mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 04:01:08 +00:00
wavpackenc: port to 0.11
This commit is contained in:
parent
8721817f6d
commit
e0c7004667
3 changed files with 66 additions and 78 deletions
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@ -29,7 +29,6 @@
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#include <string.h>
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#include <gst/gst.h>
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#include <gst/audio/multichannel.h>
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GST_DEBUG_CATEGORY_EXTERN (wavpack_debug);
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#define GST_CAT_DEFAULT wavpack_debug
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@ -147,7 +146,7 @@ static const struct
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0x00001, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, {
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0x00002, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
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0x00004, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, {
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0x00008, GST_AUDIO_CHANNEL_POSITION_LFE}, {
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0x00008, GST_AUDIO_CHANNEL_POSITION_LFE1}, {
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0x00010, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, {
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0x00020, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
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0x00040, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, {
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@ -155,37 +154,25 @@ static const struct
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0x00100, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, {
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0x00200, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, {
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0x00400, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, {
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0x00800, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_CENTER */
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{
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0x01000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_LEFT */
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{
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0x02000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_CENTER */
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{
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0x04000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_RIGHT */
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{
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0x08000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_LEFT */
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{
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0x10000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_CENTER */
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{
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0x20000, GST_AUDIO_CHANNEL_POSITION_INVALID} /* TOP_BACK_RIGHT */
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0x00800, GST_AUDIO_CHANNEL_POSITION_TOP_CENTER}, {
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0x01000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT}, {
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0x02000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER}, {
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0x04000, GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT}, {
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0x08000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT}, {
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0x10000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER}, {
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0x20000, GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT}
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};
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#define MAX_CHANNEL_POSITIONS G_N_ELEMENTS (layout_mapping)
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gboolean
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gst_wavpack_set_channel_layout (GstCaps * caps, gint layout)
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gst_wavpack_get_channel_positions (gint num_channels, gint layout,
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GstAudioChannelPosition * pos)
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{
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GstAudioChannelPosition pos[MAX_CHANNEL_POSITIONS];
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GstStructure *s;
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gint num_channels, i, p;
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s = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (s, "channels", &num_channels))
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g_return_val_if_reached (FALSE);
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gint i, p;
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if (num_channels == 1 && layout == 0x00004) {
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pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
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gst_audio_set_channel_positions (s, pos);
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pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
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return TRUE;
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}
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@ -214,7 +201,6 @@ gst_wavpack_set_channel_layout (GstCaps * caps, gint layout)
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return FALSE;
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}
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gst_audio_set_channel_positions (s, pos);
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return TRUE;
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}
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@ -242,7 +228,7 @@ gst_wavpack_get_channel_mask_from_positions (GstAudioChannelPosition * pos,
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gint channel_mask = 0;
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gint i, j;
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if (nchannels == 1 && pos[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_MONO) {
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if (nchannels == 1 && pos[0] == GST_AUDIO_CHANNEL_POSITION_MONO) {
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channel_mask = 0x00000004;
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return channel_mask;
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}
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@ -24,7 +24,7 @@
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#define __GST_WAVPACK_COMMON_H__
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#include <gst/gst.h>
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#include <gst/audio/multichannel.h>
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#include <gst/audio/audio.h>
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#include <wavpack/wavpack.h>
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typedef struct
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@ -67,7 +67,7 @@ gboolean gst_wavpack_read_header (WavpackHeader * header, guint8 * buf);
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gboolean gst_wavpack_read_metadata (GstWavpackMetadata * meta,
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guint8 * header_data, guint8 ** p_data);
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gint gst_wavpack_get_default_channel_mask (gint nchannels);
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gboolean gst_wavpack_set_channel_layout (GstCaps * caps, gint layout);
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gboolean gst_wavpack_get_channel_positions (gint nchannels, gint layout, GstAudioChannelPosition *pos);
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GstAudioChannelPosition *gst_wavpack_get_default_channel_positions (gint nchannels);
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gint gst_wavpack_get_channel_mask_from_positions (GstAudioChannelPosition *pos, gint nchannels);
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gboolean gst_wavpack_set_channel_mapping (GstAudioChannelPosition *pos, gint nchannels, gint8 *channel_mapping);
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@ -31,7 +31,7 @@
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* |[
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* gst-launch audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv
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* ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed
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* as the Wavpack encoder only accepts input with 32 bit width (and every depth between 1 and 32 bits).
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* as the Wavpack encoder only accepts input with 32 bit width.
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* |[
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* gst-launch cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv
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* ]| This pipeline encodes audio from an audio CD into a Wavpack file using
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@ -90,19 +90,17 @@ GST_DEBUG_CATEGORY_STATIC (gst_wavpack_enc_debug);
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static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"width = (int) 32, "
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"depth = (int) { 24, 32 }, "
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"endianness = (int) BYTE_ORDER, "
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"channels = (int) [ 1, 8 ], "
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"rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE")
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S32) ", "
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"layout = (string) interleaved, "
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"channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]")
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);
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static GstStaticPadTemplate src_factory = 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-wavpack, "
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"width = (int) [ 1, 32 ], "
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"depth = (int) [ 1, 32 ], "
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"channels = (int) [ 1, 8 ], "
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"rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE")
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);
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@ -202,13 +200,15 @@ gst_wavpack_enc_joint_stereo_mode_get_type (void)
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return qtype;
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}
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GST_BOILERPLATE (GstWavpackEnc, gst_wavpack_enc, GstAudioEncoder,
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GST_TYPE_AUDIO_ENCODER);
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#define gst_wavpack_enc_parent_class parent_class
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G_DEFINE_TYPE (GstWavpackEnc, gst_wavpack_enc, GST_TYPE_AUDIO_ENCODER);
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static void
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gst_wavpack_enc_base_init (gpointer klass)
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gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *element_class = (GstElementClass *) (klass);
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GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (klass);
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/* add pad templates */
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gst_element_class_add_pad_template (element_class,
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@ -223,15 +223,6 @@ gst_wavpack_enc_base_init (gpointer klass)
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"Codec/Encoder/Audio",
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"Encodes audio with the Wavpack lossless/lossy audio codec",
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"Sebastian Dröge <slomo@circular-chaos.org>");
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}
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static void
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gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (klass);
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parent_class = g_type_class_peek_parent (klass);
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/* set property handlers */
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gobject_class->set_property = gst_wavpack_enc_set_property;
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@ -327,7 +318,7 @@ gst_wavpack_enc_reset (GstWavpackEnc * enc)
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}
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static void
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gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass)
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gst_wavpack_enc_init (GstWavpackEnc * enc)
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{
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GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
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@ -383,7 +374,9 @@ gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
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{
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GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
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GstAudioChannelPosition *pos;
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GstAudioChannelPosition opos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
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GstCaps *caps;
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guint64 mask = 0;
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/* we may be configured again, but that change should have cleanup context */
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g_assert (enc->wp_context == NULL);
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gst_wavpack_set_channel_mapping (pos, enc->channels,
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enc->channel_mapping);
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/* wavpack caps hold gst mask, not wavpack mask */
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gst_audio_channel_positions_to_mask (opos, enc->channels, &mask);
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/* set fixed src pad caps now that we know what we will get */
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caps = gst_caps_new_simple ("audio/x-wavpack",
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"channels", G_TYPE_INT, enc->channels,
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"rate", G_TYPE_INT, enc->samplerate,
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"width", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
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"depth", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
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if (!gst_wavpack_set_channel_layout (caps, enc->channel_mask))
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GST_WARNING_OBJECT (enc, "setting channel layout failed");
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if (mask)
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gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, mask, NULL);
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if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps))
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goto setting_src_caps_failed;
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@ -562,11 +558,11 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
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pad = (wid->correction) ? enc->wvcsrcpad : GST_AUDIO_ENCODER_SRC_PAD (enc);
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flow =
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(wid->correction) ? &enc->wvcsrcpad_last_return : &enc->
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srcpad_last_return;
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(wid->correction) ? &enc->
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wvcsrcpad_last_return : &enc->srcpad_last_return;
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buffer = gst_buffer_new_and_alloc (count);
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g_memmove (GST_BUFFER_DATA (buffer), block, count);
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gst_buffer_fill (buffer, 0, data, count);
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if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
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/* if it's a Wavpack block set buffer timestamp and duration, etc */
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@ -612,9 +608,12 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
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/* save header for later reference, so we can re-send it later on
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* EOS with fixed up values for total sample count etc. */
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if (enc->first_block == NULL && !wid->correction) {
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enc->first_block =
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g_memdup (GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer));
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enc->first_block_size = GST_BUFFER_SIZE (buffer);
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GstMapInfo map;
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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enc->first_block = g_memdup (map.data, map.size);
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enc->first_block_size = map.size;
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gst_buffer_unmap (buffer, &map);
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}
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}
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}
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@ -634,11 +633,11 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
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if (wid->correction || wid->passthrough) {
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/* push the buffer and forward errors */
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GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes",
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GST_BUFFER_SIZE (buffer));
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gst_buffer_get_size (buffer));
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*flow = gst_pad_push (pad, buffer);
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} else {
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GST_DEBUG_OBJECT (enc, "handing frame of %d bytes",
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GST_BUFFER_SIZE (buffer));
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gst_buffer_get_size (buffer));
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*flow = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), buffer,
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samples);
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}
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@ -676,6 +675,7 @@ gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
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GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
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uint32_t sample_count;
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GstFlowReturn ret;
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GstMapInfo map;
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/* base class ensures configuration */
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g_return_val_if_fail (enc->depth != 0, GST_FLOW_NOT_NEGOTIATED);
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@ -688,7 +688,7 @@ gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
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if (G_UNLIKELY (!buf))
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return gst_wavpack_enc_drain (enc);
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sample_count = GST_BUFFER_SIZE (buf) / 4;
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sample_count = gst_buffer_get_size (buf) / 4;
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GST_DEBUG_OBJECT (enc, "got %u raw samples", sample_count);
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/* check if we already have a valid WavpackContext, otherwise make one */
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@ -716,27 +716,28 @@ gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
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if (enc->need_channel_remap) {
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buf = gst_buffer_make_writable (buf);
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gst_wavpack_enc_fix_channel_order (enc, (gint32 *) GST_BUFFER_DATA (buf),
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sample_count);
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gst_buffer_map (buf, &map, GST_MAP_WRITE);
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gst_wavpack_enc_fix_channel_order (enc, (gint32 *) map.data, sample_count);
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gst_buffer_unmap (buf, &map);
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}
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gst_buffer_map (buf, &map, GST_MAP_READ);
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/* if we want to append the MD5 sum to the stream update it here
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* with the current raw samples */
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if (enc->md5) {
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g_checksum_update (enc->md5_context, GST_BUFFER_DATA (buf),
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GST_BUFFER_SIZE (buf));
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g_checksum_update (enc->md5_context, map.data, map.size);
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}
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/* encode and handle return values from encoding */
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if (WavpackPackSamples (enc->wp_context, (int32_t *) GST_BUFFER_DATA (buf),
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if (WavpackPackSamples (enc->wp_context, (int32_t *) map.data,
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sample_count / enc->channels)) {
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GST_DEBUG_OBJECT (enc, "encoding samples successful");
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gst_buffer_unmap (buf, &map);
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ret = GST_FLOW_OK;
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} else {
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if ((enc->srcpad_last_return == GST_FLOW_RESEND) ||
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(enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) {
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ret = GST_FLOW_RESEND;
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} else if ((enc->srcpad_last_return == GST_FLOW_OK) ||
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gst_buffer_unmap (buf, &map);
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if ((enc->srcpad_last_return == GST_FLOW_OK) ||
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(enc->wvcsrcpad_last_return == GST_FLOW_OK)) {
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ret = GST_FLOW_OK;
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} else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) &&
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@ -780,8 +781,7 @@ context_failed:
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static void
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gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
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{
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GstEvent *event = gst_event_new_new_segment (TRUE, 1.0, GST_FORMAT_BYTES,
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0, GST_BUFFER_OFFSET_NONE, 0);
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GstSegment segment;
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gboolean ret;
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g_return_if_fail (enc);
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@ -791,7 +791,9 @@ gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
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WavpackUpdateNumSamples (enc->wp_context, enc->first_block);
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/* try to seek to the beginning of the output */
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ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc), event);
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gst_segment_init (&segment, GST_FORMAT_BYTES);
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ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
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gst_event_new_segment (&segment));
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if (ret) {
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/* try to rewrite the first block */
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GST_DEBUG_OBJECT (enc, "rewriting first block ...");
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@ -861,7 +863,7 @@ gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
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GST_EVENT_TYPE_NAME (event));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_NEWSEGMENT:
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case GST_EVENT_SEGMENT:
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if (enc->wp_context) {
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GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding "
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"already started");
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@ -876,7 +878,7 @@ gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
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}
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/* baseclass handles rest */
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return FALSE;
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return GST_AUDIO_ENCODER_CLASS (parent_class)->event (benc, event);
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}
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static void
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@ -980,7 +982,7 @@ gst_wavpack_enc_plugin_init (GstPlugin * plugin)
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GST_RANK_NONE, GST_TYPE_WAVPACK_ENC))
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return FALSE;
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GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpack_enc", 0,
|
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GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpackenc", 0,
|
||||
"Wavpack encoder");
|
||||
|
||||
return TRUE;
|
||||
|
|
Loading…
Reference in a new issue