--- /dev/null
+/* 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, ×tamp, 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);
+}
projects / platform / upstream / gst-plugins-bad.git / commitdiff