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33ae846607
This function might be revisited with different channel position mapping while audio source goes into play so the reorder flag needs to be reset before the checks happen.
2112 lines
56 KiB
C
2112 lines
56 KiB
C
/* GStreamer
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* Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstaudioringbuffer
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* @title: GstAudioRingBuffer
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* @short_description: Base class for audio ringbuffer implementations
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* @see_also: #GstAudioBaseSink, #GstAudioSink
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*
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* This object is the base class for audio ringbuffers used by the base
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* audio source and sink classes.
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*
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* The ringbuffer abstracts a circular buffer of data. One reader and
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* one writer can operate on the data from different threads in a lockfree
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* manner. The base class is sufficiently flexible to be used as an
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* abstraction for DMA based ringbuffers as well as a pure software
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* implementations.
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <gst/audio/audio.h>
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#include "gstaudioringbuffer.h"
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GST_DEBUG_CATEGORY_STATIC (gst_audio_ring_buffer_debug);
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#define GST_CAT_DEFAULT gst_audio_ring_buffer_debug
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static void gst_audio_ring_buffer_dispose (GObject * object);
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static void gst_audio_ring_buffer_finalize (GObject * object);
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static gboolean gst_audio_ring_buffer_pause_unlocked (GstAudioRingBuffer * buf);
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static void default_clear_all (GstAudioRingBuffer * buf);
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static guint default_commit (GstAudioRingBuffer * buf, guint64 * sample,
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guint8 * data, gint in_samples, gint out_samples, gint * accum);
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/* ringbuffer abstract base class */
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G_DEFINE_ABSTRACT_TYPE (GstAudioRingBuffer, gst_audio_ring_buffer,
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GST_TYPE_OBJECT);
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static void
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gst_audio_ring_buffer_class_init (GstAudioRingBufferClass * klass)
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{
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GObjectClass *gobject_class;
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GstAudioRingBufferClass *gstaudioringbuffer_class;
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gobject_class = (GObjectClass *) klass;
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gstaudioringbuffer_class = (GstAudioRingBufferClass *) klass;
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GST_DEBUG_CATEGORY_INIT (gst_audio_ring_buffer_debug, "ringbuffer", 0,
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"ringbuffer class");
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gobject_class->dispose = gst_audio_ring_buffer_dispose;
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gobject_class->finalize = gst_audio_ring_buffer_finalize;
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gstaudioringbuffer_class->clear_all = GST_DEBUG_FUNCPTR (default_clear_all);
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gstaudioringbuffer_class->commit = GST_DEBUG_FUNCPTR (default_commit);
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}
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static void
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gst_audio_ring_buffer_init (GstAudioRingBuffer * ringbuffer)
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{
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ringbuffer->open = FALSE;
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ringbuffer->acquired = FALSE;
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ringbuffer->state = GST_AUDIO_RING_BUFFER_STATE_STOPPED;
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g_cond_init (&ringbuffer->cond);
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ringbuffer->waiting = 0;
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ringbuffer->empty_seg = NULL;
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ringbuffer->flushing = TRUE;
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ringbuffer->segbase = 0;
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ringbuffer->segdone = 0;
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}
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static void
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gst_audio_ring_buffer_dispose (GObject * object)
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{
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GstAudioRingBuffer *ringbuffer = GST_AUDIO_RING_BUFFER (object);
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gst_caps_replace (&ringbuffer->spec.caps, NULL);
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G_OBJECT_CLASS (gst_audio_ring_buffer_parent_class)->dispose (G_OBJECT
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(ringbuffer));
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}
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static void
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gst_audio_ring_buffer_finalize (GObject * object)
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{
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GstAudioRingBuffer *ringbuffer = GST_AUDIO_RING_BUFFER (object);
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g_cond_clear (&ringbuffer->cond);
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g_free (ringbuffer->empty_seg);
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if (ringbuffer->cb_data_notify != NULL)
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ringbuffer->cb_data_notify (ringbuffer->cb_data);
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G_OBJECT_CLASS (gst_audio_ring_buffer_parent_class)->finalize (G_OBJECT
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(ringbuffer));
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}
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#ifndef GST_DISABLE_GST_DEBUG
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static const gchar *format_type_names[] = {
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"raw",
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"mu law",
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"a law",
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"ima adpcm",
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"mpeg",
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"gsm",
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"iec958",
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"ac3",
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"eac3",
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"dts",
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"aac mpeg2",
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"aac mpeg4",
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"aac mpeg2 raw",
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"aac mpeg4 raw",
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"flac"
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};
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#endif
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/**
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* gst_audio_ring_buffer_debug_spec_caps:
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* @spec: the spec to debug
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*
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* Print debug info about the parsed caps in @spec to the debug log.
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*/
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void
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gst_audio_ring_buffer_debug_spec_caps (GstAudioRingBufferSpec * spec)
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{
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#if 0
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gint i, bytes;
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#endif
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GST_DEBUG ("spec caps: %p %" GST_PTR_FORMAT, spec->caps, spec->caps);
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GST_DEBUG ("parsed caps: type: %d, '%s'", spec->type,
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format_type_names[spec->type]);
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#if 0
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GST_DEBUG ("parsed caps: width: %d", spec->width);
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GST_DEBUG ("parsed caps: sign: %d", spec->sign);
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GST_DEBUG ("parsed caps: bigend: %d", spec->bigend);
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GST_DEBUG ("parsed caps: rate: %d", spec->rate);
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GST_DEBUG ("parsed caps: channels: %d", spec->channels);
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GST_DEBUG ("parsed caps: sample bytes: %d", spec->bytes_per_sample);
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bytes = (spec->width >> 3) * spec->channels;
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for (i = 0; i < bytes; i++) {
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GST_DEBUG ("silence byte %d: %02x", i, spec->silence_sample[i]);
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}
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#endif
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}
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/**
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* gst_audio_ring_buffer_debug_spec_buff:
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* @spec: the spec to debug
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*
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* Print debug info about the buffer sized in @spec to the debug log.
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*/
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void
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gst_audio_ring_buffer_debug_spec_buff (GstAudioRingBufferSpec * spec)
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{
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gint bpf = GST_AUDIO_INFO_BPF (&spec->info);
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GST_DEBUG ("acquire ringbuffer: buffer time: %" G_GINT64_FORMAT " usec",
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spec->buffer_time);
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GST_DEBUG ("acquire ringbuffer: latency time: %" G_GINT64_FORMAT " usec",
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spec->latency_time);
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GST_DEBUG ("acquire ringbuffer: total segments: %d", spec->segtotal);
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GST_DEBUG ("acquire ringbuffer: latency segments: %d", spec->seglatency);
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GST_DEBUG ("acquire ringbuffer: segment size: %d bytes = %d samples",
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spec->segsize, (bpf != 0) ? (spec->segsize / bpf) : -1);
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GST_DEBUG ("acquire ringbuffer: buffer size: %d bytes = %d samples",
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spec->segsize * spec->segtotal,
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(bpf != 0) ? (spec->segsize * spec->segtotal / bpf) : -1);
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}
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/**
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* gst_audio_ring_buffer_parse_caps:
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* @spec: a spec
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* @caps: a #GstCaps
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*
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* Parse @caps into @spec.
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*
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* Returns: TRUE if the caps could be parsed.
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*/
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gboolean
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gst_audio_ring_buffer_parse_caps (GstAudioRingBufferSpec * spec, GstCaps * caps)
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{
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const gchar *mimetype;
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GstStructure *structure;
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gint i;
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GstAudioInfo info;
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structure = gst_caps_get_structure (caps, 0);
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gst_audio_info_init (&info);
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/* we have to differentiate between int and float formats */
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mimetype = gst_structure_get_name (structure);
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if (g_str_equal (mimetype, "audio/x-raw")) {
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if (!gst_audio_info_from_caps (&info, caps))
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goto parse_error;
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spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW;
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} else if (g_str_equal (mimetype, "audio/x-alaw")) {
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/* extract the needed information from the cap */
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if (!(gst_structure_get_int (structure, "rate", &info.rate) &&
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gst_structure_get_int (structure, "channels", &info.channels)))
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goto parse_error;
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if (!(gst_audio_channel_positions_from_mask (info.channels, 0,
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info.position)))
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goto parse_error;
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spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW;
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info.bpf = info.channels;
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} else if (g_str_equal (mimetype, "audio/x-mulaw")) {
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/* extract the needed information from the cap */
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if (!(gst_structure_get_int (structure, "rate", &info.rate) &&
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gst_structure_get_int (structure, "channels", &info.channels)))
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goto parse_error;
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if (!(gst_audio_channel_positions_from_mask (info.channels, 0,
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info.position)))
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goto parse_error;
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spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW;
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info.bpf = info.channels;
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} else if (g_str_equal (mimetype, "audio/x-iec958")) {
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/* extract the needed information from the cap */
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if (!(gst_structure_get_int (structure, "rate", &info.rate)))
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goto parse_error;
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spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958;
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info.bpf = 4;
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} else if (g_str_equal (mimetype, "audio/x-ac3")) {
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/* extract the needed information from the cap */
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if (!(gst_structure_get_int (structure, "rate", &info.rate)))
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goto parse_error;
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gst_structure_get_int (structure, "channels", &info.channels);
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spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3;
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info.bpf = 4;
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} else if (g_str_equal (mimetype, "audio/x-eac3")) {
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/* extract the needed information from the cap */
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if (!(gst_structure_get_int (structure, "rate", &info.rate)))
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goto parse_error;
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gst_structure_get_int (structure, "channels", &info.channels);
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spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3;
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info.bpf = 16;
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} else if (g_str_equal (mimetype, "audio/x-dts")) {
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/* extract the needed information from the cap */
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if (!(gst_structure_get_int (structure, "rate", &info.rate)))
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goto parse_error;
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gst_structure_get_int (structure, "channels", &info.channels);
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spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS;
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info.bpf = 4;
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} else if (g_str_equal (mimetype, "audio/mpeg") &&
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gst_structure_get_int (structure, "mpegaudioversion", &i) &&
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(i == 1 || i == 2 || i == 3)) {
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/* Now we know this is MPEG-1, MPEG-2 or MPEG-2.5 (non AAC) */
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/* extract the needed information from the cap */
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if (!(gst_structure_get_int (structure, "rate", &info.rate)))
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goto parse_error;
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gst_structure_get_int (structure, "channels", &info.channels);
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spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG;
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info.bpf = 1;
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} else if (g_str_equal (mimetype, "audio/mpeg") &&
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gst_structure_get_int (structure, "mpegversion", &i) &&
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(i == 2 || i == 4) &&
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(!g_strcmp0 (gst_structure_get_string (structure, "stream-format"),
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"adts")
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|| !g_strcmp0 (gst_structure_get_string (structure, "stream-format"),
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"raw"))) {
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/* MPEG-2 AAC or MPEG-4 AAC */
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if (!(gst_structure_get_int (structure, "rate", &info.rate)))
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goto parse_error;
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gst_structure_get_int (structure, "channels", &info.channels);
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if (!g_strcmp0 (gst_structure_get_string (structure, "stream-format"),
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"adts"))
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spec->type = (i == 2) ? GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC :
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC;
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else
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spec->type = (i == 2) ?
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW :
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GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW;
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info.bpf = 1;
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} else if (g_str_equal (mimetype, "audio/x-flac")) {
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/* extract the needed information from the cap */
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if (!(gst_structure_get_int (structure, "rate", &info.rate)))
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goto parse_error;
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gst_structure_get_int (structure, "channels", &info.channels);
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spec->type = GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC;
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info.bpf = 1;
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} else {
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goto parse_error;
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}
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gst_caps_replace (&spec->caps, caps);
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g_return_val_if_fail (spec->latency_time != 0, FALSE);
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/* calculate suggested segsize and segtotal. segsize should be one unit
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* of 'latency_time' samples, scaling for the fact that latency_time is
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* currently stored in microseconds (FIXME: in 0.11) */
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spec->segsize = gst_util_uint64_scale (info.rate * info.bpf,
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spec->latency_time, GST_SECOND / GST_USECOND);
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/* Round to an integer number of samples */
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spec->segsize -= spec->segsize % info.bpf;
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spec->segtotal = spec->buffer_time / spec->latency_time;
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/* leave the latency undefined now, implementations can change it but if it's
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* not changed, we assume the same value as segtotal */
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spec->seglatency = -1;
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spec->info = info;
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gst_audio_ring_buffer_debug_spec_caps (spec);
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gst_audio_ring_buffer_debug_spec_buff (spec);
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return TRUE;
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/* ERRORS */
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parse_error:
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{
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GST_DEBUG ("could not parse caps");
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return FALSE;
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}
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}
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|
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/**
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* gst_audio_ring_buffer_convert:
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* @buf: the #GstAudioRingBuffer
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* @src_fmt: the source format
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* @src_val: the source value
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* @dest_fmt: the destination format
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* @dest_val: (out): a location to store the converted value
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*
|
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* Convert @src_val in @src_fmt to the equivalent value in @dest_fmt. The result
|
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* will be put in @dest_val.
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*
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* Returns: TRUE if the conversion succeeded.
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*/
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gboolean
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gst_audio_ring_buffer_convert (GstAudioRingBuffer * buf,
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GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val)
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{
|
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gboolean res;
|
|
|
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GST_OBJECT_LOCK (buf);
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res =
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gst_audio_info_convert (&buf->spec.info, src_fmt, src_val, dest_fmt,
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dest_val);
|
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GST_OBJECT_UNLOCK (buf);
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|
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return res;
|
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}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_set_callback: (skip)
|
|
* @buf: the #GstAudioRingBuffer to set the callback on
|
|
* @cb: (allow-none): the callback to set
|
|
* @user_data: user data passed to the callback
|
|
*
|
|
* Sets the given callback function on the buffer. This function
|
|
* will be called every time a segment has been written to a device.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
void
|
|
gst_audio_ring_buffer_set_callback (GstAudioRingBuffer * buf,
|
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GstAudioRingBufferCallback cb, gpointer user_data)
|
|
{
|
|
gst_audio_ring_buffer_set_callback_full (buf, cb, user_data, NULL);
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_set_callback_full: (rename-to gst_audio_ring_buffer_set_callback)
|
|
* @buf: the #GstAudioRingBuffer to set the callback on
|
|
* @cb: (allow-none): the callback to set
|
|
* @user_data: user data passed to the callback
|
|
* @notify: function to be called when @user_data is no longer needed
|
|
*
|
|
* Sets the given callback function on the buffer. This function
|
|
* will be called every time a segment has been written to a device.
|
|
*
|
|
* MT safe.
|
|
*
|
|
* Since: 1.12
|
|
*/
|
|
void
|
|
gst_audio_ring_buffer_set_callback_full (GstAudioRingBuffer * buf,
|
|
GstAudioRingBufferCallback cb, gpointer user_data, GDestroyNotify notify)
|
|
{
|
|
gpointer old_data = NULL;
|
|
GDestroyNotify old_notify;
|
|
|
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g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
|
|
|
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GST_OBJECT_LOCK (buf);
|
|
old_notify = buf->cb_data_notify;
|
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old_data = buf->cb_data;
|
|
|
|
buf->callback = cb;
|
|
buf->cb_data = user_data;
|
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buf->cb_data_notify = notify;
|
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GST_OBJECT_UNLOCK (buf);
|
|
|
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if (old_notify) {
|
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old_notify (old_data);
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_open_device:
|
|
* @buf: the #GstAudioRingBuffer
|
|
*
|
|
* Open the audio device associated with the ring buffer. Does not perform any
|
|
* setup on the device. You must open the device before acquiring the ring
|
|
* buffer.
|
|
*
|
|
* Returns: TRUE if the device could be opened, FALSE on error.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
gboolean
|
|
gst_audio_ring_buffer_open_device (GstAudioRingBuffer * buf)
|
|
{
|
|
gboolean res = TRUE;
|
|
GstAudioRingBufferClass *rclass;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
|
|
|
|
GST_DEBUG_OBJECT (buf, "opening device");
|
|
|
|
GST_OBJECT_LOCK (buf);
|
|
if (G_UNLIKELY (buf->open))
|
|
goto was_opened;
|
|
|
|
buf->open = TRUE;
|
|
|
|
/* if this fails, something is wrong in this file */
|
|
g_assert (!buf->acquired);
|
|
|
|
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
|
|
if (G_LIKELY (rclass->open_device))
|
|
res = rclass->open_device (buf);
|
|
|
|
if (G_UNLIKELY (!res))
|
|
goto open_failed;
|
|
|
|
GST_DEBUG_OBJECT (buf, "opened device");
|
|
|
|
done:
|
|
GST_OBJECT_UNLOCK (buf);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
was_opened:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "Device for ring buffer already open");
|
|
g_warning ("Device for ring buffer %p already open, fix your code", buf);
|
|
res = TRUE;
|
|
goto done;
|
|
}
|
|
open_failed:
|
|
{
|
|
buf->open = FALSE;
|
|
GST_DEBUG_OBJECT (buf, "failed opening device");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_close_device:
|
|
* @buf: the #GstAudioRingBuffer
|
|
*
|
|
* Close the audio device associated with the ring buffer. The ring buffer
|
|
* should already have been released via gst_audio_ring_buffer_release().
|
|
*
|
|
* Returns: TRUE if the device could be closed, FALSE on error.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
gboolean
|
|
gst_audio_ring_buffer_close_device (GstAudioRingBuffer * buf)
|
|
{
|
|
gboolean res = TRUE;
|
|
GstAudioRingBufferClass *rclass;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
|
|
|
|
GST_DEBUG_OBJECT (buf, "closing device");
|
|
|
|
GST_OBJECT_LOCK (buf);
|
|
if (G_UNLIKELY (!buf->open))
|
|
goto was_closed;
|
|
|
|
if (G_UNLIKELY (buf->acquired))
|
|
goto was_acquired;
|
|
|
|
buf->open = FALSE;
|
|
|
|
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
|
|
if (G_LIKELY (rclass->close_device))
|
|
res = rclass->close_device (buf);
|
|
|
|
if (G_UNLIKELY (!res))
|
|
goto close_error;
|
|
|
|
GST_DEBUG_OBJECT (buf, "closed device");
|
|
|
|
done:
|
|
GST_OBJECT_UNLOCK (buf);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
was_closed:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "Device for ring buffer already closed");
|
|
g_warning ("Device for ring buffer %p already closed, fix your code", buf);
|
|
res = TRUE;
|
|
goto done;
|
|
}
|
|
was_acquired:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "Resources for ring buffer still acquired");
|
|
g_critical ("Resources for ring buffer %p still acquired", buf);
|
|
res = FALSE;
|
|
goto done;
|
|
}
|
|
close_error:
|
|
{
|
|
buf->open = TRUE;
|
|
GST_DEBUG_OBJECT (buf, "error closing device");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_device_is_open:
|
|
* @buf: the #GstAudioRingBuffer
|
|
*
|
|
* Checks the status of the device associated with the ring buffer.
|
|
*
|
|
* Returns: TRUE if the device was open, FALSE if it was closed.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
gboolean
|
|
gst_audio_ring_buffer_device_is_open (GstAudioRingBuffer * buf)
|
|
{
|
|
gboolean res = TRUE;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
|
|
|
|
GST_OBJECT_LOCK (buf);
|
|
res = buf->open;
|
|
GST_OBJECT_UNLOCK (buf);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_acquire:
|
|
* @buf: the #GstAudioRingBuffer to acquire
|
|
* @spec: the specs of the buffer
|
|
*
|
|
* Allocate the resources for the ringbuffer. This function fills
|
|
* in the data pointer of the ring buffer with a valid #GstBuffer
|
|
* to which samples can be written.
|
|
*
|
|
* Returns: TRUE if the device could be acquired, FALSE on error.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
gboolean
|
|
gst_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
|
|
GstAudioRingBufferSpec * spec)
|
|
{
|
|
gboolean res = FALSE;
|
|
GstAudioRingBufferClass *rclass;
|
|
gint segsize, bpf, i;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
|
|
|
|
GST_DEBUG_OBJECT (buf, "acquiring device %p", buf);
|
|
|
|
GST_OBJECT_LOCK (buf);
|
|
if (G_UNLIKELY (!buf->open))
|
|
goto not_opened;
|
|
|
|
if (G_UNLIKELY (buf->acquired))
|
|
goto was_acquired;
|
|
|
|
buf->acquired = TRUE;
|
|
buf->need_reorder = FALSE;
|
|
|
|
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
|
|
if (G_LIKELY (rclass->acquire))
|
|
res = rclass->acquire (buf, spec);
|
|
|
|
/* Only reorder for raw audio */
|
|
buf->need_reorder = (buf->need_reorder
|
|
&& buf->spec.type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW);
|
|
|
|
if (G_UNLIKELY (!res))
|
|
goto acquire_failed;
|
|
|
|
GST_INFO_OBJECT (buf, "Allocating an array for %d timestamps",
|
|
spec->segtotal);
|
|
buf->timestamps = g_slice_alloc0 (sizeof (GstClockTime) * spec->segtotal);
|
|
/* initialize array with invalid timestamps */
|
|
for (i = 0; i < spec->segtotal; i++) {
|
|
buf->timestamps[i] = GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
if (G_UNLIKELY ((bpf = buf->spec.info.bpf) == 0))
|
|
goto invalid_bpf;
|
|
|
|
/* if the seglatency was overwritten with something else than -1, use it, else
|
|
* assume segtotal as the latency */
|
|
if (buf->spec.seglatency == -1)
|
|
buf->spec.seglatency = buf->spec.segtotal;
|
|
|
|
segsize = buf->spec.segsize;
|
|
|
|
buf->samples_per_seg = segsize / bpf;
|
|
|
|
/* create an empty segment */
|
|
g_free (buf->empty_seg);
|
|
buf->empty_seg = g_malloc (segsize);
|
|
|
|
if (buf->spec.type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW) {
|
|
gst_audio_format_fill_silence (buf->spec.info.finfo, buf->empty_seg,
|
|
segsize);
|
|
} else {
|
|
/* FIXME, non-raw formats get 0 as the empty sample */
|
|
memset (buf->empty_seg, 0, segsize);
|
|
}
|
|
GST_DEBUG_OBJECT (buf, "acquired device");
|
|
|
|
done:
|
|
GST_OBJECT_UNLOCK (buf);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
not_opened:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "device not opened");
|
|
g_critical ("Device for %p not opened", buf);
|
|
res = FALSE;
|
|
goto done;
|
|
}
|
|
was_acquired:
|
|
{
|
|
res = TRUE;
|
|
GST_DEBUG_OBJECT (buf, "device was acquired");
|
|
goto done;
|
|
}
|
|
acquire_failed:
|
|
{
|
|
buf->acquired = FALSE;
|
|
GST_DEBUG_OBJECT (buf, "failed to acquire device");
|
|
goto done;
|
|
}
|
|
invalid_bpf:
|
|
{
|
|
g_warning
|
|
("invalid bytes_per_frame from acquire ringbuffer %p, fix the element",
|
|
buf);
|
|
buf->acquired = FALSE;
|
|
res = FALSE;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_release:
|
|
* @buf: the #GstAudioRingBuffer to release
|
|
*
|
|
* Free the resources of the ringbuffer.
|
|
*
|
|
* Returns: TRUE if the device could be released, FALSE on error.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
gboolean
|
|
gst_audio_ring_buffer_release (GstAudioRingBuffer * buf)
|
|
{
|
|
gboolean res = FALSE;
|
|
GstAudioRingBufferClass *rclass;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
|
|
|
|
GST_DEBUG_OBJECT (buf, "releasing device");
|
|
|
|
gst_audio_ring_buffer_stop (buf);
|
|
|
|
GST_OBJECT_LOCK (buf);
|
|
|
|
if (G_LIKELY (buf->timestamps)) {
|
|
GST_INFO_OBJECT (buf, "Freeing timestamp buffer, %d entries",
|
|
buf->spec.segtotal);
|
|
g_slice_free1 (sizeof (GstClockTime) * buf->spec.segtotal, buf->timestamps);
|
|
buf->timestamps = NULL;
|
|
}
|
|
|
|
if (G_UNLIKELY (!buf->acquired))
|
|
goto was_released;
|
|
|
|
buf->acquired = FALSE;
|
|
|
|
/* if this fails, something is wrong in this file */
|
|
g_assert (buf->open);
|
|
|
|
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
|
|
if (G_LIKELY (rclass->release))
|
|
res = rclass->release (buf);
|
|
|
|
/* signal any waiters */
|
|
GST_DEBUG_OBJECT (buf, "signal waiter");
|
|
GST_AUDIO_RING_BUFFER_SIGNAL (buf);
|
|
|
|
if (G_UNLIKELY (!res))
|
|
goto release_failed;
|
|
|
|
g_atomic_int_set (&buf->segdone, 0);
|
|
buf->segbase = 0;
|
|
g_free (buf->empty_seg);
|
|
buf->empty_seg = NULL;
|
|
gst_caps_replace (&buf->spec.caps, NULL);
|
|
gst_audio_info_init (&buf->spec.info);
|
|
GST_DEBUG_OBJECT (buf, "released device");
|
|
|
|
done:
|
|
GST_OBJECT_UNLOCK (buf);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
was_released:
|
|
{
|
|
res = TRUE;
|
|
GST_DEBUG_OBJECT (buf, "device was released");
|
|
goto done;
|
|
}
|
|
release_failed:
|
|
{
|
|
buf->acquired = TRUE;
|
|
GST_DEBUG_OBJECT (buf, "failed to release device");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_is_acquired:
|
|
* @buf: the #GstAudioRingBuffer to check
|
|
*
|
|
* Check if the ringbuffer is acquired and ready to use.
|
|
*
|
|
* Returns: TRUE if the ringbuffer is acquired, FALSE on error.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
gboolean
|
|
gst_audio_ring_buffer_is_acquired (GstAudioRingBuffer * buf)
|
|
{
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
|
|
|
|
GST_OBJECT_LOCK (buf);
|
|
res = buf->acquired;
|
|
GST_OBJECT_UNLOCK (buf);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_activate:
|
|
* @buf: the #GstAudioRingBuffer to activate
|
|
* @active: the new mode
|
|
*
|
|
* Activate @buf to start or stop pulling data.
|
|
*
|
|
* MT safe.
|
|
*
|
|
* Returns: TRUE if the device could be activated in the requested mode,
|
|
* FALSE on error.
|
|
*/
|
|
gboolean
|
|
gst_audio_ring_buffer_activate (GstAudioRingBuffer * buf, gboolean active)
|
|
{
|
|
gboolean res = FALSE;
|
|
GstAudioRingBufferClass *rclass;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
|
|
|
|
GST_DEBUG_OBJECT (buf, "activate device");
|
|
|
|
GST_OBJECT_LOCK (buf);
|
|
if (G_UNLIKELY (active && !buf->acquired))
|
|
goto not_acquired;
|
|
|
|
if (G_UNLIKELY (buf->active == active))
|
|
goto was_active;
|
|
|
|
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
|
|
/* if there is no activate function we assume it was started/released
|
|
* in the acquire method */
|
|
if (G_LIKELY (rclass->activate))
|
|
res = rclass->activate (buf, active);
|
|
else
|
|
res = TRUE;
|
|
|
|
if (G_UNLIKELY (!res))
|
|
goto activate_failed;
|
|
|
|
buf->active = active;
|
|
|
|
done:
|
|
GST_OBJECT_UNLOCK (buf);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
not_acquired:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "device not acquired");
|
|
g_critical ("Device for %p not acquired", buf);
|
|
res = FALSE;
|
|
goto done;
|
|
}
|
|
was_active:
|
|
{
|
|
res = TRUE;
|
|
GST_DEBUG_OBJECT (buf, "device was active in mode %d", active);
|
|
goto done;
|
|
}
|
|
activate_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "failed to activate device");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_is_active:
|
|
* @buf: the #GstAudioRingBuffer
|
|
*
|
|
* Check if @buf is activated.
|
|
*
|
|
* MT safe.
|
|
*
|
|
* Returns: TRUE if the device is active.
|
|
*/
|
|
gboolean
|
|
gst_audio_ring_buffer_is_active (GstAudioRingBuffer * buf)
|
|
{
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
|
|
|
|
GST_OBJECT_LOCK (buf);
|
|
res = buf->active;
|
|
GST_OBJECT_UNLOCK (buf);
|
|
|
|
return res;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_set_flushing:
|
|
* @buf: the #GstAudioRingBuffer to flush
|
|
* @flushing: the new mode
|
|
*
|
|
* Set the ringbuffer to flushing mode or normal mode.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
void
|
|
gst_audio_ring_buffer_set_flushing (GstAudioRingBuffer * buf, gboolean flushing)
|
|
{
|
|
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
|
|
|
|
GST_OBJECT_LOCK (buf);
|
|
buf->flushing = flushing;
|
|
|
|
if (flushing) {
|
|
gst_audio_ring_buffer_pause_unlocked (buf);
|
|
} else {
|
|
gst_audio_ring_buffer_clear_all (buf);
|
|
}
|
|
GST_OBJECT_UNLOCK (buf);
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_is_flushing:
|
|
* @buf: the #GstAudioRingBuffer
|
|
*
|
|
* Check if @buf is flushing.
|
|
*
|
|
* MT safe.
|
|
*
|
|
* Returns: TRUE if the device is flushing.
|
|
*/
|
|
gboolean
|
|
gst_audio_ring_buffer_is_flushing (GstAudioRingBuffer * buf)
|
|
{
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), TRUE);
|
|
|
|
GST_OBJECT_LOCK (buf);
|
|
res = buf->flushing;
|
|
GST_OBJECT_UNLOCK (buf);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_start:
|
|
* @buf: the #GstAudioRingBuffer to start
|
|
*
|
|
* Start processing samples from the ringbuffer.
|
|
*
|
|
* Returns: TRUE if the device could be started, FALSE on error.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
gboolean
|
|
gst_audio_ring_buffer_start (GstAudioRingBuffer * buf)
|
|
{
|
|
gboolean res = FALSE;
|
|
GstAudioRingBufferClass *rclass;
|
|
gboolean resume = FALSE;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
|
|
|
|
GST_DEBUG_OBJECT (buf, "starting ringbuffer");
|
|
|
|
GST_OBJECT_LOCK (buf);
|
|
if (G_UNLIKELY (buf->flushing))
|
|
goto flushing;
|
|
|
|
if (G_UNLIKELY (!buf->acquired))
|
|
goto not_acquired;
|
|
|
|
if (G_UNLIKELY (!g_atomic_int_get (&buf->may_start)))
|
|
goto may_not_start;
|
|
|
|
/* if stopped, set to started */
|
|
res = g_atomic_int_compare_and_exchange (&buf->state,
|
|
GST_AUDIO_RING_BUFFER_STATE_STOPPED, GST_AUDIO_RING_BUFFER_STATE_STARTED);
|
|
|
|
if (!res) {
|
|
GST_DEBUG_OBJECT (buf, "was not stopped, try paused");
|
|
/* was not stopped, try from paused */
|
|
res = g_atomic_int_compare_and_exchange (&buf->state,
|
|
GST_AUDIO_RING_BUFFER_STATE_PAUSED,
|
|
GST_AUDIO_RING_BUFFER_STATE_STARTED);
|
|
if (!res) {
|
|
/* was not paused either, must be started then */
|
|
res = TRUE;
|
|
GST_DEBUG_OBJECT (buf, "was not paused, must have been started");
|
|
goto done;
|
|
}
|
|
resume = TRUE;
|
|
GST_DEBUG_OBJECT (buf, "resuming");
|
|
}
|
|
|
|
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
|
|
if (resume) {
|
|
if (G_LIKELY (rclass->resume))
|
|
res = rclass->resume (buf);
|
|
} else {
|
|
if (G_LIKELY (rclass->start))
|
|
res = rclass->start (buf);
|
|
}
|
|
|
|
if (G_UNLIKELY (!res)) {
|
|
buf->state = GST_AUDIO_RING_BUFFER_STATE_PAUSED;
|
|
GST_DEBUG_OBJECT (buf, "failed to start");
|
|
} else {
|
|
GST_DEBUG_OBJECT (buf, "started");
|
|
}
|
|
|
|
done:
|
|
GST_OBJECT_UNLOCK (buf);
|
|
|
|
return res;
|
|
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "we are flushing");
|
|
GST_OBJECT_UNLOCK (buf);
|
|
return FALSE;
|
|
}
|
|
not_acquired:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "we are not acquired");
|
|
GST_OBJECT_UNLOCK (buf);
|
|
return FALSE;
|
|
}
|
|
may_not_start:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "we may not start");
|
|
GST_OBJECT_UNLOCK (buf);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_audio_ring_buffer_pause_unlocked (GstAudioRingBuffer * buf)
|
|
{
|
|
gboolean res = FALSE;
|
|
GstAudioRingBufferClass *rclass;
|
|
|
|
GST_DEBUG_OBJECT (buf, "pausing ringbuffer");
|
|
|
|
/* if started, set to paused */
|
|
res = g_atomic_int_compare_and_exchange (&buf->state,
|
|
GST_AUDIO_RING_BUFFER_STATE_STARTED, GST_AUDIO_RING_BUFFER_STATE_PAUSED);
|
|
|
|
if (!res)
|
|
goto not_started;
|
|
|
|
/* signal any waiters */
|
|
GST_DEBUG_OBJECT (buf, "signal waiter");
|
|
GST_AUDIO_RING_BUFFER_SIGNAL (buf);
|
|
|
|
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
|
|
if (G_LIKELY (rclass->pause))
|
|
res = rclass->pause (buf);
|
|
|
|
if (G_UNLIKELY (!res)) {
|
|
buf->state = GST_AUDIO_RING_BUFFER_STATE_STARTED;
|
|
GST_DEBUG_OBJECT (buf, "failed to pause");
|
|
} else {
|
|
GST_DEBUG_OBJECT (buf, "paused");
|
|
}
|
|
|
|
return res;
|
|
|
|
not_started:
|
|
{
|
|
/* was not started */
|
|
GST_DEBUG_OBJECT (buf, "was not started");
|
|
return TRUE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_pause:
|
|
* @buf: the #GstAudioRingBuffer to pause
|
|
*
|
|
* Pause processing samples from the ringbuffer.
|
|
*
|
|
* Returns: TRUE if the device could be paused, FALSE on error.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
gboolean
|
|
gst_audio_ring_buffer_pause (GstAudioRingBuffer * buf)
|
|
{
|
|
gboolean res = FALSE;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
|
|
|
|
GST_OBJECT_LOCK (buf);
|
|
if (G_UNLIKELY (buf->flushing))
|
|
goto flushing;
|
|
|
|
if (G_UNLIKELY (!buf->acquired))
|
|
goto not_acquired;
|
|
|
|
res = gst_audio_ring_buffer_pause_unlocked (buf);
|
|
GST_OBJECT_UNLOCK (buf);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "we are flushing");
|
|
GST_OBJECT_UNLOCK (buf);
|
|
return FALSE;
|
|
}
|
|
not_acquired:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "not acquired");
|
|
GST_OBJECT_UNLOCK (buf);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_stop:
|
|
* @buf: the #GstAudioRingBuffer to stop
|
|
*
|
|
* Stop processing samples from the ringbuffer.
|
|
*
|
|
* Returns: TRUE if the device could be stopped, FALSE on error.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
gboolean
|
|
gst_audio_ring_buffer_stop (GstAudioRingBuffer * buf)
|
|
{
|
|
gboolean res = FALSE;
|
|
GstAudioRingBufferClass *rclass;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
|
|
|
|
GST_DEBUG_OBJECT (buf, "stopping");
|
|
|
|
GST_OBJECT_LOCK (buf);
|
|
|
|
/* if started, set to stopped */
|
|
res = g_atomic_int_compare_and_exchange (&buf->state,
|
|
GST_AUDIO_RING_BUFFER_STATE_STARTED, GST_AUDIO_RING_BUFFER_STATE_STOPPED);
|
|
|
|
if (!res) {
|
|
GST_DEBUG_OBJECT (buf, "was not started, try paused");
|
|
/* was not started, try from paused */
|
|
res = g_atomic_int_compare_and_exchange (&buf->state,
|
|
GST_AUDIO_RING_BUFFER_STATE_PAUSED,
|
|
GST_AUDIO_RING_BUFFER_STATE_STOPPED);
|
|
if (!res) {
|
|
/* was not paused either, must have been stopped then */
|
|
res = TRUE;
|
|
GST_DEBUG_OBJECT (buf, "was not paused, must have been stopped");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/* signal any waiters */
|
|
GST_DEBUG_OBJECT (buf, "signal waiter");
|
|
GST_AUDIO_RING_BUFFER_SIGNAL (buf);
|
|
|
|
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
|
|
if (G_LIKELY (rclass->stop))
|
|
res = rclass->stop (buf);
|
|
|
|
if (G_UNLIKELY (!res)) {
|
|
buf->state = GST_AUDIO_RING_BUFFER_STATE_STARTED;
|
|
GST_DEBUG_OBJECT (buf, "failed to stop");
|
|
} else {
|
|
GST_DEBUG_OBJECT (buf, "stopped");
|
|
}
|
|
done:
|
|
GST_OBJECT_UNLOCK (buf);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_delay:
|
|
* @buf: the #GstAudioRingBuffer to query
|
|
*
|
|
* Get the number of samples queued in the audio device. This is
|
|
* usually less than the segment size but can be bigger when the
|
|
* implementation uses another internal buffer between the audio
|
|
* device.
|
|
*
|
|
* For playback ringbuffers this is the amount of samples transferred from the
|
|
* ringbuffer to the device but still not played.
|
|
*
|
|
* For capture ringbuffers this is the amount of samples in the device that are
|
|
* not yet transferred to the ringbuffer.
|
|
*
|
|
* Returns: The number of samples queued in the audio device.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
guint
|
|
gst_audio_ring_buffer_delay (GstAudioRingBuffer * buf)
|
|
{
|
|
GstAudioRingBufferClass *rclass;
|
|
guint res;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), 0);
|
|
|
|
/* buffer must be acquired */
|
|
if (G_UNLIKELY (!gst_audio_ring_buffer_is_acquired (buf)))
|
|
goto not_acquired;
|
|
|
|
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
|
|
if (G_LIKELY (rclass->delay))
|
|
res = rclass->delay (buf);
|
|
else
|
|
res = 0;
|
|
|
|
return res;
|
|
|
|
not_acquired:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "not acquired");
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_samples_done:
|
|
* @buf: the #GstAudioRingBuffer to query
|
|
*
|
|
* Get the number of samples that were processed by the ringbuffer
|
|
* since it was last started. This does not include the number of samples not
|
|
* yet processed (see gst_audio_ring_buffer_delay()).
|
|
*
|
|
* Returns: The number of samples processed by the ringbuffer.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
guint64
|
|
gst_audio_ring_buffer_samples_done (GstAudioRingBuffer * buf)
|
|
{
|
|
gint segdone;
|
|
guint64 samples;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), 0);
|
|
|
|
/* get the amount of segments we processed */
|
|
segdone = g_atomic_int_get (&buf->segdone);
|
|
|
|
/* convert to samples */
|
|
samples = ((guint64) segdone) * buf->samples_per_seg;
|
|
|
|
return samples;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_set_sample:
|
|
* @buf: the #GstAudioRingBuffer to use
|
|
* @sample: the sample number to set
|
|
*
|
|
* Make sure that the next sample written to the device is
|
|
* accounted for as being the @sample sample written to the
|
|
* device. This value will be used in reporting the current
|
|
* sample position of the ringbuffer.
|
|
*
|
|
* This function will also clear the buffer with silence.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
void
|
|
gst_audio_ring_buffer_set_sample (GstAudioRingBuffer * buf, guint64 sample)
|
|
{
|
|
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
|
|
|
|
if (sample == -1)
|
|
sample = 0;
|
|
|
|
if (G_UNLIKELY (buf->samples_per_seg == 0))
|
|
return;
|
|
|
|
/* FIXME, we assume the ringbuffer can restart at a random
|
|
* position, round down to the beginning and keep track of
|
|
* offset when calculating the processed samples. */
|
|
buf->segbase = buf->segdone - sample / buf->samples_per_seg;
|
|
|
|
gst_audio_ring_buffer_clear_all (buf);
|
|
|
|
GST_DEBUG_OBJECT (buf, "set sample to %" G_GUINT64_FORMAT ", segbase %d",
|
|
sample, buf->segbase);
|
|
}
|
|
|
|
/**
|
|
* default_clear_all:
|
|
* @buf: the #GstAudioRingBuffer to clear
|
|
*
|
|
* Fill the ringbuffer with silence.
|
|
*/
|
|
static void
|
|
default_clear_all (GstAudioRingBuffer * buf)
|
|
{
|
|
gint i;
|
|
|
|
/* not fatal, we just are not negotiated yet */
|
|
if (G_UNLIKELY (buf->spec.segtotal <= 0))
|
|
return;
|
|
|
|
GST_DEBUG_OBJECT (buf, "clear all segments");
|
|
|
|
for (i = 0; i < buf->spec.segtotal; i++) {
|
|
gst_audio_ring_buffer_clear (buf, i);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_clear_all:
|
|
* @buf: the #GstAudioRingBuffer to clear
|
|
*
|
|
* Clear all samples from the ringbuffer.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
void
|
|
gst_audio_ring_buffer_clear_all (GstAudioRingBuffer * buf)
|
|
{
|
|
GstAudioRingBufferClass *rclass;
|
|
|
|
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
|
|
|
|
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
|
|
|
|
if (G_LIKELY (rclass->clear_all))
|
|
rclass->clear_all (buf);
|
|
}
|
|
|
|
|
|
static gboolean
|
|
wait_segment (GstAudioRingBuffer * buf)
|
|
{
|
|
gint segments;
|
|
gboolean wait = TRUE;
|
|
|
|
/* buffer must be started now or we deadlock since nobody is reading */
|
|
if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
|
|
GST_AUDIO_RING_BUFFER_STATE_STARTED)) {
|
|
/* see if we are allowed to start it */
|
|
if (G_UNLIKELY (!g_atomic_int_get (&buf->may_start)))
|
|
goto no_start;
|
|
|
|
GST_DEBUG_OBJECT (buf, "start!");
|
|
segments = g_atomic_int_get (&buf->segdone);
|
|
gst_audio_ring_buffer_start (buf);
|
|
|
|
/* After starting, the writer may have wrote segments already and then we
|
|
* don't need to wait anymore */
|
|
if (G_LIKELY (g_atomic_int_get (&buf->segdone) != segments))
|
|
wait = FALSE;
|
|
}
|
|
|
|
/* take lock first, then update our waiting flag */
|
|
GST_OBJECT_LOCK (buf);
|
|
if (G_UNLIKELY (buf->flushing))
|
|
goto flushing;
|
|
|
|
if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
|
|
GST_AUDIO_RING_BUFFER_STATE_STARTED))
|
|
goto not_started;
|
|
|
|
if (G_LIKELY (wait)) {
|
|
if (g_atomic_int_compare_and_exchange (&buf->waiting, 0, 1)) {
|
|
GST_DEBUG_OBJECT (buf, "waiting..");
|
|
GST_AUDIO_RING_BUFFER_WAIT (buf);
|
|
|
|
if (G_UNLIKELY (buf->flushing))
|
|
goto flushing;
|
|
|
|
if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
|
|
GST_AUDIO_RING_BUFFER_STATE_STARTED))
|
|
goto not_started;
|
|
}
|
|
}
|
|
GST_OBJECT_UNLOCK (buf);
|
|
|
|
return TRUE;
|
|
|
|
/* ERROR */
|
|
not_started:
|
|
{
|
|
g_atomic_int_compare_and_exchange (&buf->waiting, 1, 0);
|
|
GST_DEBUG_OBJECT (buf, "stopped processing");
|
|
GST_OBJECT_UNLOCK (buf);
|
|
return FALSE;
|
|
}
|
|
flushing:
|
|
{
|
|
g_atomic_int_compare_and_exchange (&buf->waiting, 1, 0);
|
|
GST_DEBUG_OBJECT (buf, "flushing");
|
|
GST_OBJECT_UNLOCK (buf);
|
|
return FALSE;
|
|
}
|
|
no_start:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "not allowed to start");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
|
|
|
|
#define REORDER_SAMPLE(d, s, l) \
|
|
G_STMT_START { \
|
|
gint i; \
|
|
for (i = 0; i < channels; i++) { \
|
|
memcpy (d + reorder_map[i] * bps, s + i * bps, bps); \
|
|
} \
|
|
} G_STMT_END
|
|
|
|
#define REORDER_SAMPLES(d, s, len) \
|
|
G_STMT_START { \
|
|
gint i, len_ = len / bpf; \
|
|
guint8 *d_ = d, *s_ = s; \
|
|
for (i = 0; i < len_; i++) { \
|
|
REORDER_SAMPLE(d_, s_, bpf); \
|
|
d_ += bpf; \
|
|
s_ += bpf; \
|
|
} \
|
|
} G_STMT_END
|
|
|
|
#define FWD_SAMPLES(s,se,d,de,F) \
|
|
G_STMT_START { \
|
|
/* no rate conversion */ \
|
|
guint towrite = MIN (se + bpf - s, de - d); \
|
|
/* simple copy */ \
|
|
if (!skip) \
|
|
F (d, s, towrite); \
|
|
in_samples -= towrite / bpf; \
|
|
out_samples -= towrite / bpf; \
|
|
s += towrite; \
|
|
GST_DEBUG ("copy %u bytes", towrite); \
|
|
} G_STMT_END
|
|
|
|
/* in_samples >= out_samples, rate > 1.0 */
|
|
#define FWD_UP_SAMPLES(s,se,d,de,F) \
|
|
G_STMT_START { \
|
|
guint8 *sb = s, *db = d; \
|
|
while (s <= se && d < de) { \
|
|
if (!skip) \
|
|
F (d, s, bpf); \
|
|
s += bpf; \
|
|
*accum += outr; \
|
|
if ((*accum << 1) >= inr) { \
|
|
*accum -= inr; \
|
|
d += bpf; \
|
|
} \
|
|
} \
|
|
in_samples -= (s - sb)/bpf; \
|
|
out_samples -= (d - db)/bpf; \
|
|
GST_DEBUG ("fwd_up end %d/%d",*accum,*toprocess); \
|
|
} G_STMT_END
|
|
|
|
/* out_samples > in_samples, for rates smaller than 1.0 */
|
|
#define FWD_DOWN_SAMPLES(s,se,d,de,F) \
|
|
G_STMT_START { \
|
|
guint8 *sb = s, *db = d; \
|
|
while (s <= se && d < de) { \
|
|
if (!skip) \
|
|
F (d, s, bpf); \
|
|
d += bpf; \
|
|
*accum += inr; \
|
|
if ((*accum << 1) >= outr) { \
|
|
*accum -= outr; \
|
|
s += bpf; \
|
|
} \
|
|
} \
|
|
in_samples -= (s - sb)/bpf; \
|
|
out_samples -= (d - db)/bpf; \
|
|
GST_DEBUG ("fwd_down end %d/%d",*accum,*toprocess); \
|
|
} G_STMT_END
|
|
|
|
#define REV_UP_SAMPLES(s,se,d,de,F) \
|
|
G_STMT_START { \
|
|
guint8 *sb = se, *db = d; \
|
|
while (s <= se && d < de) { \
|
|
if (!skip) \
|
|
F (d, se, bpf); \
|
|
se -= bpf; \
|
|
*accum += outr; \
|
|
while (d < de && (*accum << 1) >= inr) { \
|
|
*accum -= inr; \
|
|
d += bpf; \
|
|
} \
|
|
} \
|
|
in_samples -= (sb - se)/bpf; \
|
|
out_samples -= (d - db)/bpf; \
|
|
GST_DEBUG ("rev_up end %d/%d",*accum,*toprocess); \
|
|
} G_STMT_END
|
|
|
|
#define REV_DOWN_SAMPLES(s,se,d,de,F) \
|
|
G_STMT_START { \
|
|
guint8 *sb = se, *db = d; \
|
|
while (s <= se && d < de) { \
|
|
if (!skip) \
|
|
F (d, se, bpf); \
|
|
d += bpf; \
|
|
*accum += inr; \
|
|
while (s <= se && (*accum << 1) >= outr) { \
|
|
*accum -= outr; \
|
|
se -= bpf; \
|
|
} \
|
|
} \
|
|
in_samples -= (sb - se)/bpf; \
|
|
out_samples -= (d - db)/bpf; \
|
|
GST_DEBUG ("rev_down end %d/%d",*accum,*toprocess); \
|
|
} G_STMT_END
|
|
|
|
static guint
|
|
default_commit (GstAudioRingBuffer * buf, guint64 * sample,
|
|
guint8 * data, gint in_samples, gint out_samples, gint * accum)
|
|
{
|
|
gint segdone;
|
|
gint segsize, segtotal, channels, bps, bpf, sps;
|
|
guint8 *dest, *data_end;
|
|
gint writeseg, sampleoff;
|
|
gint *toprocess;
|
|
gint inr, outr;
|
|
gboolean reverse;
|
|
gboolean need_reorder;
|
|
|
|
g_return_val_if_fail (buf->memory != NULL, -1);
|
|
g_return_val_if_fail (data != NULL, -1);
|
|
|
|
need_reorder = buf->need_reorder;
|
|
|
|
channels = buf->spec.info.channels;
|
|
dest = buf->memory;
|
|
segsize = buf->spec.segsize;
|
|
segtotal = buf->spec.segtotal;
|
|
bpf = buf->spec.info.bpf;
|
|
bps = bpf / channels;
|
|
sps = buf->samples_per_seg;
|
|
|
|
reverse = out_samples < 0;
|
|
out_samples = ABS (out_samples);
|
|
|
|
if (in_samples >= out_samples)
|
|
toprocess = &in_samples;
|
|
else
|
|
toprocess = &out_samples;
|
|
|
|
inr = in_samples - 1;
|
|
outr = out_samples - 1;
|
|
|
|
/* data_end points to the last sample we have to write, not past it. This is
|
|
* needed to properly handle reverse playback: it points to the last sample. */
|
|
data_end = data + (bpf * inr);
|
|
|
|
/* figure out the segment and the offset inside the segment where
|
|
* the first sample should be written. */
|
|
writeseg = *sample / sps;
|
|
sampleoff = (*sample % sps) * bpf;
|
|
|
|
GST_DEBUG_OBJECT (buf, "write %d : %d", in_samples, out_samples);
|
|
|
|
/* write out all samples */
|
|
while (*toprocess > 0) {
|
|
gint avail;
|
|
guint8 *d, *d_end;
|
|
gint ws;
|
|
gboolean skip;
|
|
|
|
while (TRUE) {
|
|
gint diff;
|
|
|
|
/* get the currently processed segment */
|
|
segdone = g_atomic_int_get (&buf->segdone) - buf->segbase;
|
|
|
|
/* see how far away it is from the write segment */
|
|
diff = writeseg - segdone;
|
|
|
|
GST_DEBUG_OBJECT (buf,
|
|
"pointer at %d, write to %d-%d, diff %d, segtotal %d, segsize %d, base %d",
|
|
segdone, writeseg, sampleoff, diff, segtotal, segsize, buf->segbase);
|
|
|
|
/* segment too far ahead, writer too slow, we need to drop, hopefully UNLIKELY */
|
|
if (G_UNLIKELY (diff < 0)) {
|
|
/* we need to drop one segment at a time, pretend we wrote a segment. */
|
|
skip = TRUE;
|
|
break;
|
|
}
|
|
|
|
/* write segment is within writable range, we can break the loop and
|
|
* start writing the data. */
|
|
if (diff < segtotal) {
|
|
skip = FALSE;
|
|
break;
|
|
}
|
|
|
|
/* else we need to wait for the segment to become writable. */
|
|
if (!wait_segment (buf))
|
|
goto not_started;
|
|
}
|
|
|
|
/* we can write now */
|
|
ws = writeseg % segtotal;
|
|
avail = MIN (segsize - sampleoff, bpf * out_samples);
|
|
|
|
d = dest + (ws * segsize) + sampleoff;
|
|
d_end = d + avail;
|
|
*sample += avail / bpf;
|
|
|
|
GST_DEBUG_OBJECT (buf, "write @%p seg %d, sps %d, off %d, avail %d",
|
|
dest + ws * segsize, ws, sps, sampleoff, avail);
|
|
|
|
if (need_reorder) {
|
|
gint *reorder_map = buf->channel_reorder_map;
|
|
|
|
if (G_LIKELY (inr == outr && !reverse)) {
|
|
/* no rate conversion, simply copy samples */
|
|
FWD_SAMPLES (data, data_end, d, d_end, REORDER_SAMPLES);
|
|
} else if (!reverse) {
|
|
if (inr >= outr)
|
|
/* forward speed up */
|
|
FWD_UP_SAMPLES (data, data_end, d, d_end, REORDER_SAMPLE);
|
|
else
|
|
/* forward slow down */
|
|
FWD_DOWN_SAMPLES (data, data_end, d, d_end, REORDER_SAMPLE);
|
|
} else {
|
|
if (inr >= outr)
|
|
/* reverse speed up */
|
|
REV_UP_SAMPLES (data, data_end, d, d_end, REORDER_SAMPLE);
|
|
else
|
|
/* reverse slow down */
|
|
REV_DOWN_SAMPLES (data, data_end, d, d_end, REORDER_SAMPLE);
|
|
}
|
|
} else {
|
|
if (G_LIKELY (inr == outr && !reverse)) {
|
|
/* no rate conversion, simply copy samples */
|
|
FWD_SAMPLES (data, data_end, d, d_end, memcpy);
|
|
} else if (!reverse) {
|
|
if (inr >= outr)
|
|
/* forward speed up */
|
|
FWD_UP_SAMPLES (data, data_end, d, d_end, memcpy);
|
|
else
|
|
/* forward slow down */
|
|
FWD_DOWN_SAMPLES (data, data_end, d, d_end, memcpy);
|
|
} else {
|
|
if (inr >= outr)
|
|
/* reverse speed up */
|
|
REV_UP_SAMPLES (data, data_end, d, d_end, memcpy);
|
|
else
|
|
/* reverse slow down */
|
|
REV_DOWN_SAMPLES (data, data_end, d, d_end, memcpy);
|
|
}
|
|
}
|
|
|
|
/* for the next iteration we write to the next segment at the beginning. */
|
|
writeseg++;
|
|
sampleoff = 0;
|
|
}
|
|
/* we consumed all samples here */
|
|
data = data_end + bpf;
|
|
|
|
done:
|
|
return inr - ((data_end - data) / bpf);
|
|
|
|
/* ERRORS */
|
|
not_started:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "stopped processing");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_commit:
|
|
* @buf: the #GstAudioRingBuffer to commit
|
|
* @sample: the sample position of the data
|
|
* @data: (array length=in_samples): the data to commit
|
|
* @in_samples: the number of samples in the data to commit
|
|
* @out_samples: the number of samples to write to the ringbuffer
|
|
* @accum: (inout): accumulator for rate conversion.
|
|
*
|
|
* Commit @in_samples samples pointed to by @data to the ringbuffer @buf.
|
|
*
|
|
* @in_samples and @out_samples define the rate conversion to perform on the
|
|
* samples in @data. For negative rates, @out_samples must be negative and
|
|
* @in_samples positive.
|
|
*
|
|
* When @out_samples is positive, the first sample will be written at position @sample
|
|
* in the ringbuffer. When @out_samples is negative, the last sample will be written to
|
|
* @sample in reverse order.
|
|
*
|
|
* @out_samples does not need to be a multiple of the segment size of the ringbuffer
|
|
* although it is recommended for optimal performance.
|
|
*
|
|
* @accum will hold a temporary accumulator used in rate conversion and should be
|
|
* set to 0 when this function is first called. In case the commit operation is
|
|
* interrupted, one can resume the processing by passing the previously returned
|
|
* @accum value back to this function.
|
|
*
|
|
* MT safe.
|
|
*
|
|
* Returns: The number of samples written to the ringbuffer or -1 on error. The
|
|
* number of samples written can be less than @out_samples when @buf was interrupted
|
|
* with a flush or stop.
|
|
*/
|
|
guint
|
|
gst_audio_ring_buffer_commit (GstAudioRingBuffer * buf, guint64 * sample,
|
|
guint8 * data, gint in_samples, gint out_samples, gint * accum)
|
|
{
|
|
GstAudioRingBufferClass *rclass;
|
|
guint res = -1;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), -1);
|
|
|
|
if (G_UNLIKELY (in_samples == 0 || out_samples == 0))
|
|
return in_samples;
|
|
|
|
rclass = GST_AUDIO_RING_BUFFER_GET_CLASS (buf);
|
|
|
|
if (G_LIKELY (rclass->commit))
|
|
res = rclass->commit (buf, sample, data, in_samples, out_samples, accum);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_read:
|
|
* @buf: the #GstAudioRingBuffer to read from
|
|
* @sample: the sample position of the data
|
|
* @data: (array length=len): where the data should be read
|
|
* @len: the number of samples in data to read
|
|
* @timestamp: (out): where the timestamp is returned
|
|
*
|
|
* Read @len samples from the ringbuffer into the memory pointed
|
|
* to by @data.
|
|
* The first sample should be read from position @sample in
|
|
* the ringbuffer.
|
|
*
|
|
* @len should not be a multiple of the segment size of the ringbuffer
|
|
* although it is recommended.
|
|
*
|
|
* @timestamp will return the timestamp associated with the data returned.
|
|
*
|
|
* Returns: The number of samples read from the ringbuffer or -1 on
|
|
* error.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
guint
|
|
gst_audio_ring_buffer_read (GstAudioRingBuffer * buf, guint64 sample,
|
|
guint8 * data, guint len, GstClockTime * timestamp)
|
|
{
|
|
gint segdone;
|
|
gint segsize, segtotal, channels, bps, bpf, sps, readseg = 0;
|
|
guint8 *dest;
|
|
guint to_read;
|
|
gboolean need_reorder;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), -1);
|
|
g_return_val_if_fail (buf->memory != NULL, -1);
|
|
g_return_val_if_fail (data != NULL, -1);
|
|
|
|
need_reorder = buf->need_reorder;
|
|
dest = buf->memory;
|
|
segsize = buf->spec.segsize;
|
|
segtotal = buf->spec.segtotal;
|
|
channels = buf->spec.info.channels;
|
|
bpf = buf->spec.info.bpf;
|
|
bps = bpf / channels;
|
|
sps = buf->samples_per_seg;
|
|
|
|
to_read = len;
|
|
/* read enough samples */
|
|
while (to_read > 0) {
|
|
gint sampleslen;
|
|
gint sampleoff;
|
|
|
|
/* figure out the segment and the offset inside the segment where
|
|
* the sample should be read from. */
|
|
readseg = sample / sps;
|
|
sampleoff = (sample % sps);
|
|
|
|
while (TRUE) {
|
|
gint diff;
|
|
|
|
/* get the currently processed segment */
|
|
segdone = g_atomic_int_get (&buf->segdone) - buf->segbase;
|
|
|
|
/* see how far away it is from the read segment, normally segdone (where
|
|
* the hardware is writing) is bigger than readseg (where software is
|
|
* reading) */
|
|
diff = segdone - readseg;
|
|
|
|
GST_DEBUG_OBJECT
|
|
(buf, "pointer at %d, sample %" G_GUINT64_FORMAT
|
|
", read from %d-%d, to_read %d, diff %d, segtotal %d, segsize %d",
|
|
segdone, sample, readseg, sampleoff, to_read, diff, segtotal,
|
|
segsize);
|
|
|
|
/* segment too far ahead, reader too slow */
|
|
if (G_UNLIKELY (diff >= segtotal)) {
|
|
/* pretend we read an empty segment. */
|
|
sampleslen = MIN (sps, to_read);
|
|
memcpy (data, buf->empty_seg, sampleslen * bpf);
|
|
goto next;
|
|
}
|
|
|
|
/* read segment is within readable range, we can break the loop and
|
|
* start reading the data. */
|
|
if (diff > 0)
|
|
break;
|
|
|
|
/* else we need to wait for the segment to become readable. */
|
|
if (!wait_segment (buf))
|
|
goto not_started;
|
|
}
|
|
|
|
/* we can read now */
|
|
readseg = readseg % segtotal;
|
|
sampleslen = MIN (sps - sampleoff, to_read);
|
|
|
|
GST_DEBUG_OBJECT (buf, "read @%p seg %d, off %d, sampleslen %d",
|
|
dest + readseg * segsize, readseg, sampleoff, sampleslen);
|
|
|
|
if (need_reorder) {
|
|
guint8 *ptr = dest + (readseg * segsize) + (sampleoff * bpf);
|
|
gint i, j;
|
|
gint *reorder_map = buf->channel_reorder_map;
|
|
|
|
/* Reorder from device order to GStreamer order */
|
|
for (i = 0; i < sampleslen; i++) {
|
|
for (j = 0; j < channels; j++) {
|
|
memcpy (data + i * bpf + reorder_map[j] * bps, ptr + j * bps, bps);
|
|
}
|
|
ptr += bpf;
|
|
}
|
|
} else {
|
|
memcpy (data, dest + (readseg * segsize) + (sampleoff * bpf),
|
|
(sampleslen * bpf));
|
|
}
|
|
|
|
next:
|
|
to_read -= sampleslen;
|
|
sample += sampleslen;
|
|
data += sampleslen * bpf;
|
|
}
|
|
|
|
if (buf->timestamps && timestamp) {
|
|
*timestamp = buf->timestamps[readseg % segtotal];
|
|
GST_DEBUG_OBJECT (buf, "Retrieved timestamp %" GST_TIME_FORMAT
|
|
" @ %d", GST_TIME_ARGS (*timestamp), readseg % segtotal);
|
|
}
|
|
|
|
return len - to_read;
|
|
|
|
/* ERRORS */
|
|
not_started:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "stopped processing");
|
|
return len - to_read;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_prepare_read:
|
|
* @buf: the #GstAudioRingBuffer to read from
|
|
* @segment: (out): the segment to read
|
|
* @readptr: (out) (array length=len):
|
|
* the pointer to the memory where samples can be read
|
|
* @len: (out): the number of bytes to read
|
|
*
|
|
* Returns a pointer to memory where the data from segment @segment
|
|
* can be found. This function is mostly used by subclasses.
|
|
*
|
|
* Returns: FALSE if the buffer is not started.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
gboolean
|
|
gst_audio_ring_buffer_prepare_read (GstAudioRingBuffer * buf, gint * segment,
|
|
guint8 ** readptr, gint * len)
|
|
{
|
|
guint8 *data;
|
|
gint segdone;
|
|
|
|
g_return_val_if_fail (GST_IS_AUDIO_RING_BUFFER (buf), FALSE);
|
|
|
|
if (buf->callback == NULL) {
|
|
/* push mode, fail when nothing is started */
|
|
if (g_atomic_int_get (&buf->state) != GST_AUDIO_RING_BUFFER_STATE_STARTED)
|
|
return FALSE;
|
|
}
|
|
|
|
g_return_val_if_fail (buf->memory != NULL, FALSE);
|
|
g_return_val_if_fail (segment != NULL, FALSE);
|
|
g_return_val_if_fail (readptr != NULL, FALSE);
|
|
g_return_val_if_fail (len != NULL, FALSE);
|
|
|
|
data = buf->memory;
|
|
|
|
/* get the position of the pointer */
|
|
segdone = g_atomic_int_get (&buf->segdone);
|
|
|
|
*segment = segdone % buf->spec.segtotal;
|
|
*len = buf->spec.segsize;
|
|
*readptr = data + *segment * *len;
|
|
|
|
GST_LOG_OBJECT (buf, "prepare read from segment %d (real %d) @%p",
|
|
*segment, segdone, *readptr);
|
|
|
|
/* callback to fill the memory with data, for pull based
|
|
* scheduling. */
|
|
if (buf->callback)
|
|
buf->callback (buf, *readptr, *len, buf->cb_data);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_advance:
|
|
* @buf: the #GstAudioRingBuffer to advance
|
|
* @advance: the number of segments written
|
|
*
|
|
* Subclasses should call this function to notify the fact that
|
|
* @advance segments are now processed by the device.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
void
|
|
gst_audio_ring_buffer_advance (GstAudioRingBuffer * buf, guint advance)
|
|
{
|
|
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
|
|
|
|
/* update counter */
|
|
g_atomic_int_add (&buf->segdone, advance);
|
|
|
|
/* the lock is already taken when the waiting flag is set,
|
|
* we grab the lock as well to make sure the waiter is actually
|
|
* waiting for the signal */
|
|
if (g_atomic_int_compare_and_exchange (&buf->waiting, 1, 0)) {
|
|
GST_OBJECT_LOCK (buf);
|
|
GST_DEBUG_OBJECT (buf, "signal waiter");
|
|
GST_AUDIO_RING_BUFFER_SIGNAL (buf);
|
|
GST_OBJECT_UNLOCK (buf);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_clear:
|
|
* @buf: the #GstAudioRingBuffer to clear
|
|
* @segment: the segment to clear
|
|
*
|
|
* Clear the given segment of the buffer with silence samples.
|
|
* This function is used by subclasses.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
void
|
|
gst_audio_ring_buffer_clear (GstAudioRingBuffer * buf, gint segment)
|
|
{
|
|
guint8 *data;
|
|
|
|
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
|
|
|
|
/* no data means it's already cleared */
|
|
if (G_UNLIKELY (buf->memory == NULL))
|
|
return;
|
|
|
|
/* no empty_seg means it's not opened */
|
|
if (G_UNLIKELY (buf->empty_seg == NULL))
|
|
return;
|
|
|
|
segment %= buf->spec.segtotal;
|
|
|
|
data = buf->memory;
|
|
data += segment * buf->spec.segsize;
|
|
|
|
GST_LOG_OBJECT (buf, "clear segment %d @%p", segment, data);
|
|
|
|
memcpy (data, buf->empty_seg, buf->spec.segsize);
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_may_start:
|
|
* @buf: the #GstAudioRingBuffer
|
|
* @allowed: the new value
|
|
*
|
|
* Tell the ringbuffer that it is allowed to start playback when
|
|
* the ringbuffer is filled with samples.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
void
|
|
gst_audio_ring_buffer_may_start (GstAudioRingBuffer * buf, gboolean allowed)
|
|
{
|
|
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
|
|
|
|
GST_LOG_OBJECT (buf, "may start: %d", allowed);
|
|
g_atomic_int_set (&buf->may_start, allowed);
|
|
}
|
|
|
|
/* GST_AUDIO_CHANNEL_POSITION_NONE is used for position-less
|
|
* mutually exclusive channels. In this case we should not attempt
|
|
* to do any reordering.
|
|
*/
|
|
static gboolean
|
|
position_less_channels (const GstAudioChannelPosition * pos, guint channels)
|
|
{
|
|
guint i;
|
|
|
|
for (i = 0; i < channels; i++) {
|
|
if (pos[i] != GST_AUDIO_CHANNEL_POSITION_NONE)
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_audio_ring_buffer_set_channel_positions:
|
|
* @buf: the #GstAudioRingBuffer
|
|
* @position: (array): the device channel positions
|
|
*
|
|
* Tell the ringbuffer about the device's channel positions. This must
|
|
* be called in when the ringbuffer is acquired.
|
|
*/
|
|
void
|
|
gst_audio_ring_buffer_set_channel_positions (GstAudioRingBuffer * buf,
|
|
const GstAudioChannelPosition * position)
|
|
{
|
|
const GstAudioChannelPosition *to;
|
|
gint channels;
|
|
gint i;
|
|
|
|
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
|
|
g_return_if_fail (buf->acquired);
|
|
|
|
channels = buf->spec.info.channels;
|
|
to = buf->spec.info.position;
|
|
|
|
buf->need_reorder = FALSE;
|
|
if (memcmp (position, to, channels * sizeof (to[0])) == 0)
|
|
return;
|
|
|
|
if (position_less_channels (position, channels)) {
|
|
GST_LOG_OBJECT (buf, "position-less channels, no need to reorder");
|
|
return;
|
|
}
|
|
|
|
if (!gst_audio_get_channel_reorder_map (channels, position, to,
|
|
buf->channel_reorder_map))
|
|
g_return_if_reached ();
|
|
|
|
for (i = 0; i < channels; i++) {
|
|
if (buf->channel_reorder_map[i] != i) {
|
|
#ifndef GST_DISABLE_GST_DEBUG
|
|
{
|
|
gchar *tmp1, *tmp2;
|
|
|
|
tmp1 = gst_audio_channel_positions_to_string (position, channels);
|
|
tmp2 = gst_audio_channel_positions_to_string (to, channels);
|
|
GST_LOG_OBJECT (buf, "may have to reorder channels: %s -> %s", tmp1,
|
|
tmp2);
|
|
g_free (tmp1);
|
|
g_free (tmp2);
|
|
}
|
|
#endif /* GST_DISABLE_GST_DEBUG */
|
|
|
|
buf->need_reorder = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_ring_buffer_set_timestamp:
|
|
* @buf: the #GstRingBuffer
|
|
* @readseg: the current data segment
|
|
* @timestamp: The new timestamp of the buffer.
|
|
*
|
|
* Set a new timestamp on the buffer.
|
|
*
|
|
* MT safe.
|
|
*/
|
|
void
|
|
gst_audio_ring_buffer_set_timestamp (GstAudioRingBuffer * buf, gint readseg,
|
|
GstClockTime timestamp)
|
|
{
|
|
g_return_if_fail (GST_IS_AUDIO_RING_BUFFER (buf));
|
|
|
|
GST_DEBUG_OBJECT (buf, "Storing timestamp %" GST_TIME_FORMAT
|
|
" @ %d", GST_TIME_ARGS (timestamp), readseg);
|
|
|
|
GST_OBJECT_LOCK (buf);
|
|
if (G_UNLIKELY (!buf->acquired))
|
|
goto not_acquired;
|
|
|
|
buf->timestamps[readseg] = timestamp;
|
|
|
|
done:
|
|
GST_OBJECT_UNLOCK (buf);
|
|
return;
|
|
|
|
not_acquired:
|
|
{
|
|
GST_DEBUG_OBJECT (buf, "we are not acquired");
|
|
goto done;
|
|
}
|
|
}
|