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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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opus: port to updated 0.11
This commit is contained in:
Mark Nauwelaerts
parent
99a4e43c76
commit
8b39a2eb0c
9 changed files with 154 additions and 94 deletions
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@ -313,7 +313,7 @@ GST_PLUGINS_NONPORTED=" aiff asfmux \
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apexsink bz2 cdaudio cog curl dc1394 dirac directfb resindvd \
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gsettings jp2k ladspa mimic \
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musepack musicbrainz nas neon ofa openal opencv rsvg sdl smooth sndfile soundtouch spandsp timidity \
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wildmidi xvid lv2 teletextdec opus dvb"
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wildmidi xvid lv2 teletextdec dvb"
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AC_SUBST(GST_PLUGINS_NONPORTED)
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dnl these are all the gst plug-ins, compilable without additional libs
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@ -25,7 +25,7 @@
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/* copy of the same structure in the vorbis plugin */
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const GstAudioChannelPosition gst_opus_channel_positions[][8] = {
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{ /* Mono */
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GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
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GST_AUDIO_CHANNEL_POSITION_MONO},
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{ /* Stereo */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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@ -52,7 +52,7 @@ const GstAudioChannelPosition gst_opus_channel_positions[][8] = {
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE,
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GST_AUDIO_CHANNEL_POSITION_LFE1,
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},
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{ /* 6.1 Surround, in Vorbis spec since 2010-01-13 */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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@ -61,7 +61,7 @@ const GstAudioChannelPosition gst_opus_channel_positions[][8] = {
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GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
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GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
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GST_AUDIO_CHANNEL_POSITION_LFE},
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GST_AUDIO_CHANNEL_POSITION_LFE1},
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{ /* 7.1 Surround, in Vorbis spec since 2010-01-13 */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
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@ -70,7 +70,7 @@ const GstAudioChannelPosition gst_opus_channel_positions[][8] = {
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GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE},
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GST_AUDIO_CHANNEL_POSITION_LFE1},
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};
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const char *gst_opus_channel_names[] = {
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@ -22,7 +22,7 @@
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#define __GST_OPUS_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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G_BEGIN_DECLS
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@ -56,6 +56,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) { " GST_AUDIO_NE (S16) " }, "
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"layout = (string) interleaved, "
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"rate = (int) { 48000, 24000, 16000, 12000, 8000 }, "
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"channels = (int) [ 1, 8 ] ")
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);
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@ -206,11 +207,12 @@ gst_opus_dec_get_r128_volume (gint16 r128_gain)
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return DB_TO_LINEAR (gst_opus_dec_get_r128_gain (r128_gain));
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}
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static GstCaps *
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gst_opus_dec_negotiate (GstOpusDec * dec)
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static void
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gst_opus_dec_negotiate (GstOpusDec * dec, const GstAudioChannelPosition * pos)
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{
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GstCaps *caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
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GstStructure *s;
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GstAudioInfo info;
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caps = gst_caps_make_writable (caps);
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gst_caps_truncate (caps);
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@ -224,19 +226,41 @@ gst_opus_dec_negotiate (GstOpusDec * dec)
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GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels,
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dec->sample_rate);
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return caps;
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/* pass valid order to audio info */
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if (pos) {
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memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
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gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels);
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}
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/* set up source format */
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gst_audio_info_init (&info);
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gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16,
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dec->sample_rate, dec->n_channels, pos ? dec->opus_pos : NULL);
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gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
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/* but we still need the opus order for later reordering */
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if (pos) {
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memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
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gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels);
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} else {
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dec->opus_pos[0] = GST_AUDIO_CHANNEL_POSITION_INVALID;
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}
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dec->info = info;
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}
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static GstFlowReturn
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gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
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{
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const guint8 *data;
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GstCaps *caps;
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const GstAudioChannelPosition *pos = NULL;
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GstAudioChannelPosition pos[64];
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const GstAudioChannelPosition *posn = NULL;
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GstMapInfo map;
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g_return_val_if_fail (gst_opus_header_is_id_header (buf), GST_FLOW_ERROR);
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data = gst_buffer_map (buf, NULL, NULL, GST_MAP_READ);
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gst_buffer_map (buf, &map, GST_MAP_READ);
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data = map.data;
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g_return_val_if_fail (dec->n_channels == 0
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|| dec->n_channels == data[9], GST_FLOW_ERROR);
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@ -274,20 +298,18 @@ gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
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case 6:
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case 7:
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case 8:
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pos = gst_opus_channel_positions[dec->n_channels - 1];
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posn = gst_opus_channel_positions[dec->n_channels - 1];
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break;
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default:{
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gint i;
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GstAudioChannelPosition *posn =
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g_new (GstAudioChannelPosition, dec->n_channels);
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GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
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(NULL), ("Using NONE channel layout for more than 8 channels"));
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for (i = 0; i < dec->n_channels; i++)
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posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
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pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
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pos = posn;
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posn = pos;
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}
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}
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} else {
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@ -296,22 +318,9 @@ gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
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}
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}
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caps = gst_opus_dec_negotiate (dec);
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gst_opus_dec_negotiate (dec, posn);
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if (pos) {
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GST_DEBUG_OBJECT (dec, "Setting channel positions on caps");
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gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
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}
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if (dec->n_channels > 8) {
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g_free ((GstAudioChannelPosition *) pos);
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}
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GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
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gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
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gst_caps_unref (caps);
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gst_buffer_unmap (buf, (guint8 *) data, -1);
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gst_buffer_unmap (buf, &map);
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return GST_FLOW_OK;
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}
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@ -327,7 +336,7 @@ static GstFlowReturn
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opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
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{
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GstFlowReturn res = GST_FLOW_OK;
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gsize size, out_size;
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gsize size;
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guint8 *data;
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GstBuffer *outbuf;
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gint16 *out_data;
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@ -335,24 +344,22 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
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int samples;
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unsigned int packet_size;
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GstBuffer *buf;
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GstMapInfo map, omap;
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if (dec->state == NULL) {
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/* If we did not get any headers, default to 2 channels */
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if (dec->n_channels == 0) {
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GstCaps *caps;
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GST_INFO_OBJECT (dec, "No header, assuming single stream");
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dec->n_channels = 2;
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dec->sample_rate = 48000;
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caps = gst_opus_dec_negotiate (dec);
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GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
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gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
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gst_caps_unref (caps);
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/* default stereo mapping */
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dec->channel_mapping_family = 0;
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dec->channel_mapping[0] = 0;
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dec->channel_mapping[1] = 1;
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dec->n_streams = 1;
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dec->n_stereo_streams = 1;
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gst_opus_dec_negotiate (dec, NULL);
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}
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GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
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@ -389,7 +396,9 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
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buf = dec->use_inband_fec && dec->last_buffer ? dec->last_buffer : buffer;
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if (buf) {
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data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
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gst_buffer_map (buf, &map, GST_MAP_READ);
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data = map.data;
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size = map.size;
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GST_DEBUG_OBJECT (dec, "Using buffer of size %u", size);
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} else {
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@ -409,7 +418,8 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
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goto buffer_failed;
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}
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out_data = (gint16 *) gst_buffer_map (outbuf, &out_size, NULL, GST_MAP_WRITE);
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gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
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out_data = (gint16 *) omap.data;
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if (dec->use_inband_fec) {
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if (dec->last_buffer) {
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@ -425,8 +435,9 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
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/* normal decode */
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n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0);
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}
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gst_buffer_unmap (buf, data, size);
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gst_buffer_unmap (outbuf, out_data, out_size);
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gst_buffer_unmap (outbuf, &omap);
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if (buf)
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gst_buffer_unmap (buf, &map);
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if (n < 0) {
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GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
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@ -451,6 +462,9 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
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if (gst_buffer_get_size (outbuf) == 0) {
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gst_buffer_unref (outbuf);
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outbuf = NULL;
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} else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {
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gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,
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dec->n_channels, dec->opus_pos, dec->info.position);
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}
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/* Apply gain */
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@ -463,16 +477,18 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
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gsize rsize;
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unsigned int i, nsamples;
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double volume = dec->r128_gain_volume;
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gint16 *samples =
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(gint16 *) gst_buffer_map (outbuf, &rsize, NULL, GST_MAP_READWRITE);
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gint16 *samples;
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gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE);
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samples = (gint16 *) omap.data;
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rsize = omap.size;
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GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
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nsamples = rsize / 2;
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for (i = 0; i < nsamples; ++i) {
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int sample = (int) (samples[i] * volume + 0.5);
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samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
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}
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gst_buffer_unmap (outbuf, samples, rsize);
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gst_buffer_unmap (outbuf, &omap);
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}
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res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
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@ -542,8 +558,8 @@ static gboolean
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memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2)
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{
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gsize size1, size2;
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gpointer data1;
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gboolean res;
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GstMapInfo map;
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size1 = gst_buffer_get_size (buf1);
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size2 = gst_buffer_get_size (buf2);
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@ -551,9 +567,9 @@ memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2)
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if (size1 != size2)
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return FALSE;
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data1 = gst_buffer_map (buf1, NULL, NULL, GST_MAP_READ);
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res = gst_buffer_memcmp (buf2, 0, data1, size1) == 0;
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gst_buffer_unmap (buf1, data1, size1);
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gst_buffer_map (buf1, &map, GST_MAP_READ);
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res = gst_buffer_memcmp (buf2, 0, map.data, map.size) == 0;
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gst_buffer_unmap (buf1, &map);
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return res;
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}
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@ -56,6 +56,9 @@ struct _GstOpusDec {
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guint32 pre_skip;
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gint16 r128_gain;
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GstAudioChannelPosition opus_pos[64];
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GstAudioInfo info;
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guint8 n_streams;
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guint8 n_stereo_streams;
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guint8 channel_mapping_family;
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@ -120,6 +120,7 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " FORMAT_STR ", "
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"layout = (string) interleaved, "
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"rate = (int) { 8000, 12000, 16000, 24000, 48000 }, "
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"channels = (int) [ 1, 2 ] ")
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);
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@ -725,7 +726,7 @@ gst_opus_enc_sink_getcaps (GstAudioEncoder * benc, GstCaps * filter)
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GST_DEBUG_OBJECT (enc, "sink getcaps");
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peercaps = gst_pad_peer_query_caps (GST_AUDIO_ENCODER_SRC_PAD (benc), filter);
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peercaps = gst_pad_peer_query_caps (GST_AUDIO_ENCODER_SRC_PAD (benc), NULL);
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if (!peercaps) {
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GST_DEBUG_OBJECT (benc, "No peercaps, returning template sink caps");
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return
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@ -787,11 +788,14 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
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gsize bsize, size;
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gsize bytes = enc->frame_samples * enc->n_channels * 2;
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gint ret = GST_FLOW_OK;
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GstMapInfo map;
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g_mutex_lock (enc->property_lock);
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if (G_LIKELY (buf)) {
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bdata = gst_buffer_map (buf, &bsize, NULL, GST_MAP_READ);
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gst_buffer_map (buf, &map, GST_MAP_READ);
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bdata = map.data;
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bsize = map.size;
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if (G_UNLIKELY (bsize % bytes)) {
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GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
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@ -799,8 +803,6 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
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size = ((bsize / bytes) + 1) * bytes;
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mdata = g_malloc0 (size);
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memcpy (mdata, bdata, bsize);
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gst_buffer_unmap (buf, bdata, bsize);
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bdata = NULL;
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data = mdata;
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} else {
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data = bdata;
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@ -813,9 +815,8 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
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while (size) {
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gint encoded_size;
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unsigned char *out_data;
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gsize out_size;
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GstBuffer *outbuf;
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GstMapInfo omap;
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outbuf = gst_buffer_new_and_alloc (enc->max_payload_size * enc->n_channels);
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if (!outbuf)
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@ -824,11 +825,11 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
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GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
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enc->frame_samples, (int) bytes);
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out_data = gst_buffer_map (outbuf, &out_size, NULL, GST_MAP_WRITE);
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gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
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encoded_size =
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opus_multistream_encode (enc->state, (const gint16 *) data,
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enc->frame_samples, out_data, enc->max_payload_size * enc->n_channels);
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gst_buffer_unmap (outbuf, out_data, out_size);
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enc->frame_samples, omap.data, enc->max_payload_size * enc->n_channels);
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gst_buffer_unmap (outbuf, &omap);
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if (encoded_size < 0) {
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GST_ERROR_OBJECT (enc, "Encoding failed: %d", encoded_size);
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|
@ -837,7 +838,7 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
|
|||
} else if (encoded_size > enc->max_payload_size) {
|
||||
GST_WARNING_OBJECT (enc,
|
||||
"Encoded size %d is higher than max payload size (%d bytes)",
|
||||
out_size, enc->max_payload_size);
|
||||
encoded_size, enc->max_payload_size);
|
||||
ret = GST_FLOW_ERROR;
|
||||
goto done;
|
||||
}
|
||||
|
|
@ -859,7 +860,7 @@ gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
|
|||
done:
|
||||
|
||||
if (bdata)
|
||||
gst_buffer_unmap (buf, bdata, bsize);
|
||||
gst_buffer_unmap (buf, &map);
|
||||
g_mutex_unlock (enc->property_lock);
|
||||
|
||||
if (mdata)
|
||||
|
|
@ -893,7 +894,7 @@ gst_opus_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
|
|||
/* negotiate with these caps */
|
||||
GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
|
||||
|
||||
gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps);
|
||||
gst_audio_encoder_set_output_format (benc, caps);
|
||||
gst_caps_unref (caps);
|
||||
|
||||
enc->header_sent = TRUE;
|
||||
|
|
|
|||
|
|
@ -127,11 +127,11 @@ _gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
|
|||
g_assert (gst_buffer_is_writable (buf));
|
||||
|
||||
/* mark buffer */
|
||||
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
|
||||
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
|
||||
|
||||
g_value_init (&value, GST_TYPE_BUFFER);
|
||||
buf = gst_buffer_copy (buf);
|
||||
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
|
||||
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
|
||||
gst_value_set_buffer (&value, buf);
|
||||
gst_buffer_unref (buf);
|
||||
gst_value_array_append_value (&array, &value);
|
||||
|
|
@ -152,14 +152,15 @@ gst_opus_header_create_caps_from_headers (GstCaps ** caps, GSList ** headers,
|
|||
{
|
||||
int n_streams, family;
|
||||
gboolean multistream;
|
||||
GstMapInfo map;
|
||||
guint8 *data;
|
||||
gsize size;
|
||||
|
||||
g_return_if_fail (caps);
|
||||
g_return_if_fail (headers && !*headers);
|
||||
g_return_if_fail (gst_buffer_get_size (buf1) >= 19);
|
||||
|
||||
data = gst_buffer_map (buf1, &size, NULL, GST_MAP_READ);
|
||||
gst_buffer_map (buf1, &map, GST_MAP_READ);
|
||||
data = map.data;
|
||||
|
||||
/* work out the number of streams */
|
||||
family = data[18];
|
||||
|
|
@ -167,16 +168,16 @@ gst_opus_header_create_caps_from_headers (GstCaps ** caps, GSList ** headers,
|
|||
n_streams = 1;
|
||||
} else {
|
||||
/* only included in the header for family > 0 */
|
||||
if (size >= 20)
|
||||
if (map.size >= 20)
|
||||
n_streams = data[19];
|
||||
else {
|
||||
g_warning ("family > 0 but header buffer size < 20");
|
||||
gst_buffer_unmap (buf1, data, size);
|
||||
gst_buffer_unmap (buf1, &map);
|
||||
return;
|
||||
}
|
||||
}
|
||||
|
||||
gst_buffer_unmap (buf1, data, size);
|
||||
gst_buffer_unmap (buf1, &map);
|
||||
|
||||
/* mark and put on caps */
|
||||
multistream = n_streams > 1;
|
||||
|
|
@ -228,13 +229,16 @@ gst_opus_header_is_id_header (GstBuffer * buf)
|
|||
guint8 *data = NULL;
|
||||
guint8 channels, channel_mapping_family, n_streams, n_stereo_streams;
|
||||
gboolean ret = FALSE;
|
||||
GstMapInfo map;
|
||||
|
||||
if (size < 19)
|
||||
goto beach;
|
||||
if (!gst_opus_header_is_header (buf, "OpusHead", 8))
|
||||
goto beach;
|
||||
|
||||
data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
|
||||
gst_buffer_map (buf, &map, GST_MAP_READ);
|
||||
data = map.data;
|
||||
size = map.size;
|
||||
|
||||
channels = data[9];
|
||||
|
||||
|
|
@ -263,7 +267,7 @@ gst_opus_header_is_id_header (GstBuffer * buf)
|
|||
|
||||
beach:
|
||||
if (data)
|
||||
gst_buffer_unmap (buf, data, size);
|
||||
gst_buffer_unmap (buf, &map);
|
||||
return ret;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -65,8 +65,8 @@ G_DEFINE_TYPE (GstOpusParse, gst_opus_parse, GST_TYPE_BASE_PARSE);
|
|||
|
||||
static gboolean gst_opus_parse_start (GstBaseParse * parse);
|
||||
static gboolean gst_opus_parse_stop (GstBaseParse * parse);
|
||||
static gboolean gst_opus_parse_check_valid_frame (GstBaseParse * base,
|
||||
GstBaseParseFrame * frame, guint * frame_size, gint * skip);
|
||||
static GstFlowReturn gst_opus_parse_handle_frame (GstBaseParse * base,
|
||||
GstBaseParseFrame * frame, gint * skip);
|
||||
static GstFlowReturn gst_opus_parse_parse_frame (GstBaseParse * base,
|
||||
GstBaseParseFrame * frame);
|
||||
|
||||
|
|
@ -81,9 +81,7 @@ gst_opus_parse_class_init (GstOpusParseClass * klass)
|
|||
|
||||
bpclass->start = GST_DEBUG_FUNCPTR (gst_opus_parse_start);
|
||||
bpclass->stop = GST_DEBUG_FUNCPTR (gst_opus_parse_stop);
|
||||
bpclass->check_valid_frame =
|
||||
GST_DEBUG_FUNCPTR (gst_opus_parse_check_valid_frame);
|
||||
bpclass->parse_frame = GST_DEBUG_FUNCPTR (gst_opus_parse_parse_frame);
|
||||
bpclass->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_parse_handle_frame);
|
||||
|
||||
gst_element_class_add_pad_template (element_class,
|
||||
gst_static_pad_template_get (&opus_parse_src_factory));
|
||||
|
|
@ -134,13 +132,13 @@ gst_opus_parse_stop (GstBaseParse * base)
|
|||
return TRUE;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_opus_parse_check_valid_frame (GstBaseParse * base,
|
||||
GstBaseParseFrame * frame, guint * frame_size, gint * skip)
|
||||
static GstFlowReturn
|
||||
gst_opus_parse_handle_frame (GstBaseParse * base,
|
||||
GstBaseParseFrame * frame, gint * skip)
|
||||
{
|
||||
GstOpusParse *parse;
|
||||
guint8 *data;
|
||||
gsize size;
|
||||
gint size;
|
||||
guint32 packet_size;
|
||||
int ret = FALSE;
|
||||
const unsigned char *frames[48];
|
||||
|
|
@ -150,10 +148,15 @@ gst_opus_parse_check_valid_frame (GstBaseParse * base,
|
|||
int nframes;
|
||||
int packet_offset = 0;
|
||||
gboolean is_header, is_idheader, is_commentheader;
|
||||
GstMapInfo map;
|
||||
|
||||
parse = GST_OPUS_PARSE (base);
|
||||
|
||||
data = gst_buffer_map (frame->buffer, &size, NULL, GST_MAP_READ);
|
||||
*skip = -1;
|
||||
|
||||
gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
|
||||
data = map.data;
|
||||
size = map.size;
|
||||
GST_DEBUG_OBJECT (parse,
|
||||
"Checking for frame, %" G_GSIZE_FORMAT " bytes in buffer", size);
|
||||
|
||||
|
|
@ -201,18 +204,53 @@ gst_opus_parse_check_valid_frame (GstBaseParse * base,
|
|||
|
||||
if (is_header) {
|
||||
*skip = 0;
|
||||
*frame_size = size;
|
||||
} else {
|
||||
*skip = packet_offset;
|
||||
*frame_size = payload_offset;
|
||||
size = payload_offset;
|
||||
}
|
||||
|
||||
GST_DEBUG_OBJECT (parse, "Got Opus packet at offset %d, %d bytes", *skip,
|
||||
*frame_size);
|
||||
size);
|
||||
ret = TRUE;
|
||||
|
||||
beach:
|
||||
gst_buffer_unmap (frame->buffer, data, size);
|
||||
gst_buffer_unmap (frame->buffer, &map);
|
||||
|
||||
/* convert old style result to new one */
|
||||
if (!ret) {
|
||||
if (*skip < 0)
|
||||
*skip = 1;
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
|
||||
/* always skip first if needed */
|
||||
if (*skip > 0)
|
||||
return GST_FLOW_OK;
|
||||
|
||||
/* normalize again */
|
||||
if (*skip < 0)
|
||||
*skip = 0;
|
||||
|
||||
/* not enough */
|
||||
if (size > map.size)
|
||||
return GST_FLOW_OK;
|
||||
|
||||
/* FIXME some day ... should not mess with buffer itself */
|
||||
if (!parse->header_sent) {
|
||||
gst_buffer_replace (&frame->buffer,
|
||||
gst_buffer_copy_region (frame->buffer, GST_BUFFER_COPY_ALL, 0, size));
|
||||
gst_buffer_unref (frame->buffer);
|
||||
}
|
||||
|
||||
ret = gst_opus_parse_parse_frame (base, frame);
|
||||
|
||||
if (ret == GST_BASE_PARSE_FLOW_DROPPED) {
|
||||
frame->flags |= GST_BASE_PARSE_FRAME_FLAG_DROP;
|
||||
ret = GST_FLOW_OK;
|
||||
}
|
||||
if (ret == GST_FLOW_OK)
|
||||
ret = gst_base_parse_finish_frame (base, frame, size);
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
|
|
@ -278,8 +316,7 @@ gst_opus_parse_parse_frame (GstBaseParse * base, GstBaseParseFrame * frame)
|
|||
guint64 duration;
|
||||
GstOpusParse *parse;
|
||||
gboolean is_idheader, is_commentheader;
|
||||
guint8 *data;
|
||||
gsize size;
|
||||
GstMapInfo map;
|
||||
|
||||
parse = GST_OPUS_PARSE (base);
|
||||
|
||||
|
|
@ -331,9 +368,9 @@ gst_opus_parse_parse_frame (GstBaseParse * base, GstBaseParseFrame * frame)
|
|||
|
||||
GST_BUFFER_TIMESTAMP (frame->buffer) = parse->next_ts;
|
||||
|
||||
data = gst_buffer_map (frame->buffer, &size, NULL, GST_MAP_READ);
|
||||
duration = packet_duration_opus (data, size);
|
||||
gst_buffer_unmap (frame->buffer, data, size);
|
||||
gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
|
||||
duration = packet_duration_opus (map.data, map.size);
|
||||
gst_buffer_unmap (frame->buffer, &map);
|
||||
parse->next_ts += duration;
|
||||
|
||||
GST_BUFFER_DURATION (frame->buffer) = duration;
|
||||
|
|
|
|||
|
|
@ -114,17 +114,16 @@ gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
|
|||
{
|
||||
GstRTPBuffer rtpbuf = { NULL, };
|
||||
GstBuffer *outbuf;
|
||||
gsize size;
|
||||
gpointer *data;
|
||||
GstMapInfo map;
|
||||
|
||||
/* Copy data and timestamp to a new output buffer
|
||||
* FIXME : Don't we have a convenience function for this ? */
|
||||
data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
|
||||
outbuf = gst_rtp_buffer_new_copy_data (data, size);
|
||||
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
||||
outbuf = gst_rtp_buffer_new_copy_data (map.data, map.size);
|
||||
GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buffer);
|
||||
|
||||
/* Unmap and free input buffer */
|
||||
gst_buffer_unmap (buffer, data, size);
|
||||
gst_buffer_unmap (buffer, &map);
|
||||
gst_buffer_unref (buffer);
|
||||
|
||||
/* Remove marker from RTP buffer */
|
||||
|
|
|
|||
Loading…
Reference in a new issue