diff options
Diffstat (limited to 'ext/opus/gstopusdec.c')
-rw-r--r-- | ext/opus/gstopusdec.c | 819 |
1 files changed, 0 insertions, 819 deletions
diff --git a/ext/opus/gstopusdec.c b/ext/opus/gstopusdec.c deleted file mode 100644 index 1470ea321..000000000 --- a/ext/opus/gstopusdec.c +++ /dev/null @@ -1,819 +0,0 @@ -/* GStreamer - * Copyright (C) 2004 Wim Taymans <wim@fluendo.com> - * Copyright (C) 2006 Tim-Philipp Müller <tim centricular net> - * Copyright (C) 2008 Sebastian Dröge <sebastian.droege@collabora.co.uk> - * Copyright (C) 2011-2012 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, - * Boston, MA 02110-1301, USA. - */ - -/* - * Based on the speexdec element. - */ - -/** - * SECTION:element-opusdec - * @see_also: opusenc, oggdemux - * - * This element decodes a OPUS stream to raw integer audio. - * - * <refsect2> - * <title>Example pipelines</title> - * |[ - * gst-launch-1.0 -v filesrc location=opus.ogg ! oggdemux ! opusdec ! audioconvert ! audioresample ! alsasink - * ]| Decode an Ogg/Opus file. To create an Ogg/Opus file refer to the documentation of opusenc. - * </refsect2> - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include <math.h> -#include <string.h> -#include "gstopusheader.h" -#include "gstopuscommon.h" -#include "gstopusdec.h" -#include <gst/pbutils/pbutils.h> - -GST_DEBUG_CATEGORY_STATIC (opusdec_debug); -#define GST_CAT_DEFAULT opusdec_debug - -static GstStaticPadTemplate opus_dec_src_factory = -GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw, " - "format = (string) " GST_AUDIO_NE (S16) ", " - "layout = (string) interleaved, " - "rate = (int) { 48000, 24000, 16000, 12000, 8000 }, " - "channels = (int) [ 1, 8 ] ") - ); - -static GstStaticPadTemplate opus_dec_sink_factory = - GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-opus, " - "channel-mapping-family = (int) 0; " - "audio/x-opus, " - "channel-mapping-family = (int) [1, 255], " - "channels = (int) [1, 255], " - "stream-count = (int) [1, 255], " "coupled-count = (int) [0, 255]") - ); - -G_DEFINE_TYPE (GstOpusDec, gst_opus_dec, GST_TYPE_AUDIO_DECODER); - -#define DB_TO_LINEAR(x) pow (10., (x) / 20.) - -#define DEFAULT_USE_INBAND_FEC FALSE -#define DEFAULT_APPLY_GAIN TRUE - -enum -{ - PROP_0, - PROP_USE_INBAND_FEC, - PROP_APPLY_GAIN -}; - - -static GstFlowReturn gst_opus_dec_parse_header (GstOpusDec * dec, - GstBuffer * buf); -static gboolean gst_opus_dec_start (GstAudioDecoder * dec); -static gboolean gst_opus_dec_stop (GstAudioDecoder * dec); -static GstFlowReturn gst_opus_dec_handle_frame (GstAudioDecoder * dec, - GstBuffer * buffer); -static gboolean gst_opus_dec_set_format (GstAudioDecoder * bdec, - GstCaps * caps); -static void gst_opus_dec_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); -static void gst_opus_dec_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); - - -static void -gst_opus_dec_class_init (GstOpusDecClass * klass) -{ - GObjectClass *gobject_class; - GstAudioDecoderClass *adclass; - GstElementClass *element_class; - - gobject_class = (GObjectClass *) klass; - adclass = (GstAudioDecoderClass *) klass; - element_class = (GstElementClass *) klass; - - gobject_class->set_property = gst_opus_dec_set_property; - gobject_class->get_property = gst_opus_dec_get_property; - - adclass->start = GST_DEBUG_FUNCPTR (gst_opus_dec_start); - adclass->stop = GST_DEBUG_FUNCPTR (gst_opus_dec_stop); - adclass->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_dec_handle_frame); - adclass->set_format = GST_DEBUG_FUNCPTR (gst_opus_dec_set_format); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&opus_dec_src_factory)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&opus_dec_sink_factory)); - gst_element_class_set_static_metadata (element_class, "Opus audio decoder", - "Codec/Decoder/Audio", - "decode opus streams to audio", - "Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>"); - g_object_class_install_property (gobject_class, PROP_USE_INBAND_FEC, - g_param_spec_boolean ("use-inband-fec", "Use in-band FEC", - "Use forward error correction if available (needs PLC enabled)", - DEFAULT_USE_INBAND_FEC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - g_object_class_install_property (gobject_class, PROP_APPLY_GAIN, - g_param_spec_boolean ("apply-gain", "Apply gain", - "Apply gain if any is specified in the header", DEFAULT_APPLY_GAIN, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - GST_DEBUG_CATEGORY_INIT (opusdec_debug, "opusdec", 0, - "opus decoding element"); -} - -static void -gst_opus_dec_reset (GstOpusDec * dec) -{ - dec->packetno = 0; - if (dec->state) { - opus_multistream_decoder_destroy (dec->state); - dec->state = NULL; - } - - gst_buffer_replace (&dec->streamheader, NULL); - gst_buffer_replace (&dec->vorbiscomment, NULL); - gst_buffer_replace (&dec->last_buffer, NULL); - dec->primed = FALSE; - - dec->pre_skip = 0; - dec->r128_gain = 0; - dec->sample_rate = 0; - dec->n_channels = 0; - dec->leftover_plc_duration = 0; -} - -static void -gst_opus_dec_init (GstOpusDec * dec) -{ - dec->use_inband_fec = FALSE; - dec->apply_gain = DEFAULT_APPLY_GAIN; - - gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE); - gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST - (dec), TRUE); - GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec)); - - gst_opus_dec_reset (dec); -} - -static gboolean -gst_opus_dec_start (GstAudioDecoder * dec) -{ - GstOpusDec *odec = GST_OPUS_DEC (dec); - - gst_opus_dec_reset (odec); - - /* we know about concealment */ - gst_audio_decoder_set_plc_aware (dec, TRUE); - - if (odec->use_inband_fec) { - /* opusdec outputs samples directly from an input buffer, except if - * FEC is on, in which case it buffers one buffer in case one buffer - * goes missing. - */ - gst_audio_decoder_set_latency (dec, 120 * GST_MSECOND, 120 * GST_MSECOND); - } - - return TRUE; -} - -static gboolean -gst_opus_dec_stop (GstAudioDecoder * dec) -{ - GstOpusDec *odec = GST_OPUS_DEC (dec); - - gst_opus_dec_reset (odec); - - return TRUE; -} - -static double -gst_opus_dec_get_r128_gain (gint16 r128_gain) -{ - return r128_gain / (double) (1 << 8); -} - -static double -gst_opus_dec_get_r128_volume (gint16 r128_gain) -{ - return DB_TO_LINEAR (gst_opus_dec_get_r128_gain (r128_gain)); -} - -static void -gst_opus_dec_negotiate (GstOpusDec * dec, const GstAudioChannelPosition * pos) -{ - GstCaps *caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec)); - GstStructure *s; - GstAudioInfo info; - - if (caps) { - gint rate, channels; - - caps = gst_caps_truncate (caps); - caps = gst_caps_make_writable (caps); - s = gst_caps_get_structure (caps, 0); - - if (gst_structure_has_field (s, "rate")) - gst_structure_fixate_field_nearest_int (s, "rate", dec->sample_rate); - else - gst_structure_set (s, "rate", G_TYPE_INT, dec->sample_rate, NULL); - gst_structure_get_int (s, "rate", &rate); - dec->sample_rate = rate; - - if (gst_structure_has_field (s, "channels")) - gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels); - else - gst_structure_set (s, "channels", G_TYPE_INT, dec->n_channels, NULL); - gst_structure_get_int (s, "channels", &channels); - dec->n_channels = channels; - - gst_caps_unref (caps); - } - - if (dec->n_channels == 0) { - GST_DEBUG_OBJECT (dec, "Using a default of 2 channels"); - dec->n_channels = 2; - pos = NULL; - } - - if (dec->sample_rate == 0) { - GST_DEBUG_OBJECT (dec, "Using a default of 48kHz sample rate"); - dec->sample_rate = 48000; - } - - GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels, - dec->sample_rate); - - /* pass valid order to audio info */ - if (pos) { - memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels); - gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels); - } - - /* set up source format */ - gst_audio_info_init (&info); - gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, - dec->sample_rate, dec->n_channels, pos ? dec->opus_pos : NULL); - gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info); - - /* but we still need the opus order for later reordering */ - if (pos) { - memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels); - } else { - dec->opus_pos[0] = GST_AUDIO_CHANNEL_POSITION_INVALID; - } - - dec->info = info; -} - -static GstFlowReturn -gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf) -{ - GstAudioChannelPosition pos[64]; - const GstAudioChannelPosition *posn = NULL; - - if (!gst_opus_header_is_id_header (buf)) { - GST_ERROR_OBJECT (dec, "Header is not an Opus ID header"); - return GST_FLOW_ERROR; - } - - if (!gst_codec_utils_opus_parse_header (buf, - &dec->sample_rate, - &dec->n_channels, - &dec->channel_mapping_family, - &dec->n_streams, - &dec->n_stereo_streams, - dec->channel_mapping, &dec->pre_skip, &dec->r128_gain)) { - GST_ERROR_OBJECT (dec, "Failed to parse Opus ID header"); - return GST_FLOW_ERROR; - } - dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain); - - GST_INFO_OBJECT (dec, - "Found pre-skip of %u samples, R128 gain %d (volume %f)", - dec->pre_skip, dec->r128_gain, dec->r128_gain_volume); - - if (dec->channel_mapping_family == 1) { - GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping"); - switch (dec->n_channels) { - case 1: - case 2: - /* nothing */ - break; - case 3: - case 4: - case 5: - case 6: - case 7: - case 8: - posn = gst_opus_channel_positions[dec->n_channels - 1]; - break; - default:{ - gint i; - - GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE, - (NULL), ("Using NONE channel layout for more than 8 channels")); - - for (i = 0; i < dec->n_channels; i++) - pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE; - - posn = pos; - } - } - } else { - GST_INFO_OBJECT (dec, "Channel mapping family %d", - dec->channel_mapping_family); - } - - gst_opus_dec_negotiate (dec, posn); - - return GST_FLOW_OK; -} - - -static GstFlowReturn -gst_opus_dec_parse_comments (GstOpusDec * dec, GstBuffer * buf) -{ - return GST_FLOW_OK; -} - -static GstFlowReturn -opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer) -{ - GstFlowReturn res = GST_FLOW_OK; - gsize size; - guint8 *data; - GstBuffer *outbuf, *bufd; - gint16 *out_data; - int n, err; - int samples; - unsigned int packet_size; - GstBuffer *buf; - GstMapInfo map, omap; - GstAudioClippingMeta *cmeta = NULL; - - if (dec->state == NULL) { - /* If we did not get any headers, default to 2 channels */ - if (dec->n_channels == 0) { - GST_INFO_OBJECT (dec, "No header, assuming single stream"); - dec->n_channels = 2; - dec->sample_rate = 48000; - /* default stereo mapping */ - dec->channel_mapping_family = 0; - dec->channel_mapping[0] = 0; - dec->channel_mapping[1] = 1; - dec->n_streams = 1; - dec->n_stereo_streams = 1; - - gst_opus_dec_negotiate (dec, NULL); - } - - GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz", - dec->n_channels, dec->sample_rate); -#ifndef GST_DISABLE_GST_DEBUG - gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug, - "Mapping table", dec->n_channels, dec->channel_mapping); -#endif - - GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams, - dec->n_stereo_streams); - dec->state = - opus_multistream_decoder_create (dec->sample_rate, dec->n_channels, - dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err); - if (!dec->state || err != OPUS_OK) - goto creation_failed; - } - - if (buffer) { - GST_DEBUG_OBJECT (dec, "Received buffer of size %" G_GSIZE_FORMAT, - gst_buffer_get_size (buffer)); - } else { - GST_DEBUG_OBJECT (dec, "Received missing buffer"); - } - - /* if using in-band FEC, we introdude one extra frame's delay as we need - to potentially wait for next buffer to decode a missing buffer */ - if (dec->use_inband_fec && !dec->primed) { - GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out"); - gst_buffer_replace (&dec->last_buffer, buffer); - dec->primed = TRUE; - goto done; - } - - /* That's the buffer we'll be sending to the opus decoder. */ - buf = (dec->use_inband_fec - && gst_buffer_get_size (dec->last_buffer) > - 0) ? dec->last_buffer : buffer; - - /* That's the buffer we get duration from */ - bufd = dec->use_inband_fec ? dec->last_buffer : buffer; - - if (buf && gst_buffer_get_size (buf) > 0) { - gst_buffer_map (buf, &map, GST_MAP_READ); - data = map.data; - size = map.size; - GST_DEBUG_OBJECT (dec, "Using buffer of size %" G_GSIZE_FORMAT, size); - } else { - /* concealment data, pass NULL as the bits parameters */ - GST_DEBUG_OBJECT (dec, "Using NULL buffer"); - data = NULL; - size = 0; - } - - if (gst_buffer_get_size (bufd) == 0) { - GstClockTime const opus_plc_alignment = 2500 * GST_USECOND; - GstClockTime aligned_missing_duration; - GstClockTime missing_duration = GST_BUFFER_DURATION (bufd); - - GST_DEBUG_OBJECT (dec, - "missing buffer, doing PLC duration %" GST_TIME_FORMAT - " plus leftover %" GST_TIME_FORMAT, GST_TIME_ARGS (missing_duration), - GST_TIME_ARGS (dec->leftover_plc_duration)); - - /* add the leftover PLC duration to that of the buffer */ - missing_duration += dec->leftover_plc_duration; - - /* align the combined buffer and leftover PLC duration to multiples - * of 2.5ms, rounding to nearest, and store excess duration for later */ - aligned_missing_duration = - ((missing_duration + - opus_plc_alignment / 2) / opus_plc_alignment) * opus_plc_alignment; - dec->leftover_plc_duration = missing_duration - aligned_missing_duration; - - /* Opus' PLC cannot operate with less than 2.5ms; skip PLC - * and accumulate the missing duration in the leftover_plc_duration - * for the next PLC attempt */ - if (aligned_missing_duration < opus_plc_alignment) { - GST_DEBUG_OBJECT (dec, - "current duration %" GST_TIME_FORMAT - " of missing data not enough for PLC (minimum needed: %" - GST_TIME_FORMAT ") - skipping", GST_TIME_ARGS (missing_duration), - GST_TIME_ARGS (opus_plc_alignment)); - goto done; - } - - /* convert the duration (in nanoseconds) to sample count */ - samples = - gst_util_uint64_scale_int (aligned_missing_duration, dec->sample_rate, - GST_SECOND); - - GST_DEBUG_OBJECT (dec, - "calculated PLC frame length: %" GST_TIME_FORMAT - " num frame samples: %d new leftover: %" GST_TIME_FORMAT, - GST_TIME_ARGS (aligned_missing_duration), samples, - GST_TIME_ARGS (dec->leftover_plc_duration)); - } else { - /* use maximum size (120 ms) as the number of returned samples is - not constant over the stream. */ - samples = 120 * dec->sample_rate / 1000; - } - - packet_size = samples * dec->n_channels * 2; - - outbuf = - gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec), - packet_size); - if (!outbuf) { - goto buffer_failed; - } - - gst_buffer_map (outbuf, &omap, GST_MAP_WRITE); - out_data = (gint16 *) omap.data; - - if (dec->use_inband_fec) { - if (gst_buffer_get_size (dec->last_buffer) > 0) { - /* normal delayed decode */ - GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer"); - n = opus_multistream_decode (dec->state, data, size, out_data, samples, - 0); - } else { - /* FEC reconstruction decode */ - GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer"); - n = opus_multistream_decode (dec->state, data, size, out_data, samples, - 1); - } - } else { - /* normal decode */ - GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer"); - n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0); - } - gst_buffer_unmap (outbuf, &omap); - if (data != NULL) - gst_buffer_unmap (buf, &map); - - if (n < 0) { - GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL)); - gst_buffer_unref (outbuf); - return GST_FLOW_ERROR; - } - GST_DEBUG_OBJECT (dec, "decoded %d samples", n); - gst_buffer_set_size (outbuf, n * 2 * dec->n_channels); - - cmeta = gst_buffer_get_audio_clipping_meta (buf); - - g_assert (!cmeta || cmeta->format == GST_FORMAT_DEFAULT); - - /* Skip any samples that need skipping */ - if (cmeta && cmeta->start) { - guint pre_skip = cmeta->start; - guint scaled_pre_skip = pre_skip * dec->sample_rate / 48000; - guint skip = scaled_pre_skip > n ? n : scaled_pre_skip; - guint scaled_skip = skip * 48000 / dec->sample_rate; - - gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1); - - GST_INFO_OBJECT (dec, - "Skipping %u samples at the beginning (%u at 48000 Hz)", - skip, scaled_skip); - } - - if (cmeta && cmeta->end) { - guint post_skip = cmeta->end; - guint scaled_post_skip = post_skip * dec->sample_rate / 48000; - guint skip = scaled_post_skip > n ? n : scaled_post_skip; - guint scaled_skip = skip * 48000 / dec->sample_rate; - guint outsize = gst_buffer_get_size (outbuf); - guint skip_bytes = skip * 2 * dec->n_channels; - - if (outsize > skip_bytes) - outsize -= skip_bytes; - else - outsize = 0; - - gst_buffer_resize (outbuf, 0, outsize); - - GST_INFO_OBJECT (dec, - "Skipping %u samples at the end (%u at 48000 Hz)", skip, scaled_skip); - } - - if (gst_buffer_get_size (outbuf) == 0) { - gst_buffer_unref (outbuf); - outbuf = NULL; - } else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) { - gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16, - dec->n_channels, dec->opus_pos, dec->info.position); - } - - /* Apply gain */ - /* Would be better off leaving this to a volume element, as this is - a naive conversion that does too many int/float conversions. - However, we don't have control over the pipeline... - So make it optional if the user program wants to use a volume, - but do it by default so the correct volume goes out by default */ - if (dec->apply_gain && outbuf && dec->r128_gain) { - gsize rsize; - unsigned int i, nsamples; - double volume = dec->r128_gain_volume; - gint16 *samples; - - gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE); - samples = (gint16 *) omap.data; - rsize = omap.size; - GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume); - nsamples = rsize / 2; - for (i = 0; i < nsamples; ++i) { - int sample = (int) (samples[i] * volume + 0.5); - samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample; - } - gst_buffer_unmap (outbuf, &omap); - } - - if (dec->use_inband_fec) { - gst_buffer_replace (&dec->last_buffer, buffer); - } - - res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1); - - if (res != GST_FLOW_OK) - GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res)); - -done: - return res; - -creation_failed: - GST_ERROR_OBJECT (dec, "Failed to create Opus decoder: %d", err); - return GST_FLOW_ERROR; - -buffer_failed: - GST_ERROR_OBJECT (dec, "Failed to create %u byte buffer", packet_size); - return GST_FLOW_ERROR; -} - -static gboolean -gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps) -{ - GstOpusDec *dec = GST_OPUS_DEC (bdec); - gboolean ret = TRUE; - GstStructure *s; - const GValue *streamheader; - GstCaps *old_caps; - - GST_DEBUG_OBJECT (dec, "set_format: %" GST_PTR_FORMAT, caps); - - if ((old_caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (bdec)))) { - if (gst_caps_is_equal (caps, old_caps)) { - gst_caps_unref (old_caps); - GST_DEBUG_OBJECT (dec, "caps didn't change"); - goto done; - } - - GST_DEBUG_OBJECT (dec, "caps have changed, resetting decoder"); - gst_opus_dec_reset (dec); - gst_caps_unref (old_caps); - } - - s = gst_caps_get_structure (caps, 0); - if ((streamheader = gst_structure_get_value (s, "streamheader")) && - G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) && - gst_value_array_get_size (streamheader) >= 2) { - const GValue *header, *vorbiscomment; - GstBuffer *buf; - GstFlowReturn res = GST_FLOW_OK; - - header = gst_value_array_get_value (streamheader, 0); - if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) { - buf = gst_value_get_buffer (header); - res = gst_opus_dec_parse_header (dec, buf); - if (res != GST_FLOW_OK) { - ret = FALSE; - goto done; - } - gst_buffer_replace (&dec->streamheader, buf); - } - - vorbiscomment = gst_value_array_get_value (streamheader, 1); - if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) { - buf = gst_value_get_buffer (vorbiscomment); - res = gst_opus_dec_parse_comments (dec, buf); - if (res != GST_FLOW_OK) { - ret = FALSE; - goto done; - } - gst_buffer_replace (&dec->vorbiscomment, buf); - } - } else { - const GstAudioChannelPosition *posn = NULL; - - if (!gst_codec_utils_opus_parse_caps (caps, &dec->sample_rate, - &dec->n_channels, &dec->channel_mapping_family, &dec->n_streams, - &dec->n_stereo_streams, dec->channel_mapping)) { - ret = FALSE; - goto done; - } - - if (dec->channel_mapping_family == 1 && dec->n_channels <= 8) - posn = gst_opus_channel_positions[dec->n_channels - 1]; - - gst_opus_dec_negotiate (dec, posn); - } - -done: - return ret; -} - -static gboolean -memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2) -{ - gsize size1, size2; - gboolean res; - GstMapInfo map; - - size1 = gst_buffer_get_size (buf1); - size2 = gst_buffer_get_size (buf2); - - if (size1 != size2) - return FALSE; - - gst_buffer_map (buf1, &map, GST_MAP_READ); - res = gst_buffer_memcmp (buf2, 0, map.data, map.size) == 0; - gst_buffer_unmap (buf1, &map); - - return res; -} - -static GstFlowReturn -gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf) -{ - GstFlowReturn res; - GstOpusDec *dec; - - /* no fancy draining */ - if (G_UNLIKELY (!buf)) - return GST_FLOW_OK; - - dec = GST_OPUS_DEC (adec); - GST_LOG_OBJECT (dec, - "Got buffer ts %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), - GST_TIME_ARGS (GST_BUFFER_DURATION (buf))); - - /* If we have the streamheader and vorbiscomment from the caps already - * ignore them here */ - if (dec->streamheader && dec->vorbiscomment) { - if (memcmp_buffers (dec->streamheader, buf)) { - GST_DEBUG_OBJECT (dec, "found streamheader"); - gst_audio_decoder_finish_frame (adec, NULL, 1); - res = GST_FLOW_OK; - } else if (memcmp_buffers (dec->vorbiscomment, buf)) { - GST_DEBUG_OBJECT (dec, "found vorbiscomments"); - gst_audio_decoder_finish_frame (adec, NULL, 1); - res = GST_FLOW_OK; - } else { - res = opus_dec_chain_parse_data (dec, buf); - } - } else { - /* Otherwise fall back to packet counting and assume that the - * first two packets might be the headers, checking magic. */ - switch (dec->packetno) { - case 0: - if (gst_opus_header_is_header (buf, "OpusHead", 8)) { - GST_DEBUG_OBJECT (dec, "found streamheader"); - res = gst_opus_dec_parse_header (dec, buf); - gst_audio_decoder_finish_frame (adec, NULL, 1); - } else { - res = opus_dec_chain_parse_data (dec, buf); - } - break; - case 1: - if (gst_opus_header_is_header (buf, "OpusTags", 8)) { - GST_DEBUG_OBJECT (dec, "counted vorbiscomments"); - res = gst_opus_dec_parse_comments (dec, buf); - gst_audio_decoder_finish_frame (adec, NULL, 1); - } else { - res = opus_dec_chain_parse_data (dec, buf); - } - break; - default: - { - res = opus_dec_chain_parse_data (dec, buf); - break; - } - } - } - - dec->packetno++; - - return res; -} - -static void -gst_opus_dec_get_property (GObject * object, guint prop_id, GValue * value, - GParamSpec * pspec) -{ - GstOpusDec *dec = GST_OPUS_DEC (object); - - switch (prop_id) { - case PROP_USE_INBAND_FEC: - g_value_set_boolean (value, dec->use_inband_fec); - break; - case PROP_APPLY_GAIN: - g_value_set_boolean (value, dec->apply_gain); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_opus_dec_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstOpusDec *dec = GST_OPUS_DEC (object); - - switch (prop_id) { - case PROP_USE_INBAND_FEC: - dec->use_inband_fec = g_value_get_boolean (value); - break; - case PROP_APPLY_GAIN: - dec->apply_gain = g_value_get_boolean (value); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} |