gstreamer/subprojects/gst-plugins-bad/sys/applemedia/atdec.c

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/* 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-atdec
* @title: atdec
*
* AudioToolbox based decoder.
*
* ## Example launch line
* |[
* gst-launch-1.0 -v filesrc location=file.mov ! qtdemux ! queue ! aacparse ! atdec ! autoaudiosink
* ]|
* Decode aac audio from a mov file
*
*/
#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,
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GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE ("S16LE") ", layout=interleaved;"
GST_AUDIO_CAPS_MAKE ("F32LE") ", layout=interleaved")
);
static GstStaticPadTemplate gst_atdec_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/mpeg, mpegversion=4, framed=true, channels=[1,max];"
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"audio/mpeg, mpegversion=1, layer=[1, 3]")
);
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 =
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GST_STATIC_CAPS ("audio/mpeg, mpegversion=1, layer=[1, 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_static_pad_template (GST_ELEMENT_CLASS (klass),
&gst_atdec_src_template);
gst_element_class_add_static_pad_template (GST_ELEMENT_CLASS (klass),
&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;
atdec->output_position = 0;
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atdec->input_position = 0;
}
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");
atdec->output_position = 0;
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atdec->input_position = 0;
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;
memset (format, 0, sizeof (AudioStreamBasicDescription));
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;
format->mFramesPerPacket = 1024;
} else if (can_intersect_static_caps (caps, &mp3_caps)) {
gint layer, mpegaudioversion = 1;
gst_structure_get_int (structure, "layer", &layer);
gst_structure_get_int (structure, "mpegaudioversion", &mpegaudioversion);
switch (layer) {
case 1:
format->mFormatID = kAudioFormatMPEGLayer1;
format->mFramesPerPacket = 384;
break;
case 2:
format->mFormatID = kAudioFormatMPEGLayer2;
format->mFramesPerPacket = 1152;
break;
case 3:
format->mFormatID = kAudioFormatMPEGLayer3;
format->mFramesPerPacket = (mpegaudioversion == 1 ? 1152 : 576);
break;
default:
g_warn_if_reached ();
format->mFormatID = kAudioFormatMPEGLayer3;
format->mFramesPerPacket = 1152;
break;
}
} 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;
AudioTimeStamp timestamp = { 0 };
AudioQueueBufferRef output_buffer;
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);
/* Remember the number of samples per frame */
atdec->spf = input_format.mFramesPerPacket;
/* negotiate output caps */
output_caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (atdec));
if (!output_caps)
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output_caps =
gst_pad_get_pad_template_caps (GST_AUDIO_DECODER_SRC_PAD (atdec));
output_caps = gst_caps_fixate (output_caps);
gst_caps_set_simple (output_caps,
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"rate", G_TYPE_INT, (int) input_format.mSampleRate,
"channels", G_TYPE_INT, input_format.mChannelsPerFrame, 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;
timestamp.mFlags = kAudioTimeStampSampleTimeValid;
timestamp.mSampleTime = 0;
status =
AudioQueueAllocateBuffer (atdec->queue, atdec->spf * output_info.bpf,
&output_buffer);
if (status)
goto allocate_output_error;
status = AudioQueueOfflineRender (atdec->queue, &timestamp, output_buffer, 0);
if (status)
goto offline_render_error;
AudioQueueFreeBuffer (atdec->queue, output_buffer);
return TRUE;
create_queue_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("AudioQueueNewOutput returned error: %d", (gint) status));
return FALSE;
set_format_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("AudioQueueSetOfflineRenderFormat returned error: %d", (gint) status));
gst_atdec_destroy_queue (atdec, FALSE);
return FALSE;
start_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("AudioQueueStart returned error: %d", (gint) status));
gst_atdec_destroy_queue (atdec, FALSE);
return FALSE;
allocate_output_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("AudioQueueAllocateBuffer returned error: %d", (gint) status));
gst_atdec_destroy_queue (atdec, FALSE);
return FALSE;
offline_render_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("AudioQueueOfflineRender returned error: %d", (gint) status));
AudioQueueFreeBuffer (atdec->queue, output_buffer);
gst_atdec_destroy_queue (atdec, FALSE);
return FALSE;
}
static void
gst_atdec_buffer_emptied (void *user_data, AudioQueueRef queue,
AudioQueueBufferRef buffer)
{
AudioQueueFreeBuffer (queue, buffer);
}
static GstFlowReturn
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gst_atdec_offline_render (GstATDec * atdec, GstAudioInfo * audio_info)
{
OSStatus status;
AudioTimeStamp timestamp = { 0 };
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AudioQueueBufferRef output_buffer;
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GstFlowReturn flow_ret = GST_FLOW_OK;
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GstBuffer *out;
guint out_frames;
/* figure out how many frames we need to pull out of the queue */
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out_frames = atdec->input_position - atdec->output_position;
if (out_frames > atdec->spf)
out_frames = atdec->spf;
status = AudioQueueAllocateBuffer (atdec->queue, out_frames * audio_info->bpf,
&output_buffer);
if (status)
goto allocate_output_failed;
/* pull the frames */
timestamp.mFlags = kAudioTimeStampSampleTimeValid;
timestamp.mSampleTime = atdec->output_position;
status =
AudioQueueOfflineRender (atdec->queue, &timestamp, output_buffer,
out_frames);
if (status)
goto offline_render_failed;
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if (output_buffer->mAudioDataByteSize) {
if (output_buffer->mAudioDataByteSize % audio_info->bpf != 0)
goto invalid_buffer_size;
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GST_DEBUG_OBJECT (atdec,
"Got output buffer of size %u at position %" G_GUINT64_FORMAT,
(guint) output_buffer->mAudioDataByteSize, atdec->output_position);
atdec->output_position +=
output_buffer->mAudioDataByteSize / audio_info->bpf;
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out =
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gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (atdec),
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output_buffer->mAudioDataByteSize);
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gst_buffer_fill (out, 0, output_buffer->mAudioData,
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output_buffer->mAudioDataByteSize);
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flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (atdec), out, 1);
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GST_DEBUG_OBJECT (atdec, "Finished buffer: %s",
gst_flow_get_name (flow_ret));
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} else {
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GST_DEBUG_OBJECT (atdec, "Got empty output buffer");
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flow_ret = GST_FLOW_CUSTOM_SUCCESS;
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}
AudioQueueFreeBuffer (atdec->queue, output_buffer);
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return flow_ret;
allocate_output_failed:
{
GST_ELEMENT_ERROR (atdec, STREAM, DECODE, (NULL),
("AudioQueueAllocateBuffer returned error: %d", (gint) status));
return GST_FLOW_ERROR;
}
offline_render_failed:
{
AudioQueueFreeBuffer (atdec->queue, output_buffer);
GST_AUDIO_DECODER_ERROR (atdec, 1, STREAM, DECODE, (NULL),
("AudioQueueOfflineRender returned error: %d", (gint) status),
flow_ret);
return flow_ret;
}
invalid_buffer_size:
{
GST_AUDIO_DECODER_ERROR (atdec, 1, STREAM, DECODE, (NULL),
("AudioQueueOfflineRender returned invalid buffer size: %u (bpf %d)",
(guint) output_buffer->mAudioDataByteSize, audio_info->bpf),
flow_ret);
AudioQueueFreeBuffer (atdec->queue, output_buffer);
return flow_ret;
}
}
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static GstFlowReturn
gst_atdec_handle_frame (GstAudioDecoder * decoder, GstBuffer * buffer)
{
OSStatus status;
AudioStreamPacketDescription packet;
AudioQueueBufferRef input_buffer;
GstAudioInfo *audio_info;
int size;
GstFlowReturn flow_ret = GST_FLOW_OK;
GstATDec *atdec = GST_ATDEC (decoder);
audio_info = gst_audio_decoder_get_audio_info (decoder);
if (buffer == NULL) {
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GST_DEBUG_OBJECT (atdec, "Draining");
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AudioQueueFlush (atdec->queue);
while (atdec->input_position > atdec->output_position
&& flow_ret == GST_FLOW_OK) {
flow_ret = gst_atdec_offline_render (atdec, audio_info);
}
if (flow_ret == GST_FLOW_CUSTOM_SUCCESS)
flow_ret = GST_FLOW_OK;
return flow_ret;
}
/* copy the input buffer into an AudioQueueBuffer */
size = gst_buffer_get_size (buffer);
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GST_DEBUG_OBJECT (atdec,
"Handling buffer of size %u at timestamp %" GST_TIME_FORMAT, (guint) size,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
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status = AudioQueueAllocateBuffer (atdec->queue, size, &input_buffer);
if (status)
goto allocate_input_failed;
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
*/
status = AudioQueueEnqueueBuffer (atdec->queue, input_buffer, 1, &packet);
if (status)
goto enqueue_buffer_failed;
atdec->input_position += atdec->spf;
flow_ret = gst_atdec_offline_render (atdec, audio_info);
if (flow_ret == GST_FLOW_CUSTOM_SUCCESS)
flow_ret = GST_FLOW_OK;
return flow_ret;
allocate_input_failed:
{
GST_ELEMENT_ERROR (atdec, STREAM, DECODE, (NULL),
("AudioQueueAllocateBuffer returned error: %d", (gint) status));
return GST_FLOW_ERROR;
}
enqueue_buffer_failed:
{
GST_AUDIO_DECODER_ERROR (atdec, 1, STREAM, DECODE, (NULL),
("AudioQueueEnqueueBuffer returned error: %d", (gint) status),
flow_ret);
return flow_ret;
}
}
static void
gst_atdec_flush (GstAudioDecoder * decoder, gboolean hard)
{
GstATDec *atdec = GST_ATDEC (decoder);
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GST_DEBUG_OBJECT (atdec, "Flushing");
AudioQueueReset (atdec->queue);
atdec->output_position = 0;
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atdec->input_position = 0;
}