECT_LOCK (self); self->buffer_time = g_value_get_uint64 (value) * 1000; GST_OBJECT_UNLOCK (self); break; case PROP_PERSISTENT_ID: self->persistent_id = g_value_get_int64 (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec); break; } } void gst_decklink_audio_sink_get_property (GObject * object, guint property_id, GValue * value, GParamSpec * pspec) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (object); switch (property_id) { case PROP_DEVICE_NUMBER: g_value_set_int (value, self->device_number); break; case PROP_HW_SERIAL_NUMBER: if (self->output) g_value_set_string (value, self->output->hw_serial_number); else g_value_set_string (value, NULL); break; case PROP_ALIGNMENT_THRESHOLD: GST_OBJECT_LOCK (self); g_value_set_uint64 (value, gst_audio_stream_align_get_alignment_threshold (self->stream_align)); GST_OBJECT_UNLOCK (self); break; case PROP_DISCONT_WAIT: GST_OBJECT_LOCK (self); g_value_set_uint64 (value, gst_audio_stream_align_get_discont_wait (self->stream_align)); GST_OBJECT_UNLOCK (self); break; case PROP_BUFFER_TIME: GST_OBJECT_LOCK (self); g_value_set_uint64 (value, self->buffer_time / 1000); GST_OBJECT_UNLOCK (self); break; case PROP_PERSISTENT_ID: g_value_set_int64 (value, self->persistent_id); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec); break; } } void gst_decklink_audio_sink_finalize (GObject * object) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (object); if (self->stream_align) { gst_audio_stream_align_free (self->stream_align); self->stream_align = NULL; } G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_decklink_audio_sink_set_caps (GstBaseSink * bsink, GstCaps * caps) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink); HRESULT ret; BMDAudioSampleType sample_depth; GstAudioInfo info; GST_DEBUG_OBJECT (self, "Setting caps %" GST_PTR_FORMAT, caps); if (!gst_audio_info_from_caps (&info, caps)) return FALSE; if (self->output->audio_enabled && (self->info.finfo->format != info.finfo->format || self->info.channels != info.channels)) { GST_ERROR_OBJECT (self, "Reconfiguration not supported"); return FALSE; } else if (self->output->audio_enabled) { return TRUE; } if (info.finfo->format == GST_AUDIO_FORMAT_S16LE) { sample_depth = bmdAudioSampleType16bitInteger; } else { sample_depth = bmdAudioSampleType32bitInteger; } ret = self->output->output->EnableAudioOutput (bmdAudioSampleRate48kHz, sample_depth, info.channels, bmdAudioOutputStreamContinuous); if (ret != S_OK) { GST_WARNING_OBJECT (self, "Failed to enable audio output 0x%08lx", (unsigned long) ret); return FALSE; } self->output->audio_enabled = TRUE; self->info = info; // Create a new resampler as needed if (self->resampler) gst_audio_resampler_free (self->resampler); self->resampler = NULL; return TRUE; } static GstCaps * gst_decklink_audio_sink_get_caps (GstBaseSink * bsink, GstCaps * filter) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink); GstCaps *caps; if ((caps = gst_pad_get_current_caps (GST_BASE_SINK_PAD (bsink)))) return caps; caps = gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD (bsink)); GST_OBJECT_LOCK (self); if (self->output && self->output->attributes) { int64_t max_channels = 0; HRESULT ret; GstStructure *s; GValue arr = G_VALUE_INIT; GValue v = G_VALUE_INIT; ret = self->output->attributes->GetInt (BMDDeckLinkMaximumAudioChannels, &max_channels); /* 2 should always be supported */ if (ret != S_OK) { max_channels = 2; } caps = gst_caps_make_writable (caps); s = gst_caps_get_structure (caps, 0); g_value_init (&arr, GST_TYPE_LIST); g_value_init (&v, G_TYPE_INT); if (max_channels >= 16) { g_value_set_int (&v, 16); gst_value_list_append_value (&arr, &v); } if (max_channels >= 8) { g_value_set_int (&v, 8); gst_value_list_append_value (&arr, &v); } g_value_set_int (&v, 2); gst_value_list_append_value (&arr, &v); gst_structure_set_value (s, "channels", &arr); g_value_unset (&v); g_value_unset (&arr); } GST_OBJECT_UNLOCK (self); if (filter) { GstCaps *intersection = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (caps); caps = intersection; } return caps; } static gboolean gst_decklink_audio_sink_query (GstBaseSink * bsink, GstQuery * query) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK (bsink); gboolean res = FALSE; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_LATENCY: { gboolean live, us_live; GstClockTime min_l, max_l; GST_DEBUG_OBJECT (self, "latency query"); /* ask parent first, it will do an upstream query for us. */ if ((res = gst_base_sink_query_latency (GST_BASE_SINK_CAST (self), &live, &us_live, &min_l, &max_l))) { GstClockTime base_latency, min_latency, max_latency; /* we and upstream are both live, adjust the min_latency */ if (live && us_live) { GST_OBJECT_LOCK (self); if (!self->info.rate) { GST_OBJECT_UNLOCK (self); GST_DEBUG_OBJECT (self, "we are not negotiated, can't report latency yet"); res = FALSE; goto done; } base_latency = self->buffer_time * 1000; GST_OBJECT_UNLOCK (self); /* we cannot go lower than the buffer size and the min peer latency */ min_latency = base_latency + min_l; /* the max latency is the max of the peer, we can delay an infinite * amount of time. */ max_latency = (max_l == GST_CLOCK_TIME_NONE) ? GST_CLOCK_TIME_NONE : (base_latency + max_l); GST_DEBUG_OBJECT (self, "peer min %" GST_TIME_FORMAT ", our min latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (min_l), GST_TIME_ARGS (min_latency)); GST_DEBUG_OBJECT (self, "peer max %" GST_TIME_FORMAT ", our max latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (max_l), GST_TIME_ARGS (max_latency)); } else { GST_DEBUG_OBJECT (self, "peer or we are not live, don't care about latency"); min_latency = min_l; max_latency = max_l; } gst_query_set_latency (query, live, min_latency, max_latency); } break; } default: res = GST_BASE_SINK_CLASS (parent_class)->query (bsink, query); break; } done: return res; } static gboolean gst_decklink_audio_sink_event (GstBaseSink * bsink, GstEvent * event) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink); if (GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT) { const GstSegment *new_segment; gst_event_parse_segment (event, &new_segment); if (ABS (new_segment->rate) != 1.0) { guint out_rate = self->info.rate / ABS (new_segment->rate); if (self->resampler && (self->resampler_out_rate != out_rate || self->resampler_in_rate != (guint) self->info.rate)) gst_audio_resampler_update (self->resampler, self->info.rate, out_rate, NULL); else if (!self->resampler) self->resampler = gst_audio_resampler_new (GST_AUDIO_RESAMPLER_METHOD_LINEAR, GST_AUDIO_RESAMPLER_FLAG_NONE, self->info.finfo->format, self->info.channels, self->info.rate, out_rate, NULL); self->resampler_in_rate = self->info.rate; self->resampler_out_rate = out_rate; } else if (self->resampler) { gst_audio_resampler_free (self->resampler); self->resampler = NULL; } if (new_segment->rate < 0) gst_audio_stream_align_set_rate (self->stream_align, -48000); } return GST_BASE_SINK_CLASS (parent_class)->event (bsink, event); } static GstFlowReturn gst_decklink_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink); GstDecklinkVideoSink *video_sink; GstFlowReturn flow_ret; HRESULT ret; GstClockTime timestamp, duration; GstClockTime running_time, running_time_duration; GstClockTime schedule_time, schedule_time_duration; GstClockTime latency, render_delay; GstClockTimeDiff ts_offset; GstMapInfo map_info; const guint8 *data; gsize len, written_all; gboolean discont; GST_DEBUG_OBJECT (self, "Rendering buffer %p", buffer); // FIXME: Handle no timestamps if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) { return GST_FLOW_ERROR; } if (GST_BASE_SINK_CAST (self)->flushing) { return GST_FLOW_FLUSHING; } // If we're called before output is actually started, start pre-rolling if (!self->output->started) { self->output->output->BeginAudioPreroll (); } video_sink = GST_DECKLINK_VIDEO_SINK (gst_object_ref (self->output->videosink)); timestamp = GST_BUFFER_TIMESTAMP (buffer); duration = GST_BUFFER_DURATION (buffer); discont = gst_audio_stream_align_process (self->stream_align, GST_BUFFER_IS_DISCONT (buffer), timestamp, gst_buffer_get_size (buffer) / self->info.bpf, ×tamp, &duration, NULL); if (discont && self->resampler) gst_audio_resampler_reset (self->resampler); if (GST_BASE_SINK_CAST (self)->segment.rate < 0.0) { GstMapInfo out_map; gint out_frames = gst_buffer_get_size (buffer) / self->info.bpf; buffer = gst_buffer_make_writable (gst_buffer_ref (buffer)); gst_buffer_map (buffer, &out_map, GST_MAP_READWRITE); if (self->info.finfo->format == GST_AUDIO_FORMAT_S16) { gint16 *swap_data = (gint16 *) out_map.data; gint16 *swap_data_end = swap_data + (out_frames - 1) * self->info.channels; gint16 swap_tmp[16]; while (out_frames > 0) { memcpy (&swap_tmp, swap_data, self->info.bpf); memcpy (swap_data, swap_data_end, self->info.bpf); memcpy (swap_data_end, &swap_tmp, self->info.bpf); swap_data += self->info.channels; swap_data_end -= self->info.channels; out_frames -= 2; } } else { gint32 *swap_data = (gint32 *) out_map.data; gint32 *swap_data_end = swap_data + (out_frames - 1) * self->info.channels; gint32 swap_tmp[16]; while (out_frames > 0) { memcpy (&swap_tmp, swap_data, self->info.bpf); memcpy (swap_data, swap_data_end, self->info.bpf); memcpy (swap_data_end, &swap_tmp, self->info.bpf); swap_data += self->info.channels; swap_data_end -= self->info.channels; out_frames -= 2; } } gst_buffer_unmap (buffer, &out_map); } else { gst_buffer_ref (buffer); } if (self->resampler) { gint in_frames = gst_buffer_get_size (buffer) / self->info.bpf; gint out_frames = gst_audio_resampler_get_out_frames (self->resampler, in_frames); GstBuffer *out_buf = gst_buffer_new_and_alloc (out_frames * self->info.bpf); GstMapInfo out_map; gst_buffer_map (buffer, &map_info, GST_MAP_READ); gst_buffer_map (out_buf, &out_map, GST_MAP_READWRITE); gst_audio_resampler_resample (self->resampler, (gpointer *) & map_info.data, in_frames, (gpointer *) & out_map.data, out_frames); gst_buffer_unmap (out_buf, &out_map); gst_buffer_unmap (buffer, &map_info); buffer = out_buf; } gst_buffer_map (buffer, &map_info, GST_MAP_READ); data = map_info.data; len = map_info.size / self->info.bpf; written_all = 0; do { GstClockTime timestamp_now = timestamp + gst_util_uint64_scale (written_all, GST_SECOND, self->info.rate); guint32 buffered_samples; GstClockTime buffered_time; guint32 written = 0; GstClock *clock; GstClockTimeDiff clock_ahead; if (GST_BASE_SINK_CAST (self)->flushing) { flow_ret = GST_FLOW_FLUSHING; break; } running_time = gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment, GST_FORMAT_TIME, timestamp_now); running_time_duration = gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment, GST_FORMAT_TIME, timestamp_now + duration) - running_time; /* See gst_base_sink_adjust_time() */ latency = gst_base_sink_get_latency (bsink); render_delay = gst_base_sink_get_render_delay (bsink); ts_offset = gst_base_sink_get_ts_offset (bsink); running_time += latency; if (ts_offset < 0) { ts_offset = -ts_offset; if ((GstClockTime) ts_offset < running_time) running_time -= ts_offset; else running_time = 0; } else { running_time += ts_offset; } if (running_time > render_delay) running_time -= render_delay; else running_time = 0; clock = gst_element_get_clock (GST_ELEMENT_CAST (self)); clock_ahead = 0; if (clock) { GstClockTime clock_now = gst_clock_get_time (clock); GstClockTime base_time = gst_element_get_base_time (GST_ELEMENT_CAST (self)); gst_object_unref (clock); clock = NULL; if (clock_now != GST_CLOCK_TIME_NONE && base_time != GST_CLOCK_TIME_NONE) { GST_DEBUG_OBJECT (self, "Clock time %" GST_TIME_FORMAT ", base time %" GST_TIME_FORMAT ", target running time %" GST_TIME_FORMAT, GST_TIME_ARGS (clock_now), GST_TIME_ARGS (base_time), GST_TIME_ARGS (running_time)); if (clock_now > base_time) clock_now -= base_time; else clock_now = 0; clock_ahead = running_time - clock_now; } } GST_DEBUG_OBJECT (self, "Ahead %" GST_STIME_FORMAT " of the clock running time", GST_STIME_ARGS (clock_ahead)); if (self->output-> output->GetBufferedAudioSampleFrameCount (&buffered_samples) != S_OK) buffered_samples = 0; buffered_time = gst_util_uint64_scale (buffered_samples, GST_SECOND, self->info.rate); buffered_time /= ABS (GST_BASE_SINK_CAST (self)->segment.rate); GST_DEBUG_OBJECT (self, "Buffered %" GST_TIME_FORMAT " in the driver (%u samples)", GST_TIME_ARGS (buffered_time), buffered_samples); { GstClockTimeDiff buffered_ahead_of_clock_ahead = GST_CLOCK_DIFF (clock_ahead, buffered_time); GST_DEBUG_OBJECT (self, "driver is %" GST_STIME_FORMAT " ahead of the " "expected clock", GST_STIME_ARGS (buffered_ahead_of_clock_ahead)); /* we don't want to store too much data in the driver as decklink * doesn't seem to actually use our provided timestamps to perform its * own synchronisation. It seems to count samples instead. */ /* FIXME: do we need to split buffers? */ if (buffered_ahead_of_clock_ahead > 0 && buffered_ahead_of_clock_ahead > gst_base_sink_get_max_lateness (bsink)) { GST_DEBUG_OBJECT (self, "Dropping buffer that is %" GST_STIME_FORMAT " too late", GST_STIME_ARGS (buffered_ahead_of_clock_ahead)); if (self->resampler) gst_audio_resampler_reset (self->resampler); flow_ret = GST_FLOW_OK; break; } } // We start waiting once we have more than buffer-time buffered if (((GstClockTime) clock_ahead) > self->buffer_time) { GstClockReturn clock_ret; GstClockTime wait_time = running_time; GST_DEBUG_OBJECT (self, "Buffered enough, wait for preroll or the clock or flushing. " "Configured buffer time: %" GST_TIME_FORMAT, GST_TIME_ARGS (self->buffer_time)); if (wait_time < self->buffer_time) wait_time = 0; else wait_time -= self->buffer_time; flow_ret = gst_base_sink_do_preroll (GST_BASE_SINK_CAST (self), GST_MINI_OBJECT_CAST (buffer)); if (flow_ret != GST_FLOW_OK) break; clock_ret = gst_base_sink_wait_clock (GST_BASE_SINK_CAST (self), wait_time, NULL); if (GST_BASE_SINK_CAST (self)->flushing) { flow_ret = GST_FLOW_FLUSHING; break; } // Rerun the whole loop again if (clock_ret == GST_CLOCK_UNSCHEDULED) continue; } schedule_time = running_time; schedule_time_duration = running_time_duration; gst_decklink_video_sink_convert_to_internal_clock (video_sink, &schedule_time, &schedule_time_duration); GST_LOG_OBJECT (self, "Scheduling audio samples at %" GST_TIME_FORMAT " with duration %" GST_TIME_FORMAT, GST_TIME_ARGS (schedule_time), GST_TIME_ARGS (schedule_time_duration)); ret = self->output->output->ScheduleAudioSamples ((void *) data, len, schedule_time, GST_SECOND, &written); if (ret != S_OK) { bool is_running = true; self->output->output->IsScheduledPlaybackRunning (&is_running); if (is_running && !GST_BASE_SINK_CAST (self)->flushing && self->output->started) { GST_ELEMENT_ERROR (self, STREAM, FAILED, (NULL), ("Failed to schedule frame: 0x%08lx", (unsigned long) ret)); flow_ret = GST_FLOW_ERROR; break; } else { // Ignore the error and go out of the loop here, we're shutting down // or are not started yet and there's nothing we can do at this point GST_INFO_OBJECT (self, "Ignoring scheduling error 0x%08x because we're not started yet" " or not anymore", (guint) ret); flow_ret = GST_FLOW_OK; break; } } len -= written; data += written * self->info.bpf; if (self->resampler) written_all += written * ABS (GST_BASE_SINK_CAST (self)->segment.rate); else written_all += written; flow_ret = GST_FLOW_OK; } while (len > 0); gst_buffer_unmap (buffer, &map_info); gst_buffer_unref (buffer); GST_DEBUG_OBJECT (self, "Returning %s", gst_flow_get_name (flow_ret)); return flow_ret; } static gboolean gst_decklink_audio_sink_open (GstBaseSink * bsink) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink); GST_DEBUG_OBJECT (self, "Starting"); self->output = gst_decklink_acquire_nth_output (self->device_number, self->persistent_id, GST_ELEMENT_CAST (self), TRUE); if (!self->output) { GST_ERROR_OBJECT (self, "Failed to acquire output"); return FALSE; } g_object_notify (G_OBJECT (self), "hw-serial-number"); return TRUE; } static gboolean gst_decklink_audio_sink_close (GstBaseSink * bsink) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink); GST_DEBUG_OBJECT (self, "Closing"); if (self->output) { g_mutex_lock (&self->output->lock); self->output->mode = NULL; self->output->audio_enabled = FALSE; if (self->output->start_scheduled_playback && self->output->videosink) self->output->start_scheduled_playback (self->output->videosink); g_mutex_unlock (&self->output->lock); self->output->output->DisableAudioOutput (); gst_decklink_release_nth_output (self->device_number, self->persistent_id, GST_ELEMENT_CAST (self), TRUE); self->output = NULL; } return TRUE; } static gboolean gst_decklink_audio_sink_stop (GstDecklinkAudioSink * self) { GST_DEBUG_OBJECT (self, "Stopping"); if (self->output && self->output->audio_enabled) { g_mutex_lock (&self->output->lock); self->output->audio_enabled = FALSE; g_mutex_unlock (&self->output->lock); self->output->output->DisableAudioOutput (); } if (self->resampler) { gst_audio_resampler_free (self->resampler); self->resampler = NULL; } return TRUE; } static gboolean gst_decklink_audio_sink_unlock_stop (GstBaseSink * bsink) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK (bsink); if (self->output) { self->output->output->FlushBufferedAudioSamples (); } return TRUE; } static void gst_decklink_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer, GstClockTime * start, GstClockTime * end) { /* our clock sync is a bit too much for the base class to handle so * we implement it ourselves. */ *start = GST_CLOCK_TIME_NONE; *end = GST_CLOCK_TIME_NONE; } static GstStateChangeReturn gst_decklink_audio_sink_change_state (GstElement * element, GstStateChange transition) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (element); GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: GST_OBJECT_LOCK (self); gst_audio_stream_align_mark_discont (self->stream_align); GST_OBJECT_UNLOCK (self); g_mutex_lock (&self->output->lock); if (self->output->start_scheduled_playback) self->output->start_scheduled_playback (self->output->videosink); g_mutex_unlock (&self->output->lock); break; case GST_STATE_CHANGE_PAUSED_TO_READY: gst_decklink_audio_sink_stop (self); break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); if (ret == GST_STATE_CHANGE_FAILURE) return ret; switch (transition) { default: break; } return ret; } static GstClock * gst_decklink_audio_sink_provide_clock (GstElement * element) { GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (element); if (!self->output) return NULL; return GST_CLOCK_CAST (gst_object_ref (self->output->clock)); }