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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1698d3254d
Just use plain memmove(), g_memmove() is deprecated in recent GLib versions. https://bugzilla.gnome.org/show_bug.cgi?id=712811
1101 lines
34 KiB
C
1101 lines
34 KiB
C
/*
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* GStreamer QuickTime audio decoder codecs wrapper
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* Copyright <2006, 2007> Fluendo <gstreamer@fluendo.com>
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* Copyright <2006, 2007, 2008> Pioneers of the Inevitable
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* <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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <gst/gst.h>
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#include "qtwrapper.h"
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#include "codecmapping.h"
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#include "qtutils.h"
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#ifdef G_OS_WIN32
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#include <QuickTimeComponents.h>
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#else
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#include <QuickTime/QuickTimeComponents.h>
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#endif
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#define QTWRAPPER_ADEC_PARAMS_QDATA g_quark_from_static_string("qtwrapper-adec-params")
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#define NO_MORE_INPUT_DATA 42
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static GstStaticPadTemplate src_templ = 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-float, "
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"endianness = (int) {" G_STRINGIFY (G_BYTE_ORDER) " }, "
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"signed = (boolean) { TRUE }, "
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"width = (int) 32, "
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"depth = (int) 32, " "rate = (int) [1, MAX], "
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"channels = (int) [1, MAX]")
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);
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typedef struct _QTWrapperAudioDecoder QTWrapperAudioDecoder;
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typedef struct _QTWrapperAudioDecoderClass QTWrapperAudioDecoderClass;
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struct _QTWrapperAudioDecoder
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{
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GstElement parent;
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GstPad *sinkpad;
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GstPad *srcpad;
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/* FIXME : all following should be protected by a mutex */
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ComponentInstance adec; /* The Audio Decoder component */
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AudioStreamBasicDescription indesc, outdesc;
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guint samplerate;
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guint channels;
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AudioBufferList *bufferlist;
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AudioStreamPacketDescription aspd[1];
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/* first time received after NEWSEGMENT */
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GstClockTime initial_time;
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/* offset in samples from the initial time */
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guint64 cur_offset;
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/* TRUE just after receiving a NEWSEGMENT */
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gboolean gotnewsegment;
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/* Data for StdAudio callbacks */
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GstBuffer *input_buffer;
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};
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struct _QTWrapperAudioDecoderClass
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{
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GstElementClass parent_class;
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/* fourcc of the format */
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guint32 componentSubType;
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GstPadTemplate *sinktempl;
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};
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typedef struct _QTWrapperAudioDecoderParams QTWrapperAudioDecoderParams;
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struct _QTWrapperAudioDecoderParams
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{
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Component component;
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GstCaps *sinkcaps;
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};
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static gboolean qtwrapper_audio_decoder_sink_setcaps (GstPad * pad,
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GstCaps * caps);
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static GstFlowReturn qtwrapper_audio_decoder_chain (GstPad * pad,
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GstBuffer * buf);
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static gboolean qtwrapper_audio_decoder_sink_event (GstPad * pad,
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GstEvent * event);
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static void
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qtwrapper_audio_decoder_init (QTWrapperAudioDecoder * qtwrapper)
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{
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QTWrapperAudioDecoderClass *oclass;
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oclass = (QTWrapperAudioDecoderClass *) (G_OBJECT_GET_CLASS (qtwrapper));
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/* Sink pad */
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qtwrapper->sinkpad = gst_pad_new_from_template (oclass->sinktempl, "sink");
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gst_pad_set_setcaps_function (qtwrapper->sinkpad,
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GST_DEBUG_FUNCPTR (qtwrapper_audio_decoder_sink_setcaps));
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gst_pad_set_chain_function (qtwrapper->sinkpad,
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GST_DEBUG_FUNCPTR (qtwrapper_audio_decoder_chain));
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gst_pad_set_event_function (qtwrapper->sinkpad,
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GST_DEBUG_FUNCPTR (qtwrapper_audio_decoder_sink_event));
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gst_element_add_pad (GST_ELEMENT (qtwrapper), qtwrapper->sinkpad);
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/* Source pad */
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qtwrapper->srcpad = gst_pad_new_from_static_template (&src_templ, "src");
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gst_element_add_pad (GST_ELEMENT (qtwrapper), qtwrapper->srcpad);
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}
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static void
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clear_AudioStreamBasicDescription (AudioStreamBasicDescription * desc)
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{
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desc->mSampleRate = 0;
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desc->mFormatID = 0;
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desc->mFormatFlags = 0;
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desc->mBytesPerPacket = 0;
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desc->mFramesPerPacket = 0;
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desc->mBytesPerFrame = 0;
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desc->mChannelsPerFrame = 0;
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desc->mBitsPerChannel = 0;
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desc->mReserved = 0;
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}
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static void
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fill_indesc_mp3 (QTWrapperAudioDecoder * qtwrapper, guint32 fourcc, gint rate,
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gint channels)
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{
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GST_INFO_OBJECT (qtwrapper, "Filling input description for MP3 data");
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clear_AudioStreamBasicDescription (&qtwrapper->indesc);
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/* only the samplerate is needed apparently */
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qtwrapper->indesc.mSampleRate = (double) rate;
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qtwrapper->indesc.mFormatID = kAudioFormatMPEGLayer3;
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qtwrapper->indesc.mChannelsPerFrame = channels;
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}
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static void
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fill_indesc_aac (QTWrapperAudioDecoder * qtwrapper, guint32 fourcc, gint rate,
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gint channels)
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{
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clear_AudioStreamBasicDescription (&qtwrapper->indesc);
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qtwrapper->indesc.mSampleRate = (double) rate;
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qtwrapper->indesc.mFormatID = kAudioFormatMPEG4AAC;
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/* aac always has 1024 frames per packet */
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qtwrapper->indesc.mFramesPerPacket = 1024;
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qtwrapper->indesc.mChannelsPerFrame = channels;
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}
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static void
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fill_indesc_samr (QTWrapperAudioDecoder * qtwrapper, guint32 fourcc,
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gint channels)
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{
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clear_AudioStreamBasicDescription (&qtwrapper->indesc);
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qtwrapper->indesc.mSampleRate = 8000;
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qtwrapper->indesc.mFormatID = fourcc;
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qtwrapper->indesc.mChannelsPerFrame = 1;
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qtwrapper->indesc.mFramesPerPacket = 160;
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}
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static void
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fill_indesc_generic (QTWrapperAudioDecoder * qtwrapper, guint32 fourcc,
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gint rate, gint channels)
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{
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clear_AudioStreamBasicDescription (&qtwrapper->indesc);
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qtwrapper->indesc.mSampleRate = rate;
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qtwrapper->indesc.mFormatID = fourcc;
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qtwrapper->indesc.mChannelsPerFrame = channels;
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}
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static void
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fill_indesc_alac (QTWrapperAudioDecoder * qtwrapper, guint32 fourcc,
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gint rate, gint channels)
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{
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clear_AudioStreamBasicDescription (&qtwrapper->indesc);
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qtwrapper->indesc.mSampleRate = rate;
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qtwrapper->indesc.mFormatID = fourcc;
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qtwrapper->indesc.mChannelsPerFrame = channels;
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// This has to be set, but the particular value doesn't seem to matter much
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qtwrapper->indesc.mFramesPerPacket = 4096;
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}
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static gpointer
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make_alac_magic_cookie (GstBuffer * codec_data, gsize * len)
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{
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guint8 *res;
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if (GST_BUFFER_SIZE (codec_data) < 4)
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return NULL;
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*len = 20 + GST_BUFFER_SIZE (codec_data);
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res = g_malloc0 (*len);
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/* 12 first bytes are 'frma' (format) atom with 'alac' value */
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GST_WRITE_UINT32_BE (res, 0xc); /* Atom length: 12 bytes */
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GST_WRITE_UINT32_LE (res + 4, QT_MAKE_FOURCC_BE ('f', 'r', 'm', 'a'));
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GST_WRITE_UINT32_LE (res + 8, QT_MAKE_FOURCC_BE ('a', 'l', 'a', 'c'));
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/* Write the codec_data, but with the first four bytes reversed (different
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endianness). This is the 'alac' atom. */
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GST_WRITE_UINT32_BE (res + 12,
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GST_READ_UINT32_LE (GST_BUFFER_DATA (codec_data)));
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memcpy (res + 16, GST_BUFFER_DATA (codec_data) + 4,
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GST_BUFFER_SIZE (codec_data) - 4);
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/* Terminator atom */
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GST_WRITE_UINT32_BE (res + 12 + GST_BUFFER_SIZE (codec_data), 8);
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GST_WRITE_UINT32_BE (res + 12 + GST_BUFFER_SIZE (codec_data) + 4, 0);
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return res;
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}
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static gpointer
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make_samr_magic_cookie (GstBuffer * codec_data, gsize * len)
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{
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guint8 *res;
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*len = 48;
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res = g_malloc0 (0x30);
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/* 12 first bytes are 'frma' (format) atom with 'samr' value */
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GST_WRITE_UINT32_BE (res, 0xc);
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GST_WRITE_UINT32_LE (res + 4, QT_MAKE_FOURCC_BE ('f', 'r', 'm', 'a'));
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GST_WRITE_UINT32_LE (res + 8, QT_MAKE_FOURCC_BE ('s', 'a', 'm', 'r'));
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/* 10 bytes for 'enda' atom with 0 */
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GST_WRITE_UINT32_BE (res + 12, 10);
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GST_WRITE_UINT32_LE (res + 16, QT_MAKE_FOURCC_BE ('e', 'n', 'd', 'a'));
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/* 17(+1) bytes for the codec_data contents */
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GST_WRITE_UINT32_BE (res + 22, 18);
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memcpy (res + 26, GST_BUFFER_DATA (codec_data) + 4, 17);
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/* yes... we need to replace 'damr' by 'samr'. Blame Apple ! */
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GST_WRITE_UINT8 (res + 26, 's');
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/* Terminator atom */
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GST_WRITE_UINT32_BE (res + 40, 8);
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#if DEBUG_DUMP
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gst_util_dump_mem (res, 48);
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#endif
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return res;
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}
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static int
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write_len (guint8 * buf, int val)
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{
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/* This is some sort of variable-length coding, but the quicktime
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* file(s) I have here all just use a 4-byte version, so we'll do that.
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* Return the number of bytes written;
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*/
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buf[0] = ((val >> 21) & 0x7f) | 0x80;
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buf[1] = ((val >> 14) & 0x7f) | 0x80;
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buf[2] = ((val >> 7) & 0x7f) | 0x80;
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buf[3] = ((val >> 0) & 0x7f);
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return 4;
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}
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static void
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aac_parse_codec_data (GstBuffer * codec_data, gint * channels)
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{
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guint8 *data = GST_BUFFER_DATA (codec_data);
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guint codec_channels;
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if (GST_BUFFER_SIZE (codec_data) < 2) {
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GST_WARNING ("Cannot parse codec_data for channel count");
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return;
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}
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codec_channels = (data[1] & 0x7f) >> 3;
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if (*channels != codec_channels) {
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GST_INFO ("Overwriting channels %d with %d", *channels, codec_channels);
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*channels = (gint) codec_channels;
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} else {
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GST_INFO ("Retaining channel count %d", codec_channels);
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}
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}
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/* The AAC decoder requires the entire mpeg4 audio elementary stream
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* descriptor, which is the body (except the 4-byte version field) of
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* the quicktime 'esds' atom. However, qtdemux only passes through the
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* (two byte, normally) payload, so we need to reconstruct the ESD */
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/* TODO: Get the AAC spec, and verify this implementation */
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static gpointer
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make_aac_magic_cookie (GstBuffer * codec_data, gsize * len)
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{
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guint8 *cookie;
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int offset = 0;
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int decoder_specific_len = GST_BUFFER_SIZE (codec_data);
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int config_len = 13 + 5 + decoder_specific_len;
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int es_len = 3 + 5 + config_len + 5 + 1;
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int total_len = es_len + 5;
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cookie = g_malloc0 (total_len);
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*len = total_len;
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/* Structured something like this:
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* [ES Descriptor
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* [Config Descriptor
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* [Specific Descriptor]]
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* [Unknown]]
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*/
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QT_WRITE_UINT8 (cookie + offset, 0x03);
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offset += 1; /* ES Descriptor tag */
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offset += write_len (cookie + offset, es_len);
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QT_WRITE_UINT16 (cookie + offset, 0);
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offset += 2; /* Track ID */
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QT_WRITE_UINT8 (cookie + offset, 0);
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offset += 1; /* Flags */
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QT_WRITE_UINT8 (cookie + offset, 0x04);
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offset += 1; /* Config Descriptor tag */
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offset += write_len (cookie + offset, config_len);
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/* TODO: Fix these up */
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QT_WRITE_UINT8 (cookie + offset, 0x40);
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offset += 1; /* object_type_id */
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QT_WRITE_UINT8 (cookie + offset, 0x15);
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offset += 1; /* stream_type */
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QT_WRITE_UINT24 (cookie + offset, 0x1800);
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offset += 3; /* buffer_size_db */
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QT_WRITE_UINT32 (cookie + offset, 128000);
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offset += 4; /* max_bitrate */
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QT_WRITE_UINT32 (cookie + offset, 128000);
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offset += 4; /* avg_bitrate */
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QT_WRITE_UINT8 (cookie + offset, 0x05);
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offset += 1; /* Specific Descriptor tag */
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offset += write_len (cookie + offset, decoder_specific_len);
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memcpy (cookie + offset, GST_BUFFER_DATA (codec_data), decoder_specific_len);
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offset += decoder_specific_len;
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/* TODO: What is this? 'SL descriptor' apparently, but what does that mean? */
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QT_WRITE_UINT8 (cookie + offset, 0x06);
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offset += 1; /* SL Descriptor tag */
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offset += write_len (cookie + offset, 1);
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QT_WRITE_UINT8 (cookie + offset, 2);
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offset += 1;
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return cookie;
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}
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static void
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close_decoder (QTWrapperAudioDecoder * qtwrapper)
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{
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if (qtwrapper->adec) {
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CloseComponent (qtwrapper->adec);
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qtwrapper->adec = NULL;
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}
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if (qtwrapper->bufferlist) {
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DestroyAudioBufferList (qtwrapper->bufferlist);
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qtwrapper->bufferlist = NULL;
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}
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}
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static gboolean
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open_decoder (QTWrapperAudioDecoder * qtwrapper, GstCaps * caps,
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GstCaps ** othercaps)
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{
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gboolean ret = FALSE;
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QTWrapperAudioDecoderClass *oclass;
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/* TODO: these will be used as the output rate/channels for formats that
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* don't supply these in the caps. This isn't very nice!
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*/
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gint channels = 2;
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gint rate = 44100;
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OSStatus status;
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GstStructure *s;
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gchar *tmp;
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const GValue *value;
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GstBuffer *codec_data = NULL;
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gboolean have_esds = FALSE;
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/* Clean up any existing decoder */
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close_decoder (qtwrapper);
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tmp = gst_caps_to_string (caps);
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GST_LOG_OBJECT (qtwrapper, "caps: %s", tmp);
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g_free (tmp);
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/* extract rate/channels information from the caps */
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s = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (s, "rate", &rate);
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gst_structure_get_int (s, "channels", &channels);
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/* get codec_data */
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if ((value = gst_structure_get_value (s, "codec_data"))) {
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codec_data = GST_BUFFER_CAST (gst_value_get_mini_object (value));
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}
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oclass = (QTWrapperAudioDecoderClass *) (G_OBJECT_GET_CLASS (qtwrapper));
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if (codec_data
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&& oclass->componentSubType == QT_MAKE_FOURCC_LE ('m', 'p', '4', 'a')) {
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/* QuickTime/iTunes creates AAC files with the wrong channel count in the header,
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so parse that out of the codec data if we can.
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*/
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aac_parse_codec_data (codec_data, &channels);
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}
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/* If the quicktime demuxer gives us a full esds atom, use that instead of
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* the codec_data */
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if ((value = gst_structure_get_value (s, "quicktime_esds"))) {
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have_esds = TRUE;
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codec_data = GST_BUFFER_CAST (gst_value_get_mini_object (value));
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}
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#if DEBUG_DUMP
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if (codec_data)
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gst_util_dump_mem (GST_BUFFER_DATA (codec_data),
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GST_BUFFER_SIZE (codec_data));
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#endif
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GST_INFO_OBJECT (qtwrapper, "rate:%d, channels:%d", rate, channels);
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GST_INFO_OBJECT (qtwrapper, "componentSubType is %" GST_FOURCC_FORMAT,
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QT_FOURCC_ARGS (oclass->componentSubType));
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/* Setup the input format description, some format require special handling */
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switch (oclass->componentSubType) {
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case QT_MAKE_FOURCC_LE ('.', 'm', 'p', '3'):
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|
fill_indesc_mp3 (qtwrapper, oclass->componentSubType, rate, channels);
|
|
break;
|
|
case QT_MAKE_FOURCC_LE ('m', 'p', '4', 'a'):
|
|
fill_indesc_aac (qtwrapper, oclass->componentSubType, rate, channels);
|
|
break;
|
|
case QT_MAKE_FOURCC_LE ('s', 'a', 'm', 'r'):
|
|
fill_indesc_samr (qtwrapper, oclass->componentSubType, channels);
|
|
rate = 8000;
|
|
break;
|
|
case QT_MAKE_FOURCC_LE ('a', 'l', 'a', 'c'):
|
|
fill_indesc_alac (qtwrapper, oclass->componentSubType, rate, channels);
|
|
break;
|
|
default:
|
|
fill_indesc_generic (qtwrapper, oclass->componentSubType, rate, channels);
|
|
break;
|
|
}
|
|
|
|
#if DEBUG_DUMP
|
|
gst_util_dump_mem ((gpointer) & qtwrapper->indesc,
|
|
sizeof (AudioStreamBasicDescription));
|
|
#endif
|
|
|
|
qtwrapper->samplerate = rate;
|
|
qtwrapper->channels = channels;
|
|
|
|
/* Create an instance of SCAudio */
|
|
status = OpenADefaultComponent (StandardCompressionType,
|
|
StandardCompressionSubTypeAudio, &qtwrapper->adec);
|
|
if (status) {
|
|
GST_WARNING_OBJECT (qtwrapper,
|
|
"Error instantiating SCAudio component: %ld", status);
|
|
qtwrapper->adec = NULL;
|
|
goto beach;
|
|
}
|
|
|
|
/* This is necessary to make setting the InputBasicDescription succeed;
|
|
without it SCAudio only accepts PCM as input. Presumably a bug in
|
|
QuickTime. Thanks to Arek for figuring this one out!
|
|
*/
|
|
{
|
|
QTAtomContainer audiosettings = NULL;
|
|
|
|
SCGetSettingsAsAtomContainer (qtwrapper->adec, &audiosettings);
|
|
SCSetSettingsFromAtomContainer (qtwrapper->adec, audiosettings);
|
|
|
|
/* TODO: Figure out if disposing of the QTAtomContainer is needed here */
|
|
}
|
|
|
|
/* Set the input description info on the SCAudio instance */
|
|
status = QTSetComponentProperty (qtwrapper->adec, kQTPropertyClass_SCAudio,
|
|
kQTSCAudioPropertyID_InputBasicDescription,
|
|
sizeof (qtwrapper->indesc), &qtwrapper->indesc);
|
|
if (status) {
|
|
GST_WARNING_OBJECT (qtwrapper,
|
|
"Error setting input description on SCAudio: %ld", status);
|
|
|
|
GST_ELEMENT_ERROR (qtwrapper, STREAM, NOT_IMPLEMENTED,
|
|
("A QuickTime error occurred trying to decode this stream"),
|
|
("QuickTime returned error status %lx", status));
|
|
goto beach;
|
|
}
|
|
|
|
/* TODO: we can select a channel layout here, figure out if we want to */
|
|
|
|
/* if we have codec_data, give it to the converter ! */
|
|
if (codec_data) {
|
|
gsize len = 0;
|
|
gpointer magiccookie;
|
|
|
|
switch (oclass->componentSubType) {
|
|
/* Some decoders want the 'magic cookie' in a different format from how
|
|
* gstreamer represents it. So, convert...
|
|
*/
|
|
case QT_MAKE_FOURCC_LE ('s', 'a', 'm', 'r'):
|
|
magiccookie = make_samr_magic_cookie (codec_data, &len);
|
|
break;
|
|
case QT_MAKE_FOURCC_LE ('a', 'l', 'a', 'c'):
|
|
magiccookie = make_alac_magic_cookie (codec_data, &len);
|
|
break;
|
|
case QT_MAKE_FOURCC_LE ('m', 'p', '4', 'a'):
|
|
if (!have_esds) {
|
|
magiccookie = make_aac_magic_cookie (codec_data, &len);
|
|
break;
|
|
}
|
|
/* Else: fallthrough */
|
|
default:
|
|
len = GST_BUFFER_SIZE (codec_data);
|
|
magiccookie = GST_BUFFER_DATA (codec_data);
|
|
break;
|
|
}
|
|
|
|
if (magiccookie) {
|
|
GST_LOG_OBJECT (qtwrapper, "Setting magic cookie %p of size %"
|
|
G_GSIZE_FORMAT, magiccookie, len);
|
|
|
|
#if DEBUG_DUMP
|
|
gst_util_dump_mem (magiccookie, len);
|
|
#endif
|
|
|
|
status =
|
|
QTSetComponentProperty (qtwrapper->adec, kQTPropertyClass_SCAudio,
|
|
kQTSCAudioPropertyID_InputMagicCookie, len, magiccookie);
|
|
if (status) {
|
|
GST_WARNING_OBJECT (qtwrapper, "Error setting extra codec data: %ld",
|
|
status);
|
|
goto beach;
|
|
}
|
|
|
|
g_free (magiccookie);
|
|
}
|
|
}
|
|
|
|
/* Set output to be interleaved raw PCM */
|
|
{
|
|
OSType outputFormat = kAudioFormatLinearPCM;
|
|
SCAudioFormatFlagsRestrictions restrictions = { 0 };
|
|
|
|
/* Set the mask in order to set this flag to zero */
|
|
restrictions.formatFlagsMask =
|
|
kAudioFormatFlagIsFloat | kAudioFormatFlagIsBigEndian;
|
|
restrictions.formatFlagsValues = kAudioFormatFlagIsFloat;
|
|
|
|
status = QTSetComponentProperty (qtwrapper->adec, kQTPropertyClass_SCAudio,
|
|
kQTSCAudioPropertyID_ClientRestrictedLPCMFlags,
|
|
sizeof (restrictions), &restrictions);
|
|
if (status) {
|
|
GST_WARNING_OBJECT (qtwrapper, "Error setting PCM to interleaved: %ld",
|
|
status);
|
|
goto beach;
|
|
}
|
|
|
|
status = QTSetComponentProperty (qtwrapper->adec, kQTPropertyClass_SCAudio,
|
|
kQTSCAudioPropertyID_ClientRestrictedCompressionFormatList,
|
|
sizeof (outputFormat), &outputFormat);
|
|
if (status) {
|
|
GST_WARNING_OBJECT (qtwrapper, "Error setting output to PCM: %ld",
|
|
status);
|
|
goto beach;
|
|
}
|
|
}
|
|
|
|
qtwrapper->outdesc.mSampleRate = 0; /* Use recommended; we read this out later */
|
|
qtwrapper->outdesc.mFormatID = kAudioFormatLinearPCM;
|
|
qtwrapper->outdesc.mFormatFlags = kAudioFormatFlagIsFloat;
|
|
qtwrapper->outdesc.mBytesPerPacket = 0;
|
|
qtwrapper->outdesc.mFramesPerPacket = 0;
|
|
qtwrapper->outdesc.mBytesPerFrame = 4 * channels;
|
|
qtwrapper->outdesc.mChannelsPerFrame = channels;
|
|
qtwrapper->outdesc.mBitsPerChannel = 32;
|
|
qtwrapper->outdesc.mReserved = 0;
|
|
|
|
status = QTSetComponentProperty (qtwrapper->adec, kQTPropertyClass_SCAudio,
|
|
kQTSCAudioPropertyID_BasicDescription,
|
|
sizeof (qtwrapper->outdesc), &qtwrapper->outdesc);
|
|
if (status) {
|
|
GST_WARNING_OBJECT (qtwrapper, "Error setting output description: %ld",
|
|
status);
|
|
goto beach;
|
|
}
|
|
|
|
status = QTGetComponentProperty (qtwrapper->adec, kQTPropertyClass_SCAudio,
|
|
kQTSCAudioPropertyID_BasicDescription,
|
|
sizeof (qtwrapper->outdesc), &qtwrapper->outdesc, NULL);
|
|
|
|
if (status) {
|
|
GST_WARNING_OBJECT (qtwrapper,
|
|
"Failed to get output audio description: %ld", status);
|
|
ret = FALSE;
|
|
goto beach;
|
|
}
|
|
|
|
if (qtwrapper->outdesc.mFormatID != kAudioFormatLinearPCM /*||
|
|
(qtwrapper->outdesc.mFormatFlags & kAudioFormatFlagIsFloat) !=
|
|
kAudioFormatFlagIsFloat */ ) {
|
|
GST_WARNING_OBJECT (qtwrapper, "Output is not floating point PCM");
|
|
ret = FALSE;
|
|
goto beach;
|
|
}
|
|
|
|
qtwrapper->samplerate = (int) qtwrapper->outdesc.mSampleRate;
|
|
qtwrapper->channels = qtwrapper->outdesc.mChannelsPerFrame;
|
|
GST_DEBUG_OBJECT (qtwrapper, "Output is %d Hz, %d channels",
|
|
qtwrapper->samplerate, qtwrapper->channels);
|
|
|
|
/* Create output bufferlist, big enough for 200ms of audio */
|
|
GST_DEBUG_OBJECT (qtwrapper, "Allocating bufferlist for %d channels",
|
|
channels);
|
|
qtwrapper->bufferlist =
|
|
AllocateAudioBufferList (channels,
|
|
qtwrapper->samplerate / 5 * qtwrapper->channels * 4);
|
|
|
|
/* Create output caps matching the format the component is giving us */
|
|
*othercaps = gst_caps_new_simple ("audio/x-raw-float",
|
|
"endianness", G_TYPE_INT, G_BYTE_ORDER,
|
|
"signed", G_TYPE_BOOLEAN, TRUE,
|
|
"width", G_TYPE_INT, 32,
|
|
"depth", G_TYPE_INT, 32,
|
|
"rate", G_TYPE_INT, qtwrapper->samplerate, "channels", G_TYPE_INT,
|
|
qtwrapper->channels, NULL);
|
|
|
|
ret = TRUE;
|
|
|
|
beach:
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
qtwrapper_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
|
|
{
|
|
QTWrapperAudioDecoder *qtwrapper;
|
|
gboolean ret = FALSE;
|
|
GstCaps *othercaps = NULL;
|
|
|
|
qtwrapper = (QTWrapperAudioDecoder *) gst_pad_get_parent (pad);
|
|
|
|
GST_LOG_OBJECT (qtwrapper, "caps:%" GST_PTR_FORMAT, caps);
|
|
|
|
/* 1. open decoder */
|
|
if (!(open_decoder (qtwrapper, caps, &othercaps)))
|
|
goto beach;
|
|
|
|
/* 2. set caps downstream */
|
|
ret = gst_pad_set_caps (qtwrapper->srcpad, othercaps);
|
|
|
|
beach:
|
|
if (othercaps)
|
|
gst_caps_unref (othercaps);
|
|
gst_object_unref (qtwrapper);
|
|
return ret;
|
|
}
|
|
|
|
static OSStatus
|
|
process_buffer_cb (ComponentInstance inAudioConverter,
|
|
UInt32 * ioNumberDataPackets,
|
|
AudioBufferList * ioData,
|
|
AudioStreamPacketDescription ** outDataPacketDescription,
|
|
QTWrapperAudioDecoder * qtwrapper)
|
|
{
|
|
GST_LOG_OBJECT (qtwrapper,
|
|
"ioNumberDataPackets:%lu, iodata:%p, outDataPacketDescription:%p",
|
|
*ioNumberDataPackets, ioData, outDataPacketDescription);
|
|
if (outDataPacketDescription)
|
|
GST_LOG ("*outDataPacketDescription:%p", *outDataPacketDescription);
|
|
|
|
GST_LOG ("mNumberBuffers : %u", (guint32) ioData->mNumberBuffers);
|
|
GST_LOG ("mData:%p , mDataByteSize:%u",
|
|
ioData->mBuffers[0].mData, (guint32) ioData->mBuffers[0].mDataByteSize);
|
|
|
|
ioData->mBuffers[0].mData = NULL;
|
|
ioData->mBuffers[0].mDataByteSize = 0;
|
|
|
|
*ioNumberDataPackets = 1;
|
|
|
|
if (qtwrapper->input_buffer && GST_BUFFER_SIZE (qtwrapper->input_buffer)) {
|
|
ioData->mBuffers[0].mData = GST_BUFFER_DATA (qtwrapper->input_buffer);
|
|
ioData->mBuffers[0].mDataByteSize =
|
|
GST_BUFFER_SIZE (qtwrapper->input_buffer);
|
|
|
|
/* if we have a valid outDataPacketDescription, we need to fill it */
|
|
if (outDataPacketDescription) {
|
|
qtwrapper->aspd[0].mStartOffset = 0;
|
|
qtwrapper->aspd[0].mVariableFramesInPacket = 0;
|
|
qtwrapper->aspd[0].mDataByteSize =
|
|
GST_BUFFER_SIZE (qtwrapper->input_buffer);
|
|
*outDataPacketDescription = qtwrapper->aspd;
|
|
}
|
|
|
|
GST_LOG_OBJECT (qtwrapper, "returning %d bytes at %p",
|
|
GST_BUFFER_SIZE (qtwrapper->input_buffer), ioData->mBuffers[0].mData);
|
|
|
|
qtwrapper->input_buffer = 0;
|
|
return noErr;
|
|
}
|
|
|
|
GST_LOG_OBJECT (qtwrapper,
|
|
"No remaining input data, returning NO_MORE_INPUT_DATA");
|
|
|
|
return NO_MORE_INPUT_DATA;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
qtwrapper_audio_decoder_chain (GstPad * pad, GstBuffer * buf)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
QTWrapperAudioDecoder *qtwrapper;
|
|
GstBuffer *outbuf;
|
|
OSStatus status;
|
|
guint32 outsamples;
|
|
guint32 savedbytes;
|
|
guint32 realbytes;
|
|
|
|
qtwrapper = (QTWrapperAudioDecoder *) gst_pad_get_parent (pad);
|
|
|
|
if (!qtwrapper->adec) {
|
|
GST_WARNING_OBJECT (qtwrapper, "QTWrapper not initialised");
|
|
goto beach;
|
|
}
|
|
|
|
GST_LOG_OBJECT (qtwrapper,
|
|
"buffer:%p , timestamp:%" GST_TIME_FORMAT " ,size:%d", buf,
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), GST_BUFFER_SIZE (buf));
|
|
|
|
#if DEBUG_DUMP
|
|
gst_util_dump_mem (GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
|
|
#endif
|
|
|
|
if (qtwrapper->gotnewsegment) {
|
|
|
|
GST_DEBUG_OBJECT (qtwrapper, "SCAudioReset()");
|
|
|
|
SCAudioReset (qtwrapper->adec);
|
|
|
|
/* some formats can give us a better initial time using the buffer
|
|
* timestamp. */
|
|
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
|
|
qtwrapper->initial_time = GST_BUFFER_TIMESTAMP (buf);
|
|
|
|
qtwrapper->gotnewsegment = FALSE;
|
|
}
|
|
|
|
outsamples = qtwrapper->bufferlist->mBuffers[0].mDataByteSize / 8;
|
|
savedbytes = qtwrapper->bufferlist->mBuffers[0].mDataByteSize;
|
|
|
|
qtwrapper->input_buffer = buf;
|
|
|
|
do {
|
|
GST_LOG_OBJECT (qtwrapper,
|
|
"Calling SCAudioFillBuffer(outsamples:%d , outdata:%p)", outsamples,
|
|
qtwrapper->bufferlist->mBuffers[0].mData);
|
|
|
|
/* Ask SCAudio to give us data ! */
|
|
status = SCAudioFillBuffer (qtwrapper->adec,
|
|
(SCAudioInputDataProc) process_buffer_cb,
|
|
qtwrapper, (UInt32 *) & outsamples, qtwrapper->bufferlist, NULL);
|
|
|
|
if ((status != noErr) && (status != NO_MORE_INPUT_DATA)) {
|
|
if (status < 0)
|
|
GST_WARNING_OBJECT (qtwrapper,
|
|
"Error in SCAudioFillBuffer() : %d", (gint32) status);
|
|
else
|
|
GST_WARNING_OBJECT (qtwrapper,
|
|
"Error in SCAudioFillBuffer() : %" GST_FOURCC_FORMAT,
|
|
QT_FOURCC_ARGS (status));
|
|
ret = GST_FLOW_ERROR;
|
|
goto beach;
|
|
}
|
|
|
|
realbytes = qtwrapper->bufferlist->mBuffers[0].mDataByteSize;
|
|
|
|
GST_LOG_OBJECT (qtwrapper, "We now have %d samples [%d bytes]",
|
|
outsamples, realbytes);
|
|
|
|
qtwrapper->bufferlist->mBuffers[0].mDataByteSize = savedbytes;
|
|
|
|
if (!outsamples)
|
|
goto beach;
|
|
|
|
/* 4. Create buffer and copy data in it */
|
|
ret = gst_pad_alloc_buffer (qtwrapper->srcpad, qtwrapper->cur_offset,
|
|
realbytes, GST_PAD_CAPS (qtwrapper->srcpad), &outbuf);
|
|
if (ret != GST_FLOW_OK)
|
|
goto beach;
|
|
|
|
/* copy data from bufferlist to output buffer */
|
|
memmove (GST_BUFFER_DATA (outbuf),
|
|
qtwrapper->bufferlist->mBuffers[0].mData, realbytes);
|
|
|
|
/* 5. calculate timestamp and duration */
|
|
GST_BUFFER_TIMESTAMP (outbuf) =
|
|
qtwrapper->initial_time + gst_util_uint64_scale_int (GST_SECOND,
|
|
(gint) qtwrapper->cur_offset, qtwrapper->samplerate);
|
|
GST_BUFFER_SIZE (outbuf) = realbytes;
|
|
GST_BUFFER_DURATION (outbuf) =
|
|
gst_util_uint64_scale_int (GST_SECOND,
|
|
realbytes / (qtwrapper->channels * 4), qtwrapper->samplerate);
|
|
|
|
GST_LOG_OBJECT (qtwrapper,
|
|
"timestamp:%" GST_TIME_FORMAT ", duration:%" GST_TIME_FORMAT
|
|
"offset:%lld, offset_end:%lld",
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
|
|
GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
|
|
|
|
qtwrapper->cur_offset += outsamples;
|
|
|
|
/* 6. push buffer downstream */
|
|
|
|
ret = gst_pad_push (qtwrapper->srcpad, outbuf);
|
|
if (ret != GST_FLOW_OK)
|
|
goto beach;
|
|
|
|
GST_DEBUG_OBJECT (qtwrapper,
|
|
"Read %d bytes, could have read up to %d bytes", realbytes, savedbytes);
|
|
} while (status != NO_MORE_INPUT_DATA);
|
|
|
|
beach:
|
|
gst_buffer_unref (buf);
|
|
gst_object_unref (qtwrapper);
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
qtwrapper_audio_decoder_sink_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
QTWrapperAudioDecoder *qtwrapper;
|
|
gboolean ret = FALSE;
|
|
|
|
qtwrapper = (QTWrapperAudioDecoder *) gst_pad_get_parent (pad);
|
|
|
|
GST_LOG_OBJECT (qtwrapper, "event:%s", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
/* TODO: Flush events should reset the decoder component */
|
|
case GST_EVENT_NEWSEGMENT:{
|
|
gint64 start, stop, position;
|
|
gboolean update;
|
|
gdouble rate;
|
|
GstFormat format;
|
|
|
|
GST_LOG ("We've got a newsegment");
|
|
gst_event_parse_new_segment (event, &update, &rate, &format, &start,
|
|
&stop, &position);
|
|
|
|
/* if the format isn't time, we need to create a new time newsegment */
|
|
/* FIXME : This is really bad, we should convert the values properly to time */
|
|
if (format != GST_FORMAT_TIME) {
|
|
GstEvent *newevent;
|
|
|
|
GST_WARNING_OBJECT (qtwrapper,
|
|
"Original event wasn't in GST_FORMAT_TIME, creating new fake one.");
|
|
|
|
start = 0;
|
|
|
|
newevent =
|
|
gst_event_new_new_segment (update, rate, GST_FORMAT_TIME, start,
|
|
GST_CLOCK_TIME_NONE, start);
|
|
gst_event_unref (event);
|
|
event = newevent;
|
|
}
|
|
|
|
qtwrapper->initial_time = start;
|
|
qtwrapper->cur_offset = 0;
|
|
|
|
GST_LOG ("initial_time is now %" GST_TIME_FORMAT, GST_TIME_ARGS (start));
|
|
|
|
if (qtwrapper->adec)
|
|
qtwrapper->gotnewsegment = TRUE;
|
|
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = gst_pad_push_event (qtwrapper->srcpad, event);
|
|
|
|
gst_object_unref (qtwrapper);
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
qtwrapper_audio_decoder_base_init (QTWrapperAudioDecoderClass * klass)
|
|
{
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
|
|
gchar *name = NULL;
|
|
gchar *info = NULL;
|
|
char *longname, *description;
|
|
ComponentDescription desc;
|
|
QTWrapperAudioDecoderParams *params;
|
|
|
|
params = (QTWrapperAudioDecoderParams *)
|
|
g_type_get_qdata (G_OBJECT_CLASS_TYPE (klass),
|
|
QTWRAPPER_ADEC_PARAMS_QDATA);
|
|
g_assert (params);
|
|
|
|
get_name_info_from_component (params->component, &desc, &name, &info);
|
|
|
|
/* Fill in details */
|
|
longname =
|
|
g_strdup_printf ("QTWrapper SCAudio Audio Decoder : %s",
|
|
GST_STR_NULL (name));
|
|
description =
|
|
g_strdup_printf ("QTWrapper SCAudio wrapper for decoder: %s",
|
|
GST_STR_NULL (info));
|
|
gst_element_class_set_metadata (element_class,
|
|
longname, "Codec/Decoder/Audio", description,
|
|
"Fluendo <gstreamer@fluendo.com>, "
|
|
"Pioneers of the Inevitable <songbird@songbirdnest.com>");
|
|
|
|
g_free (longname);
|
|
g_free (description);
|
|
g_free (name);
|
|
g_free (info);
|
|
|
|
/* Add pad templates */
|
|
klass->sinktempl = gst_pad_template_new ("sink", GST_PAD_SINK,
|
|
GST_PAD_ALWAYS, params->sinkcaps);
|
|
gst_element_class_add_pad_template (element_class, klass->sinktempl);
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&src_templ));
|
|
|
|
/* Store class-global values */
|
|
klass->componentSubType = desc.componentSubType;
|
|
}
|
|
|
|
static void
|
|
qtwrapper_audio_decoder_dispose (GObject * object)
|
|
{
|
|
QTWrapperAudioDecoder *qtwrapper = (QTWrapperAudioDecoder *) object;
|
|
QTWrapperAudioDecoderClass *oclass =
|
|
(QTWrapperAudioDecoderClass *) (G_OBJECT_GET_CLASS (qtwrapper));
|
|
GObjectClass *parent_class = g_type_class_peek_parent (oclass);
|
|
|
|
close_decoder (qtwrapper);
|
|
|
|
G_OBJECT_CLASS (parent_class)->dispose (object);
|
|
}
|
|
|
|
static void
|
|
qtwrapper_audio_decoder_class_init (QTWrapperAudioDecoderClass * klass)
|
|
{
|
|
GObjectClass *object_class;
|
|
|
|
object_class = (GObjectClass *) klass;
|
|
|
|
object_class->dispose = qtwrapper_audio_decoder_dispose;
|
|
}
|
|
|
|
gboolean
|
|
qtwrapper_audio_decoders_register (GstPlugin * plugin)
|
|
{
|
|
gboolean res = TRUE;
|
|
Component componentID = NULL;
|
|
|
|
ComponentDescription desc = {
|
|
kSoundDecompressor, 0, 0, 0, 0
|
|
};
|
|
|
|
GTypeInfo typeinfo = {
|
|
sizeof (QTWrapperAudioDecoderClass),
|
|
(GBaseInitFunc) qtwrapper_audio_decoder_base_init,
|
|
NULL,
|
|
(GClassInitFunc) qtwrapper_audio_decoder_class_init,
|
|
NULL,
|
|
NULL,
|
|
sizeof (QTWrapperAudioDecoder),
|
|
0,
|
|
(GInstanceInitFunc) qtwrapper_audio_decoder_init,
|
|
};
|
|
|
|
/* Find all SoundDecompressors ! */
|
|
GST_DEBUG ("There are %ld decompressors available", CountComponents (&desc));
|
|
|
|
/* loop over SoundDecompressors */
|
|
do {
|
|
componentID = FindNextComponent (componentID, &desc);
|
|
|
|
GST_LOG ("componentID : %p", componentID);
|
|
|
|
if (componentID) {
|
|
ComponentDescription thisdesc;
|
|
gchar *name = NULL, *info = NULL;
|
|
GstCaps *caps = NULL;
|
|
gchar *type_name = NULL;
|
|
GType type;
|
|
QTWrapperAudioDecoderParams *params = NULL;
|
|
|
|
if (!(get_name_info_from_component (componentID, &thisdesc, &name,
|
|
&info)))
|
|
goto next;
|
|
|
|
GST_LOG (" name:%s", GST_STR_NULL (name));
|
|
GST_LOG (" info:%s", GST_STR_NULL (info));
|
|
|
|
GST_LOG (" type:%" GST_FOURCC_FORMAT,
|
|
QT_FOURCC_ARGS (thisdesc.componentType));
|
|
GST_LOG (" subtype:%" GST_FOURCC_FORMAT,
|
|
QT_FOURCC_ARGS (thisdesc.componentSubType));
|
|
GST_LOG (" manufacturer:%" GST_FOURCC_FORMAT,
|
|
QT_FOURCC_ARGS (thisdesc.componentManufacturer));
|
|
|
|
if (!(caps =
|
|
fourcc_to_caps (QT_READ_UINT32 (&thisdesc.componentSubType))))
|
|
goto next;
|
|
|
|
type_name = g_strdup_printf ("qtwrapperaudiodec_%" GST_FOURCC_FORMAT,
|
|
QT_FOURCC_ARGS (thisdesc.componentSubType));
|
|
g_strdelimit (type_name, " .", '_');
|
|
|
|
if (g_type_from_name (type_name)) {
|
|
GST_WARNING ("We already have a registered plugin for %s", type_name);
|
|
goto next;
|
|
}
|
|
|
|
params = g_new0 (QTWrapperAudioDecoderParams, 1);
|
|
params->component = componentID;
|
|
params->sinkcaps = gst_caps_ref (caps);
|
|
|
|
type = g_type_register_static (GST_TYPE_ELEMENT, type_name, &typeinfo, 0);
|
|
/* Store params in type qdata */
|
|
g_type_set_qdata (type, QTWRAPPER_ADEC_PARAMS_QDATA, (gpointer) params);
|
|
|
|
/* register type */
|
|
if (!gst_element_register (plugin, type_name, GST_RANK_MARGINAL, type)) {
|
|
g_warning ("Failed to register %s", type_name);;
|
|
g_type_set_qdata (type, QTWRAPPER_ADEC_PARAMS_QDATA, NULL);
|
|
g_free (params);
|
|
res = FALSE;
|
|
goto next;
|
|
}
|
|
|
|
next:
|
|
if (name)
|
|
g_free (name);
|
|
if (info)
|
|
g_free (info);
|
|
if (type_name)
|
|
g_free (type_name);
|
|
if (caps)
|
|
gst_caps_unref (caps);
|
|
}
|
|
|
|
} while (componentID && res);
|
|
|
|
return res;
|
|
}
|