mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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8ca40fa86f
Fixes stuttering audio when iOS AU is resampling. To make AU resample, one has to request a rate that differs from AVAudioSession's sampleRate. The resampling itself is not the culprit, but rather our API misuse. AudioUnitRender modifies the mDataByteSize members with the actual read bytes count. Therefore, they must be reinitialized before each AudioUnitRender. (The buffers themselves can be preallocated.) The "stutter" was caused by one AudioUnitRender making the buffer too small for other AudioUnitRender invocations, making them fail with -50 (paramErr). By way of luck, when AU didn't resample, all AudioUnitRender invocations read the same number of bytes. (This patch addresses some non-interleaved audio concerns, but at this moment the elements do not support non-interleaved audio and non-interleaved is untested.) https://bugzilla.gnome.org/show_bug.cgi?id=744922
444 lines
13 KiB
C
444 lines
13 KiB
C
/*
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* GStreamer
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* Copyright (C) 2005,2006 Zaheer Abbas Merali <zaheerabbas at merali dot org>
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* Copyright (C) 2008 Pioneers of the Inevitable <songbird@songbirdnest.com>
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*
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* Permission is hereby granted, free of charge, to any person obtaining a
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* copy of this software and associated documentation files (the "Software"),
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* to deal in the Software without restriction, including without limitation
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* the rights to use, copy, modify, merge, publish, distribute, sublicense,
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* and/or sell copies of the Software, and to permit persons to whom the
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* Software is furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
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* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
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* DEALINGS IN THE SOFTWARE.
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*
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* Alternatively, the contents of this file may be used under the
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* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
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* which case the following provisions apply instead of the ones
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* mentioned above:
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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-osxaudiosrc
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*
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* This element captures raw audio samples using the CoreAudio api.
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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-1.0 osxaudiosrc ! wavenc ! filesink location=audio.wav
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* ]|
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* </refsect2>
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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 <gst/gst.h>
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#include "gstosxaudiosrc.h"
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#include "gstosxaudioelement.h"
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GST_DEBUG_CATEGORY_STATIC (osx_audiosrc_debug);
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#define GST_CAT_DEFAULT osx_audiosrc_debug
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/* Filter signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0,
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ARG_DEVICE
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};
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static GstStaticPadTemplate src_factory = 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 ("audio/x-raw, "
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"format = (string) " GST_AUDIO_FORMATS_ALL ", "
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"layout = (string) interleaved, "
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"rate = (int) [1, MAX], " "channels = (int) [1, MAX]")
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);
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static void gst_osx_audio_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_osx_audio_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstStateChangeReturn
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gst_osx_audio_src_change_state (GstElement * element,
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GstStateChange transition);
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static void gst_osx_audio_src_probe_caps (GstOsxAudioSrc * src);
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static GstCaps *gst_osx_audio_src_get_caps (GstBaseSrc * src, GstCaps * filter);
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static GstAudioRingBuffer *gst_osx_audio_src_create_ringbuffer (GstAudioBaseSrc
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* src);
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static void gst_osx_audio_src_osxelement_init (gpointer g_iface,
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gpointer iface_data);
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static OSStatus gst_osx_audio_src_io_proc (GstOsxAudioRingBuffer * buf,
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AudioUnitRenderActionFlags * ioActionFlags,
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const AudioTimeStamp * inTimeStamp, UInt32 inBusNumber,
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UInt32 inNumberFrames, AudioBufferList * bufferList);
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static void
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gst_osx_audio_src_do_init (GType type)
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{
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static const GInterfaceInfo osxelement_info = {
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gst_osx_audio_src_osxelement_init,
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NULL,
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NULL
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};
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GST_DEBUG_CATEGORY_INIT (osx_audiosrc_debug, "osxaudiosrc", 0,
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"OSX Audio Src");
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g_type_add_interface_static (type, GST_OSX_AUDIO_ELEMENT_TYPE,
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&osxelement_info);
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}
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#define gst_osx_audio_src_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstOsxAudioSrc, gst_osx_audio_src,
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GST_TYPE_AUDIO_BASE_SRC, gst_osx_audio_src_do_init (g_define_type_id));
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static void
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gst_osx_audio_src_class_init (GstOsxAudioSrcClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseSrcClass *gstbasesrc_class;
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GstAudioBaseSrcClass *gstaudiobasesrc_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasesrc_class = (GstBaseSrcClass *) klass;
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gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass;
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gobject_class->set_property = gst_osx_audio_src_set_property;
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gobject_class->get_property = gst_osx_audio_src_get_property;
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_osx_audio_src_change_state);
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gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_osx_audio_src_get_caps);
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g_object_class_install_property (gobject_class, ARG_DEVICE,
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g_param_spec_int ("device", "Device ID", "Device ID of input device",
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0, G_MAXINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gstaudiobasesrc_class->create_ringbuffer =
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GST_DEBUG_FUNCPTR (gst_osx_audio_src_create_ringbuffer);
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&src_factory));
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gst_element_class_set_static_metadata (gstelement_class, "Audio Source (OSX)",
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"Source/Audio",
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"Input from a sound card in OS X",
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"Zaheer Abbas Merali <zaheerabbas at merali dot org>");
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}
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static void
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gst_osx_audio_src_init (GstOsxAudioSrc * src)
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{
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gst_base_src_set_live (GST_BASE_SRC (src), TRUE);
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src->device_id = kAudioDeviceUnknown;
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src->cached_caps = NULL;
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}
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static void
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gst_osx_audio_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstOsxAudioSrc *src = GST_OSX_AUDIO_SRC (object);
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switch (prop_id) {
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case ARG_DEVICE:
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src->device_id = g_value_get_int (value);
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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_osx_audio_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstOsxAudioSrc *src = GST_OSX_AUDIO_SRC (object);
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switch (prop_id) {
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case ARG_DEVICE:
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g_value_set_int (value, src->device_id);
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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 GstStateChangeReturn
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gst_osx_audio_src_change_state (GstElement * element, GstStateChange transition)
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{
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GstOsxAudioSrc *osxsrc = GST_OSX_AUDIO_SRC (element);
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GstOsxAudioRingBuffer *ringbuffer;
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GstStateChangeReturn ret;
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switch (transition) {
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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if (ret == GST_STATE_CHANGE_FAILURE)
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goto out;
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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/* The device is open now, so fix our device_id if it changed */
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ringbuffer =
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GST_OSX_AUDIO_RING_BUFFER (GST_AUDIO_BASE_SRC (osxsrc)->ringbuffer);
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if (ringbuffer->core_audio->device_id != osxsrc->device_id) {
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osxsrc->device_id = ringbuffer->core_audio->device_id;
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g_object_notify (G_OBJECT (osxsrc), "device");
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}
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break;
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default:
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break;
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}
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out:
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return ret;
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}
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static void
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gst_osx_audio_src_probe_caps (GstOsxAudioSrc * osxsrc)
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{
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GstOsxAudioRingBuffer *ringbuffer =
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GST_OSX_AUDIO_RING_BUFFER (GST_AUDIO_BASE_SRC (osxsrc)->ringbuffer);
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GstCoreAudio *core_audio = ringbuffer->core_audio;
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GstCaps *caps;
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gint channels;
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AudioChannelLayout *layout;
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AudioStreamBasicDescription asbd_in;
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UInt32 propertySize;
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OSStatus status;
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propertySize = sizeof (asbd_in);
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status = AudioUnitGetProperty (core_audio->audiounit,
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kAudioUnitProperty_StreamFormat,
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kAudioUnitScope_Input, 1, &asbd_in, &propertySize);
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if (status)
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goto fail;
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layout = gst_core_audio_audio_device_get_channel_layout (osxsrc->device_id,
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FALSE);
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if (layout) {
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channels = MIN (layout->mNumberChannelDescriptions,
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GST_OSX_AUDIO_MAX_CHANNEL);
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} else {
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GST_WARNING_OBJECT (osxsrc, "This driver does not support "
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"kAudioDevicePropertyPreferredChannelLayout.");
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channels = 2;
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}
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caps = gst_core_audio_asbd_to_caps (&asbd_in, layout);
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if (!caps) {
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GST_WARNING_OBJECT (osxsrc, "Could not get caps from stream description");
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g_free (layout);
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goto fail;
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} else {
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GST_DEBUG_OBJECT (osxsrc, "Got caps on device: %p", caps);
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}
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g_free (layout);
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if (osxsrc->cached_caps)
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gst_caps_unref (osxsrc->cached_caps);
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osxsrc->cached_caps = caps;
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return;
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fail:
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AudioComponentInstanceDispose (core_audio->audiounit);
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core_audio->audiounit = NULL;
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GST_WARNING_OBJECT (osxsrc,
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"Unable to obtain device properties: %d", (int) status);
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}
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static GstCaps *
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gst_osx_audio_src_get_caps (GstBaseSrc * src, GstCaps * filter)
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{
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GstElementClass *gstelement_class;
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GstOsxAudioSrc *osxsrc;
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GstAudioRingBuffer *buf;
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GstCaps *ret = NULL;
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gstelement_class = GST_ELEMENT_GET_CLASS (src);
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osxsrc = GST_OSX_AUDIO_SRC (src);
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GST_OBJECT_LOCK (osxsrc);
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buf = GST_AUDIO_BASE_SRC (src)->ringbuffer;
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if (buf)
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gst_object_ref (buf);
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GST_OBJECT_UNLOCK (osxsrc);
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if (buf) {
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GST_OBJECT_LOCK (buf);
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if (buf->acquired && buf->spec.caps) {
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/* Caps are fixed, use what we have */
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ret = gst_caps_ref (buf->spec.caps);
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}
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if (!ret && buf->open && !osxsrc->cached_caps) {
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/* Device is open, let's probe its caps */
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gst_osx_audio_src_probe_caps (osxsrc);
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}
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GST_OBJECT_UNLOCK (buf);
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gst_object_unref (buf);
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}
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if (!ret && osxsrc->cached_caps)
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ret = gst_caps_ref (osxsrc->cached_caps);
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if (filter) {
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GstCaps *tmp;
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tmp = gst_caps_intersect_full (filter, ret, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (ret);
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ret = tmp;
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}
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return ret;
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}
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static GstAudioRingBuffer *
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gst_osx_audio_src_create_ringbuffer (GstAudioBaseSrc * src)
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{
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GstOsxAudioSrc *osxsrc;
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GstOsxAudioRingBuffer *ringbuffer;
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osxsrc = GST_OSX_AUDIO_SRC (src);
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GST_DEBUG_OBJECT (osxsrc, "Creating ringbuffer");
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ringbuffer = g_object_new (GST_TYPE_OSX_AUDIO_RING_BUFFER, NULL);
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GST_DEBUG_OBJECT (osxsrc, "osx src 0x%p element 0x%p ioproc 0x%p", osxsrc,
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GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsrc),
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(void *) gst_osx_audio_src_io_proc);
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ringbuffer->core_audio->element =
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GST_OSX_AUDIO_ELEMENT_GET_INTERFACE (osxsrc);
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ringbuffer->core_audio->is_src = TRUE;
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/* By default the coreaudio instance created by the ringbuffer
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* has device_id==kAudioDeviceUnknown. The user might have
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* selected a different one here
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*/
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if (ringbuffer->core_audio->device_id != osxsrc->device_id)
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ringbuffer->core_audio->device_id = osxsrc->device_id;
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return GST_AUDIO_RING_BUFFER (ringbuffer);
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}
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static OSStatus
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gst_osx_audio_src_io_proc (GstOsxAudioRingBuffer * buf,
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AudioUnitRenderActionFlags * ioActionFlags,
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const AudioTimeStamp * inTimeStamp,
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UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList * bufferList)
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{
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OSStatus status;
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guint8 *writeptr;
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gint writeseg;
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gint len;
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gint remaining;
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UInt32 n;
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gint offset = 0;
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/* Previous invoke of AudioUnitRender changed mDataByteSize into
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* number of bytes actually read. Reset the members. */
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for (n = 0; n < buf->core_audio->recBufferList->mNumberBuffers; ++n) {
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buf->core_audio->recBufferList->mBuffers[n].mDataByteSize =
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buf->core_audio->recBufferSize;
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}
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status = AudioUnitRender (buf->core_audio->audiounit, ioActionFlags,
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inTimeStamp, inBusNumber, inNumberFrames, buf->core_audio->recBufferList);
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if (status) {
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GST_WARNING_OBJECT (buf, "AudioUnitRender returned %d", (int) status);
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return status;
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}
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/* TODO: To support non-interleaved audio, go over all mBuffers,
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* not just the first one. */
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remaining = buf->core_audio->recBufferList->mBuffers[0].mDataByteSize;
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while (remaining) {
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if (!gst_audio_ring_buffer_prepare_read (GST_AUDIO_RING_BUFFER (buf),
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&writeseg, &writeptr, &len))
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return 0;
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len -= buf->segoffset;
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if (len > remaining)
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len = remaining;
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memcpy (writeptr + buf->segoffset,
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(char *) buf->core_audio->recBufferList->mBuffers[0].mData + offset,
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len);
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buf->segoffset += len;
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offset += len;
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remaining -= len;
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if ((gint) buf->segoffset == GST_AUDIO_RING_BUFFER (buf)->spec.segsize) {
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/* we wrote one segment */
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gst_audio_ring_buffer_advance (GST_AUDIO_RING_BUFFER (buf), 1);
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buf->segoffset = 0;
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}
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}
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return 0;
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}
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static void
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gst_osx_audio_src_osxelement_init (gpointer g_iface, gpointer iface_data)
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{
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GstOsxAudioElementInterface *iface = (GstOsxAudioElementInterface *) g_iface;
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iface->io_proc = (AURenderCallback) gst_osx_audio_src_io_proc;
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}
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