*
* 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., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-wavpackenc
*
- * <refsect2>
* WavpackEnc encodes raw audio into a framed Wavpack stream.
* <ulink url="http://www.wavpack.com/">Wavpack</ulink> is an open-source
* audio codec that features both lossless and lossy encoding.
+ *
+ * <refsect2>
* <title>Example launch line</title>
- * <para>
- * <programlisting>
- * gst-launch audiotestsrc num-buffers=500 ! wavpackenc ! filesink location=sinewave.wv
- * </programlisting>
- * This pipeline encodes audio from audiotestsrc into a Wavpack file.
- * </para>
- * <para>
- * <programlisting>
- * gst-launch cdda://1 ! wavpackenc ! filesink location=track1.wv
- * </programlisting>
- * This pipeline encodes audio from an audio CD into a Wavpack file using
+ * |[
+ * gst-launch-1.0 audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv
+ * ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed
+ * as the Wavpack encoder only accepts input with 32 bit width.
+ * |[
+ * gst-launch-1.0 cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv
+ * ]| This pipeline encodes audio from an audio CD into a Wavpack file using
* lossless encoding (the file output will be fairly large).
- * </para>
- * <para>
- * <programlisting>
- * gst-launch cdda://1 ! wavpackenc bitrate=128000 ! filesink location=track1.wv
- * </programlisting>
- * This pipeline encodes audio from an audio CD into a Wavpack file using
+ * |[
+ * gst-launch-1.0 cdda://1 ! audioconvert ! wavpackenc bitrate=128000 ! filesink location=track1.wv
+ * ]| This pipeline encodes audio from an audio CD into a Wavpack file using
* lossy encoding at a certain bitrate (the file will be fairly small).
- * </para>
* </refsect2>
*/
/*
- * TODO: - add multichannel handling. channel_mask is:
- * front left
- * front right
- * center
- * LFE
- * back left
- * back right
- * front left center
- * front right center
- * back left
- * back center
- * side left
- * side right
- * ...
- * - add 32 bit float mode. CONFIG_FLOAT_DATA
+ * TODO: - add 32 bit float mode. CONFIG_FLOAT_DATA
*/
#include <string.h>
#include <wavpack/wavpack.h>
#include "gstwavpackenc.h"
#include "gstwavpackcommon.h"
-#include "md5.h"
-static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buffer);
-static gboolean gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps);
+static gboolean gst_wavpack_enc_start (GstAudioEncoder * enc);
+static gboolean gst_wavpack_enc_stop (GstAudioEncoder * enc);
+static gboolean gst_wavpack_enc_set_format (GstAudioEncoder * enc,
+ GstAudioInfo * info);
+static GstFlowReturn gst_wavpack_enc_handle_frame (GstAudioEncoder * enc,
+ GstBuffer * in_buf);
+static gboolean gst_wavpack_enc_sink_event (GstAudioEncoder * enc,
+ GstEvent * event);
+
static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count);
-static gboolean gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event);
-static GstStateChangeReturn gst_wavpack_enc_change_state (GstElement * element,
- GstStateChange transition);
+static GstFlowReturn gst_wavpack_enc_drain (GstWavpackEnc * enc);
+
static void gst_wavpack_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_wavpack_enc_get_property (GObject * object, guint prop_id,
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "width = (int) 32, "
- "depth = (int) 32, "
- "endianness = (int) LITTLE_ENDIAN, "
- "channels = (int) [ 1, 2 ], "
- "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;"
- "audio/x-raw-int, "
- "width = (int) 24, "
- "depth = (int) 24, "
- "endianness = (int) LITTLE_ENDIAN, "
- "channels = (int) [ 1, 2 ], "
- "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;"
- "audio/x-raw-int, "
- "width = (int) 16, "
- "depth = (int) 16, "
- "endianness = (int) LITTLE_ENDIAN, "
- "channels = (int) [ 1, 2 ], "
- "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;"
- "audio/x-raw-int, "
- "width = (int) 8, "
- "depth = (int) 8, "
- "endianness = (int) LITTLE_ENDIAN, "
- "channels = (int) [ 1, 2 ], "
- "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE")
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " GST_AUDIO_NE (S32) ", "
+ "layout = (string) interleaved, "
+ "channels = (int) [ 1, 8 ], " "rate = (int) [ 6000, 192000 ]")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
- "width = (int) { 8, 16, 24, 32 }, "
- "channels = (int) [ 1, 2 ], "
+ "depth = (int) [ 1, 32 ], "
+ "channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE")
);
{GST_WAVPACK_ENC_MODE_FAST, "Fast Compression", "fast"},
{GST_WAVPACK_ENC_MODE_DEFAULT, "Normal Compression", "normal"},
{GST_WAVPACK_ENC_MODE_HIGH, "High Compression", "high"},
-#ifndef WAVPACK_OLD_API
{GST_WAVPACK_ENC_MODE_VERY_HIGH, "Very High Compression", "veryhigh"},
-#endif
{0, NULL, NULL}
};
return qtype;
}
-GST_BOILERPLATE (GstWavpackEnc, gst_wavpack_enc, GstElement, GST_TYPE_ELEMENT);
-
-static void
-gst_wavpack_enc_base_init (gpointer klass)
-{
- static const GstElementDetails element_details = {
- "Wavpack audio encoder",
- "Codec/Encoder/Audio",
- "Encodes audio with the Wavpack lossless/lossy audio codec",
- "Sebastian Dröge <slomo@circular-chaos.org>"
- };
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
- /* add pad templates */
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&sink_factory));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&src_factory));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&wvcsrc_factory));
-
- /* set element details */
- gst_element_class_set_details (element_class, &element_details);
-}
-
+#define gst_wavpack_enc_parent_class parent_class
+G_DEFINE_TYPE (GstWavpackEnc, gst_wavpack_enc, GST_TYPE_AUDIO_ENCODER);
static void
gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
- GstElementClass *gstelement_class = (GstElementClass *) klass;
+ GstElementClass *element_class = (GstElementClass *) (klass);
+ GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (klass);
- parent_class = g_type_class_peek_parent (klass);
+ /* add pad templates */
+ gst_element_class_add_static_pad_template (element_class, &sink_factory);
+ gst_element_class_add_static_pad_template (element_class, &src_factory);
+ gst_element_class_add_static_pad_template (element_class, &wvcsrc_factory);
- /* set state change handler */
- gstelement_class->change_state =
- GST_DEBUG_FUNCPTR (gst_wavpack_enc_change_state);
+ /* set element details */
+ gst_element_class_set_static_metadata (element_class, "Wavpack audio encoder",
+ "Codec/Encoder/Audio",
+ "Encodes audio with the Wavpack lossless/lossy audio codec",
+ "Sebastian Dröge <slomo@circular-chaos.org>");
/* set property handlers */
- gobject_class->set_property =
- GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_property);
- gobject_class->get_property =
- GST_DEBUG_FUNCPTR (gst_wavpack_enc_get_property);
+ gobject_class->set_property = gst_wavpack_enc_set_property;
+ gobject_class->get_property = gst_wavpack_enc_get_property;
+
+ base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_enc_start);
+ base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_enc_stop);
+ base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_format);
+ base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_enc_handle_frame);
+ base_class->sink_event = GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event);
/* install all properties */
g_object_class_install_property (gobject_class, ARG_MODE,
g_param_spec_enum ("mode", "Encoding mode",
"Speed versus compression tradeoff.",
GST_TYPE_WAVPACK_ENC_MODE, GST_WAVPACK_ENC_MODE_DEFAULT,
- G_PARAM_READWRITE));
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, ARG_BITRATE,
- g_param_spec_double ("bitrate", "Bitrate",
+ g_param_spec_uint ("bitrate", "Bitrate",
"Try to encode with this average bitrate (bits/sec). "
- "This enables lossy encoding! A value smaller than 24000.0 disables this.",
- 0.0, 9600000.0, 0.0, G_PARAM_READWRITE));
+ "This enables lossy encoding, values smaller than 24000 disable it again.",
+ 0, 9600000, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, ARG_BITSPERSAMPLE,
g_param_spec_double ("bits-per-sample", "Bits per sample",
"Try to encode with this amount of bits per sample. "
- "This enables lossy encoding! A value smaller than 2.0 disables this.",
- 0.0, 24.0, 0.0, G_PARAM_READWRITE));
+ "This enables lossy encoding, values smaller than 2.0 disable it again.",
+ 0.0, 24.0, 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, ARG_CORRECTION_MODE,
- g_param_spec_enum ("correction-mode", "Correction file mode",
- "Use this mode for correction file creation. Only works in lossy mode!",
+ g_param_spec_enum ("correction-mode", "Correction stream mode",
+ "Use this mode for the correction stream. Only works in lossy mode!",
GST_TYPE_WAVPACK_ENC_CORRECTION_MODE, GST_WAVPACK_CORRECTION_MODE_OFF,
- G_PARAM_READWRITE));
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, ARG_MD5,
g_param_spec_boolean ("md5", "MD5",
"Store MD5 hash of raw samples within the file.", FALSE,
- G_PARAM_READWRITE));
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, ARG_EXTRA_PROCESSING,
- g_param_spec_boolean ("extra-processing", "Extra processing",
- "Extra encode processing.", FALSE, G_PARAM_READWRITE));
+ g_param_spec_uint ("extra-processing", "Extra processing",
+ "Use better but slower filters for better compression/quality.",
+ 0, 6, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, ARG_JOINT_STEREO_MODE,
g_param_spec_enum ("joint-stereo-mode", "Joint-Stereo mode",
"Use this joint-stereo mode.", GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE,
- GST_WAVPACK_JS_MODE_AUTO, G_PARAM_READWRITE));
+ GST_WAVPACK_JS_MODE_AUTO,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
}
enc->first_block_size = 0;
if (enc->md5_context) {
- g_free (enc->md5_context);
+ g_checksum_free (enc->md5_context);
enc->md5_context = NULL;
}
+ if (enc->pending_segment)
+ gst_event_unref (enc->pending_segment);
+ enc->pending_segment = NULL;
+
+ if (enc->pending_buffer) {
+ gst_buffer_unref (enc->pending_buffer);
+ enc->pending_buffer = NULL;
+ enc->pending_offset = 0;
+ }
/* reset the last returns to GST_FLOW_OK. This is only set to something else
* while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
/* reset stream information */
enc->samplerate = 0;
- enc->width = 0;
+ enc->depth = 0;
enc->channels = 0;
+ enc->channel_mask = 0;
+ enc->need_channel_remap = FALSE;
+
+ enc->timestamp_offset = GST_CLOCK_TIME_NONE;
+ enc->next_ts = GST_CLOCK_TIME_NONE;
}
static void
-gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass)
+gst_wavpack_enc_init (GstWavpackEnc * enc)
{
- enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
- gst_pad_set_setcaps_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_set_caps));
- gst_pad_set_chain_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_wavpack_enc_chain));
- gst_pad_set_event_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event));
- gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
-
- /* setup src pad */
- enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
- gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
+ GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
/* initialize object attributes */
enc->wp_config = NULL;
enc->wv_id.correction = FALSE;
enc->wv_id.wavpack_enc = enc;
+ enc->wv_id.passthrough = FALSE;
enc->wvc_id.correction = TRUE;
enc->wvc_id.wavpack_enc = enc;
+ enc->wvc_id.passthrough = FALSE;
/* set default values of params */
enc->mode = GST_WAVPACK_ENC_MODE_DEFAULT;
- enc->bitrate = 0.0;
+ enc->bitrate = 0;
+ enc->bps = 0.0;
enc->correction_mode = GST_WAVPACK_CORRECTION_MODE_OFF;
enc->md5 = FALSE;
- enc->extra_processing = FALSE;
+ enc->extra_processing = 0;
enc->joint_stereo_mode = GST_WAVPACK_JS_MODE_AUTO;
+
+ /* require perfect ts */
+ gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
+
+ GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
}
+
static gboolean
-gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
+gst_wavpack_enc_start (GstAudioEncoder * enc)
{
- GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
- GstStructure *structure = gst_caps_get_structure (caps, 0);
- int depth = 0;
-
- /* check caps and put relevant parts into our object attributes */
- if (!gst_structure_get_int (structure, "channels", &enc->channels) ||
- !gst_structure_get_int (structure, "rate", &enc->samplerate) ||
- !gst_structure_get_int (structure, "width", &enc->width) ||
- !(gst_structure_get_int (structure, "depth", &depth) ||
- depth != enc->width)) {
- GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
- ("got invalid caps: %" GST_PTR_FORMAT, caps));
- gst_object_unref (enc);
- return FALSE;
+ GST_DEBUG_OBJECT (enc, "start");
+
+ return TRUE;
+}
+
+static gboolean
+gst_wavpack_enc_stop (GstAudioEncoder * enc)
+{
+ GstWavpackEnc *wpenc = GST_WAVPACK_ENC (enc);
+
+ GST_DEBUG_OBJECT (enc, "stop");
+ gst_wavpack_enc_reset (wpenc);
+
+ return TRUE;
+}
+
+static gboolean
+gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
+{
+ GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
+ GstAudioChannelPosition *pos;
+ GstAudioChannelPosition opos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
+ GstCaps *caps;
+ guint64 mask = 0;
+
+ /* we may be configured again, but that change should have cleanup context */
+ g_assert (enc->wp_context == NULL);
+
+ enc->channels = GST_AUDIO_INFO_CHANNELS (info);
+ enc->depth = GST_AUDIO_INFO_DEPTH (info);
+ enc->samplerate = GST_AUDIO_INFO_RATE (info);
+
+ pos = info->position;
+ g_assert (pos);
+
+ /* If one channel is NONE they'll be all undefined */
+ if (pos != NULL && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE) {
+ goto invalid_channels;
}
+ enc->channel_mask =
+ gst_wavpack_get_channel_mask_from_positions (pos, enc->channels);
+ enc->need_channel_remap =
+ gst_wavpack_set_channel_mapping (pos, enc->channels,
+ enc->channel_mapping);
+
+ /* wavpack caps hold gst mask, not wavpack mask */
+ gst_audio_channel_positions_to_mask (opos, enc->channels, FALSE, &mask);
+
/* set fixed src pad caps now that we know what we will get */
caps = gst_caps_new_simple ("audio/x-wavpack",
"channels", G_TYPE_INT, enc->channels,
"rate", G_TYPE_INT, enc->samplerate,
- "width", G_TYPE_INT, enc->width, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
+ "depth", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
- if (!gst_pad_set_caps (enc->srcpad, caps)) {
- GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
- ("setting caps failed: %" GST_PTR_FORMAT, caps));
- gst_caps_unref (caps);
- gst_object_unref (enc);
- return FALSE;
- }
- gst_pad_use_fixed_caps (enc->srcpad);
+ if (mask)
+ gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, mask, NULL);
+
+ if (!gst_audio_encoder_set_output_format (benc, caps))
+ goto setting_src_caps_failed;
gst_caps_unref (caps);
- gst_object_unref (enc);
+
+ /* no special feedback to base class; should provide all available samples */
+
return TRUE;
+
+ /* ERRORS */
+setting_src_caps_failed:
+ {
+ GST_DEBUG_OBJECT (enc,
+ "Couldn't set caps on source pad: %" GST_PTR_FORMAT, caps);
+ gst_caps_unref (caps);
+ return FALSE;
+ }
+invalid_channels:
+ {
+ GST_DEBUG_OBJECT (enc, "input has invalid channel layout");
+ return FALSE;
+ }
}
static void
{
enc->wp_config = g_new0 (WavpackConfig, 1);
/* set general stream informations in the WavpackConfig */
- enc->wp_config->bytes_per_sample = (enc->width + 7) >> 3;
- enc->wp_config->bits_per_sample = enc->width;
+ enc->wp_config->bytes_per_sample = GST_ROUND_UP_8 (enc->depth) / 8;
+ enc->wp_config->bits_per_sample = enc->depth;
enc->wp_config->num_channels = enc->channels;
-
- /* TODO: handle more than 2 channels correctly! */
- if (enc->channels == 1) {
- enc->wp_config->channel_mask = 0x4;
- } else if (enc->channels == 2) {
- enc->wp_config->channel_mask = 0x2 | 0x1;
- }
+ enc->wp_config->channel_mask = enc->channel_mask;
enc->wp_config->sample_rate = enc->samplerate;
/*
case GST_WAVPACK_ENC_MODE_HIGH:
enc->wp_config->flags |= CONFIG_HIGH_FLAG;
break;
-#ifndef WAVPACK_OLD_API
case GST_WAVPACK_ENC_MODE_VERY_HIGH:
enc->wp_config->flags |= CONFIG_HIGH_FLAG;
enc->wp_config->flags |= CONFIG_VERY_HIGH_FLAG;
break;
-#endif
}
/* Bitrate, enables lossy mode */
- if (enc->bitrate >= 2.0) {
+ if (enc->bitrate) {
enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
- if (enc->bitrate >= 24000.0) {
- enc->wp_config->bitrate = enc->bitrate / 1000.0;
- enc->wp_config->flags |= CONFIG_BITRATE_KBPS;
- } else {
- enc->wp_config->bitrate = enc->bitrate;
- }
+ enc->wp_config->flags |= CONFIG_BITRATE_KBPS;
+ enc->wp_config->bitrate = enc->bitrate / 1000.0;
+ } else if (enc->bps) {
+ enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
+ enc->wp_config->bitrate = enc->bps;
}
/* Correction Mode, only in lossy mode */
if (enc->wp_config->flags & CONFIG_HYBRID_FLAG) {
if (enc->correction_mode > GST_WAVPACK_CORRECTION_MODE_OFF) {
+ GstCaps *caps = gst_caps_new_simple ("audio/x-wavpack-correction",
+ "framed", G_TYPE_BOOLEAN, TRUE, NULL);
+
enc->wvcsrcpad =
gst_pad_new_from_static_template (&wvcsrc_factory, "wvcsrc");
/* try to add correction src pad, don't set correction mode on failure */
- GstCaps *caps = gst_caps_new_simple ("audio/x-wavpack-correction",
- "framed", G_TYPE_BOOLEAN, TRUE, NULL);
-
GST_DEBUG_OBJECT (enc, "Adding correction pad with caps %"
GST_PTR_FORMAT, caps);
if (!gst_pad_set_caps (enc->wvcsrcpad, caps)) {
/* MD5, setup MD5 context */
if ((enc->md5) && !(enc->md5_context)) {
enc->wp_config->flags |= CONFIG_MD5_CHECKSUM;
- enc->md5_context = g_new0 (MD5_CTX, 1);
- MD5Init (enc->md5_context);
+ enc->md5_context = g_checksum_new (G_CHECKSUM_MD5);
}
/* Extra encode processing */
if (enc->extra_processing) {
enc->wp_config->flags |= CONFIG_EXTRA_MODE;
+ enc->wp_config->xmode = enc->extra_processing;
}
/* Joint stereo mode */
}
}
-static int32_t *
-gst_wavpack_enc_format_samples (const uchar * src_data, uint32_t sample_count,
- guint width)
-{
- int32_t *data = g_new0 (int32_t, sample_count);
-
- /* put all samples into an int32_t*, no matter what
- * width we have and convert them from little endian
- * to host byte order */
-
- switch (width) {
- int i;
-
- case 8:
- for (i = 0; i < sample_count; i++)
- data[i] = (int32_t) (int8_t) src_data[i];
- break;
- case 16:
- for (i = 0; i < sample_count; i++)
- data[i] = (int32_t) src_data[2 * i]
- | ((int32_t) (int8_t) src_data[2 * i + 1] << 8);
- break;
- case 24:
- for (i = 0; i < sample_count; i++)
- data[i] = (int32_t) src_data[3 * i]
- | ((int32_t) src_data[3 * i + 1] << 8)
- | ((int32_t) (int8_t) src_data[3 * i + 2] << 16);
- break;
- case 32:
- for (i = 0; i < sample_count; i++)
- data[i] = (int32_t) src_data[4 * i]
- | ((int32_t) src_data[4 * i + 1] << 8)
- | ((int32_t) src_data[4 * i + 2] << 16)
- | ((int32_t) (int8_t) src_data[4 * i + 3] << 24);
- break;
- }
-
- return data;
-}
-
static int
gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
{
GstBuffer *buffer;
GstPad *pad;
guchar *block = (guchar *) data;
+ gint samples = 0;
- pad = (wid->correction) ? enc->wvcsrcpad : enc->srcpad;
+ pad = (wid->correction) ? enc->wvcsrcpad : GST_AUDIO_ENCODER_SRC_PAD (enc);
flow =
- (wid->correction) ? &enc->wvcsrcpad_last_return : &enc->
- srcpad_last_return;
-
- *flow = gst_pad_alloc_buffer_and_set_caps (pad, GST_BUFFER_OFFSET_NONE,
- count, GST_PAD_CAPS (pad), &buffer);
-
- if (*flow != GST_FLOW_OK) {
- GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
- GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow));
- return FALSE;
- }
+ (wid->correction) ? &enc->
+ wvcsrcpad_last_return : &enc->srcpad_last_return;
- g_memmove (GST_BUFFER_DATA (buffer), block, count);
+ buffer = gst_buffer_new_and_alloc (count);
+ gst_buffer_fill (buffer, 0, data, count);
if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
/* if it's a Wavpack block set buffer timestamp and duration, etc */
gst_wavpack_read_header (&wph, block);
- /* if it's the first wavpack block, send a NEW_SEGMENT event */
- if (wph.block_index == 0) {
- gst_pad_push_event (pad,
- gst_event_new_new_segment (FALSE,
- 1.0, GST_FORMAT_TIME, 0, GST_BUFFER_OFFSET_NONE, 0));
-
- /* save header for later reference, so we can re-send it later on
- * EOS with fixed up values for total sample count etc. */
- if (enc->first_block == NULL && !wid->correction) {
- enc->first_block = g_memdup (block, count);
- enc->first_block_size = count;
+ /* Only set when pushing the first buffer again, in that case
+ * we don't want to delay the buffer or push newsegment events
+ */
+ if (!wid->passthrough) {
+ /* Only push complete blocks */
+ if (enc->pending_buffer == NULL) {
+ enc->pending_buffer = buffer;
+ enc->pending_offset = wph.block_index;
+ } else if (enc->pending_offset == wph.block_index) {
+ enc->pending_buffer = gst_buffer_append (enc->pending_buffer, buffer);
+ } else {
+ GST_ERROR ("Got incomplete block, dropping");
+ gst_buffer_unref (enc->pending_buffer);
+ enc->pending_buffer = buffer;
+ enc->pending_offset = wph.block_index;
+ }
+
+ /* Is this the not-final block of multi-channel data? If so, just
+ * accumulate and return here. */
+ if (!(wph.flags & FINAL_BLOCK) && ((block[32] & ID_OPTIONAL_DATA) == 0))
+ return TRUE;
+
+ buffer = enc->pending_buffer;
+ enc->pending_buffer = NULL;
+ enc->pending_offset = 0;
+
+ /* only send segment on correction pad,
+ * regular pad is handled normally by baseclass */
+ if (wid->correction && enc->pending_segment) {
+ gst_pad_push_event (pad, enc->pending_segment);
+ enc->pending_segment = NULL;
+ }
+
+ if (wph.block_index == 0) {
+ /* save header for later reference, so we can re-send it later on
+ * EOS with fixed up values for total sample count etc. */
+ if (enc->first_block == NULL && !wid->correction) {
+ GstMapInfo map;
+
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
+ enc->first_block = g_memdup (map.data, map.size);
+ enc->first_block_size = map.size;
+ gst_buffer_unmap (buffer, &map);
+ }
}
}
+ samples = wph.block_samples;
- /* set buffer timestamp, duration, offset, offset_end from
- * the wavpack header */
- GST_BUFFER_TIMESTAMP (buffer) =
- gst_util_uint64_scale_int (GST_SECOND, wph.block_index,
- enc->samplerate);
- GST_BUFFER_DURATION (buffer) =
- gst_util_uint64_scale_int (GST_SECOND, wph.block_samples,
- enc->samplerate);
+ /* decorate buffer */
+ /* NOTE: this will get overwritten by baseclass, but stay for those
+ * that are pushed directly
+ * FIXME: add setting to baseclass to avoid overwriting it ?? */
GST_BUFFER_OFFSET (buffer) = wph.block_index;
GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples;
} else {
/* if it's something else set no timestamp and duration on the buffer */
GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count);
-
- GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
- GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE;
}
- /* push the buffer and forward errors */
- *flow = gst_pad_push (pad, buffer);
+ if (wid->correction || wid->passthrough) {
+ /* push the buffer and forward errors */
+ GST_DEBUG_OBJECT (enc, "pushing buffer with %" G_GSIZE_FORMAT " bytes",
+ gst_buffer_get_size (buffer));
+ *flow = gst_pad_push (pad, buffer);
+ } else {
+ GST_DEBUG_OBJECT (enc, "handing frame of %" G_GSIZE_FORMAT " bytes",
+ gst_buffer_get_size (buffer));
+ *flow = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), buffer,
+ samples);
+ }
if (*flow != GST_FLOW_OK) {
GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
return TRUE;
}
+static void
+gst_wavpack_enc_fix_channel_order (GstWavpackEnc * enc, gint32 * data,
+ gint nsamples)
+{
+ gint i, j;
+ gint32 tmp[8];
+
+ for (i = 0; i < nsamples / enc->channels; i++) {
+ for (j = 0; j < enc->channels; j++) {
+ tmp[enc->channel_mapping[j]] = data[j];
+ }
+ for (j = 0; j < enc->channels; j++) {
+ data[j] = tmp[j];
+ }
+ data += enc->channels;
+ }
+}
+
static GstFlowReturn
-gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
+gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
- GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
- uint32_t sample_count = GST_BUFFER_SIZE (buf) / ((enc->width + 7) >> 3);
- int32_t *data;
+ GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
+ uint32_t sample_count;
GstFlowReturn ret;
+ GstMapInfo map;
+
+ /* base class ensures configuration */
+ g_return_val_if_fail (enc->depth != 0, GST_FLOW_NOT_NEGOTIATED);
/* reset the last returns to GST_FLOW_OK. This is only set to something else
* while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
* so not valid anymore */
enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
- GST_DEBUG ("got %u raw samples", sample_count);
+ if (G_UNLIKELY (!buf))
+ return gst_wavpack_enc_drain (enc);
+
+ sample_count = gst_buffer_get_size (buf) / 4;
+ GST_DEBUG_OBJECT (enc, "got %u raw samples", sample_count);
/* check if we already have a valid WavpackContext, otherwise make one */
if (!enc->wp_context) {
enc->wp_context =
WavpackOpenFileOutput (gst_wavpack_enc_push_block, &enc->wv_id,
(enc->correction_mode > 0) ? &enc->wvc_id : NULL);
- if (!enc->wp_context) {
- GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
- ("error creating Wavpack context"));
- gst_object_unref (enc);
- gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
- }
+ if (!enc->wp_context)
+ goto context_failed;
/* set the WavpackConfig according to our parameters */
gst_wavpack_enc_set_wp_config (enc);
if (!WavpackSetConfiguration (enc->wp_context,
enc->wp_config, (uint32_t) (-1))
|| !WavpackPackInit (enc->wp_context)) {
- GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
- ("error setting up wavpack encoding context"));
WavpackCloseFile (enc->wp_context);
- gst_object_unref (enc);
- gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
+ goto config_failed;
}
- GST_DEBUG ("setup of encoding context successfull");
+ GST_DEBUG_OBJECT (enc, "setup of encoding context successfull");
}
+ if (enc->need_channel_remap) {
+ buf = gst_buffer_make_writable (buf);
+ gst_buffer_map (buf, &map, GST_MAP_WRITE);
+ gst_wavpack_enc_fix_channel_order (enc, (gint32 *) map.data, sample_count);
+ gst_buffer_unmap (buf, &map);
+ }
+
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+
/* if we want to append the MD5 sum to the stream update it here
* with the current raw samples */
if (enc->md5) {
- MD5Update (enc->md5_context, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
+ g_checksum_update (enc->md5_context, map.data, map.size);
}
- /* put all samples into an int32_t*, no matter what
- * width we have and convert them from little endian
- * to host byte order */
- data =
- gst_wavpack_enc_format_samples (GST_BUFFER_DATA (buf), sample_count,
- enc->width);
-
- gst_buffer_unref (buf);
-
/* encode and handle return values from encoding */
- if (WavpackPackSamples (enc->wp_context, data, sample_count / enc->channels)) {
- GST_DEBUG ("encoding samples successful");
+ if (WavpackPackSamples (enc->wp_context, (int32_t *) map.data,
+ sample_count / enc->channels)) {
+ GST_DEBUG_OBJECT (enc, "encoding samples successful");
+ gst_buffer_unmap (buf, &map);
ret = GST_FLOW_OK;
} else {
- if ((enc->srcpad_last_return == GST_FLOW_RESEND) ||
- (enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) {
- ret = GST_FLOW_RESEND;
- } else if ((enc->srcpad_last_return == GST_FLOW_OK) ||
+ gst_buffer_unmap (buf, &map);
+ if ((enc->srcpad_last_return == GST_FLOW_OK) ||
(enc->wvcsrcpad_last_return == GST_FLOW_OK)) {
ret = GST_FLOW_OK;
} else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) &&
(enc->wvcsrcpad_last_return == GST_FLOW_NOT_LINKED)) {
ret = GST_FLOW_NOT_LINKED;
- } else if ((enc->srcpad_last_return == GST_FLOW_WRONG_STATE) &&
- (enc->wvcsrcpad_last_return == GST_FLOW_WRONG_STATE)) {
- ret = GST_FLOW_WRONG_STATE;
+ } else if ((enc->srcpad_last_return == GST_FLOW_FLUSHING) &&
+ (enc->wvcsrcpad_last_return == GST_FLOW_FLUSHING)) {
+ ret = GST_FLOW_FLUSHING;
} else {
- GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
- ("encoding samples failed"));
- ret = GST_FLOW_ERROR;
+ goto encoding_failed;
}
}
- g_free (data);
- gst_object_unref (enc);
+exit:
return ret;
+
+ /* ERRORS */
+encoding_failed:
+ {
+ GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
+ ("encoding samples failed"));
+ ret = GST_FLOW_ERROR;
+ goto exit;
+ }
+config_failed:
+ {
+ GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
+ ("error setting up wavpack encoding context"));
+ ret = GST_FLOW_ERROR;
+ goto exit;
+ }
+context_failed:
+ {
+ GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
+ ("error creating Wavpack context"));
+ ret = GST_FLOW_ERROR;
+ goto exit;
+ }
}
static void
gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
{
- GstEvent *event = gst_event_new_new_segment (TRUE, 1.0, GST_FORMAT_BYTES,
- 0, GST_BUFFER_OFFSET_NONE, 0);
+ GstSegment segment;
gboolean ret;
+ GstQuery *query;
+ gboolean seekable = FALSE;
g_return_if_fail (enc);
g_return_if_fail (enc->first_block);
WavpackUpdateNumSamples (enc->wp_context, enc->first_block);
/* try to seek to the beginning of the output */
- ret = gst_pad_push_event (enc->srcpad, event);
+ query = gst_query_new_seeking (GST_FORMAT_BYTES);
+ if (gst_pad_peer_query (GST_AUDIO_ENCODER_SRC_PAD (enc), query)) {
+ GstFormat format;
+
+ gst_query_parse_seeking (query, &format, &seekable, NULL, NULL);
+ if (format != GST_FORMAT_BYTES)
+ seekable = FALSE;
+ } else {
+ GST_LOG_OBJECT (enc, "SEEKING query not handled");
+ }
+ gst_query_unref (query);
+
+ if (!seekable) {
+ GST_DEBUG_OBJECT (enc, "downstream not seekable; not rewriting");
+ return;
+ }
+
+ gst_segment_init (&segment, GST_FORMAT_BYTES);
+ ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
+ gst_event_new_segment (&segment));
if (ret) {
/* try to rewrite the first block */
GST_DEBUG_OBJECT (enc, "rewriting first block ...");
+ enc->wv_id.passthrough = TRUE;
ret = gst_wavpack_enc_push_block (&enc->wv_id,
enc->first_block, enc->first_block_size);
+ enc->wv_id.passthrough = FALSE;
+ g_free (enc->first_block);
+ enc->first_block = NULL;
} else {
GST_WARNING_OBJECT (enc, "rewriting of first block failed. "
"Seeking to first block failed!");
}
}
-static gboolean
-gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event)
+static GstFlowReturn
+gst_wavpack_enc_drain (GstWavpackEnc * enc)
{
- GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
- gboolean ret = TRUE;
+ if (!enc->wp_context)
+ return GST_FLOW_OK;
- GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event));
+ GST_DEBUG_OBJECT (enc, "draining");
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_EOS:
- /* Encode all remaining samples and flush them to the src pads */
- WavpackFlushSamples (enc->wp_context);
+ /* Encode all remaining samples and flush them to the src pads */
+ WavpackFlushSamples (enc->wp_context);
- /* write the MD5 sum if we have to write one */
- if ((enc->md5) && (enc->md5_context)) {
- guchar md5_digest[16];
+ /* Drop all remaining data, this is no complete block otherwise
+ * it would've been pushed already */
+ if (enc->pending_buffer) {
+ gst_buffer_unref (enc->pending_buffer);
+ enc->pending_buffer = NULL;
+ enc->pending_offset = 0;
+ }
- MD5Final (md5_digest, enc->md5_context);
- WavpackStoreMD5Sum (enc->wp_context, md5_digest);
- }
+ /* write the MD5 sum if we have to write one */
+ if ((enc->md5) && (enc->md5_context)) {
+ guint8 md5_digest[16];
+ gsize digest_len = sizeof (md5_digest);
- /* Try to rewrite the first frame with the correct sample number */
- if (enc->first_block)
- gst_wavpack_enc_rewrite_first_block (enc);
+ g_checksum_get_digest (enc->md5_context, md5_digest, &digest_len);
+ if (digest_len == sizeof (md5_digest)) {
+ WavpackStoreMD5Sum (enc->wp_context, md5_digest);
+ WavpackFlushSamples (enc->wp_context);
+ } else
+ GST_WARNING_OBJECT (enc, "Calculating MD5 digest failed");
+ }
- /* close the context if not already happened */
- if (enc->wp_context) {
- WavpackCloseFile (enc->wp_context);
- enc->wp_context = NULL;
- }
+ /* Try to rewrite the first frame with the correct sample number */
+ if (enc->first_block)
+ gst_wavpack_enc_rewrite_first_block (enc);
- ret = gst_pad_event_default (pad, event);
- break;
- case GST_EVENT_NEWSEGMENT:
- if (enc->wp_context) {
- GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding "
- "already started");
- }
- /* drop NEWSEGMENT events, we create our own when pushing
- * the first buffer to the pads */
- gst_event_unref (event);
- ret = TRUE;
- break;
- default:
- ret = gst_pad_event_default (pad, event);
- break;
+ /* close the context if not already happened */
+ if (enc->wp_context) {
+ WavpackCloseFile (enc->wp_context);
+ enc->wp_context = NULL;
}
- gst_object_unref (enc);
- return ret;
+ return GST_FLOW_OK;
}
-static GstStateChangeReturn
-gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition)
+static gboolean
+gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
{
- GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
- GstWavpackEnc *enc = GST_WAVPACK_ENC (element);
-
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:
- /* set the last returned GstFlowReturns of the two pads to GST_FLOW_OK
- * as they're only set to something else in WavpackPackSamples() or more
- * specific gst_wavpack_enc_push_block() and nothing happened there yet */
- enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
- break;
- case GST_STATE_CHANGE_READY_TO_PAUSED:
- break;
- case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
- default:
- break;
- }
+ GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
- ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+ GST_DEBUG_OBJECT (enc, "Received %s event on sinkpad",
+ GST_EVENT_TYPE_NAME (event));
- switch (transition) {
- case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
- break;
- case GST_STATE_CHANGE_PAUSED_TO_READY:
- gst_wavpack_enc_reset (enc);
- break;
- case GST_STATE_CHANGE_READY_TO_NULL:
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_SEGMENT:
+ if (enc->wp_context) {
+ GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding "
+ "already started");
+ }
+ /* peek and hold NEWSEGMENT events for sending on correction pad */
+ if (enc->pending_segment)
+ gst_event_unref (enc->pending_segment);
+ enc->pending_segment = gst_event_ref (event);
break;
default:
break;
}
- return ret;
+ /* baseclass handles rest */
+ return GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (benc, event);
}
static void
enc->mode = g_value_get_enum (value);
break;
case ARG_BITRATE:{
- gdouble val = g_value_get_double (value);
+ guint val = g_value_get_uint (value);
- if ((val >= 24000.0) && (val <= 9600000.0)) {
+ if ((val >= 24000) && (val <= 9600000)) {
enc->bitrate = val;
+ enc->bps = 0.0;
} else {
- enc->bitrate = 0.0;
+ enc->bitrate = 0;
+ enc->bps = 0.0;
}
break;
}
gdouble val = g_value_get_double (value);
if ((val >= 2.0) && (val <= 24.0)) {
- enc->bitrate = val;
+ enc->bps = val;
+ enc->bitrate = 0;
} else {
- enc->bitrate = 0.0;
+ enc->bps = 0.0;
+ enc->bitrate = 0;
}
break;
}
enc->md5 = g_value_get_boolean (value);
break;
case ARG_EXTRA_PROCESSING:
- enc->extra_processing = g_value_get_boolean (value);
+ enc->extra_processing = g_value_get_uint (value);
break;
case ARG_JOINT_STEREO_MODE:
enc->joint_stereo_mode = g_value_get_enum (value);
g_value_set_enum (value, enc->mode);
break;
case ARG_BITRATE:
- if (enc->bitrate >= 24000.0) {
- g_value_set_double (value, enc->bitrate);
+ if (enc->bps == 0.0) {
+ g_value_set_uint (value, enc->bitrate);
} else {
- g_value_set_double (value, 0.0);
+ g_value_set_uint (value, 0);
}
break;
case ARG_BITSPERSAMPLE:
- if (enc->bitrate <= 24.0) {
- g_value_set_double (value, enc->bitrate);
+ if (enc->bitrate == 0) {
+ g_value_set_double (value, enc->bps);
} else {
g_value_set_double (value, 0.0);
}
g_value_set_boolean (value, enc->md5);
break;
case ARG_EXTRA_PROCESSING:
- g_value_set_boolean (value, enc->extra_processing);
+ g_value_set_uint (value, enc->extra_processing);
break;
case ARG_JOINT_STEREO_MODE:
g_value_set_enum (value, enc->joint_stereo_mode);
GST_RANK_NONE, GST_TYPE_WAVPACK_ENC))
return FALSE;
- GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpack_enc", 0,
+ GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpackenc", 0,
"Wavpack encoder");
return TRUE;
projects / platform / upstream / gst-plugins-good.git / blobdiff