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1999 lines
64 KiB
C
1999 lines
64 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2001 Thomas <thomas@apestaart.org>
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* 2005,2006 Wim Taymans <wim@fluendo.com>
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* 2013 Sebastian Dröge <sebastian@centricular.com>
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*
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* audiomixer.c: AudioMixer element, N in, one out, samples are added
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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:element-audiomixer
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*
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* The audiomixer allows to mix several streams into one by adding the data.
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* Mixed data is clamped to the min/max values of the data format.
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*
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* The audiomixer currently mixes all data received on the sinkpads as soon as
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* possible without trying to synchronize the streams.
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch audiotestsrc freq=100 ! audiomixer name=mix ! audioconvert ! alsasink audiotestsrc freq=500 ! mix.
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* ]| This pipeline produces two sine waves mixed together.
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* </refsect2>
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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 "gstaudiomixer.h"
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#include <gst/audio/audio.h>
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#include <string.h> /* strcmp */
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#include "gstaudiomixerorc.h"
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#define GST_CAT_DEFAULT gst_audiomixer_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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typedef struct _GstAudioMixerCollect GstAudioMixerCollect;
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struct _GstAudioMixerCollect
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{
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GstCollectData collect; /* we extend the CollectData */
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GstBuffer *buffer; /* current buffer we're mixing,
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for comparison with collect.buffer
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to see if we need to update our
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cached values. */
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guint position, size;
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guint64 output_offset; /* Offset in output segment that
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collect.pos refers to in the
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current buffer. */
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guint64 next_offset; /* Next expected offset in the input segment */
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};
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#define DEFAULT_PAD_VOLUME (1.0)
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#define DEFAULT_PAD_MUTE (FALSE)
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/* some defines for audio processing */
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/* the volume factor is a range from 0.0 to (arbitrary) VOLUME_MAX_DOUBLE = 10.0
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* we map 1.0 to VOLUME_UNITY_INT*
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*/
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#define VOLUME_UNITY_INT8 8 /* internal int for unity 2^(8-5) */
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#define VOLUME_UNITY_INT8_BIT_SHIFT 3 /* number of bits to shift for unity */
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#define VOLUME_UNITY_INT16 2048 /* internal int for unity 2^(16-5) */
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#define VOLUME_UNITY_INT16_BIT_SHIFT 11 /* number of bits to shift for unity */
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#define VOLUME_UNITY_INT24 524288 /* internal int for unity 2^(24-5) */
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#define VOLUME_UNITY_INT24_BIT_SHIFT 19 /* number of bits to shift for unity */
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#define VOLUME_UNITY_INT32 134217728 /* internal int for unity 2^(32-5) */
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#define VOLUME_UNITY_INT32_BIT_SHIFT 27
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enum
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{
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PROP_PAD_0,
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PROP_PAD_VOLUME,
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PROP_PAD_MUTE
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};
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G_DEFINE_TYPE (GstAudioMixerPad, gst_audiomixer_pad, GST_TYPE_PAD);
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static void
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gst_audiomixer_pad_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (object);
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switch (prop_id) {
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case PROP_PAD_VOLUME:
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g_value_set_double (value, pad->volume);
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break;
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case PROP_PAD_MUTE:
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g_value_set_boolean (value, pad->mute);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_audiomixer_pad_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (object);
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switch (prop_id) {
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case PROP_PAD_VOLUME:
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GST_OBJECT_LOCK (pad);
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pad->volume = g_value_get_double (value);
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pad->volume_i8 = pad->volume * VOLUME_UNITY_INT8;
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pad->volume_i16 = pad->volume * VOLUME_UNITY_INT16;
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pad->volume_i32 = pad->volume * VOLUME_UNITY_INT32;
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GST_OBJECT_UNLOCK (pad);
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break;
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case PROP_PAD_MUTE:
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GST_OBJECT_LOCK (pad);
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pad->mute = g_value_get_boolean (value);
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GST_OBJECT_UNLOCK (pad);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_audiomixer_pad_class_init (GstAudioMixerPadClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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gobject_class->set_property = gst_audiomixer_pad_set_property;
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gobject_class->get_property = gst_audiomixer_pad_get_property;
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g_object_class_install_property (gobject_class, PROP_PAD_VOLUME,
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g_param_spec_double ("volume", "Volume", "Volume of this pad",
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0.0, 10.0, DEFAULT_PAD_VOLUME,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_PAD_MUTE,
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g_param_spec_boolean ("mute", "Mute", "Mute this pad",
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DEFAULT_PAD_MUTE,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_audiomixer_pad_init (GstAudioMixerPad * pad)
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{
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pad->volume = DEFAULT_PAD_VOLUME;
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pad->mute = DEFAULT_PAD_MUTE;
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}
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#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
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#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
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#define DEFAULT_BLOCKSIZE (1024)
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enum
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{
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PROP_0,
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PROP_FILTER_CAPS,
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PROP_ALIGNMENT_THRESHOLD,
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PROP_DISCONT_WAIT,
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PROP_BLOCKSIZE
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};
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/* elementfactory information */
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#if G_BYTE_ORDER == G_LITTLE_ENDIAN
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#define CAPS \
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GST_AUDIO_CAPS_MAKE ("{ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }") \
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", layout = (string) { interleaved, non-interleaved }"
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#else
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#define CAPS \
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GST_AUDIO_CAPS_MAKE ("{ S32BE, U32BE, S16BE, U16BE, S8, U8, F32BE, F64BE }") \
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", layout = (string) { interleaved, non-interleaved }"
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#endif
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static GstStaticPadTemplate gst_audiomixer_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (CAPS)
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);
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static GstStaticPadTemplate gst_audiomixer_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink_%u",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS (CAPS)
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);
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static void gst_audiomixer_child_proxy_init (gpointer g_iface,
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gpointer iface_data);
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#define gst_audiomixer_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstAudioMixer, gst_audiomixer, GST_TYPE_ELEMENT,
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G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY,
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gst_audiomixer_child_proxy_init));
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static void gst_audiomixer_dispose (GObject * object);
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static void gst_audiomixer_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audiomixer_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_audiomixer_setcaps (GstAudioMixer * audiomixer,
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GstPad * pad, GstCaps * caps);
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static gboolean gst_audiomixer_src_query (GstPad * pad, GstObject * parent,
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GstQuery * query);
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static gboolean gst_audiomixer_sink_query (GstCollectPads * pads,
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GstCollectData * pad, GstQuery * query, gpointer user_data);
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static gboolean gst_audiomixer_src_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static gboolean gst_audiomixer_sink_event (GstCollectPads * pads,
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GstCollectData * pad, GstEvent * event, gpointer user_data);
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static GstPad *gst_audiomixer_request_new_pad (GstElement * element,
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GstPadTemplate * temp, const gchar * unused, const GstCaps * caps);
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static void gst_audiomixer_release_pad (GstElement * element, GstPad * pad);
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static GstStateChangeReturn gst_audiomixer_change_state (GstElement * element,
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GstStateChange transition);
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static GstFlowReturn gst_audiomixer_do_clip (GstCollectPads * pads,
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GstCollectData * data, GstBuffer * buffer, GstBuffer ** out,
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gpointer user_data);
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static GstFlowReturn gst_audiomixer_collected (GstCollectPads * pads,
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gpointer user_data);
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/* we can only accept caps that we and downstream can handle.
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* if we have filtercaps set, use those to constrain the target caps.
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*/
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static GstCaps *
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gst_audiomixer_sink_getcaps (GstPad * pad, GstCaps * filter)
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{
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GstAudioMixer *audiomixer;
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GstCaps *result, *peercaps, *current_caps, *filter_caps;
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GstStructure *s;
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gint i, n;
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audiomixer = GST_AUDIO_MIXER (GST_PAD_PARENT (pad));
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GST_OBJECT_LOCK (audiomixer);
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/* take filter */
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if ((filter_caps = audiomixer->filter_caps)) {
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if (filter)
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filter_caps =
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gst_caps_intersect_full (filter, filter_caps,
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GST_CAPS_INTERSECT_FIRST);
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else
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gst_caps_ref (filter_caps);
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} else {
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filter_caps = filter ? gst_caps_ref (filter) : NULL;
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}
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GST_OBJECT_UNLOCK (audiomixer);
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if (filter_caps && gst_caps_is_empty (filter_caps)) {
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GST_WARNING_OBJECT (pad, "Empty filter caps");
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return filter_caps;
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}
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/* get the downstream possible caps */
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peercaps = gst_pad_peer_query_caps (audiomixer->srcpad, filter_caps);
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/* get the allowed caps on this sinkpad */
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GST_OBJECT_LOCK (audiomixer);
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current_caps =
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audiomixer->current_caps ? gst_caps_ref (audiomixer->current_caps) : NULL;
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if (current_caps == NULL) {
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current_caps = gst_pad_get_pad_template_caps (pad);
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if (!current_caps)
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current_caps = gst_caps_new_any ();
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}
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GST_OBJECT_UNLOCK (audiomixer);
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if (peercaps) {
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/* if the peer has caps, intersect */
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GST_DEBUG_OBJECT (audiomixer, "intersecting peer and our caps");
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result =
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gst_caps_intersect_full (peercaps, current_caps,
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GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (peercaps);
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gst_caps_unref (current_caps);
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} else {
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/* the peer has no caps (or there is no peer), just use the allowed caps
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* of this sinkpad. */
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/* restrict with filter-caps if any */
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if (filter_caps) {
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GST_DEBUG_OBJECT (audiomixer, "no peer caps, using filtered caps");
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result =
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gst_caps_intersect_full (filter_caps, current_caps,
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GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (current_caps);
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} else {
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GST_DEBUG_OBJECT (audiomixer, "no peer caps, using our caps");
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result = current_caps;
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}
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}
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result = gst_caps_make_writable (result);
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n = gst_caps_get_size (result);
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for (i = 0; i < n; i++) {
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GstStructure *sref;
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s = gst_caps_get_structure (result, i);
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sref = gst_structure_copy (s);
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gst_structure_set (sref, "channels", GST_TYPE_INT_RANGE, 0, 2, NULL);
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if (gst_structure_is_subset (s, sref)) {
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/* This field is irrelevant when in mono or stereo */
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gst_structure_remove_field (s, "channel-mask");
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}
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gst_structure_free (sref);
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}
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if (filter_caps)
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gst_caps_unref (filter_caps);
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GST_LOG_OBJECT (audiomixer, "getting caps on pad %p,%s to %" GST_PTR_FORMAT,
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pad, GST_PAD_NAME (pad), result);
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return result;
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}
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static gboolean
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gst_audiomixer_sink_query (GstCollectPads * pads, GstCollectData * pad,
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GstQuery * query, gpointer user_data)
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{
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gboolean res = FALSE;
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_CAPS:
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{
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GstCaps *filter, *caps;
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gst_query_parse_caps (query, &filter);
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caps = gst_audiomixer_sink_getcaps (pad->pad, filter);
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gst_query_set_caps_result (query, caps);
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gst_caps_unref (caps);
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res = TRUE;
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break;
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}
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default:
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res = gst_collect_pads_query_default (pads, pad, query, FALSE);
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break;
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}
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return res;
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}
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/* the first caps we receive on any of the sinkpads will define the caps for all
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* the other sinkpads because we can only mix streams with the same caps.
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*/
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static gboolean
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gst_audiomixer_setcaps (GstAudioMixer * audiomixer, GstPad * pad,
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GstCaps * orig_caps)
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{
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GstCaps *caps;
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GstAudioInfo info;
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GstStructure *s;
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gint channels;
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caps = gst_caps_copy (orig_caps);
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s = gst_caps_get_structure (caps, 0);
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if (gst_structure_get_int (s, "channels", &channels))
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if (channels <= 2)
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gst_structure_remove_field (s, "channel-mask");
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if (!gst_audio_info_from_caps (&info, caps))
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goto invalid_format;
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GST_OBJECT_LOCK (audiomixer);
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/* don't allow reconfiguration for now; there's still a race between the
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* different upstream threads doing query_caps + accept_caps + sending
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* (possibly different) CAPS events, but there's not much we can do about
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* that, upstream needs to deal with it. */
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if (audiomixer->current_caps != NULL) {
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if (gst_audio_info_is_equal (&info, &audiomixer->info)) {
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GST_OBJECT_UNLOCK (audiomixer);
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gst_caps_unref (caps);
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return TRUE;
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} else {
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GST_DEBUG_OBJECT (pad, "got input caps %" GST_PTR_FORMAT ", but "
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"current caps are %" GST_PTR_FORMAT, caps, audiomixer->current_caps);
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GST_OBJECT_UNLOCK (audiomixer);
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gst_pad_push_event (pad, gst_event_new_reconfigure ());
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gst_caps_unref (caps);
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return FALSE;
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}
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}
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GST_INFO_OBJECT (pad, "setting caps to %" GST_PTR_FORMAT, caps);
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gst_caps_replace (&audiomixer->current_caps, caps);
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memcpy (&audiomixer->info, &info, sizeof (info));
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audiomixer->send_caps = TRUE;
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GST_OBJECT_UNLOCK (audiomixer);
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/* send caps event later, after stream-start event */
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GST_INFO_OBJECT (pad, "handle caps change to %" GST_PTR_FORMAT, caps);
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gst_caps_unref (caps);
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return TRUE;
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/* ERRORS */
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invalid_format:
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{
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gst_caps_unref (caps);
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GST_WARNING_OBJECT (audiomixer, "invalid format set as caps");
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return FALSE;
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}
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}
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/* FIXME, the duration query should reflect how long you will produce
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* data, that is the amount of stream time until you will emit EOS.
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*
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* For synchronized mixing this is always the max of all the durations
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* of upstream since we emit EOS when all of them finished.
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*
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* We don't do synchronized mixing so this really depends on where the
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* streams where punched in and what their relative offsets are against
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* eachother which we can get from the first timestamps we see.
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*
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* When we add a new stream (or remove a stream) the duration might
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* also become invalid again and we need to post a new DURATION
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* message to notify this fact to the parent.
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* For now we take the max of all the upstream elements so the simple
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* cases work at least somewhat.
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*/
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static gboolean
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gst_audiomixer_query_duration (GstAudioMixer * audiomixer, GstQuery * query)
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{
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gint64 max;
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gboolean res;
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GstFormat format;
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GstIterator *it;
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gboolean done;
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GValue item = { 0, };
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/* parse format */
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gst_query_parse_duration (query, &format, NULL);
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max = -1;
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res = TRUE;
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done = FALSE;
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|
|
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (audiomixer));
|
|
while (!done) {
|
|
GstIteratorResult ires;
|
|
|
|
ires = gst_iterator_next (it, &item);
|
|
switch (ires) {
|
|
case GST_ITERATOR_DONE:
|
|
done = TRUE;
|
|
break;
|
|
case GST_ITERATOR_OK:
|
|
{
|
|
GstPad *pad = g_value_get_object (&item);
|
|
gint64 duration;
|
|
|
|
/* ask sink peer for duration */
|
|
res &= gst_pad_peer_query_duration (pad, format, &duration);
|
|
/* take max from all valid return values */
|
|
if (res) {
|
|
/* valid unknown length, stop searching */
|
|
if (duration == -1) {
|
|
max = duration;
|
|
done = TRUE;
|
|
}
|
|
/* else see if bigger than current max */
|
|
else if (duration > max)
|
|
max = duration;
|
|
}
|
|
g_value_reset (&item);
|
|
break;
|
|
}
|
|
case GST_ITERATOR_RESYNC:
|
|
max = -1;
|
|
res = TRUE;
|
|
gst_iterator_resync (it);
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
done = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
g_value_unset (&item);
|
|
gst_iterator_free (it);
|
|
|
|
if (res) {
|
|
/* and store the max */
|
|
GST_DEBUG_OBJECT (audiomixer, "Total duration in format %s: %"
|
|
GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max));
|
|
gst_query_set_duration (query, format, max);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_audiomixer_query_latency (GstAudioMixer * audiomixer, GstQuery * query)
|
|
{
|
|
GstClockTime min, max;
|
|
gboolean live;
|
|
gboolean res;
|
|
GstIterator *it;
|
|
gboolean done;
|
|
GValue item = { 0, };
|
|
|
|
res = TRUE;
|
|
done = FALSE;
|
|
|
|
live = FALSE;
|
|
min = 0;
|
|
max = GST_CLOCK_TIME_NONE;
|
|
|
|
/* Take maximum of all latency values */
|
|
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (audiomixer));
|
|
while (!done) {
|
|
GstIteratorResult ires;
|
|
|
|
ires = gst_iterator_next (it, &item);
|
|
switch (ires) {
|
|
case GST_ITERATOR_DONE:
|
|
done = TRUE;
|
|
break;
|
|
case GST_ITERATOR_OK:
|
|
{
|
|
GstPad *pad = g_value_get_object (&item);
|
|
GstQuery *peerquery;
|
|
GstClockTime min_cur, max_cur;
|
|
gboolean live_cur;
|
|
|
|
peerquery = gst_query_new_latency ();
|
|
|
|
/* Ask peer for latency */
|
|
res &= gst_pad_peer_query (pad, peerquery);
|
|
|
|
/* take max from all valid return values */
|
|
if (res) {
|
|
gst_query_parse_latency (peerquery, &live_cur, &min_cur, &max_cur);
|
|
|
|
if (min_cur > min)
|
|
min = min_cur;
|
|
|
|
if (max_cur != GST_CLOCK_TIME_NONE &&
|
|
((max != GST_CLOCK_TIME_NONE && max_cur > max) ||
|
|
(max == GST_CLOCK_TIME_NONE)))
|
|
max = max_cur;
|
|
|
|
live = live || live_cur;
|
|
}
|
|
|
|
gst_query_unref (peerquery);
|
|
g_value_reset (&item);
|
|
break;
|
|
}
|
|
case GST_ITERATOR_RESYNC:
|
|
live = FALSE;
|
|
min = 0;
|
|
max = GST_CLOCK_TIME_NONE;
|
|
res = TRUE;
|
|
gst_iterator_resync (it);
|
|
break;
|
|
default:
|
|
res = FALSE;
|
|
done = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
g_value_unset (&item);
|
|
gst_iterator_free (it);
|
|
|
|
if (res) {
|
|
/* store the results */
|
|
GST_DEBUG_OBJECT (audiomixer, "Calculated total latency: live %s, min %"
|
|
GST_TIME_FORMAT ", max %" GST_TIME_FORMAT,
|
|
(live ? "yes" : "no"), GST_TIME_ARGS (min), GST_TIME_ARGS (max));
|
|
gst_query_set_latency (query, live, min, max);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_audiomixer_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
|
|
{
|
|
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (parent);
|
|
gboolean res = FALSE;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_POSITION:
|
|
{
|
|
GstFormat format;
|
|
|
|
gst_query_parse_position (query, &format, NULL);
|
|
|
|
switch (format) {
|
|
case GST_FORMAT_TIME:
|
|
/* FIXME, bring to stream time, might be tricky */
|
|
gst_query_set_position (query, format, audiomixer->segment.position);
|
|
res = TRUE;
|
|
break;
|
|
case GST_FORMAT_DEFAULT:
|
|
gst_query_set_position (query, format, audiomixer->offset);
|
|
res = TRUE;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_DURATION:
|
|
res = gst_audiomixer_query_duration (audiomixer, query);
|
|
break;
|
|
case GST_QUERY_LATENCY:
|
|
res = gst_audiomixer_query_latency (audiomixer, query);
|
|
break;
|
|
default:
|
|
/* FIXME, needs a custom query handler because we have multiple
|
|
* sinkpads */
|
|
res = gst_pad_query_default (pad, parent, query);
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
/* event handling */
|
|
|
|
typedef struct
|
|
{
|
|
GstEvent *event;
|
|
gboolean flush;
|
|
} EventData;
|
|
|
|
/* FIXME: What is this supposed to solve? */
|
|
static gboolean
|
|
forward_event_func (const GValue * val, GValue * ret, EventData * data)
|
|
{
|
|
GstPad *pad = g_value_get_object (val);
|
|
GstEvent *event = data->event;
|
|
GstPad *peer;
|
|
|
|
gst_event_ref (event);
|
|
GST_LOG_OBJECT (pad, "About to send event %s", GST_EVENT_TYPE_NAME (event));
|
|
peer = gst_pad_get_peer (pad);
|
|
/* collect pad might have been set flushing,
|
|
* so bypass core checking that and send directly to peer */
|
|
if (!peer || !gst_pad_send_event (peer, event)) {
|
|
if (!peer)
|
|
gst_event_unref (event);
|
|
GST_WARNING_OBJECT (pad, "Sending event %p (%s) failed.",
|
|
event, GST_EVENT_TYPE_NAME (event));
|
|
/* quick hack to unflush the pads, ideally we need a way to just unflush
|
|
* this single collect pad */
|
|
if (data->flush)
|
|
gst_pad_send_event (pad, gst_event_new_flush_stop (TRUE));
|
|
} else {
|
|
g_value_set_boolean (ret, TRUE);
|
|
GST_LOG_OBJECT (pad, "Sent event %p (%s).",
|
|
event, GST_EVENT_TYPE_NAME (event));
|
|
}
|
|
if (peer)
|
|
gst_object_unref (peer);
|
|
|
|
/* continue on other pads, even if one failed */
|
|
return TRUE;
|
|
}
|
|
|
|
/* forwards the event to all sinkpads, takes ownership of the
|
|
* event
|
|
*
|
|
* Returns: TRUE if the event could be forwarded on all
|
|
* sinkpads.
|
|
*/
|
|
static gboolean
|
|
forward_event (GstAudioMixer * audiomixer, GstEvent * event, gboolean flush)
|
|
{
|
|
gboolean ret;
|
|
GstIterator *it;
|
|
GstIteratorResult ires;
|
|
GValue vret = { 0 };
|
|
EventData data;
|
|
|
|
GST_LOG_OBJECT (audiomixer, "Forwarding event %p (%s)", event,
|
|
GST_EVENT_TYPE_NAME (event));
|
|
|
|
data.event = event;
|
|
data.flush = flush;
|
|
|
|
g_value_init (&vret, G_TYPE_BOOLEAN);
|
|
g_value_set_boolean (&vret, FALSE);
|
|
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (audiomixer));
|
|
while (TRUE) {
|
|
ires =
|
|
gst_iterator_fold (it, (GstIteratorFoldFunction) forward_event_func,
|
|
&vret, &data);
|
|
switch (ires) {
|
|
case GST_ITERATOR_RESYNC:
|
|
GST_WARNING ("resync");
|
|
gst_iterator_resync (it);
|
|
g_value_set_boolean (&vret, TRUE);
|
|
break;
|
|
case GST_ITERATOR_OK:
|
|
case GST_ITERATOR_DONE:
|
|
ret = g_value_get_boolean (&vret);
|
|
goto done;
|
|
default:
|
|
ret = FALSE;
|
|
goto done;
|
|
}
|
|
}
|
|
done:
|
|
gst_iterator_free (it);
|
|
GST_LOG_OBJECT (audiomixer, "Forwarded event %p (%s), ret=%d", event,
|
|
GST_EVENT_TYPE_NAME (event), ret);
|
|
gst_event_unref (event);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_audiomixer_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
|
|
{
|
|
GstAudioMixer *audiomixer;
|
|
gboolean result;
|
|
|
|
audiomixer = GST_AUDIO_MIXER (parent);
|
|
|
|
GST_DEBUG_OBJECT (pad, "Got %s event on src pad",
|
|
GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
/* TODO: Update from videomixer */
|
|
case GST_EVENT_SEEK:
|
|
{
|
|
GstSeekFlags flags;
|
|
gdouble rate;
|
|
GstSeekType start_type, stop_type;
|
|
gint64 start, stop;
|
|
GstFormat seek_format, dest_format;
|
|
gboolean flush;
|
|
|
|
/* parse the seek parameters */
|
|
gst_event_parse_seek (event, &rate, &seek_format, &flags, &start_type,
|
|
&start, &stop_type, &stop);
|
|
|
|
if ((start_type != GST_SEEK_TYPE_NONE)
|
|
&& (start_type != GST_SEEK_TYPE_SET)) {
|
|
result = FALSE;
|
|
GST_DEBUG_OBJECT (audiomixer,
|
|
"seeking failed, unhandled seek type for start: %d", start_type);
|
|
goto done;
|
|
}
|
|
if ((stop_type != GST_SEEK_TYPE_NONE) && (stop_type != GST_SEEK_TYPE_SET)) {
|
|
result = FALSE;
|
|
GST_DEBUG_OBJECT (audiomixer,
|
|
"seeking failed, unhandled seek type for end: %d", stop_type);
|
|
goto done;
|
|
}
|
|
|
|
dest_format = audiomixer->segment.format;
|
|
if (seek_format != dest_format) {
|
|
result = FALSE;
|
|
GST_DEBUG_OBJECT (audiomixer,
|
|
"seeking failed, unhandled seek format: %d", seek_format);
|
|
goto done;
|
|
}
|
|
|
|
flush = (flags & GST_SEEK_FLAG_FLUSH) == GST_SEEK_FLAG_FLUSH;
|
|
|
|
/* check if we are flushing */
|
|
if (flush) {
|
|
/* flushing seek, start flush downstream, the flush will be done
|
|
* when all pads received a FLUSH_STOP.
|
|
* Make sure we accept nothing anymore and return WRONG_STATE.
|
|
* We send a flush-start before, to ensure no streaming is done
|
|
* as we need to take the stream lock.
|
|
*/
|
|
gst_pad_push_event (audiomixer->srcpad, gst_event_new_flush_start ());
|
|
gst_collect_pads_set_flushing (audiomixer->collect, TRUE);
|
|
|
|
/* We can't send FLUSH_STOP here since upstream could start pushing data
|
|
* after we unlock audiomixer->collect.
|
|
* We set flush_stop_pending to TRUE instead and send FLUSH_STOP after
|
|
* forwarding the seek upstream or from gst_audiomixer_collected,
|
|
* whichever happens first.
|
|
*/
|
|
GST_COLLECT_PADS_STREAM_LOCK (audiomixer->collect);
|
|
audiomixer->flush_stop_pending = TRUE;
|
|
GST_COLLECT_PADS_STREAM_UNLOCK (audiomixer->collect);
|
|
GST_DEBUG_OBJECT (audiomixer, "mark pending flush stop event");
|
|
}
|
|
GST_DEBUG_OBJECT (audiomixer, "handling seek event: %" GST_PTR_FORMAT,
|
|
event);
|
|
|
|
/* now wait for the collected to be finished and mark a new
|
|
* segment. After we have the lock, no collect function is running and no
|
|
* new collect function will be called for as long as we're flushing. */
|
|
GST_COLLECT_PADS_STREAM_LOCK (audiomixer->collect);
|
|
/* clip position and update our segment */
|
|
if (audiomixer->segment.stop != -1) {
|
|
audiomixer->segment.position = audiomixer->segment.stop;
|
|
}
|
|
gst_segment_do_seek (&audiomixer->segment, rate, seek_format, flags,
|
|
start_type, start, stop_type, stop, NULL);
|
|
|
|
if (flush) {
|
|
/* Yes, we need to call _set_flushing again *WHEN* the streaming threads
|
|
* have stopped so that the cookie gets properly updated. */
|
|
gst_collect_pads_set_flushing (audiomixer->collect, TRUE);
|
|
}
|
|
GST_COLLECT_PADS_STREAM_UNLOCK (audiomixer->collect);
|
|
GST_DEBUG_OBJECT (audiomixer, "forwarding seek event: %" GST_PTR_FORMAT,
|
|
event);
|
|
GST_DEBUG_OBJECT (audiomixer, "updated segment: %" GST_SEGMENT_FORMAT,
|
|
&audiomixer->segment);
|
|
|
|
/* we're forwarding seek to all upstream peers and wait for one to reply
|
|
* with a newsegment-event before we send a newsegment-event downstream */
|
|
g_atomic_int_set (&audiomixer->segment_pending, TRUE);
|
|
result = forward_event (audiomixer, event, flush);
|
|
/* FIXME: We should use the seek segment and forward that downstream next time
|
|
* not any upstream segment event */
|
|
if (!result) {
|
|
/* seek failed. maybe source is a live source. */
|
|
GST_DEBUG_OBJECT (audiomixer, "seeking failed");
|
|
}
|
|
if (g_atomic_int_compare_and_exchange (&audiomixer->flush_stop_pending,
|
|
TRUE, FALSE)) {
|
|
GST_DEBUG_OBJECT (audiomixer, "pending flush stop");
|
|
if (!gst_pad_push_event (audiomixer->srcpad,
|
|
gst_event_new_flush_stop (TRUE))) {
|
|
GST_WARNING_OBJECT (audiomixer, "Sending flush stop event failed");
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
case GST_EVENT_QOS:
|
|
/* QoS might be tricky */
|
|
result = FALSE;
|
|
gst_event_unref (event);
|
|
break;
|
|
case GST_EVENT_NAVIGATION:
|
|
/* navigation is rather pointless. */
|
|
result = FALSE;
|
|
gst_event_unref (event);
|
|
break;
|
|
default:
|
|
/* just forward the rest for now */
|
|
GST_DEBUG_OBJECT (audiomixer, "forward unhandled event: %s",
|
|
GST_EVENT_TYPE_NAME (event));
|
|
result = forward_event (audiomixer, event, FALSE);
|
|
break;
|
|
}
|
|
|
|
done:
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_audiomixer_sink_event (GstCollectPads * pads, GstCollectData * pad,
|
|
GstEvent * event, gpointer user_data)
|
|
{
|
|
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (user_data);
|
|
GstAudioMixerCollect *adata = (GstAudioMixerCollect *) pad;
|
|
gboolean res = TRUE, discard = FALSE;
|
|
|
|
GST_DEBUG_OBJECT (pad->pad, "Got %s event on sink pad",
|
|
GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_CAPS:
|
|
{
|
|
GstCaps *caps;
|
|
|
|
gst_event_parse_caps (event, &caps);
|
|
res = gst_audiomixer_setcaps (audiomixer, pad->pad, caps);
|
|
gst_event_unref (event);
|
|
event = NULL;
|
|
break;
|
|
}
|
|
/* FIXME: Who cares about flushes from upstream? We should
|
|
* not forward them at all */
|
|
case GST_EVENT_FLUSH_START:
|
|
/* ensure that we will send a flush stop */
|
|
GST_COLLECT_PADS_STREAM_LOCK (audiomixer->collect);
|
|
audiomixer->flush_stop_pending = TRUE;
|
|
res = gst_collect_pads_event_default (pads, pad, event, discard);
|
|
event = NULL;
|
|
GST_COLLECT_PADS_STREAM_UNLOCK (audiomixer->collect);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
/* we received a flush-stop. We will only forward it when
|
|
* flush_stop_pending is set, and we will unset it then.
|
|
*/
|
|
g_atomic_int_set (&audiomixer->segment_pending, TRUE);
|
|
GST_COLLECT_PADS_STREAM_LOCK (audiomixer->collect);
|
|
if (audiomixer->flush_stop_pending) {
|
|
GST_DEBUG_OBJECT (pad->pad, "forwarding flush stop");
|
|
res = gst_collect_pads_event_default (pads, pad, event, discard);
|
|
audiomixer->flush_stop_pending = FALSE;
|
|
event = NULL;
|
|
gst_buffer_replace (&audiomixer->current_buffer, NULL);
|
|
audiomixer->discont_time = GST_CLOCK_TIME_NONE;
|
|
} else {
|
|
discard = TRUE;
|
|
GST_DEBUG_OBJECT (pad->pad, "eating flush stop");
|
|
}
|
|
GST_COLLECT_PADS_STREAM_UNLOCK (audiomixer->collect);
|
|
/* Clear pending tags */
|
|
if (audiomixer->pending_events) {
|
|
g_list_foreach (audiomixer->pending_events, (GFunc) gst_event_unref,
|
|
NULL);
|
|
g_list_free (audiomixer->pending_events);
|
|
audiomixer->pending_events = NULL;
|
|
}
|
|
adata->position = adata->size = 0;
|
|
adata->output_offset = adata->next_offset = -1;
|
|
gst_buffer_replace (&adata->buffer, NULL);
|
|
break;
|
|
case GST_EVENT_TAG:
|
|
/* collect tags here so we can push them out when we collect data */
|
|
audiomixer->pending_events =
|
|
g_list_append (audiomixer->pending_events, event);
|
|
event = NULL;
|
|
break;
|
|
case GST_EVENT_SEGMENT:{
|
|
const GstSegment *segment;
|
|
gst_event_parse_segment (event, &segment);
|
|
if (segment->rate != audiomixer->segment.rate) {
|
|
GST_ERROR_OBJECT (pad->pad,
|
|
"Got segment event with wrong rate %lf, expected %lf",
|
|
segment->rate, audiomixer->segment.rate);
|
|
res = FALSE;
|
|
gst_event_unref (event);
|
|
event = NULL;
|
|
} else if (segment->rate < 0.0) {
|
|
GST_ERROR_OBJECT (pad->pad, "Negative rates not supported yet");
|
|
res = FALSE;
|
|
gst_event_unref (event);
|
|
event = NULL;
|
|
}
|
|
discard = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (G_LIKELY (event))
|
|
return gst_collect_pads_event_default (pads, pad, event, discard);
|
|
else
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_audiomixer_class_init (GstAudioMixerClass * klass)
|
|
{
|
|
GObjectClass *gobject_class = (GObjectClass *) klass;
|
|
GstElementClass *gstelement_class = (GstElementClass *) klass;
|
|
|
|
gobject_class->set_property = gst_audiomixer_set_property;
|
|
gobject_class->get_property = gst_audiomixer_get_property;
|
|
gobject_class->dispose = gst_audiomixer_dispose;
|
|
|
|
g_object_class_install_property (gobject_class, PROP_FILTER_CAPS,
|
|
g_param_spec_boxed ("caps", "Target caps",
|
|
"Set target format for mixing (NULL means ANY). "
|
|
"Setting this property takes a reference to the supplied GstCaps "
|
|
"object", GST_TYPE_CAPS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
|
|
g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
|
|
"Timestamp alignment threshold in nanoseconds", 0,
|
|
G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
|
|
g_param_spec_uint64 ("discont-wait", "Discont Wait",
|
|
"Window of time in nanoseconds to wait before "
|
|
"creating a discontinuity", 0,
|
|
G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_BLOCKSIZE,
|
|
g_param_spec_uint ("blocksize", "Block Size",
|
|
"Output block size in number of samples", 0,
|
|
G_MAXUINT, DEFAULT_BLOCKSIZE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&gst_audiomixer_src_template));
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&gst_audiomixer_sink_template));
|
|
gst_element_class_set_static_metadata (gstelement_class, "AudioMixer",
|
|
"Generic/Audio",
|
|
"Mixes multiple audio streams",
|
|
"Sebastian Dröge <sebastian@centricular.com>");
|
|
|
|
gstelement_class->request_new_pad =
|
|
GST_DEBUG_FUNCPTR (gst_audiomixer_request_new_pad);
|
|
gstelement_class->release_pad =
|
|
GST_DEBUG_FUNCPTR (gst_audiomixer_release_pad);
|
|
gstelement_class->change_state =
|
|
GST_DEBUG_FUNCPTR (gst_audiomixer_change_state);
|
|
}
|
|
|
|
static void
|
|
gst_audiomixer_init (GstAudioMixer * audiomixer)
|
|
{
|
|
GstPadTemplate *template;
|
|
|
|
template = gst_static_pad_template_get (&gst_audiomixer_src_template);
|
|
audiomixer->srcpad = gst_pad_new_from_template (template, "src");
|
|
gst_object_unref (template);
|
|
|
|
gst_pad_set_query_function (audiomixer->srcpad,
|
|
GST_DEBUG_FUNCPTR (gst_audiomixer_src_query));
|
|
gst_pad_set_event_function (audiomixer->srcpad,
|
|
GST_DEBUG_FUNCPTR (gst_audiomixer_src_event));
|
|
GST_PAD_SET_PROXY_CAPS (audiomixer->srcpad);
|
|
gst_element_add_pad (GST_ELEMENT (audiomixer), audiomixer->srcpad);
|
|
|
|
audiomixer->current_caps = NULL;
|
|
gst_audio_info_init (&audiomixer->info);
|
|
audiomixer->padcount = 0;
|
|
|
|
audiomixer->filter_caps = NULL;
|
|
audiomixer->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
|
|
audiomixer->discont_wait = DEFAULT_DISCONT_WAIT;
|
|
audiomixer->blocksize = DEFAULT_BLOCKSIZE;
|
|
|
|
/* keep track of the sinkpads requested */
|
|
audiomixer->collect = gst_collect_pads_new ();
|
|
gst_collect_pads_set_function (audiomixer->collect,
|
|
GST_DEBUG_FUNCPTR (gst_audiomixer_collected), audiomixer);
|
|
gst_collect_pads_set_clip_function (audiomixer->collect,
|
|
GST_DEBUG_FUNCPTR (gst_audiomixer_do_clip), audiomixer);
|
|
gst_collect_pads_set_event_function (audiomixer->collect,
|
|
GST_DEBUG_FUNCPTR (gst_audiomixer_sink_event), audiomixer);
|
|
gst_collect_pads_set_query_function (audiomixer->collect,
|
|
GST_DEBUG_FUNCPTR (gst_audiomixer_sink_query), audiomixer);
|
|
}
|
|
|
|
static void
|
|
gst_audiomixer_dispose (GObject * object)
|
|
{
|
|
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (object);
|
|
|
|
if (audiomixer->collect) {
|
|
gst_object_unref (audiomixer->collect);
|
|
audiomixer->collect = NULL;
|
|
}
|
|
gst_caps_replace (&audiomixer->filter_caps, NULL);
|
|
gst_caps_replace (&audiomixer->current_caps, NULL);
|
|
|
|
if (audiomixer->pending_events) {
|
|
g_list_foreach (audiomixer->pending_events, (GFunc) gst_event_unref, NULL);
|
|
g_list_free (audiomixer->pending_events);
|
|
audiomixer->pending_events = NULL;
|
|
}
|
|
|
|
G_OBJECT_CLASS (parent_class)->dispose (object);
|
|
}
|
|
|
|
static void
|
|
gst_audiomixer_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_FILTER_CAPS:{
|
|
GstCaps *new_caps = NULL;
|
|
GstCaps *old_caps;
|
|
const GstCaps *new_caps_val = gst_value_get_caps (value);
|
|
|
|
if (new_caps_val != NULL) {
|
|
new_caps = (GstCaps *) new_caps_val;
|
|
gst_caps_ref (new_caps);
|
|
}
|
|
|
|
GST_OBJECT_LOCK (audiomixer);
|
|
old_caps = audiomixer->filter_caps;
|
|
audiomixer->filter_caps = new_caps;
|
|
GST_OBJECT_UNLOCK (audiomixer);
|
|
|
|
if (old_caps)
|
|
gst_caps_unref (old_caps);
|
|
|
|
GST_DEBUG_OBJECT (audiomixer, "set new caps %" GST_PTR_FORMAT, new_caps);
|
|
break;
|
|
}
|
|
case PROP_ALIGNMENT_THRESHOLD:
|
|
audiomixer->alignment_threshold = g_value_get_uint64 (value);
|
|
break;
|
|
case PROP_DISCONT_WAIT:
|
|
audiomixer->discont_wait = g_value_get_uint64 (value);
|
|
break;
|
|
case PROP_BLOCKSIZE:
|
|
audiomixer->blocksize = g_value_get_uint (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_audiomixer_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_FILTER_CAPS:
|
|
GST_OBJECT_LOCK (audiomixer);
|
|
gst_value_set_caps (value, audiomixer->filter_caps);
|
|
GST_OBJECT_UNLOCK (audiomixer);
|
|
break;
|
|
case PROP_ALIGNMENT_THRESHOLD:
|
|
g_value_set_uint64 (value, audiomixer->alignment_threshold);
|
|
break;
|
|
case PROP_DISCONT_WAIT:
|
|
g_value_set_uint64 (value, audiomixer->discont_wait);
|
|
break;
|
|
case PROP_BLOCKSIZE:
|
|
g_value_set_uint (value, audiomixer->blocksize);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
free_pad (GstCollectData * data)
|
|
{
|
|
GstAudioMixerCollect *adata = (GstAudioMixerCollect *) data;
|
|
|
|
gst_buffer_replace (&adata->buffer, NULL);
|
|
}
|
|
|
|
static GstPad *
|
|
gst_audiomixer_request_new_pad (GstElement * element, GstPadTemplate * templ,
|
|
const gchar * unused, const GstCaps * caps)
|
|
{
|
|
gchar *name;
|
|
GstAudioMixer *audiomixer;
|
|
GstPad *newpad;
|
|
gint padcount;
|
|
GstCollectData *cdata;
|
|
GstAudioMixerCollect *adata;
|
|
|
|
if (templ->direction != GST_PAD_SINK)
|
|
goto not_sink;
|
|
|
|
audiomixer = GST_AUDIO_MIXER (element);
|
|
|
|
/* increment pad counter */
|
|
padcount = g_atomic_int_add (&audiomixer->padcount, 1);
|
|
|
|
name = g_strdup_printf ("sink_%u", padcount);
|
|
newpad = g_object_new (GST_TYPE_AUDIO_MIXER_PAD, "name", name, "direction",
|
|
templ->direction, "template", templ, NULL);
|
|
GST_DEBUG_OBJECT (audiomixer, "request new pad %s", name);
|
|
g_free (name);
|
|
|
|
cdata =
|
|
gst_collect_pads_add_pad (audiomixer->collect, newpad,
|
|
sizeof (GstAudioMixerCollect), free_pad, TRUE);
|
|
adata = (GstAudioMixerCollect *) cdata;
|
|
adata->buffer = NULL;
|
|
adata->position = 0;
|
|
adata->size = 0;
|
|
adata->output_offset = -1;
|
|
adata->next_offset = -1;
|
|
|
|
/* takes ownership of the pad */
|
|
if (!gst_element_add_pad (GST_ELEMENT (audiomixer), newpad))
|
|
goto could_not_add;
|
|
|
|
gst_child_proxy_child_added (GST_CHILD_PROXY (audiomixer), G_OBJECT (newpad),
|
|
GST_OBJECT_NAME (newpad));
|
|
|
|
return newpad;
|
|
|
|
/* errors */
|
|
not_sink:
|
|
{
|
|
g_warning ("gstaudiomixer: request new pad that is not a SINK pad\n");
|
|
return NULL;
|
|
}
|
|
could_not_add:
|
|
{
|
|
GST_DEBUG_OBJECT (audiomixer, "could not add pad");
|
|
gst_collect_pads_remove_pad (audiomixer->collect, newpad);
|
|
gst_object_unref (newpad);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_audiomixer_release_pad (GstElement * element, GstPad * pad)
|
|
{
|
|
GstAudioMixer *audiomixer;
|
|
|
|
audiomixer = GST_AUDIO_MIXER (element);
|
|
|
|
GST_DEBUG_OBJECT (audiomixer, "release pad %s:%s", GST_DEBUG_PAD_NAME (pad));
|
|
|
|
gst_child_proxy_child_removed (GST_CHILD_PROXY (audiomixer), G_OBJECT (pad),
|
|
GST_OBJECT_NAME (pad));
|
|
if (audiomixer->collect)
|
|
gst_collect_pads_remove_pad (audiomixer->collect, pad);
|
|
gst_element_remove_pad (element, pad);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_audiomixer_do_clip (GstCollectPads * pads, GstCollectData * data,
|
|
GstBuffer * buffer, GstBuffer ** out, gpointer user_data)
|
|
{
|
|
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (user_data);
|
|
gint rate, bpf;
|
|
|
|
rate = GST_AUDIO_INFO_RATE (&audiomixer->info);
|
|
bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
|
|
|
|
buffer = gst_audio_buffer_clip (buffer, &data->segment, rate, bpf);
|
|
|
|
*out = buffer;
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static gboolean
|
|
gst_audio_mixer_fill_buffer (GstAudioMixer * audiomixer, GstCollectPads * pads,
|
|
GstCollectData * collect_data, GstAudioMixerCollect * adata,
|
|
GstBuffer * inbuf)
|
|
{
|
|
GstClockTime start_time, end_time;
|
|
gboolean discont = FALSE;
|
|
guint64 start_offset, end_offset;
|
|
GstClockTime timestamp, stream_time;
|
|
gint rate, bpf;
|
|
|
|
g_assert (adata->buffer == NULL);
|
|
|
|
rate = GST_AUDIO_INFO_RATE (&audiomixer->info);
|
|
bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
|
|
|
|
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
|
|
stream_time =
|
|
gst_segment_to_stream_time (&collect_data->segment, GST_FORMAT_TIME,
|
|
timestamp);
|
|
|
|
/* sync object properties on stream time */
|
|
/* TODO: Ideally we would want to do that on every sample */
|
|
if (GST_CLOCK_TIME_IS_VALID (stream_time))
|
|
gst_object_sync_values (GST_OBJECT (collect_data->pad), stream_time);
|
|
|
|
adata->position = 0;
|
|
adata->size = gst_buffer_get_size (inbuf);
|
|
|
|
start_time = GST_BUFFER_TIMESTAMP (inbuf);
|
|
end_time =
|
|
start_time + gst_util_uint64_scale_ceil (adata->size / bpf,
|
|
GST_SECOND, rate);
|
|
|
|
start_offset = gst_util_uint64_scale (start_time, rate, GST_SECOND);
|
|
end_offset = start_offset + adata->size / bpf;
|
|
|
|
if (GST_BUFFER_IS_DISCONT (inbuf) || adata->next_offset == -1) {
|
|
discont = TRUE;
|
|
} else {
|
|
guint64 diff, max_sample_diff;
|
|
|
|
/* Check discont, based on audiobasesink */
|
|
if (start_offset <= adata->next_offset)
|
|
diff = adata->next_offset - start_offset;
|
|
else
|
|
diff = start_offset - adata->next_offset;
|
|
|
|
max_sample_diff =
|
|
gst_util_uint64_scale_int (audiomixer->alignment_threshold, rate,
|
|
GST_SECOND);
|
|
|
|
/* Discont! */
|
|
if (G_UNLIKELY (diff >= max_sample_diff)) {
|
|
if (audiomixer->discont_wait > 0) {
|
|
if (audiomixer->discont_time == GST_CLOCK_TIME_NONE) {
|
|
audiomixer->discont_time = start_time;
|
|
} else if (start_time - audiomixer->discont_time >=
|
|
audiomixer->discont_wait) {
|
|
discont = TRUE;
|
|
audiomixer->discont_time = GST_CLOCK_TIME_NONE;
|
|
}
|
|
} else {
|
|
discont = TRUE;
|
|
}
|
|
} else if (G_UNLIKELY (audiomixer->discont_time != GST_CLOCK_TIME_NONE)) {
|
|
/* we have had a discont, but are now back on track! */
|
|
audiomixer->discont_time = GST_CLOCK_TIME_NONE;
|
|
}
|
|
}
|
|
|
|
if (discont) {
|
|
/* Have discont, need resync */
|
|
if (adata->next_offset != -1)
|
|
GST_INFO_OBJECT (collect_data->pad, "Have discont. Expected %"
|
|
G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
|
|
adata->next_offset, start_offset);
|
|
adata->output_offset = -1;
|
|
} else {
|
|
audiomixer->discont_time = GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
adata->next_offset = end_offset;
|
|
|
|
if (adata->output_offset == -1) {
|
|
GstClockTime start_running_time;
|
|
GstClockTime end_running_time;
|
|
guint64 start_running_time_offset;
|
|
guint64 end_running_time_offset;
|
|
|
|
start_running_time =
|
|
gst_segment_to_running_time (&collect_data->segment,
|
|
GST_FORMAT_TIME, start_time);
|
|
end_running_time =
|
|
gst_segment_to_running_time (&collect_data->segment,
|
|
GST_FORMAT_TIME, end_time);
|
|
start_running_time_offset =
|
|
gst_util_uint64_scale (start_running_time, rate, GST_SECOND);
|
|
end_running_time_offset =
|
|
gst_util_uint64_scale (end_running_time, rate, GST_SECOND);
|
|
|
|
if (end_running_time_offset < audiomixer->offset) {
|
|
/* Before output segment, drop */
|
|
gst_buffer_unref (inbuf);
|
|
adata->buffer = NULL;
|
|
gst_buffer_unref (gst_collect_pads_pop (pads, collect_data));
|
|
adata->position = 0;
|
|
adata->size = 0;
|
|
adata->output_offset = -1;
|
|
GST_DEBUG_OBJECT (collect_data->pad,
|
|
"Buffer before segment or current position: %" G_GUINT64_FORMAT " < %"
|
|
G_GUINT64_FORMAT, end_running_time_offset, audiomixer->offset);
|
|
return FALSE;
|
|
}
|
|
|
|
if (start_running_time_offset < audiomixer->offset) {
|
|
guint diff = (audiomixer->offset - start_running_time_offset) * bpf;
|
|
adata->position += diff;
|
|
adata->size -= diff;
|
|
/* FIXME: This could only happen due to rounding errors */
|
|
if (adata->size == 0) {
|
|
/* Empty buffer, drop */
|
|
gst_buffer_unref (inbuf);
|
|
adata->buffer = NULL;
|
|
gst_buffer_unref (gst_collect_pads_pop (pads, collect_data));
|
|
adata->position = 0;
|
|
adata->size = 0;
|
|
adata->output_offset = -1;
|
|
GST_DEBUG_OBJECT (collect_data->pad,
|
|
"Buffer before segment or current position: %" G_GUINT64_FORMAT
|
|
" < %" G_GUINT64_FORMAT, end_running_time_offset,
|
|
audiomixer->offset);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
adata->output_offset = MAX (start_running_time_offset, audiomixer->offset);
|
|
GST_DEBUG_OBJECT (collect_data->pad,
|
|
"Buffer resynced: Pad offset %" G_GUINT64_FORMAT
|
|
", current mixer offset %" G_GUINT64_FORMAT, adata->output_offset,
|
|
audiomixer->offset);
|
|
}
|
|
|
|
GST_LOG_OBJECT (collect_data->pad,
|
|
"Queued new buffer at offset %" G_GUINT64_FORMAT, adata->output_offset);
|
|
adata->buffer = inbuf;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_audio_mixer_mix_buffer (GstAudioMixer * audiomixer, GstCollectPads * pads,
|
|
GstCollectData * collect_data, GstAudioMixerCollect * adata,
|
|
GstMapInfo * outmap)
|
|
{
|
|
GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (adata->collect.pad);
|
|
guint overlap;
|
|
guint out_start;
|
|
GstBuffer *inbuf;
|
|
GstMapInfo inmap;
|
|
gint bpf;
|
|
|
|
bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
|
|
|
|
/* Overlap => mix */
|
|
if (audiomixer->offset < adata->output_offset)
|
|
out_start = adata->output_offset - audiomixer->offset;
|
|
else
|
|
out_start = 0;
|
|
|
|
if (audiomixer->offset + audiomixer->blocksize + adata->position / bpf <
|
|
adata->output_offset + adata->size / bpf + out_start)
|
|
overlap = audiomixer->blocksize - out_start;
|
|
else
|
|
overlap = adata->size / bpf - adata->position / bpf;
|
|
|
|
inbuf = gst_collect_pads_peek (pads, collect_data);
|
|
g_assert (inbuf != NULL && inbuf == adata->buffer);
|
|
|
|
GST_OBJECT_LOCK (pad);
|
|
if (pad->mute || pad->volume < G_MINDOUBLE) {
|
|
GST_DEBUG_OBJECT (pad, "Skipping muted pad");
|
|
gst_buffer_unref (inbuf);
|
|
adata->position += adata->size;
|
|
adata->output_offset += adata->size / bpf;
|
|
if (adata->position >= adata->size) {
|
|
/* Buffer done, drop it */
|
|
gst_buffer_replace (&adata->buffer, NULL);
|
|
gst_buffer_unref (gst_collect_pads_pop (pads, collect_data));
|
|
}
|
|
GST_OBJECT_UNLOCK (pad);
|
|
return;
|
|
}
|
|
|
|
if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
|
|
/* skip gap buffer */
|
|
GST_LOG_OBJECT (pad, "skipping GAP buffer");
|
|
gst_buffer_unref (inbuf);
|
|
adata->position += adata->size;
|
|
adata->output_offset += adata->size / bpf;
|
|
/* Buffer done, drop it */
|
|
gst_buffer_replace (&adata->buffer, NULL);
|
|
gst_buffer_unref (gst_collect_pads_pop (pads, collect_data));
|
|
GST_OBJECT_UNLOCK (pad);
|
|
return;
|
|
}
|
|
|
|
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
|
|
GST_LOG_OBJECT (pad, "mixing %u bytes at offset %u from offset %u",
|
|
overlap * bpf, out_start * bpf, adata->position);
|
|
/* further buffers, need to add them */
|
|
if (pad->volume == 1.0) {
|
|
switch (audiomixer->info.finfo->format) {
|
|
case GST_AUDIO_FORMAT_U8:
|
|
audiomixer_orc_add_u8 ((gpointer) (outmap->data + out_start * bpf),
|
|
(gpointer) (inmap.data + adata->position),
|
|
overlap * audiomixer->info.channels);
|
|
break;
|
|
case GST_AUDIO_FORMAT_S8:
|
|
audiomixer_orc_add_s8 ((gpointer) (outmap->data + out_start * bpf),
|
|
(gpointer) (inmap.data + adata->position),
|
|
overlap * audiomixer->info.channels);
|
|
break;
|
|
case GST_AUDIO_FORMAT_U16:
|
|
audiomixer_orc_add_u16 ((gpointer) (outmap->data + out_start * bpf),
|
|
(gpointer) (inmap.data + adata->position),
|
|
overlap * audiomixer->info.channels);
|
|
break;
|
|
case GST_AUDIO_FORMAT_S16:
|
|
audiomixer_orc_add_s16 ((gpointer) (outmap->data + out_start * bpf),
|
|
(gpointer) (inmap.data + adata->position),
|
|
overlap * audiomixer->info.channels);
|
|
break;
|
|
case GST_AUDIO_FORMAT_U32:
|
|
audiomixer_orc_add_u32 ((gpointer) (outmap->data + out_start * bpf),
|
|
(gpointer) (inmap.data + adata->position),
|
|
overlap * audiomixer->info.channels);
|
|
break;
|
|
case GST_AUDIO_FORMAT_S32:
|
|
audiomixer_orc_add_s32 ((gpointer) (outmap->data + out_start * bpf),
|
|
(gpointer) (inmap.data + adata->position),
|
|
overlap * audiomixer->info.channels);
|
|
break;
|
|
case GST_AUDIO_FORMAT_F32:
|
|
audiomixer_orc_add_f32 ((gpointer) (outmap->data + out_start * bpf),
|
|
(gpointer) (inmap.data + adata->position),
|
|
overlap * audiomixer->info.channels);
|
|
break;
|
|
case GST_AUDIO_FORMAT_F64:
|
|
audiomixer_orc_add_f64 ((gpointer) (outmap->data + out_start * bpf),
|
|
(gpointer) (inmap.data + adata->position),
|
|
overlap * audiomixer->info.channels);
|
|
break;
|
|
default:
|
|
g_assert_not_reached ();
|
|
break;
|
|
}
|
|
} else {
|
|
switch (audiomixer->info.finfo->format) {
|
|
case GST_AUDIO_FORMAT_U8:
|
|
audiomixer_orc_add_volume_u8 ((gpointer) (outmap->data +
|
|
out_start * bpf), (gpointer) (inmap.data + adata->position),
|
|
pad->volume_i8, overlap * audiomixer->info.channels);
|
|
break;
|
|
case GST_AUDIO_FORMAT_S8:
|
|
audiomixer_orc_add_volume_s8 ((gpointer) (outmap->data +
|
|
out_start * bpf), (gpointer) (inmap.data + adata->position),
|
|
pad->volume_i8, overlap * audiomixer->info.channels);
|
|
break;
|
|
case GST_AUDIO_FORMAT_U16:
|
|
audiomixer_orc_add_volume_u16 ((gpointer) (outmap->data +
|
|
out_start * bpf), (gpointer) (inmap.data + adata->position),
|
|
pad->volume_i16, overlap * audiomixer->info.channels);
|
|
break;
|
|
case GST_AUDIO_FORMAT_S16:
|
|
audiomixer_orc_add_volume_s16 ((gpointer) (outmap->data +
|
|
out_start * bpf), (gpointer) (inmap.data + adata->position),
|
|
pad->volume_i16, overlap * audiomixer->info.channels);
|
|
break;
|
|
case GST_AUDIO_FORMAT_U32:
|
|
audiomixer_orc_add_volume_u32 ((gpointer) (outmap->data +
|
|
out_start * bpf), (gpointer) (inmap.data + adata->position),
|
|
pad->volume_i32, overlap * audiomixer->info.channels);
|
|
break;
|
|
case GST_AUDIO_FORMAT_S32:
|
|
audiomixer_orc_add_volume_s32 ((gpointer) (outmap->data +
|
|
out_start * bpf), (gpointer) (inmap.data + adata->position),
|
|
pad->volume_i32, overlap * audiomixer->info.channels);
|
|
break;
|
|
case GST_AUDIO_FORMAT_F32:
|
|
audiomixer_orc_add_volume_f32 ((gpointer) (outmap->data +
|
|
out_start * bpf), (gpointer) (inmap.data + adata->position),
|
|
pad->volume, overlap * audiomixer->info.channels);
|
|
break;
|
|
case GST_AUDIO_FORMAT_F64:
|
|
audiomixer_orc_add_volume_f64 ((gpointer) (outmap->data +
|
|
out_start * bpf), (gpointer) (inmap.data + adata->position),
|
|
pad->volume, overlap * audiomixer->info.channels);
|
|
break;
|
|
default:
|
|
g_assert_not_reached ();
|
|
break;
|
|
}
|
|
}
|
|
gst_buffer_unmap (inbuf, &inmap);
|
|
gst_buffer_unref (inbuf);
|
|
|
|
adata->position += overlap * bpf;
|
|
adata->output_offset += overlap;
|
|
|
|
if (adata->position == adata->size) {
|
|
/* Buffer done, drop it */
|
|
gst_buffer_replace (&adata->buffer, NULL);
|
|
gst_buffer_unref (gst_collect_pads_pop (pads, collect_data));
|
|
GST_DEBUG_OBJECT (pad, "Finished mixing buffer, waiting for next");
|
|
}
|
|
|
|
GST_OBJECT_UNLOCK (pad);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_audiomixer_collected (GstCollectPads * pads, gpointer user_data)
|
|
{
|
|
/* Get all pads that have data for us and store them in a
|
|
* new list.
|
|
*
|
|
* Calculate the current output offset/timestamp and
|
|
* offset_end/timestamp_end. Allocate a silence buffer
|
|
* for this and store it.
|
|
*
|
|
* For all pads:
|
|
* 1) Once per input buffer (cached)
|
|
* 1) Check discont (flag and timestamp with tolerance)
|
|
* 2) If discont or new, resync. That means:
|
|
* 1) Drop all start data of the buffer that comes before
|
|
* the current position/offset.
|
|
* 2) Calculate the offset (output segment!) that the first
|
|
* frame of the input buffer corresponds to. Base this on
|
|
* the running time.
|
|
*
|
|
* 2) If the current pad's offset/offset_end overlaps with the output
|
|
* offset/offset_end, mix it at the appropiate position in the output
|
|
* buffer and advance the pad's position. Remember if this pad needs
|
|
* a new buffer to advance behind the output offset_end.
|
|
*
|
|
* 3) If we had no pad with a buffer, go EOS.
|
|
*
|
|
* 4) If we had at least one pad that did not advance behind output
|
|
* offset_end, let collected be called again for the current
|
|
* output offset/offset_end.
|
|
*/
|
|
GstAudioMixer *audiomixer;
|
|
GSList *collected;
|
|
GstFlowReturn ret;
|
|
GstBuffer *outbuf = NULL;
|
|
GstMapInfo outmap;
|
|
gint64 next_offset;
|
|
gint64 next_timestamp;
|
|
gint rate, bpf;
|
|
gboolean dropped = FALSE;
|
|
gboolean is_eos = TRUE;
|
|
gboolean is_done = TRUE;
|
|
|
|
audiomixer = GST_AUDIO_MIXER (user_data);
|
|
|
|
/* this is fatal */
|
|
if (G_UNLIKELY (audiomixer->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN))
|
|
goto not_negotiated;
|
|
|
|
if (audiomixer->flush_stop_pending == TRUE) {
|
|
GST_INFO_OBJECT (audiomixer->srcpad, "send pending flush stop event");
|
|
if (!gst_pad_push_event (audiomixer->srcpad,
|
|
gst_event_new_flush_stop (TRUE))) {
|
|
GST_WARNING_OBJECT (audiomixer->srcpad,
|
|
"Sending flush stop event failed");
|
|
}
|
|
|
|
audiomixer->flush_stop_pending = FALSE;
|
|
gst_buffer_replace (&audiomixer->current_buffer, NULL);
|
|
audiomixer->discont_time = GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
if (audiomixer->send_stream_start) {
|
|
gchar s_id[32];
|
|
|
|
GST_INFO_OBJECT (audiomixer->srcpad, "send pending stream start event");
|
|
/* stream-start (FIXME: create id based on input ids) */
|
|
g_snprintf (s_id, sizeof (s_id), "audiomixer-%08x", g_random_int ());
|
|
if (!gst_pad_push_event (audiomixer->srcpad,
|
|
gst_event_new_stream_start (s_id))) {
|
|
GST_WARNING_OBJECT (audiomixer->srcpad,
|
|
"Sending stream start event failed");
|
|
}
|
|
audiomixer->send_stream_start = FALSE;
|
|
}
|
|
|
|
if (audiomixer->send_caps) {
|
|
GstEvent *caps_event;
|
|
|
|
caps_event = gst_event_new_caps (audiomixer->current_caps);
|
|
GST_INFO_OBJECT (audiomixer->srcpad,
|
|
"send pending caps event %" GST_PTR_FORMAT, caps_event);
|
|
if (!gst_pad_push_event (audiomixer->srcpad, caps_event)) {
|
|
GST_WARNING_OBJECT (audiomixer->srcpad, "Sending caps event failed");
|
|
}
|
|
audiomixer->send_caps = FALSE;
|
|
}
|
|
|
|
rate = GST_AUDIO_INFO_RATE (&audiomixer->info);
|
|
bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
|
|
|
|
if (g_atomic_int_compare_and_exchange (&audiomixer->segment_pending, TRUE,
|
|
FALSE)) {
|
|
GstEvent *event;
|
|
|
|
/*
|
|
* When seeking we set the start and stop positions as given in the seek
|
|
* event. We also adjust offset & timestamp accordingly.
|
|
* This basically ignores all newsegments sent by upstream.
|
|
*
|
|
* FIXME: We require that all inputs have the same rate currently
|
|
* as we do no rate conversion!
|
|
*/
|
|
event = gst_event_new_segment (&audiomixer->segment);
|
|
if (audiomixer->segment.rate > 0.0) {
|
|
audiomixer->segment.position = audiomixer->segment.start;
|
|
} else {
|
|
audiomixer->segment.position = audiomixer->segment.stop;
|
|
}
|
|
audiomixer->offset = gst_util_uint64_scale (audiomixer->segment.position,
|
|
rate, GST_SECOND);
|
|
|
|
GST_INFO_OBJECT (audiomixer->srcpad, "sending pending new segment event %"
|
|
GST_SEGMENT_FORMAT, &audiomixer->segment);
|
|
if (event) {
|
|
if (!gst_pad_push_event (audiomixer->srcpad, event)) {
|
|
GST_WARNING_OBJECT (audiomixer->srcpad,
|
|
"Sending new segment event failed");
|
|
}
|
|
} else {
|
|
GST_WARNING_OBJECT (audiomixer->srcpad, "Creating new segment event for "
|
|
"start:%" G_GINT64_FORMAT " end:%" G_GINT64_FORMAT " failed",
|
|
audiomixer->segment.start, audiomixer->segment.stop);
|
|
}
|
|
}
|
|
|
|
if (G_UNLIKELY (audiomixer->pending_events)) {
|
|
GList *tmp = audiomixer->pending_events;
|
|
|
|
while (tmp) {
|
|
GstEvent *ev = (GstEvent *) tmp->data;
|
|
|
|
gst_pad_push_event (audiomixer->srcpad, ev);
|
|
tmp = g_list_next (tmp);
|
|
}
|
|
g_list_free (audiomixer->pending_events);
|
|
audiomixer->pending_events = NULL;
|
|
}
|
|
|
|
/* for the next timestamp, use the sample counter, which will
|
|
* never accumulate rounding errors */
|
|
|
|
/* FIXME: Reverse mixing does not work at all yet */
|
|
if (audiomixer->segment.rate > 0.0) {
|
|
next_offset = audiomixer->offset + audiomixer->blocksize;
|
|
} else {
|
|
next_offset = audiomixer->offset - audiomixer->blocksize;
|
|
}
|
|
|
|
next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate);
|
|
|
|
if (audiomixer->current_buffer) {
|
|
outbuf = audiomixer->current_buffer;
|
|
} else {
|
|
outbuf = gst_buffer_new_and_alloc (audiomixer->blocksize * bpf);
|
|
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
|
|
gst_audio_format_fill_silence (audiomixer->info.finfo, outmap.data,
|
|
outmap.size);
|
|
gst_buffer_unmap (outbuf, &outmap);
|
|
audiomixer->current_buffer = outbuf;
|
|
}
|
|
|
|
GST_LOG_OBJECT (audiomixer,
|
|
"Starting to mix %u samples for offset %" G_GUINT64_FORMAT
|
|
" with timestamp %" GST_TIME_FORMAT, audiomixer->blocksize,
|
|
audiomixer->offset, GST_TIME_ARGS (audiomixer->segment.position));
|
|
|
|
gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE);
|
|
|
|
for (collected = pads->data; collected; collected = collected->next) {
|
|
GstCollectData *collect_data;
|
|
GstAudioMixerCollect *adata;
|
|
GstBuffer *inbuf;
|
|
|
|
collect_data = (GstCollectData *) collected->data;
|
|
adata = (GstAudioMixerCollect *) collect_data;
|
|
|
|
inbuf = gst_collect_pads_peek (pads, collect_data);
|
|
if (!inbuf)
|
|
continue;
|
|
|
|
/* New buffer? */
|
|
if (!adata->buffer || adata->buffer != inbuf) {
|
|
/* Takes ownership of buffer */
|
|
if (!gst_audio_mixer_fill_buffer (audiomixer, pads, collect_data, adata,
|
|
inbuf)) {
|
|
dropped = TRUE;
|
|
continue;
|
|
}
|
|
} else {
|
|
gst_buffer_unref (inbuf);
|
|
}
|
|
|
|
if (!adata->buffer && !dropped
|
|
&& GST_COLLECT_PADS_STATE_IS_SET (&adata->collect,
|
|
GST_COLLECT_PADS_STATE_EOS)) {
|
|
GST_DEBUG_OBJECT (collect_data->pad, "Pad is in EOS state");
|
|
} else {
|
|
is_eos = FALSE;
|
|
}
|
|
|
|
/* At this point adata->output_offset >= audiomixer->offset or we have no buffer anymore */
|
|
if (adata->output_offset >= audiomixer->offset
|
|
&& adata->output_offset <
|
|
audiomixer->offset + audiomixer->blocksize && adata->buffer) {
|
|
GST_LOG_OBJECT (collect_data->pad, "Mixing buffer for current offset");
|
|
gst_audio_mixer_mix_buffer (audiomixer, pads, collect_data, adata,
|
|
&outmap);
|
|
if (adata->output_offset >= next_offset) {
|
|
GST_DEBUG_OBJECT (collect_data->pad,
|
|
"Pad is after current offset: %" G_GUINT64_FORMAT " >= %"
|
|
G_GUINT64_FORMAT, adata->output_offset, next_offset);
|
|
} else {
|
|
is_done = FALSE;
|
|
}
|
|
}
|
|
}
|
|
|
|
gst_buffer_unmap (outbuf, &outmap);
|
|
|
|
if (dropped) {
|
|
/* We dropped a buffer, retry */
|
|
GST_DEBUG_OBJECT (audiomixer,
|
|
"A pad dropped a buffer, wait for the next one");
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
if (!is_done && !is_eos) {
|
|
/* Get more buffers */
|
|
GST_DEBUG_OBJECT (audiomixer,
|
|
"We're not done yet for the current offset," " waiting for more data");
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
if (is_eos) {
|
|
gint64 max_offset = 0;
|
|
gboolean empty_buffer = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (audiomixer, "We're EOS");
|
|
|
|
|
|
for (collected = pads->data; collected; collected = collected->next) {
|
|
GstCollectData *collect_data;
|
|
GstAudioMixerCollect *adata;
|
|
|
|
collect_data = (GstCollectData *) collected->data;
|
|
adata = (GstAudioMixerCollect *) collect_data;
|
|
|
|
max_offset = MAX (max_offset, adata->output_offset);
|
|
if (adata->output_offset > audiomixer->offset)
|
|
empty_buffer = FALSE;
|
|
}
|
|
|
|
/* This means EOS or no pads at all */
|
|
if (empty_buffer) {
|
|
gst_buffer_replace (&audiomixer->current_buffer, NULL);
|
|
goto eos;
|
|
}
|
|
|
|
if (max_offset <= next_offset) {
|
|
GST_DEBUG_OBJECT (audiomixer,
|
|
"Last buffer is incomplete: %" G_GUINT64_FORMAT " <= %"
|
|
G_GUINT64_FORMAT, max_offset, next_offset);
|
|
next_offset = max_offset;
|
|
|
|
gst_buffer_resize (outbuf, 0, (next_offset - audiomixer->offset) * bpf);
|
|
next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate);
|
|
}
|
|
}
|
|
|
|
/* set timestamps on the output buffer */
|
|
if (audiomixer->segment.rate > 0.0) {
|
|
GST_BUFFER_TIMESTAMP (outbuf) = audiomixer->segment.position;
|
|
GST_BUFFER_OFFSET (outbuf) = audiomixer->offset;
|
|
GST_BUFFER_OFFSET_END (outbuf) = next_offset;
|
|
GST_BUFFER_DURATION (outbuf) =
|
|
next_timestamp - audiomixer->segment.position;
|
|
} else {
|
|
GST_BUFFER_TIMESTAMP (outbuf) = next_timestamp;
|
|
GST_BUFFER_OFFSET (outbuf) = next_offset;
|
|
GST_BUFFER_OFFSET_END (outbuf) = audiomixer->offset;
|
|
GST_BUFFER_DURATION (outbuf) =
|
|
audiomixer->segment.position - next_timestamp;
|
|
}
|
|
|
|
audiomixer->offset = next_offset;
|
|
audiomixer->segment.position = next_timestamp;
|
|
|
|
/* send it out */
|
|
GST_LOG_OBJECT (audiomixer,
|
|
"pushing outbuf %p, timestamp %" GST_TIME_FORMAT " offset %"
|
|
G_GINT64_FORMAT, outbuf, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
|
|
GST_BUFFER_OFFSET (outbuf));
|
|
|
|
ret = gst_pad_push (audiomixer->srcpad, outbuf);
|
|
audiomixer->current_buffer = NULL;
|
|
|
|
GST_LOG_OBJECT (audiomixer, "pushed outbuf, result = %s",
|
|
gst_flow_get_name (ret));
|
|
|
|
if (ret == GST_FLOW_OK && is_eos)
|
|
goto eos;
|
|
|
|
return ret;
|
|
/* ERRORS */
|
|
not_negotiated:
|
|
{
|
|
GST_ELEMENT_ERROR (audiomixer, STREAM, FORMAT, (NULL),
|
|
("Unknown data received, not negotiated"));
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
}
|
|
|
|
eos:
|
|
{
|
|
GST_DEBUG_OBJECT (audiomixer, "EOS");
|
|
gst_pad_push_event (audiomixer->srcpad, gst_event_new_eos ());
|
|
return GST_FLOW_EOS;
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_audiomixer_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstAudioMixer *audiomixer;
|
|
GstStateChangeReturn ret;
|
|
|
|
audiomixer = GST_AUDIO_MIXER (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
audiomixer->offset = 0;
|
|
audiomixer->flush_stop_pending = FALSE;
|
|
audiomixer->segment_pending = TRUE;
|
|
audiomixer->send_stream_start = TRUE;
|
|
audiomixer->send_caps = TRUE;
|
|
gst_caps_replace (&audiomixer->current_caps, NULL);
|
|
gst_segment_init (&audiomixer->segment, GST_FORMAT_TIME);
|
|
gst_collect_pads_start (audiomixer->collect);
|
|
audiomixer->discont_time = GST_CLOCK_TIME_NONE;
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
/* need to unblock the collectpads before calling the
|
|
* parent change_state so that streaming can finish */
|
|
gst_collect_pads_stop (audiomixer->collect);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_buffer_replace (&audiomixer->current_buffer, NULL);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* GstChildProxy implementation */
|
|
static GObject *
|
|
gst_audiomixer_child_proxy_get_child_by_index (GstChildProxy * child_proxy,
|
|
guint index)
|
|
{
|
|
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (child_proxy);
|
|
GObject *obj = NULL;
|
|
|
|
GST_OBJECT_LOCK (audiomixer);
|
|
obj = g_list_nth_data (GST_ELEMENT_CAST (audiomixer)->sinkpads, index);
|
|
if (obj)
|
|
gst_object_ref (obj);
|
|
GST_OBJECT_UNLOCK (audiomixer);
|
|
|
|
return obj;
|
|
}
|
|
|
|
static guint
|
|
gst_audiomixer_child_proxy_get_children_count (GstChildProxy * child_proxy)
|
|
{
|
|
guint count = 0;
|
|
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (child_proxy);
|
|
|
|
GST_OBJECT_LOCK (audiomixer);
|
|
count = GST_ELEMENT_CAST (audiomixer)->numsinkpads;
|
|
GST_OBJECT_UNLOCK (audiomixer);
|
|
GST_INFO_OBJECT (audiomixer, "Children Count: %d", count);
|
|
|
|
return count;
|
|
}
|
|
|
|
static void
|
|
gst_audiomixer_child_proxy_init (gpointer g_iface, gpointer iface_data)
|
|
{
|
|
GstChildProxyInterface *iface = g_iface;
|
|
|
|
GST_INFO ("intializing child proxy interface");
|
|
iface->get_child_by_index = gst_audiomixer_child_proxy_get_child_by_index;
|
|
iface->get_children_count = gst_audiomixer_child_proxy_get_children_count;
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "audiomixer", 0,
|
|
"audio mixing element");
|
|
|
|
if (!gst_element_register (plugin, "audiomixer", GST_RANK_NONE,
|
|
GST_TYPE_AUDIO_MIXER))
|
|
return FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
audiomixer,
|
|
"Mixes multiple audio streams",
|
|
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
|