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applemedia: add AudioToolbox based audio decoder

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This commit is contained in:
Alessandro Decina 2013-07-23 10:30:39 +02:00
parent 60c2c123be
commit 92bcdd9c4b
4 changed files with 465 additions and 2 deletions
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8
sys/applemedia/Makefile.am
View file

@ -7,6 +7,7 @@ libgstapplemedia_la_SOURCES = \
coremediabuffer.c \
coremediactx.c \
vtapi.c \
atdec.c \
dynapi.c
libgstapplemedia_la_CPPFLAGS = \
@ -38,7 +39,8 @@ endif
libgstapplemedia_la_LIBADD = \
$(GST_BASE_LIBS) \
$(GST_PLUGINS_BASE_LIBS) \
-lgstvideo-$(GST_API_VERSION)
-lgstvideo-$(GST_API_VERSION) \
-lgstaudio-$(GST_API_VERSION)
libgstapplemedia_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
@ -48,7 +50,8 @@ libgstapplemedia_la_LDFLAGS = \
$(GST_PLUGINS_BASE_LDFLAGS) \
-Wl,-framework -Wl,CoreFoundation \
-Wl,-framework -Wl,CoreMedia \
-Wl,-framework -Wl,CoreVideo
-Wl,-framework -Wl,CoreVideo \
-Wl,-framework -Wl,AudioToolbox
noinst_HEADERS = \
qtkitvideosrc.h \
@ -62,6 +65,7 @@ noinst_HEADERS = \
coremediabuffer.h \
coremediactx.h \
vtapi.h \
atdec.h \
dynapi.h \
dynapi-internal.h

405
sys/applemedia/atdec.c Normal file
View file

@ -0,0 +1,405 @@
/* GStreamer
* Copyright (C) 2013 Alessandro Decina <alessandro.d@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin Street, Suite 500,
* Boston, MA 02110-1335, USA.
*/
/**
* SECTION:element-gstatdec
*
* AudioToolbox based decoder.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch -v filesrc location=file.mov ! qtdemux ! queue ! aacparse ! atdec ! autoaudiosink
* ]|
* Decode aac audio from a mov file
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/gstaudiodecoder.h>
#include "atdec.h"
GST_DEBUG_CATEGORY_STATIC (gst_atdec_debug_category);
#define GST_CAT_DEFAULT gst_atdec_debug_category
static void gst_atdec_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
static void gst_atdec_get_property (GObject * object,
guint property_id, GValue * value, GParamSpec * pspec);
static void gst_atdec_finalize (GObject * object);
static gboolean gst_atdec_start (GstAudioDecoder * decoder);
static gboolean gst_atdec_stop (GstAudioDecoder * decoder);
static gboolean gst_atdec_set_format (GstAudioDecoder * decoder,
GstCaps * caps);
static GstFlowReturn gst_atdec_handle_frame (GstAudioDecoder * decoder,
GstBuffer * buffer);
static void gst_atdec_flush (GstAudioDecoder * decoder, gboolean hard);
static void gst_atdec_buffer_emptied (void *user_data,
AudioQueueRef queue, AudioQueueBufferRef buffer);
enum
{
PROP_0
};
static GstStaticPadTemplate gst_atdec_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE ("S16LE") ";"
GST_AUDIO_CAPS_MAKE ("F32LE")
)
);
static GstStaticPadTemplate gst_atdec_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, mpegversion=4, framed=true,"
"channels=[1,max]")
);
G_DEFINE_TYPE_WITH_CODE (GstATDec, gst_atdec, GST_TYPE_AUDIO_DECODER,
GST_DEBUG_CATEGORY_INIT (gst_atdec_debug_category, "atdec", 0,
"debug category for atdec element"));
static GstStaticCaps aac_caps = GST_STATIC_CAPS ("audio/mpeg, mpegversion=4");
static GstStaticCaps mp3_caps =
GST_STATIC_CAPS ("audio/mpeg, mpegversion=1, layer=3");
static GstStaticCaps raw_caps = GST_STATIC_CAPS ("audio/x-raw");
static void
gst_atdec_class_init (GstATDecClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstAudioDecoderClass *audio_decoder_class = GST_AUDIO_DECODER_CLASS (klass);
gst_element_class_add_pad_template (GST_ELEMENT_CLASS (klass),
gst_static_pad_template_get (&gst_atdec_src_template));
gst_element_class_add_pad_template (GST_ELEMENT_CLASS (klass),
gst_static_pad_template_get (&gst_atdec_sink_template));
gst_element_class_set_static_metadata (GST_ELEMENT_CLASS (klass),
"AudioToolbox based audio decoder",
"Codec/Decoder/Audio",
"AudioToolbox based audio decoder",
"Alessandro Decina <alessandro.d@gmail.com>");
gobject_class->set_property = gst_atdec_set_property;
gobject_class->get_property = gst_atdec_get_property;
gobject_class->finalize = gst_atdec_finalize;
audio_decoder_class->start = GST_DEBUG_FUNCPTR (gst_atdec_start);
audio_decoder_class->stop = GST_DEBUG_FUNCPTR (gst_atdec_stop);
audio_decoder_class->set_format = GST_DEBUG_FUNCPTR (gst_atdec_set_format);
audio_decoder_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_atdec_handle_frame);
audio_decoder_class->flush = GST_DEBUG_FUNCPTR (gst_atdec_flush);
}
static void
gst_atdec_init (GstATDec * atdec)
{
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (atdec), TRUE);
atdec->queue = NULL;
}
void
gst_atdec_set_property (GObject * object, guint property_id,
const GValue * value, GParamSpec * pspec)
{
GstATDec *atdec = GST_ATDEC (object);
GST_DEBUG_OBJECT (atdec, "set_property");
switch (property_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_atdec_get_property (GObject * object, guint property_id,
GValue * value, GParamSpec * pspec)
{
GstATDec *atdec = GST_ATDEC (object);
GST_DEBUG_OBJECT (atdec, "get_property");
switch (property_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
static void
gst_atdec_destroy_queue (GstATDec * atdec, gboolean drain)
{
AudioQueueStop (atdec->queue, drain);
AudioQueueDispose (atdec->queue, true);
atdec->queue = NULL;
}
void
gst_atdec_finalize (GObject * object)
{
GstATDec *atdec = GST_ATDEC (object);
GST_DEBUG_OBJECT (atdec, "finalize");
if (atdec->queue)
gst_atdec_destroy_queue (atdec, FALSE);
G_OBJECT_CLASS (gst_atdec_parent_class)->finalize (object);
}
static gboolean
gst_atdec_start (GstAudioDecoder * decoder)
{
GstATDec *atdec = GST_ATDEC (decoder);
GST_DEBUG_OBJECT (atdec, "start");
return TRUE;
}
static gboolean
gst_atdec_stop (GstAudioDecoder * decoder)
{
GstATDec *atdec = GST_ATDEC (decoder);
gst_atdec_destroy_queue (atdec, FALSE);
return TRUE;
}
static gboolean
can_intersect_static_caps (GstCaps * caps, GstStaticCaps * caps1)
{
GstCaps *tmp;
gboolean ret;
tmp = gst_static_caps_get (caps1);
ret = gst_caps_can_intersect (caps, tmp);
gst_caps_unref (tmp);
return ret;
}
static gboolean
gst_caps_to_at_format (GstCaps * caps, AudioStreamBasicDescription * format)
{
int channels = 0;
int rate = 0;
GstStructure *structure;
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "rate", &rate);
gst_structure_get_int (structure, "channels", &channels);
format->mSampleRate = rate;
format->mChannelsPerFrame = channels;
if (can_intersect_static_caps (caps, &aac_caps))
format->mFormatID = kAudioFormatMPEG4AAC;
else if (can_intersect_static_caps (caps, &mp3_caps))
format->mFormatID = kAudioFormatMPEGLayer3;
else if (can_intersect_static_caps (caps, &raw_caps)) {
GstAudioFormat audio_format;
const char *audio_format_str;
format->mFormatID = kAudioFormatLinearPCM;
format->mFramesPerPacket = 1;
audio_format_str = gst_structure_get_string (structure, "format");
if (!audio_format_str)
audio_format_str = "S16LE";
audio_format = gst_audio_format_from_string (audio_format_str);
switch (audio_format) {
case GST_AUDIO_FORMAT_S16LE:
format->mFormatFlags =
kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsSignedInteger;
format->mBitsPerChannel = 16;
format->mBytesPerPacket = format->mBytesPerFrame = 2 * channels;
break;
case GST_AUDIO_FORMAT_F32LE:
format->mFormatFlags =
kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsFloat;
format->mBitsPerChannel = 32;
format->mBytesPerPacket = format->mBytesPerFrame = 4 * channels;
break;
default:
g_warn_if_reached ();
break;
}
}
return TRUE;
}
static gboolean
gst_atdec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
OSStatus status;
AudioStreamBasicDescription input_format = { 0 };
AudioStreamBasicDescription output_format = { 0 };
GstAudioInfo output_info = { 0 };
AudioChannelLayout output_layout = { 0 };
GstCaps *output_caps;
GstATDec *atdec = GST_ATDEC (decoder);
GST_DEBUG_OBJECT (atdec, "set_format");
if (atdec->queue)
gst_atdec_destroy_queue (atdec, TRUE);
// configure input_format from caps
gst_caps_to_at_format (caps, &input_format);
// negotiate output caps
output_caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (atdec));
output_caps = gst_caps_fixate (output_caps);
if (!output_caps)
goto negotiation_error;
gst_caps_set_simple (output_caps,
"rate", G_TYPE_INT, (int) input_format.mSampleRate, NULL);
// configure output_format from caps
gst_caps_to_at_format (output_caps, &output_format);
// set the format we want to negotiate downstream
gst_audio_info_from_caps (&output_info, output_caps);
gst_audio_info_set_format (&output_info,
output_format.mFormatFlags & kLinearPCMFormatFlagIsSignedInteger ?
GST_AUDIO_FORMAT_S16LE : GST_AUDIO_FORMAT_F32LE,
output_format.mSampleRate, output_format.mChannelsPerFrame, NULL);
gst_audio_decoder_set_output_format (decoder, &output_info);
gst_caps_unref (output_caps);
status = AudioQueueNewOutput (&input_format, gst_atdec_buffer_emptied,
atdec, NULL, NULL, 0, &atdec->queue);
if (status)
goto create_queue_error;
// FIXME: figure out how to map this properly
if (output_format.mChannelsPerFrame == 1)
output_layout.mChannelLayoutTag = kAudioChannelLayoutTag_Mono;
else
output_layout.mChannelLayoutTag = kAudioChannelLayoutTag_Stereo;
status = AudioQueueSetOfflineRenderFormat (atdec->queue,
&output_format, &output_layout);
if (status)
goto set_format_error;
status = AudioQueueStart (atdec->queue, NULL);
if (status)
goto start_error;
return TRUE;
negotiation_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("no compatible downstream caps"));
return FALSE;
create_queue_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("AudioQueueNewOutput returned error: %d", status));
return FALSE;
set_format_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("AudioQueueSetOfflineRenderFormat returned error: %d", status));
return FALSE;
start_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("AudioQueueStart returned error: %d", status));
return FALSE;
}
static void
gst_atdec_buffer_emptied (void *user_data, AudioQueueRef queue,
AudioQueueBufferRef buffer)
{
AudioQueueFreeBuffer (queue, buffer);
}
static GstFlowReturn
gst_atdec_handle_frame (GstAudioDecoder * decoder, GstBuffer * buffer)
{
AudioTimeStamp timestamp = { 0 };
AudioStreamPacketDescription packet;
AudioQueueBufferRef input_buffer, output_buffer;
GstBuffer *out;
GstMapInfo info;
GstAudioInfo *audio_info;
int size, out_frames;
GstATDec *atdec = GST_ATDEC (decoder);
// copy the input buffer into an AudioQueueBuffer
size = gst_buffer_get_size (buffer);
AudioQueueAllocateBuffer (atdec->queue, size, &input_buffer);
gst_buffer_extract (buffer, 0, input_buffer->mAudioData, size);
input_buffer->mAudioDataByteSize = size;
// assume framed input
packet.mStartOffset = 0;
packet.mVariableFramesInPacket = 1;
packet.mDataByteSize = size;
// enqueue the buffer. It will get free'd once the gst_atdec_buffer_emptied
// callback is called
AudioQueueEnqueueBuffer (atdec->queue, input_buffer, 1, &packet);
// figure out how many frames we need to pull out of the queue
audio_info = gst_audio_decoder_get_audio_info (decoder);
out_frames =
GST_CLOCK_TIME_TO_FRAMES (GST_BUFFER_DURATION (buffer), audio_info->rate);
size = out_frames * audio_info->bpf;
AudioQueueAllocateBuffer (atdec->queue, size, &output_buffer);
// pull the frames
AudioQueueOfflineRender (atdec->queue, &timestamp, output_buffer, out_frames);
out =
gst_audio_decoder_allocate_output_buffer (decoder,
output_buffer->mAudioDataByteSize);
gst_buffer_map (out, &info, GST_MAP_WRITE);
memcpy (info.data, output_buffer->mAudioData,
output_buffer->mAudioDataByteSize);
gst_buffer_unmap (out, &info);
AudioQueueFreeBuffer (atdec->queue, output_buffer);
return gst_audio_decoder_finish_frame (decoder, out, 1);
}
static void
gst_atdec_flush (GstAudioDecoder * decoder, gboolean hard)
{
GstATDec *atdec = GST_ATDEC (decoder);
AudioQueueFlush (atdec->queue);
}

52
sys/applemedia/atdec.h Normal file
View file

@ -0,0 +1,52 @@
/* GStreamer
* Copyright (C) 2013 FIXME <alessandro.d@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef _GST_ATDEC_H_
#define _GST_ATDEC_H_
#include <gst/audio/gstaudiodecoder.h>
#include <AudioToolbox/AudioToolbox.h>
G_BEGIN_DECLS
#define GST_TYPE_ATDEC (gst_atdec_get_type())
#define GST_ATDEC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_ATDEC,GstATDec))
#define GST_ATDEC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_ATDEC,GstATDecClass))
#define GST_IS_ATDEC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_ATDEC))
#define GST_IS_ATDEC_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_ATDEC))
typedef struct _GstATDec GstATDec;
typedef struct _GstATDecClass GstATDecClass;
struct _GstATDec
{
GstAudioDecoder decoder;
AudioQueueRef queue;
};
struct _GstATDecClass
{
GstAudioDecoderClass decoder_class;
};
GType gst_atdec_get_type (void);
G_END_DECLS
#endif

2
sys/applemedia/plugin.m
View file

@ -31,6 +31,7 @@
#include "vth264encbin.h"
#include "vtenc.h"
#include "vtdec.h"
#include "atdec.h"
#ifndef HAVE_IOS
static void
@ -64,6 +65,7 @@ plugin_init (GstPlugin * plugin)
res &= gst_element_register (plugin, "vth264encbin", GST_RANK_NONE,
GST_TYPE_VT_H264_ENC_BIN);
#endif
res &= gst_element_register (plugin, "atdec", GST_RANK_PRIMARY, GST_TYPE_ATDEC);
gst_vtenc_register_elements (plugin);
gst_vtdec_register_elements (plugin);