mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-05 16:06:43 +00:00
474 lines
13 KiB
C++
474 lines
13 KiB
C++
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/* GStreamer
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* Copyright (C) 2021 Seungha Yang <seungha@centricular.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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstasioringbuffer.h"
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#include <string.h>
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#include "gstasioutils.h"
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#include "gstasioobject.h"
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GST_DEBUG_CATEGORY_STATIC (gst_asio_ring_buffer_debug);
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#define GST_CAT_DEFAULT gst_asio_ring_buffer_debug
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struct _GstAsioRingBuffer
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{
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GstAudioRingBuffer parent;
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GstAsioDeviceClassType type;
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GstAsioObject *asio_object;
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guint *channel_indices;
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guint num_channels;
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ASIOBufferInfo **infos;
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guint64 callback_id;
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gboolean callback_installed;
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gboolean running;
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guint buffer_size;
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/* Used to detect sample gap */
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gboolean is_first;
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guint64 expected_sample_position;
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gboolean trace_sample_position;
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};
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enum
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{
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PROP_0,
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PROP_DEVICE_INFO,
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};
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static void gst_asio_ring_buffer_dispose (GObject * object);
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static gboolean gst_asio_ring_buffer_open_device (GstAudioRingBuffer * buf);
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static gboolean gst_asio_ring_buffer_close_device (GstAudioRingBuffer * buf);
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static gboolean gst_asio_ring_buffer_acquire (GstAudioRingBuffer * buf,
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GstAudioRingBufferSpec * spec);
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static gboolean gst_asio_ring_buffer_release (GstAudioRingBuffer * buf);
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static gboolean gst_asio_ring_buffer_start (GstAudioRingBuffer * buf);
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static gboolean gst_asio_ring_buffer_stop (GstAudioRingBuffer * buf);
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static guint gst_asio_ring_buffer_delay (GstAudioRingBuffer * buf);
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static gboolean gst_asio_buffer_switch_cb (GstAsioObject * obj,
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glong index, ASIOBufferInfo * infos, guint num_infos,
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ASIOChannelInfo * input_channel_infos,
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ASIOChannelInfo * output_channel_infos,
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ASIOSampleRate sample_rate, glong buffer_size, gpointer user_data);
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#define gst_asio_ring_buffer_parent_class parent_class
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G_DEFINE_TYPE (GstAsioRingBuffer, gst_asio_ring_buffer,
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GST_TYPE_AUDIO_RING_BUFFER);
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static void
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gst_asio_ring_buffer_class_init (GstAsioRingBufferClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstAudioRingBufferClass *ring_buffer_class =
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GST_AUDIO_RING_BUFFER_CLASS (klass);
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gobject_class->dispose = gst_asio_ring_buffer_dispose;
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ring_buffer_class->open_device =
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GST_DEBUG_FUNCPTR (gst_asio_ring_buffer_open_device);
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ring_buffer_class->close_device =
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GST_DEBUG_FUNCPTR (gst_asio_ring_buffer_close_device);
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ring_buffer_class->acquire = GST_DEBUG_FUNCPTR (gst_asio_ring_buffer_acquire);
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ring_buffer_class->release = GST_DEBUG_FUNCPTR (gst_asio_ring_buffer_release);
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ring_buffer_class->start = GST_DEBUG_FUNCPTR (gst_asio_ring_buffer_start);
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ring_buffer_class->resume = GST_DEBUG_FUNCPTR (gst_asio_ring_buffer_start);
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ring_buffer_class->stop = GST_DEBUG_FUNCPTR (gst_asio_ring_buffer_stop);
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ring_buffer_class->delay = GST_DEBUG_FUNCPTR (gst_asio_ring_buffer_delay);
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GST_DEBUG_CATEGORY_INIT (gst_asio_ring_buffer_debug,
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"asioringbuffer", 0, "asioringbuffer");
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}
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static void
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gst_asio_ring_buffer_init (GstAsioRingBuffer * self)
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{
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}
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static void
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gst_asio_ring_buffer_dispose (GObject * object)
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{
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GstAsioRingBuffer *self = GST_ASIO_RING_BUFFER (object);
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gst_clear_object (&self->asio_object);
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g_clear_pointer (&self->channel_indices, g_free);
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static gboolean
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gst_asio_ring_buffer_open_device (GstAudioRingBuffer * buf)
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{
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GstAsioRingBuffer *self = GST_ASIO_RING_BUFFER (buf);
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GST_DEBUG_OBJECT (self, "Open");
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return TRUE;
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}
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static gboolean
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gst_asio_ring_buffer_close_device (GstAudioRingBuffer * buf)
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{
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GstAsioRingBuffer *self = GST_ASIO_RING_BUFFER (buf);
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GST_DEBUG_OBJECT (self, "Close");
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return TRUE;
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}
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#define PACK_ASIO_64(v) ((v).lo | ((guint64)((v).hi) << 32))
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static gboolean
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gst_asio_buffer_switch_cb (GstAsioObject * obj, glong index,
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ASIOBufferInfo * infos, guint num_infos,
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ASIOChannelInfo * input_channel_infos,
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ASIOChannelInfo * output_channel_infos,
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ASIOSampleRate sample_rate, glong buffer_size,
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ASIOTime * time_info, gpointer user_data)
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{
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GstAsioRingBuffer *self = (GstAsioRingBuffer *) user_data;
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GstAudioRingBuffer *ringbuffer = GST_AUDIO_RING_BUFFER_CAST (self);
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gint segment;
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guint8 *readptr;
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gint len;
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guint i, j;
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guint num_channels = 0;
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guint bps = GST_AUDIO_INFO_WIDTH (&ringbuffer->spec.info) >> 3;
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g_assert (index == 0 || index == 1);
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g_assert (num_infos >= self->num_channels);
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GST_TRACE_OBJECT (self, "Buffer Switch callback, index %ld", index);
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if (!gst_audio_ring_buffer_prepare_read (ringbuffer,
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&segment, &readptr, &len)) {
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GST_WARNING_OBJECT (self, "No segment available");
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return TRUE;
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}
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GST_TRACE_OBJECT (self, "segment %d, length %d", segment, len);
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/* Check missing frames */
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if (self->type == GST_ASIO_DEVICE_CLASS_CAPTURE) {
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if (self->is_first) {
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if (time_info) {
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self->expected_sample_position =
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PACK_ASIO_64 (time_info->timeInfo.samplePosition) + buffer_size;
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self->trace_sample_position = TRUE;
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} else {
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GST_WARNING_OBJECT (self, "ASIOTime is not available");
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self->trace_sample_position = FALSE;
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}
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self->is_first = FALSE;
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} else if (self->trace_sample_position) {
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if (!time_info) {
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GST_WARNING_OBJECT (self, "ASIOTime is not available");
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self->trace_sample_position = FALSE;
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} else {
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guint64 sample_position =
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PACK_ASIO_64 (time_info->timeInfo.samplePosition);
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if (self->expected_sample_position < sample_position) {
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guint64 gap_frames = sample_position - self->expected_sample_position;
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gint gap_size = gap_frames * bps;
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GST_WARNING_OBJECT (self, "%" G_GUINT64_FORMAT " frames are missing");
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while (gap_size >= len) {
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gst_audio_format_info_fill_silence (ringbuffer->spec.info.finfo,
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readptr, len);
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gst_audio_ring_buffer_advance (ringbuffer, 1);
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gst_audio_ring_buffer_prepare_read (ringbuffer,
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&segment, &readptr, &len);
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gap_size -= len;
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}
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}
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self->expected_sample_position = sample_position + buffer_size;
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GST_TRACE_OBJECT (self, "Sample Position %" G_GUINT64_FORMAT
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", next: %" G_GUINT64_FORMAT, sample_position,
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self->expected_sample_position);
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}
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}
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}
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/* Given @infos might contain more channel data, pick channels what we want to
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* read */
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for (i = 0; i < num_infos; i++) {
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ASIOBufferInfo *info = &infos[i];
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if (self->type == GST_ASIO_DEVICE_CLASS_CAPTURE) {
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if (!info->isInput)
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continue;
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} else {
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if (info->isInput)
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continue;
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}
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for (j = 0; j < self->num_channels; j++) {
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if (self->channel_indices[j] != info->channelNum)
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continue;
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g_assert (num_channels < self->num_channels);
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self->infos[num_channels++] = info;
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break;
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}
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}
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if (num_channels < self->num_channels) {
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GST_ERROR_OBJECT (self, "Too small number of channel %d (expected %d)",
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num_channels, self->num_channels);
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} else {
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if (self->type == GST_ASIO_DEVICE_CLASS_CAPTURE ||
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self->type == GST_ASIO_DEVICE_CLASS_LOOPBACK_CAPTURE) {
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if (num_channels == 1) {
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memcpy (readptr, self->infos[0]->buffers[index], len);
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} else {
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guint gst_offset = 0, asio_offset = 0;
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/* Interleaves audio */
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while (gst_offset < len) {
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for (i = 0; i < num_channels; i++) {
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ASIOBufferInfo *info = self->infos[i];
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memcpy (readptr + gst_offset,
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((guint8 *) info->buffers[index]) + asio_offset, bps);
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gst_offset += bps;
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}
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asio_offset += bps;
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}
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}
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} else {
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if (num_channels == 1) {
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memcpy (self->infos[0]->buffers[index], readptr, len);
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} else {
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guint gst_offset = 0, asio_offset = 0;
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/* Interleaves audio */
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while (gst_offset < len) {
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for (i = 0; i < num_channels; i++) {
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ASIOBufferInfo *info = self->infos[i];
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memcpy (((guint8 *) info->buffers[index]) + asio_offset,
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readptr + gst_offset, bps);
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gst_offset += bps;
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}
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asio_offset += bps;
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}
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}
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}
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}
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if (self->type == GST_ASIO_DEVICE_CLASS_RENDER)
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gst_audio_ring_buffer_clear (ringbuffer, segment);
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gst_audio_ring_buffer_advance (ringbuffer, 1);
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return TRUE;
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}
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static gboolean
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gst_asio_ring_buffer_acquire (GstAudioRingBuffer * buf,
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GstAudioRingBufferSpec * spec)
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{
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GstAsioRingBuffer *self = GST_ASIO_RING_BUFFER (buf);
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if (!self->asio_object) {
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GST_ERROR_OBJECT (self, "No configured ASIO object");
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return FALSE;
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}
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if (!self->channel_indices || self->num_channels == 0) {
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GST_ERROR_OBJECT (self, "No configured channels");
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return FALSE;
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}
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if (!gst_asio_object_set_sample_rate (self->asio_object,
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GST_AUDIO_INFO_RATE (&spec->info))) {
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GST_ERROR_OBJECT (self, "Failed to set sample rate");
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return FALSE;
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}
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spec->segsize = self->buffer_size *
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(GST_AUDIO_INFO_WIDTH (&spec->info) >> 3) *
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GST_AUDIO_INFO_CHANNELS (&spec->info);
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spec->segtotal = 2;
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buf->size = spec->segtotal * spec->segsize;
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buf->memory = (guint8 *) g_malloc (buf->size);
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gst_audio_format_info_fill_silence (buf->spec.info.finfo,
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buf->memory, buf->size);
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return TRUE;
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}
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static gboolean
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gst_asio_ring_buffer_release (GstAudioRingBuffer * buf)
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{
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GST_DEBUG_OBJECT (buf, "Release");
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g_clear_pointer (&buf->memory, g_free);
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return TRUE;
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}
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static gboolean
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gst_asio_ring_buffer_start (GstAudioRingBuffer * buf)
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{
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GstAsioRingBuffer *self = GST_ASIO_RING_BUFFER (buf);
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GstAsioObjectCallbacks callbacks;
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GST_DEBUG_OBJECT (buf, "Start");
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callbacks.buffer_switch = gst_asio_buffer_switch_cb;
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callbacks.user_data = self;
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self->is_first = TRUE;
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self->expected_sample_position = 0;
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if (!gst_asio_object_install_callback (self->asio_object, self->type,
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&callbacks, &self->callback_id)) {
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GST_ERROR_OBJECT (self, "Failed to install callback");
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return FALSE;
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}
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self->callback_installed = TRUE;
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if (!gst_asio_object_start (self->asio_object)) {
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GST_ERROR_OBJECT (self, "Failed to start");
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gst_asio_ring_buffer_stop (buf);
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return FALSE;
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}
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self->running = TRUE;
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return TRUE;
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}
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static gboolean
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gst_asio_ring_buffer_stop (GstAudioRingBuffer * buf)
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{
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GstAsioRingBuffer *self = GST_ASIO_RING_BUFFER (buf);
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GST_DEBUG_OBJECT (buf, "Stop");
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self->running = FALSE;
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if (!self->asio_object)
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return TRUE;
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if (self->callback_installed)
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gst_asio_object_uninstall_callback (self->asio_object, self->callback_id);
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self->callback_installed = FALSE;
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self->callback_id = 0;
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self->is_first = TRUE;
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self->expected_sample_position = 0;
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return TRUE;
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}
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static guint
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gst_asio_ring_buffer_delay (GstAudioRingBuffer * buf)
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{
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/* FIXME: impl. */
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return 0;
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}
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GstAsioRingBuffer *
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gst_asio_ring_buffer_new (GstAsioObject * object, GstAsioDeviceClassType type,
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const gchar * name)
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{
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GstAsioRingBuffer *self;
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g_return_val_if_fail (GST_IS_ASIO_OBJECT (object), nullptr);
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self =
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(GstAsioRingBuffer *) g_object_new (GST_TYPE_ASIO_RING_BUFFER,
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"name", name, nullptr);
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g_assert (self);
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self->type = type;
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self->asio_object = (GstAsioObject *) gst_object_ref (object);
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return self;
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}
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gboolean
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gst_asio_ring_buffer_configure (GstAsioRingBuffer * buf,
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||
|
guint * channel_indices, guint num_channles, guint preferred_buffer_size)
|
||
|
{
|
||
|
g_return_val_if_fail (GST_IS_ASIO_RING_BUFFER (buf), FALSE);
|
||
|
g_return_val_if_fail (buf->asio_object != nullptr, FALSE);
|
||
|
g_return_val_if_fail (num_channles > 0, FALSE);
|
||
|
|
||
|
GST_DEBUG_OBJECT (buf, "Configure");
|
||
|
|
||
|
buf->buffer_size = preferred_buffer_size;
|
||
|
|
||
|
if (!gst_asio_object_create_buffers (buf->asio_object, buf->type,
|
||
|
channel_indices, num_channles, &buf->buffer_size)) {
|
||
|
GST_ERROR_OBJECT (buf, "Failed to configure");
|
||
|
|
||
|
g_clear_pointer (&buf->channel_indices, g_free);
|
||
|
buf->num_channels = 0;
|
||
|
|
||
|
return FALSE;
|
||
|
}
|
||
|
|
||
|
GST_DEBUG_OBJECT (buf, "configured buffer size: %d", buf->buffer_size);
|
||
|
|
||
|
g_free (buf->channel_indices);
|
||
|
buf->channel_indices = g_new0 (guint, num_channles);
|
||
|
|
||
|
for (guint i = 0; i < num_channles; i++)
|
||
|
buf->channel_indices[i] = channel_indices[i];
|
||
|
|
||
|
buf->num_channels = num_channles;
|
||
|
|
||
|
g_clear_pointer (&buf->infos, g_free);
|
||
|
buf->infos = g_new0 (ASIOBufferInfo *, num_channles);
|
||
|
|
||
|
return TRUE;
|
||
|
}
|
||
|
|
||
|
GstCaps *
|
||
|
gst_asio_ring_buffer_get_caps (GstAsioRingBuffer * buf)
|
||
|
{
|
||
|
g_return_val_if_fail (GST_IS_ASIO_RING_BUFFER (buf), nullptr);
|
||
|
g_assert (buf->asio_object != nullptr);
|
||
|
|
||
|
return gst_asio_object_get_caps (buf->asio_object,
|
||
|
buf->type, buf->num_channels, buf->num_channels);
|
||
|
}
|