mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 20:21:24 +00:00
Merge branch 'master' into 0.11
Conflicts: ext/flac/gstflacenc.c
This commit is contained in:
Wim Taymans
commit
762602d56a
3 changed files with 166 additions and 248 deletions
|
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@ -1,7 +1,8 @@
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plugin_LTLIBRARIES = libgstflac.la
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libgstflac_la_SOURCES = gstflac.c gstflacdec.c gstflacenc.c gstflactag.c
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libgstflac_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(FLAC_CFLAGS)
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libgstflac_la_CFLAGS = -DGST_USE_UNSTABLE_API \
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$(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(FLAC_CFLAGS)
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libgstflac_la_LIBADD = \
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$(GST_PLUGINS_BASE_LIBS) -lgsttag-$(GST_MAJORMINOR) \
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-lgstaudio-$(GST_MAJORMINOR) \
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@ -151,24 +151,26 @@ GST_DEBUG_CATEGORY_STATIC (flacenc_debug);
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#define gst_flac_enc_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstFlacEnc, gst_flac_enc, GST_TYPE_ELEMENT,
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G_IMPLEMENT_INTERFACE (GST_TYPE_TAG_SETTER, NULL);
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G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, NULL));
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G_IMPLEMENT_INTERFACE (GST_TYPE_TAG_SETTER, NULL));
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static gboolean gst_flac_enc_start (GstAudioEncoder * enc);
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static gboolean gst_flac_enc_stop (GstAudioEncoder * enc);
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static gboolean gst_flac_enc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_flac_enc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static GstCaps *gst_flac_enc_getcaps (GstAudioEncoder * enc, GstCaps * filter);
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static gboolean gst_flac_enc_sink_event (GstAudioEncoder * enc,
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GstEvent * event);
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static void gst_flac_enc_finalize (GObject * object);
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static gboolean gst_flac_enc_sink_setcaps (GstPad * pad, GstCaps * caps);
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static GstCaps *gst_flac_enc_sink_getcaps (GstPad * pad);
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static gboolean gst_flac_enc_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_flac_enc_chain (GstPad * pad, GstBuffer * buffer);
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static gboolean gst_flac_enc_update_quality (GstFlacEnc * flacenc,
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gint quality);
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static void gst_flac_enc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_flac_enc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstStateChangeReturn gst_flac_enc_change_state (GstElement * element,
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GstStateChange transition);
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static FLAC__StreamEncoderWriteStatus
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gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder,
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@ -245,9 +247,11 @@ gst_flac_enc_class_init (GstFlacEncClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstAudioEncoderClass *base_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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base_class = (GstAudioEncoderClass *) (klass);
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GST_DEBUG_CATEGORY_INIT (flacenc_debug, "flacenc", 0,
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"Flac encoding element");
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@ -369,8 +373,6 @@ gst_flac_enc_class_init (GstFlacEncClass * klass)
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DEFAULT_SEEKPOINTS,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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gstelement_class->change_state = gst_flac_enc_change_state;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&src_factory));
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gst_element_class_add_pad_template (gstelement_class,
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@ -380,36 +382,26 @@ gst_flac_enc_class_init (GstFlacEncClass * klass)
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"Codec/Encoder/Audio",
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"Encodes audio with the FLAC lossless audio encoder",
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"Wim Taymans <wim.taymans@chello.be>");
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base_class->start = GST_DEBUG_FUNCPTR (gst_flac_enc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_flac_enc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_flac_enc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_flac_enc_handle_frame);
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base_class->getcaps = GST_DEBUG_FUNCPTR (gst_flac_enc_getcaps);
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base_class->event = GST_DEBUG_FUNCPTR (gst_flac_enc_sink_event);
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}
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static void
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gst_flac_enc_init (GstFlacEnc * flacenc)
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{
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flacenc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
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gst_pad_set_chain_function (flacenc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_flac_enc_chain));
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gst_pad_set_event_function (flacenc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_flac_enc_sink_event));
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gst_pad_set_getcaps_function (flacenc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_flac_enc_sink_getcaps));
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gst_pad_set_setcaps_function (flacenc->sinkpad,
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GST_DEBUG_FUNCPTR (gst_flac_enc_sink_setcaps));
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gst_element_add_pad (GST_ELEMENT (flacenc), flacenc->sinkpad);
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flacenc->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
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gst_pad_use_fixed_caps (flacenc->srcpad);
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gst_element_add_pad (GST_ELEMENT (flacenc), flacenc->srcpad);
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GstAudioEncoder *enc = GST_AUDIO_ENCODER (flacenc);
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flacenc->encoder = FLAC__stream_encoder_new ();
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flacenc->offset = 0;
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flacenc->samples_written = 0;
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flacenc->channels = 0;
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gst_flac_enc_update_quality (flacenc, DEFAULT_QUALITY);
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flacenc->tags = gst_tag_list_new ();
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flacenc->got_headers = FALSE;
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flacenc->headers = NULL;
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flacenc->last_flow = GST_FLOW_OK;
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/* arrange granulepos marking (and required perfect ts) */
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gst_audio_encoder_set_mark_granule (enc, TRUE);
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gst_audio_encoder_set_perfect_timestamp (enc, TRUE);
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}
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static void
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@ -417,12 +409,62 @@ gst_flac_enc_finalize (GObject * object)
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{
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GstFlacEnc *flacenc = GST_FLAC_ENC (object);
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gst_tag_list_free (flacenc->tags);
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FLAC__stream_encoder_delete (flacenc->encoder);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_flac_enc_start (GstAudioEncoder * enc)
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{
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GstFlacEnc *flacenc = GST_FLAC_ENC (enc);
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GST_DEBUG_OBJECT (enc, "start");
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flacenc->stopped = TRUE;
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flacenc->got_headers = FALSE;
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flacenc->last_flow = GST_FLOW_OK;
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flacenc->offset = 0;
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flacenc->channels = 0;
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flacenc->depth = 0;
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flacenc->sample_rate = 0;
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flacenc->eos = FALSE;
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flacenc->tags = gst_tag_list_new ();
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return TRUE;
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}
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static gboolean
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gst_flac_enc_stop (GstAudioEncoder * enc)
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{
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GstFlacEnc *flacenc = GST_FLAC_ENC (enc);
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GST_DEBUG_OBJECT (enc, "stop");
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gst_tag_list_free (flacenc->tags);
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flacenc->tags = NULL;
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if (FLAC__stream_encoder_get_state (flacenc->encoder) !=
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FLAC__STREAM_ENCODER_UNINITIALIZED) {
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flacenc->stopped = TRUE;
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FLAC__stream_encoder_finish (flacenc->encoder);
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}
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if (flacenc->meta) {
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FLAC__metadata_object_delete (flacenc->meta[0]);
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if (flacenc->meta[1])
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FLAC__metadata_object_delete (flacenc->meta[1]);
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if (flacenc->meta[2])
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FLAC__metadata_object_delete (flacenc->meta[2]);
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g_free (flacenc->meta);
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flacenc->meta = NULL;
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}
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g_list_foreach (flacenc->headers, (GFunc) gst_mini_object_unref, NULL);
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g_list_free (flacenc->headers);
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flacenc->headers = NULL;
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return TRUE;
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}
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static void
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add_one_tag (const GstTagList * list, const gchar * tag, gpointer user_data)
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{
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@ -478,9 +520,11 @@ gst_flac_enc_set_metadata (GstFlacEnc * flacenc, guint64 total_samples)
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if (n_images + n_preview_images > 0) {
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GstBuffer *buffer;
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#if 0
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GstCaps *caps;
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GstStructure *structure;
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GstTagImageType image_type = GST_TAG_IMAGE_TYPE_NONE;
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#endif
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gint i;
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guint8 *data;
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gsize size;
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@ -498,6 +542,7 @@ gst_flac_enc_set_metadata (GstFlacEnc * flacenc, guint64 total_samples)
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flacenc->meta[entries] =
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FLAC__metadata_object_new (FLAC__METADATA_TYPE_PICTURE);
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#if 0
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caps = gst_buffer_get_caps (buffer);
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structure = gst_caps_get_structure (caps, 0);
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@ -508,18 +553,21 @@ gst_flac_enc_set_metadata (GstFlacEnc * flacenc, guint64 total_samples)
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image_type = (i < n_images) ? 0x00 : 0x01;
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else
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image_type = image_type + 2;
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#endif
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data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
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FLAC__metadata_object_picture_set_data (flacenc->meta[entries],
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data, size, TRUE);
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gst_buffer_unmap (buffer, data, size);
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#if 0
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/* FIXME: There's no way to set the picture type in libFLAC */
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flacenc->meta[entries]->data.picture.type = image_type;
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FLAC__metadata_object_picture_set_mime_type (flacenc->meta[entries],
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(char *) gst_structure_get_name (structure), TRUE);
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gst_caps_unref (caps);
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#endif
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gst_buffer_unref (buffer);
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entries++;
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}
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@ -611,14 +659,17 @@ gst_flac_enc_caps_append_structure_with_widths (GstCaps * caps,
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}
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static GstCaps *
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gst_flac_enc_sink_getcaps (GstPad * pad)
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gst_flac_enc_getcaps (GstAudioEncoder * enc, GstCaps * filter)
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{
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GstCaps *ret = NULL;
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GstCaps *ret = NULL, *caps = NULL;
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GstPad *pad;
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|
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pad = GST_AUDIO_ENCODER_SINK_PAD (enc);
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|
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GST_OBJECT_LOCK (pad);
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|
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if (GST_PAD_CAPS (pad)) {
|
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ret = gst_caps_ref (GST_PAD_CAPS (pad));
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if (gst_pad_has_current_caps (pad)) {
|
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ret = gst_pad_get_current_caps (pad);
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} else {
|
||||
gint i, c;
|
||||
|
||||
|
|
@ -661,25 +712,26 @@ gst_flac_enc_sink_getcaps (GstPad * pad)
|
|||
|
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GST_DEBUG_OBJECT (pad, "Return caps %" GST_PTR_FORMAT, ret);
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|
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return ret;
|
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caps = gst_audio_encoder_proxy_getcaps (enc, ret);
|
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gst_caps_unref (ret);
|
||||
|
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return caps;
|
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}
|
||||
|
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static guint64
|
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gst_flac_enc_query_peer_total_samples (GstFlacEnc * flacenc, GstPad * pad)
|
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{
|
||||
GstFormat fmt = GST_FORMAT_DEFAULT;
|
||||
gint64 duration;
|
||||
|
||||
GST_DEBUG_OBJECT (flacenc, "querying peer for DEFAULT format duration");
|
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if (gst_pad_query_peer_duration (pad, &fmt, &duration)
|
||||
&& fmt == GST_FORMAT_DEFAULT && duration != GST_CLOCK_TIME_NONE)
|
||||
if (gst_pad_query_peer_duration (pad, GST_FORMAT_DEFAULT, &duration)
|
||||
&& duration != GST_CLOCK_TIME_NONE)
|
||||
goto done;
|
||||
|
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fmt = GST_FORMAT_TIME;
|
||||
GST_DEBUG_OBJECT (flacenc, "querying peer for TIME format duration");
|
||||
|
||||
if (gst_pad_query_peer_duration (pad, &fmt, &duration) &&
|
||||
fmt == GST_FORMAT_TIME && duration != GST_CLOCK_TIME_NONE) {
|
||||
if (gst_pad_query_peer_duration (pad, GST_FORMAT_TIME, &duration)
|
||||
&& duration != GST_CLOCK_TIME_NONE) {
|
||||
GST_DEBUG_OBJECT (flacenc, "peer reported duration %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (duration));
|
||||
duration = GST_CLOCK_TIME_TO_FRAMES (duration, flacenc->sample_rate);
|
||||
|
|
@ -698,45 +750,36 @@ done:
|
|||
}
|
||||
|
||||
static gboolean
|
||||
gst_flac_enc_sink_setcaps (GstPad * pad, GstCaps * caps)
|
||||
gst_flac_enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
|
||||
{
|
||||
GstFlacEnc *flacenc;
|
||||
GstStructure *structure;
|
||||
guint64 total_samples = GST_CLOCK_TIME_NONE;
|
||||
FLAC__StreamEncoderInitStatus init_status;
|
||||
gint depth, chans, rate, width;
|
||||
GstCaps *caps;
|
||||
|
||||
flacenc = GST_FLAC_ENC (gst_pad_get_parent (pad));
|
||||
flacenc = GST_FLAC_ENC (enc);
|
||||
|
||||
/* if configured again, means something changed, can't handle that */
|
||||
if (FLAC__stream_encoder_get_state (flacenc->encoder) !=
|
||||
FLAC__STREAM_ENCODER_UNINITIALIZED)
|
||||
goto encoder_already_initialized;
|
||||
|
||||
structure = gst_caps_get_structure (caps, 0);
|
||||
|
||||
if (!gst_structure_get_int (structure, "channels", &chans) ||
|
||||
!gst_structure_get_int (structure, "width", &width) ||
|
||||
!gst_structure_get_int (structure, "depth", &depth) ||
|
||||
!gst_structure_get_int (structure, "rate", &rate)) {
|
||||
GST_DEBUG_OBJECT (flacenc, "incomplete caps: %" GST_PTR_FORMAT, caps);
|
||||
return FALSE;
|
||||
}
|
||||
|
||||
flacenc->channels = chans;
|
||||
flacenc->width = width;
|
||||
flacenc->depth = depth;
|
||||
flacenc->sample_rate = rate;
|
||||
flacenc->channels = GST_AUDIO_INFO_CHANNELS (info);
|
||||
flacenc->width = GST_AUDIO_INFO_WIDTH (info);
|
||||
flacenc->depth = GST_AUDIO_INFO_DEPTH (info);
|
||||
flacenc->sample_rate = GST_AUDIO_INFO_RATE (info);
|
||||
|
||||
caps = gst_caps_new_simple ("audio/x-flac",
|
||||
"channels", G_TYPE_INT, flacenc->channels,
|
||||
"rate", G_TYPE_INT, flacenc->sample_rate, NULL);
|
||||
|
||||
if (!gst_pad_set_caps (flacenc->srcpad, caps))
|
||||
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps))
|
||||
goto setting_src_caps_failed;
|
||||
|
||||
gst_caps_unref (caps);
|
||||
|
||||
total_samples = gst_flac_enc_query_peer_total_samples (flacenc, pad);
|
||||
total_samples = gst_flac_enc_query_peer_total_samples (flacenc,
|
||||
GST_AUDIO_ENCODER_SINK_PAD (enc));
|
||||
|
||||
FLAC__stream_encoder_set_bits_per_sample (flacenc->encoder, flacenc->depth);
|
||||
FLAC__stream_encoder_set_sample_rate (flacenc->encoder, flacenc->sample_rate);
|
||||
|
|
@ -748,13 +791,17 @@ gst_flac_enc_sink_setcaps (GstPad * pad, GstCaps * caps)
|
|||
|
||||
gst_flac_enc_set_metadata (flacenc, total_samples);
|
||||
|
||||
/* callbacks clear to go now;
|
||||
* write callbacks receives headers during init */
|
||||
flacenc->stopped = FALSE;
|
||||
|
||||
init_status = FLAC__stream_encoder_init_stream (flacenc->encoder,
|
||||
gst_flac_enc_write_callback, gst_flac_enc_seek_callback,
|
||||
gst_flac_enc_tell_callback, NULL, flacenc);
|
||||
if (init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK)
|
||||
goto failed_to_initialize;
|
||||
|
||||
gst_object_unref (flacenc);
|
||||
/* no special feedback to base class; should provide all available samples */
|
||||
|
||||
return TRUE;
|
||||
|
||||
|
|
@ -832,16 +879,20 @@ gst_flac_enc_seek_callback (const FLAC__StreamEncoder * encoder,
|
|||
GstFlacEnc *flacenc;
|
||||
GstEvent *event;
|
||||
GstPad *peerpad;
|
||||
GstSegment seg;
|
||||
|
||||
flacenc = GST_FLAC_ENC (client_data);
|
||||
|
||||
if (flacenc->stopped)
|
||||
return FLAC__STREAM_ENCODER_SEEK_STATUS_OK;
|
||||
|
||||
event = gst_event_new_new_segment (TRUE, 1.0, GST_FORMAT_BYTES,
|
||||
absolute_byte_offset, GST_BUFFER_OFFSET_NONE, 0);
|
||||
gst_segment_init (&seg, GST_FORMAT_BYTES);
|
||||
seg.start = absolute_byte_offset;
|
||||
seg.stop = GST_BUFFER_OFFSET_NONE;
|
||||
seg.time = 0;
|
||||
event = gst_event_new_segment (&seg);
|
||||
|
||||
if ((peerpad = gst_pad_get_peer (flacenc->srcpad))) {
|
||||
if ((peerpad = gst_pad_get_peer (GST_AUDIO_ENCODER_SRC_PAD (flacenc)))) {
|
||||
gboolean ret = gst_pad_send_event (peerpad, event);
|
||||
|
||||
gst_object_unref (peerpad);
|
||||
|
|
@ -882,7 +933,7 @@ notgst_value_array_append_buffer (GValue * array_val, GstBuffer * buf)
|
|||
#define HDR_TYPE_STREAMINFO 0
|
||||
#define HDR_TYPE_VORBISCOMMENT 4
|
||||
|
||||
static void
|
||||
static GstFlowReturn
|
||||
gst_flac_enc_process_stream_headers (GstFlacEnc * enc)
|
||||
{
|
||||
GstBuffer *vorbiscomment = NULL;
|
||||
|
|
@ -891,6 +942,7 @@ gst_flac_enc_process_stream_headers (GstFlacEnc * enc)
|
|||
GValue array = { 0, };
|
||||
GstCaps *caps;
|
||||
GList *l;
|
||||
GstFlowReturn ret = GST_FLOW_OK;
|
||||
|
||||
caps = gst_caps_new_simple ("audio/x-flac",
|
||||
"channels", G_TYPE_INT, enc->channels,
|
||||
|
|
@ -976,28 +1028,27 @@ gst_flac_enc_process_stream_headers (GstFlacEnc * enc)
|
|||
|
||||
push_headers:
|
||||
|
||||
gst_pad_set_caps (enc->srcpad, caps);
|
||||
|
||||
/* push header buffers; update caps, so when we push the first buffer the
|
||||
* negotiated caps will change to caps that include the streamheader field */
|
||||
for (l = enc->headers; l != NULL; l = l->next) {
|
||||
GstBuffer *buf;
|
||||
|
||||
buf = GST_BUFFER (l->data);
|
||||
gst_buffer_set_caps (buf, caps);
|
||||
GST_LOG_OBJECT (enc, "Pushing header buffer, size %u bytes",
|
||||
gst_buffer_get_size (buf));
|
||||
#if 0
|
||||
GST_MEMDUMP_OBJECT (enc, "header buffer", GST_BUFFER_DATA (buf),
|
||||
GST_BUFFER_SIZE (buf));
|
||||
#endif
|
||||
(void) gst_pad_push (enc->srcpad, buf);
|
||||
ret = gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), buf);
|
||||
l->data = NULL;
|
||||
}
|
||||
g_list_free (enc->headers);
|
||||
enc->headers = NULL;
|
||||
|
||||
gst_caps_unref (caps);
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static FLAC__StreamEncoderWriteStatus
|
||||
|
|
@ -1017,31 +1068,6 @@ gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder,
|
|||
outbuf = gst_buffer_new_and_alloc (bytes);
|
||||
gst_buffer_fill (outbuf, 0, buffer, bytes);
|
||||
|
||||
if (samples > 0 && flacenc->samples_written != (guint64) - 1) {
|
||||
guint64 granulepos;
|
||||
|
||||
GST_BUFFER_TIMESTAMP (outbuf) = flacenc->start_ts +
|
||||
GST_FRAMES_TO_CLOCK_TIME (flacenc->samples_written,
|
||||
flacenc->sample_rate);
|
||||
GST_BUFFER_DURATION (outbuf) =
|
||||
GST_FRAMES_TO_CLOCK_TIME (samples, flacenc->sample_rate);
|
||||
/* offset_end = granulepos for ogg muxer */
|
||||
granulepos =
|
||||
flacenc->granulepos_offset + flacenc->samples_written + samples;
|
||||
GST_BUFFER_OFFSET_END (outbuf) = granulepos;
|
||||
/* offset = timestamp corresponding to granulepos for ogg muxer
|
||||
* (see vorbisenc for a much more elaborate version of this) */
|
||||
GST_BUFFER_OFFSET (outbuf) =
|
||||
GST_FRAMES_TO_CLOCK_TIME (granulepos, flacenc->sample_rate);
|
||||
} else {
|
||||
GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE;
|
||||
GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
|
||||
GST_BUFFER_OFFSET (outbuf) =
|
||||
flacenc->samples_written * flacenc->width * flacenc->channels;
|
||||
GST_BUFFER_OFFSET_END (outbuf) = 0;
|
||||
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_IN_CAPS);
|
||||
}
|
||||
|
||||
/* we assume libflac passes us stuff neatly framed */
|
||||
if (!flacenc->got_headers) {
|
||||
if (samples == 0) {
|
||||
|
|
@ -1052,34 +1078,34 @@ gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder,
|
|||
goto out;
|
||||
} else {
|
||||
GST_INFO_OBJECT (flacenc, "Non-header packet, we have all headers now");
|
||||
gst_flac_enc_process_stream_headers (flacenc);
|
||||
ret = gst_flac_enc_process_stream_headers (flacenc);
|
||||
flacenc->got_headers = TRUE;
|
||||
}
|
||||
} else if (flacenc->got_headers && samples == 0) {
|
||||
/* header fixup, push downstream directly */
|
||||
GST_DEBUG_OBJECT (flacenc, "Fixing up headers at pos=%" G_GUINT64_FORMAT
|
||||
", size=%u", flacenc->offset, (guint) bytes);
|
||||
#if 0
|
||||
GST_MEMDUMP_OBJECT (flacenc, "Presumed header fragment",
|
||||
GST_BUFFER_DATA (outbuf), GST_BUFFER_SIZE (outbuf));
|
||||
#endif
|
||||
ret = gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (flacenc), outbuf);
|
||||
} else {
|
||||
/* regular frame data, pass to base class */
|
||||
GST_LOG ("Pushing buffer: ts=%" GST_TIME_FORMAT ", samples=%u, size=%u, "
|
||||
"pos=%" G_GUINT64_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
|
||||
samples, (guint) bytes, flacenc->offset);
|
||||
ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (flacenc),
|
||||
outbuf, samples);
|
||||
}
|
||||
|
||||
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (flacenc->srcpad));
|
||||
ret = gst_pad_push (flacenc->srcpad, outbuf);
|
||||
|
||||
if (ret != GST_FLOW_OK)
|
||||
GST_DEBUG_OBJECT (flacenc, "flow: %s", gst_flow_get_name (ret));
|
||||
|
||||
flacenc->last_flow = ret;
|
||||
|
||||
out:
|
||||
|
||||
flacenc->offset += bytes;
|
||||
flacenc->samples_written += samples;
|
||||
|
||||
if (ret != GST_FLOW_OK)
|
||||
return FLAC__STREAM_ENCODER_WRITE_STATUS_FATAL_ERROR;
|
||||
|
|
@ -1099,24 +1125,25 @@ gst_flac_enc_tell_callback (const FLAC__StreamEncoder * encoder,
|
|||
}
|
||||
|
||||
static gboolean
|
||||
gst_flac_enc_sink_event (GstPad * pad, GstEvent * event)
|
||||
gst_flac_enc_sink_event (GstAudioEncoder * enc, GstEvent * event)
|
||||
{
|
||||
GstFlacEnc *flacenc;
|
||||
GstTagList *taglist;
|
||||
gboolean ret = TRUE;
|
||||
gboolean ret = FALSE;
|
||||
|
||||
flacenc = GST_FLAC_ENC (gst_pad_get_parent (pad));
|
||||
flacenc = GST_FLAC_ENC (enc);
|
||||
|
||||
GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event));
|
||||
|
||||
switch (GST_EVENT_TYPE (event)) {
|
||||
case GST_EVENT_NEWSEGMENT:{
|
||||
GstFormat format;
|
||||
case GST_EVENT_SEGMENT:{
|
||||
GstSegment seg;
|
||||
gint64 start, stream_time;
|
||||
|
||||
if (flacenc->offset == 0) {
|
||||
gst_event_parse_new_segment (event, NULL, NULL, &format, &start, NULL,
|
||||
&stream_time);
|
||||
gst_event_copy_segment (event, &seg);
|
||||
start = seg.start;
|
||||
stream_time = seg.time;
|
||||
} else {
|
||||
start = -1;
|
||||
stream_time = -1;
|
||||
|
|
@ -1128,23 +1155,24 @@ gst_flac_enc_sink_event (GstPad * pad, GstEvent * event)
|
|||
else
|
||||
GST_DEBUG ("Not handling newsegment event with non-zero start");
|
||||
} else {
|
||||
GstEvent *e = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_BYTES,
|
||||
0, -1, 0);
|
||||
GstEvent *e;
|
||||
|
||||
ret = gst_pad_push_event (flacenc->srcpad, e);
|
||||
gst_segment_init (&seg, GST_FORMAT_BYTES);
|
||||
e = gst_event_new_segment (&seg);
|
||||
ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc), e);
|
||||
}
|
||||
|
||||
if (stream_time > 0) {
|
||||
GST_DEBUG ("Not handling non-zero stream time");
|
||||
}
|
||||
|
||||
gst_event_unref (event);
|
||||
/* don't push it downstream, we'll generate our own via seek to 0 */
|
||||
gst_event_unref (event);
|
||||
ret = TRUE;
|
||||
break;
|
||||
}
|
||||
case GST_EVENT_EOS:
|
||||
FLAC__stream_encoder_finish (flacenc->encoder);
|
||||
ret = gst_pad_event_default (pad, event);
|
||||
flacenc->eos = TRUE;
|
||||
break;
|
||||
case GST_EVENT_TAG:
|
||||
if (flacenc->tags) {
|
||||
|
|
@ -1154,42 +1182,16 @@ gst_flac_enc_sink_event (GstPad * pad, GstEvent * event)
|
|||
} else {
|
||||
g_assert_not_reached ();
|
||||
}
|
||||
ret = gst_pad_event_default (pad, event);
|
||||
break;
|
||||
default:
|
||||
ret = gst_pad_event_default (pad, event);
|
||||
break;
|
||||
}
|
||||
|
||||
gst_object_unref (flacenc);
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_flac_enc_check_discont (GstFlacEnc * flacenc, GstClockTime expected,
|
||||
GstClockTime timestamp)
|
||||
{
|
||||
guint allowed_diff = GST_SECOND / flacenc->sample_rate / 2;
|
||||
|
||||
if ((timestamp + allowed_diff < expected)
|
||||
|| (timestamp > expected + allowed_diff)) {
|
||||
GST_ELEMENT_WARNING (flacenc, STREAM, FORMAT, (NULL),
|
||||
("Stream discontinuity detected (wanted %" GST_TIME_FORMAT " got %"
|
||||
GST_TIME_FORMAT "). The output will have wrong timestamps,"
|
||||
" consider using audiorate to handle discontinuities",
|
||||
GST_TIME_ARGS (expected), GST_TIME_ARGS (timestamp)));
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
/* TODO: Do something to handle discontinuities in the stream. The FLAC encoder
|
||||
* unfortunately doesn't have any way to flush it's internal buffers */
|
||||
|
||||
return FALSE;
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
gst_flac_enc_chain (GstPad * pad, GstBuffer * buffer)
|
||||
gst_flac_enc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
|
||||
{
|
||||
GstFlacEnc *flacenc;
|
||||
FLAC__int32 *data;
|
||||
|
|
@ -1199,42 +1201,26 @@ gst_flac_enc_chain (GstPad * pad, GstBuffer * buffer)
|
|||
FLAC__bool res;
|
||||
gpointer bdata;
|
||||
|
||||
flacenc = GST_FLAC_ENC (GST_PAD_PARENT (pad));
|
||||
flacenc = GST_FLAC_ENC (enc);
|
||||
|
||||
/* make sure setcaps has been called and the encoder is set up */
|
||||
if (G_UNLIKELY (flacenc->depth == 0))
|
||||
return GST_FLOW_NOT_NEGOTIATED;
|
||||
/* base class ensures configuration */
|
||||
g_return_val_if_fail (flacenc->depth != 0, GST_FLOW_NOT_NEGOTIATED);
|
||||
|
||||
width = flacenc->width;
|
||||
|
||||
/* Save the timestamp of the first buffer. This will be later
|
||||
* used as offset for all following buffers */
|
||||
if (flacenc->start_ts == GST_CLOCK_TIME_NONE) {
|
||||
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
|
||||
flacenc->start_ts = GST_BUFFER_TIMESTAMP (buffer);
|
||||
flacenc->granulepos_offset = gst_util_uint64_scale
|
||||
(GST_BUFFER_TIMESTAMP (buffer), flacenc->sample_rate, GST_SECOND);
|
||||
if (G_UNLIKELY (!buffer)) {
|
||||
if (flacenc->eos) {
|
||||
FLAC__stream_encoder_finish (flacenc->encoder);
|
||||
} else {
|
||||
flacenc->start_ts = 0;
|
||||
flacenc->granulepos_offset = 0;
|
||||
/* can't handle intermittent draining/resyncing */
|
||||
GST_ELEMENT_WARNING (flacenc, STREAM, FORMAT, (NULL),
|
||||
("Stream discontinuity detected. "
|
||||
"The output may have wrong timestamps, "
|
||||
"consider using audiorate to handle discontinuities"));
|
||||
}
|
||||
return flacenc->last_flow;
|
||||
}
|
||||
|
||||
/* Check if we have a continous stream, if not drop some samples or the buffer or
|
||||
* insert some silence samples */
|
||||
if (flacenc->next_ts != GST_CLOCK_TIME_NONE
|
||||
&& GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
|
||||
gst_flac_enc_check_discont (flacenc, flacenc->next_ts,
|
||||
GST_BUFFER_TIMESTAMP (buffer));
|
||||
}
|
||||
|
||||
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)
|
||||
&& GST_BUFFER_DURATION_IS_VALID (buffer))
|
||||
flacenc->next_ts =
|
||||
GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer);
|
||||
else
|
||||
flacenc->next_ts = GST_CLOCK_TIME_NONE;
|
||||
|
||||
bdata = gst_buffer_map (buffer, &bsize, NULL, GST_MAP_READ);
|
||||
samples = bsize / (width >> 3);
|
||||
|
||||
|
|
@ -1259,7 +1245,6 @@ gst_flac_enc_chain (GstPad * pad, GstBuffer * buffer)
|
|||
g_assert_not_reached ();
|
||||
}
|
||||
gst_buffer_unmap (buffer, bdata, bsize);
|
||||
gst_buffer_unref (buffer);
|
||||
|
||||
res = FLAC__stream_encoder_process_interleaved (flacenc->encoder,
|
||||
(const FLAC__int32 *) data, samples / flacenc->channels);
|
||||
|
|
@ -1425,64 +1410,3 @@ gst_flac_enc_get_property (GObject * object, guint prop_id,
|
|||
|
||||
GST_OBJECT_UNLOCK (this);
|
||||
}
|
||||
|
||||
static GstStateChangeReturn
|
||||
gst_flac_enc_change_state (GstElement * element, GstStateChange transition)
|
||||
{
|
||||
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
||||
GstFlacEnc *flacenc = GST_FLAC_ENC (element);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_NULL_TO_READY:
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
flacenc->stopped = FALSE;
|
||||
flacenc->start_ts = GST_CLOCK_TIME_NONE;
|
||||
flacenc->next_ts = GST_CLOCK_TIME_NONE;
|
||||
flacenc->granulepos_offset = 0;
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
||||
if (FLAC__stream_encoder_get_state (flacenc->encoder) !=
|
||||
FLAC__STREAM_ENCODER_UNINITIALIZED) {
|
||||
flacenc->stopped = TRUE;
|
||||
FLAC__stream_encoder_finish (flacenc->encoder);
|
||||
}
|
||||
flacenc->offset = 0;
|
||||
flacenc->samples_written = 0;
|
||||
flacenc->channels = 0;
|
||||
flacenc->depth = 0;
|
||||
flacenc->sample_rate = 0;
|
||||
if (flacenc->meta) {
|
||||
FLAC__metadata_object_delete (flacenc->meta[0]);
|
||||
|
||||
if (flacenc->meta[1])
|
||||
FLAC__metadata_object_delete (flacenc->meta[1]);
|
||||
|
||||
if (flacenc->meta[2])
|
||||
FLAC__metadata_object_delete (flacenc->meta[2]);
|
||||
|
||||
g_free (flacenc->meta);
|
||||
flacenc->meta = NULL;
|
||||
}
|
||||
g_list_foreach (flacenc->headers, (GFunc) gst_mini_object_unref, NULL);
|
||||
g_list_free (flacenc->headers);
|
||||
flacenc->headers = NULL;
|
||||
flacenc->got_headers = FALSE;
|
||||
flacenc->last_flow = GST_FLOW_OK;
|
||||
break;
|
||||
case GST_STATE_CHANGE_READY_TO_NULL:
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -22,6 +22,7 @@
|
|||
#define __GST_FLAC_ENC_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/audio/gstaudioencoder.h>
|
||||
|
||||
#include <FLAC/all.h>
|
||||
|
||||
|
|
@ -37,19 +38,15 @@ typedef struct _GstFlacEnc GstFlacEnc;
|
|||
typedef struct _GstFlacEncClass GstFlacEncClass;
|
||||
|
||||
struct _GstFlacEnc {
|
||||
GstElement element;
|
||||
GstAudioEncoder element;
|
||||
|
||||
/* < private > */
|
||||
|
||||
GstPad *sinkpad;
|
||||
GstPad *srcpad;
|
||||
|
||||
GstFlowReturn last_flow; /* save flow from last push so we can pass the
|
||||
* correct flow return upstream in case the push
|
||||
* fails for some reason */
|
||||
|
||||
guint64 offset;
|
||||
guint64 samples_written;
|
||||
gint channels;
|
||||
gint width;
|
||||
gint depth;
|
||||
|
|
@ -68,18 +65,14 @@ struct _GstFlacEnc {
|
|||
|
||||
GstTagList * tags;
|
||||
|
||||
gboolean eos;
|
||||
/* queue headers until we have them all so we can add streamheaders to caps */
|
||||
gboolean got_headers;
|
||||
GList *headers;
|
||||
|
||||
/* Timestamp and granulepos tracking */
|
||||
GstClockTime start_ts;
|
||||
GstClockTime next_ts;
|
||||
guint64 granulepos_offset;
|
||||
};
|
||||
|
||||
struct _GstFlacEncClass {
|
||||
GstElementClass parent_class;
|
||||
GstAudioEncoderClass parent_class;
|
||||
};
|
||||
|
||||
GType gst_flac_enc_get_type(void);
|
||||
|
|
|
|||
Loading…
Reference in a new issue