/* GStreamer * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> * Copyright (C) <2006,2011> Tim-Philipp Müller <tim centricular net> * Copyright (C) <2006> Jan Schmidt <thaytan at mad scientist com> * * 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. */ /** * SECTION:element-flacdec * @see_also: #GstFlacEnc * * flacdec decodes FLAC streams. * <ulink url="http://flac.sourceforge.net/">FLAC</ulink> * is a Free Lossless Audio Codec. * * <refsect2> * <title>Example launch line</title> * |[ * gst-launch-1.0 filesrc location=media/small/dark.441-16-s.flac ! flacparse ! flacdec ! audioconvert ! audioresample ! autoaudiosink * ]| * |[ * gst-launch-1.0 souphttpsrc location=http://gstreamer.freedesktop.org/media/small/dark.441-16-s.flac ! flacparse ! flacdec ! audioconvert ! audioresample ! queue min-threshold-buffers=10 ! autoaudiosink * ]| * </refsect2> */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include <string.h> #include "gstflacdec.h" #include <gst/gst-i18n-plugin.h> #include <gst/tag/tag.h> /* Taken from http://flac.sourceforge.net/format.html#frame_header */ static const GstAudioChannelPosition channel_positions[8][8] = { {GST_AUDIO_CHANNEL_POSITION_MONO}, {GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE1, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, /* FIXME: 7/8 channel layouts are not defined in the FLAC specs */ { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE1, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE1, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT} }; GST_DEBUG_CATEGORY_STATIC (flacdec_debug); #define GST_CAT_DEFAULT flacdec_debug static FLAC__StreamDecoderReadStatus gst_flac_dec_read_stream (const FLAC__StreamDecoder * decoder, FLAC__byte buffer[], size_t * bytes, void *client_data); static FLAC__StreamDecoderWriteStatus gst_flac_dec_write_stream (const FLAC__StreamDecoder * decoder, const FLAC__Frame * frame, const FLAC__int32 * const buffer[], void *client_data); static gboolean gst_flac_dec_handle_decoder_error (GstFlacDec * dec, gboolean msg); static void gst_flac_dec_metadata_cb (const FLAC__StreamDecoder * decoder, const FLAC__StreamMetadata * metadata, void *client_data); static void gst_flac_dec_error_cb (const FLAC__StreamDecoder * decoder, FLAC__StreamDecoderErrorStatus status, void *client_data); static void gst_flac_dec_flush (GstAudioDecoder * audio_dec, gboolean hard); static gboolean gst_flac_dec_set_format (GstAudioDecoder * dec, GstCaps * caps); static gboolean gst_flac_dec_start (GstAudioDecoder * dec); static gboolean gst_flac_dec_stop (GstAudioDecoder * dec); static GstFlowReturn gst_flac_dec_handle_frame (GstAudioDecoder * audio_dec, GstBuffer * buf); G_DEFINE_TYPE (GstFlacDec, gst_flac_dec, GST_TYPE_AUDIO_DECODER); #if G_BYTE_ORDER == G_LITTLE_ENDIAN #define FORMATS "{ S8, S16LE, S24_32LE, S32LE } " #else #define FORMATS "{ S8, S16BE, S24_32BE, S32BE } " #endif #define GST_FLAC_DEC_SRC_CAPS \ "audio/x-raw, " \ "format = (string) " FORMATS ", " \ "layout = (string) interleaved, " \ "rate = (int) [ 1, 655350 ], " \ "channels = (int) [ 1, 8 ]" #define GST_FLAC_DEC_SINK_CAPS \ "audio/x-flac, " \ "framed = (boolean) true, " \ "rate = (int) [ 1, 655350 ], " \ "channels = (int) [ 1, 8 ]" static GstStaticPadTemplate flac_dec_src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_FLAC_DEC_SRC_CAPS)); static GstStaticPadTemplate flac_dec_sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_FLAC_DEC_SINK_CAPS)); static void gst_flac_dec_class_init (GstFlacDecClass * klass) { GstAudioDecoderClass *audiodecoder_class; GstElementClass *gstelement_class; audiodecoder_class = (GstAudioDecoderClass *) klass; gstelement_class = (GstElementClass *) klass; GST_DEBUG_CATEGORY_INIT (flacdec_debug, "flacdec", 0, "flac decoder"); audiodecoder_class->stop = GST_DEBUG_FUNCPTR (gst_flac_dec_stop); audiodecoder_class->start = GST_DEBUG_FUNCPTR (gst_flac_dec_start); audiodecoder_class->flush = GST_DEBUG_FUNCPTR (gst_flac_dec_flush); audiodecoder_class->set_format = GST_DEBUG_FUNCPTR (gst_flac_dec_set_format); audiodecoder_class->handle_frame = GST_DEBUG_FUNCPTR (gst_flac_dec_handle_frame); gst_element_class_add_static_pad_template (gstelement_class, &flac_dec_src_factory); gst_element_class_add_static_pad_template (gstelement_class, &flac_dec_sink_factory); gst_element_class_set_static_metadata (gstelement_class, "FLAC audio decoder", "Codec/Decoder/Audio", "Decodes FLAC lossless audio streams", "Tim-Philipp Müller <tim@centricular.net>, " "Wim Taymans <wim.taymans@gmail.com>"); } static void gst_flac_dec_init (GstFlacDec * flacdec) { flacdec->do_resync = FALSE; gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (flacdec), TRUE); gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST (flacdec), TRUE); GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (flacdec)); } static gboolean gst_flac_dec_start (GstAudioDecoder * audio_dec) { FLAC__StreamDecoderInitStatus s; GstFlacDec *dec; dec = GST_FLAC_DEC (audio_dec); dec->adapter = gst_adapter_new (); dec->decoder = FLAC__stream_decoder_new (); gst_audio_info_init (&dec->info); dec->depth = 0; /* no point calculating MD5 since it's never checked here */ FLAC__stream_decoder_set_md5_checking (dec->decoder, false); GST_DEBUG_OBJECT (dec, "initializing decoder"); s = FLAC__stream_decoder_init_stream (dec->decoder, gst_flac_dec_read_stream, NULL, NULL, NULL, NULL, gst_flac_dec_write_stream, gst_flac_dec_metadata_cb, gst_flac_dec_error_cb, dec); if (s != FLAC__STREAM_DECODER_INIT_STATUS_OK) { GST_ELEMENT_ERROR (GST_ELEMENT (dec), LIBRARY, INIT, (NULL), (NULL)); return FALSE; } dec->got_headers = FALSE; return TRUE; } static gboolean gst_flac_dec_stop (GstAudioDecoder * dec) { GstFlacDec *flacdec = GST_FLAC_DEC (dec); if (flacdec->decoder) { FLAC__stream_decoder_delete (flacdec->decoder); flacdec->decoder = NULL; } if (flacdec->adapter) { gst_adapter_clear (flacdec->adapter); g_object_unref (flacdec->adapter); flacdec->adapter = NULL; } return TRUE; } static gboolean gst_flac_dec_set_format (GstAudioDecoder * dec, GstCaps * caps) { const GValue *headers; GstFlacDec *flacdec; GstStructure *s; guint i, num; flacdec = GST_FLAC_DEC (dec); GST_LOG_OBJECT (dec, "sink caps: %" GST_PTR_FORMAT, caps); s = gst_caps_get_structure (caps, 0); headers = gst_structure_get_value (s, "streamheader"); if (headers == NULL || !GST_VALUE_HOLDS_ARRAY (headers)) { GST_WARNING_OBJECT (dec, "no 'streamheader' field in input caps, try " "adding a flacparse element upstream"); return FALSE; } if (gst_adapter_available (flacdec->adapter) > 0) { GST_WARNING_OBJECT (dec, "unexpected data left in adapter"); gst_adapter_clear (flacdec->adapter); } num = gst_value_array_get_size (headers); for (i = 0; i < num; ++i) { const GValue *header_val; GstBuffer *header_buf; header_val = gst_value_array_get_value (headers, i); if (header_val == NULL || !GST_VALUE_HOLDS_BUFFER (header_val)) return FALSE; header_buf = g_value_dup_boxed (header_val); GST_INFO_OBJECT (dec, "pushing header buffer of %" G_GSIZE_FORMAT " bytes " "into adapter", gst_buffer_get_size (header_buf)); gst_adapter_push (flacdec->adapter, header_buf); } GST_DEBUG_OBJECT (dec, "Processing headers and metadata"); if (!FLAC__stream_decoder_process_until_end_of_metadata (flacdec->decoder)) { GST_WARNING_OBJECT (dec, "process_until_end_of_metadata failed"); if (FLAC__stream_decoder_get_state (flacdec->decoder) == FLAC__STREAM_DECODER_ABORTED) { GST_WARNING_OBJECT (flacdec, "Read callback caused internal abort"); /* allow recovery */ gst_adapter_clear (flacdec->adapter); FLAC__stream_decoder_flush (flacdec->decoder); gst_flac_dec_handle_decoder_error (flacdec, TRUE); } } GST_INFO_OBJECT (dec, "headers and metadata are now processed"); return TRUE; } /* CRC-8, poly = x^8 + x^2 + x^1 + x^0, init = 0 */ static const guint8 crc8_table[256] = { 0x00, 0x07, 0x0E, 0x09, 0x1C, 0x1B, 0x12, 0x15, 0x38, 0x3F, 0x36, 0x31, 0x24, 0x23, 0x2A, 0x2D, 0x70, 0x77, 0x7E, 0x79, 0x6C, 0x6B, 0x62, 0x65, 0x48, 0x4F, 0x46, 0x41, 0x54, 0x53, 0x5A, 0x5D, 0xE0, 0xE7, 0xEE, 0xE9, 0xFC, 0xFB, 0xF2, 0xF5, 0xD8, 0xDF, 0xD6, 0xD1, 0xC4, 0xC3, 0xCA, 0xCD, 0x90, 0x97, 0x9E, 0x99, 0x8C, 0x8B, 0x82, 0x85, 0xA8, 0xAF, 0xA6, 0xA1, 0xB4, 0xB3, 0xBA, 0xBD, 0xC7, 0xC0, 0xC9, 0xCE, 0xDB, 0xDC, 0xD5, 0xD2, 0xFF, 0xF8, 0xF1, 0xF6, 0xE3, 0xE4, 0xED, 0xEA, 0xB7, 0xB0, 0xB9, 0xBE, 0xAB, 0xAC, 0xA5, 0xA2, 0x8F, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9D, 0x9A, 0x27, 0x20, 0x29, 0x2E, 0x3B, 0x3C, 0x35, 0x32, 0x1F, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0D, 0x0A, 0x57, 0x50, 0x59, 0x5E, 0x4B, 0x4C, 0x45, 0x42, 0x6F, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7D, 0x7A, 0x89, 0x8E, 0x87, 0x80, 0x95, 0x92, 0x9B, 0x9C, 0xB1, 0xB6, 0xBF, 0xB8, 0xAD, 0xAA, 0xA3, 0xA4, 0xF9, 0xFE, 0xF7, 0xF0, 0xE5, 0xE2, 0xEB, 0xEC, 0xC1, 0xC6, 0xCF, 0xC8, 0xDD, 0xDA, 0xD3, 0xD4, 0x69, 0x6E, 0x67, 0x60, 0x75, 0x72, 0x7B, 0x7C, 0x51, 0x56, 0x5F, 0x58, 0x4D, 0x4A, 0x43, 0x44, 0x19, 0x1E, 0x17, 0x10, 0x05, 0x02, 0x0B, 0x0C, 0x21, 0x26, 0x2F, 0x28, 0x3D, 0x3A, 0x33, 0x34, 0x4E, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5C, 0x5B, 0x76, 0x71, 0x78, 0x7F, 0x6A, 0x6D, 0x64, 0x63, 0x3E, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2C, 0x2B, 0x06, 0x01, 0x08, 0x0F, 0x1A, 0x1D, 0x14, 0x13, 0xAE, 0xA9, 0xA0, 0xA7, 0xB2, 0xB5, 0xBC, 0xBB, 0x96, 0x91, 0x98, 0x9F, 0x8A, 0x8D, 0x84, 0x83, 0xDE, 0xD9, 0xD0, 0xD7, 0xC2, 0xC5, 0xCC, 0xCB, 0xE6, 0xE1, 0xE8, 0xEF, 0xFA, 0xFD, 0xF4, 0xF3 }; static guint8 gst_flac_calculate_crc8 (const guint8 * data, guint length) { guint8 crc = 0; while (length--) { crc = crc8_table[crc ^ *data]; ++data; } return crc; } /* FIXME: for our purposes it's probably enough to just check for the sync * marker - we just want to know if it's a header frame or not */ static gboolean gst_flac_dec_scan_got_frame (GstFlacDec * flacdec, const guint8 * data, guint size) { guint headerlen; guint sr_from_end = 0; /* can be 0, 8 or 16 */ guint bs_from_end = 0; /* can be 0, 8 or 16 */ guint32 val = 0; guint8 bs, sr, ca, ss, pb; gboolean vbs; if (size < 10) return FALSE; /* sync */ if (data[0] != 0xFF || (data[1] & 0xFC) != 0xF8) return FALSE; vbs = ! !(data[1] & 1); /* variable blocksize */ bs = (data[2] & 0xF0) >> 4; /* blocksize marker */ sr = (data[2] & 0x0F); /* samplerate marker */ ca = (data[3] & 0xF0) >> 4; /* channel assignment */ ss = (data[3] & 0x0F) >> 1; /* sample size marker */ pb = (data[3] & 0x01); /* padding bit */ GST_LOG_OBJECT (flacdec, "got sync, vbs=%d,bs=%x,sr=%x,ca=%x,ss=%x,pb=%x", vbs, bs, sr, ca, ss, pb); if (bs == 0 || sr == 0x0F || ca >= 0x0B || ss == 0x03 || ss == 0x07) { return FALSE; } /* read block size from end of header? */ if (bs == 6) bs_from_end = 8; else if (bs == 7) bs_from_end = 16; /* read sample rate from end of header? */ if (sr == 0x0C) sr_from_end = 8; else if (sr == 0x0D || sr == 0x0E) sr_from_end = 16; val = data[4]; /* This is slightly faster than a loop */ if (!(val & 0x80)) { val = 0; } else if ((val & 0xc0) && !(val & 0x20)) { val = 1; } else if ((val & 0xe0) && !(val & 0x10)) { val = 2; } else if ((val & 0xf0) && !(val & 0x08)) { val = 3; } else if ((val & 0xf8) && !(val & 0x04)) { val = 4; } else if ((val & 0xfc) && !(val & 0x02)) { val = 5; } else if ((val & 0xfe) && !(val & 0x01)) { val = 6; } else { GST_LOG_OBJECT (flacdec, "failed to read sample/frame"); return FALSE; } val++; headerlen = 4 + val + (bs_from_end / 8) + (sr_from_end / 8); if (gst_flac_calculate_crc8 (data, headerlen) != data[headerlen]) { GST_LOG_OBJECT (flacdec, "invalid checksum"); return FALSE; } return TRUE; } static gboolean gst_flac_dec_handle_decoder_error (GstFlacDec * dec, gboolean msg) { gboolean ret; dec->error_count++; if (dec->error_count > 10) { if (msg) GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), (NULL)); dec->last_flow = GST_FLOW_ERROR; ret = TRUE; } else { GST_DEBUG_OBJECT (dec, "ignoring error for now at count %d", dec->error_count); ret = FALSE; } return ret; } static void gst_flac_dec_metadata_cb (const FLAC__StreamDecoder * decoder, const FLAC__StreamMetadata * metadata, void *client_data) { GstFlacDec *flacdec = GST_FLAC_DEC (client_data); GstAudioChannelPosition position[8]; GST_LOG_OBJECT (flacdec, "metadata type: %d", metadata->type); switch (metadata->type) { case FLAC__METADATA_TYPE_STREAMINFO:{ gint64 samples; guint depth, width, gdepth, channels; samples = metadata->data.stream_info.total_samples; flacdec->min_blocksize = metadata->data.stream_info.min_blocksize; flacdec->max_blocksize = metadata->data.stream_info.max_blocksize; flacdec->depth = depth = metadata->data.stream_info.bits_per_sample; if (depth < 9) { gdepth = width = 8; } else if (depth < 17) { gdepth = width = 16; } else if (depth < 25) { gdepth = 24; width = 32; } else { gdepth = width = 32; } channels = metadata->data.stream_info.channels; memcpy (position, channel_positions[channels - 1], sizeof (position)); gst_audio_channel_positions_to_valid_order (position, channels); /* Note: we create the inverse reordering map here */ gst_audio_get_channel_reorder_map (channels, position, channel_positions[channels - 1], flacdec->channel_reorder_map); gst_audio_info_set_format (&flacdec->info, gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, width, gdepth), metadata->data.stream_info.sample_rate, metadata->data.stream_info.channels, position); GST_DEBUG_OBJECT (flacdec, "blocksize: min=%u, max=%u", flacdec->min_blocksize, flacdec->max_blocksize); GST_DEBUG_OBJECT (flacdec, "sample rate: %u, channels: %u", flacdec->info.rate, flacdec->info.channels); GST_DEBUG_OBJECT (flacdec, "depth: %u, width: %u", flacdec->depth, flacdec->info.finfo->width); GST_DEBUG_OBJECT (flacdec, "total samples = %" G_GINT64_FORMAT, samples); break; } default: break; } } static void gst_flac_dec_error_cb (const FLAC__StreamDecoder * d, FLAC__StreamDecoderErrorStatus status, void *client_data) { const gchar *error; GstFlacDec *dec; dec = GST_FLAC_DEC (client_data); switch (status) { case FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC: dec->do_resync = TRUE; return; case FLAC__STREAM_DECODER_ERROR_STATUS_BAD_HEADER: error = "bad header"; break; case FLAC__STREAM_DECODER_ERROR_STATUS_FRAME_CRC_MISMATCH: error = "CRC mismatch"; break; default: error = "unknown error"; break; } if (gst_flac_dec_handle_decoder_error (dec, FALSE)) GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("%s (%d)", error, status)); } static FLAC__StreamDecoderReadStatus gst_flac_dec_read_stream (const FLAC__StreamDecoder * decoder, FLAC__byte buffer[], size_t * bytes, void *client_data) { GstFlacDec *dec = GST_FLAC_DEC (client_data); guint len; len = MIN (gst_adapter_available (dec->adapter), *bytes); if (len == 0) { GST_LOG_OBJECT (dec, "0 bytes available at the moment"); return FLAC__STREAM_DECODER_READ_STATUS_ABORT; } GST_LOG_OBJECT (dec, "feeding %u bytes to decoder " "(available=%" G_GSIZE_FORMAT ", bytes=%u)", len, gst_adapter_available (dec->adapter), (guint) * bytes); gst_adapter_copy (dec->adapter, buffer, 0, len); *bytes = len; gst_adapter_flush (dec->adapter, len); return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE; } static FLAC__StreamDecoderWriteStatus gst_flac_dec_write (GstFlacDec * flacdec, const FLAC__Frame * frame, const FLAC__int32 * const buffer[]) { GstFlowReturn ret = GST_FLOW_OK; GstBuffer *outbuf; guint depth = frame->header.bits_per_sample; guint width, gdepth; guint sample_rate = frame->header.sample_rate; guint channels = frame->header.channels; guint samples = frame->header.blocksize; guint j, i; GstMapInfo map; gboolean caps_changed; GstAudioChannelPosition chanpos[8]; GST_LOG_OBJECT (flacdec, "samples in frame header: %d", samples); if (depth == 0) { if (flacdec->depth < 4 || flacdec->depth > 32) { GST_ERROR_OBJECT (flacdec, "unsupported depth %d from STREAMINFO", flacdec->depth); ret = GST_FLOW_ERROR; goto done; } depth = flacdec->depth; } switch (depth) { case 8: gdepth = width = 8; break; case 12: case 16: gdepth = width = 16; break; case 20: case 24: gdepth = 24; width = 32; break; case 32: gdepth = width = 32; break; default: GST_ERROR_OBJECT (flacdec, "unsupported depth %d", depth); ret = GST_FLOW_ERROR; goto done; } if (sample_rate == 0) { if (flacdec->info.rate != 0) { sample_rate = flacdec->info.rate; } else { GST_ERROR_OBJECT (flacdec, "unknown sample rate"); ret = GST_FLOW_ERROR; goto done; } } caps_changed = (sample_rate != GST_AUDIO_INFO_RATE (&flacdec->info)) || (width != GST_AUDIO_INFO_WIDTH (&flacdec->info)) || (gdepth != GST_AUDIO_INFO_DEPTH (&flacdec->info)) || (channels != GST_AUDIO_INFO_CHANNELS (&flacdec->info)); if (caps_changed || !gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (flacdec))) { GST_DEBUG_OBJECT (flacdec, "Negotiating %d Hz @ %d channels", sample_rate, channels); memcpy (chanpos, channel_positions[flacdec->info.channels - 1], sizeof (chanpos)); gst_audio_channel_positions_to_valid_order (chanpos, flacdec->info.channels); gst_audio_info_set_format (&flacdec->info, gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, width, gdepth), sample_rate, channels, chanpos); /* Note: we create the inverse reordering map here */ gst_audio_get_channel_reorder_map (flacdec->info.channels, flacdec->info.position, channel_positions[flacdec->info.channels - 1], flacdec->channel_reorder_map); flacdec->depth = depth; gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (flacdec), &flacdec->info); } outbuf = gst_buffer_new_allocate (NULL, samples * channels * (width / 8), NULL); gst_buffer_map (outbuf, &map, GST_MAP_WRITE); if (width == 8) { gint8 *outbuffer = (gint8 *) map.data; gint *reorder_map = flacdec->channel_reorder_map; g_assert (gdepth == 8 && depth == 8); for (i = 0; i < samples; i++) { for (j = 0; j < channels; j++) { *outbuffer++ = (gint8) buffer[reorder_map[j]][i]; } } } else if (width == 16) { gint16 *outbuffer = (gint16 *) map.data; gint *reorder_map = flacdec->channel_reorder_map; if (gdepth != depth) { for (i = 0; i < samples; i++) { for (j = 0; j < channels; j++) { *outbuffer++ = (gint16) (buffer[reorder_map[j]][i] << (gdepth - depth)); } } } else { for (i = 0; i < samples; i++) { for (j = 0; j < channels; j++) { *outbuffer++ = (gint16) buffer[reorder_map[j]][i]; } } } } else if (width == 32) { gint32 *outbuffer = (gint32 *) map.data; gint *reorder_map = flacdec->channel_reorder_map; if (gdepth != depth) { for (i = 0; i < samples; i++) { for (j = 0; j < channels; j++) { *outbuffer++ = (gint32) (buffer[reorder_map[j]][i] << (gdepth - depth)); } } } else { for (i = 0; i < samples; i++) { for (j = 0; j < channels; j++) { *outbuffer++ = (gint32) buffer[reorder_map[j]][i]; } } } } else { g_assert_not_reached (); } gst_buffer_unmap (outbuf, &map); GST_DEBUG_OBJECT (flacdec, "pushing %d samples", samples); if (flacdec->error_count) flacdec->error_count--; ret = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (flacdec), outbuf, 1); if (G_UNLIKELY (ret != GST_FLOW_OK)) { GST_DEBUG_OBJECT (flacdec, "finish_frame flow %s", gst_flow_get_name (ret)); } done: /* we act on the flow return value later in the handle_frame function, as we * don't want to mess up the internal decoder state by returning ABORT when * the error is in fact non-fatal (like a pad in flushing mode) and we want * to continue later. So just pretend everything's dandy and act later. */ flacdec->last_flow = ret; return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; } static FLAC__StreamDecoderWriteStatus gst_flac_dec_write_stream (const FLAC__StreamDecoder * decoder, const FLAC__Frame * frame, const FLAC__int32 * const buffer[], void *client_data) { return gst_flac_dec_write (GST_FLAC_DEC (client_data), frame, buffer); } static void gst_flac_dec_flush (GstAudioDecoder * audio_dec, gboolean hard) { GstFlacDec *dec = GST_FLAC_DEC (audio_dec); if (!hard) { guint available = gst_adapter_available (dec->adapter); if (available > 0) { GST_INFO_OBJECT (dec, "draining, %u bytes left in adapter", available); FLAC__stream_decoder_process_until_end_of_stream (dec->decoder); } } dec->do_resync = FALSE; FLAC__stream_decoder_flush (dec->decoder); gst_adapter_clear (dec->adapter); } static GstFlowReturn gst_flac_dec_handle_frame (GstAudioDecoder * audio_dec, GstBuffer * buf) { GstFlacDec *dec; dec = GST_FLAC_DEC (audio_dec); /* drain remaining data? */ if (G_UNLIKELY (buf == NULL)) { gst_flac_dec_flush (audio_dec, FALSE); return GST_FLOW_OK; } if (dec->do_resync) { GST_WARNING_OBJECT (dec, "Lost sync, flushing decoder"); FLAC__stream_decoder_flush (dec->decoder); dec->do_resync = FALSE; } GST_LOG_OBJECT (dec, "frame: ts %" GST_TIME_FORMAT ", flags 0x%04x, " "%" G_GSIZE_FORMAT " bytes", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), GST_BUFFER_FLAGS (buf), gst_buffer_get_size (buf)); /* drop any in-stream headers, we've processed those in set_format already */ if (G_UNLIKELY (!dec->got_headers)) { gboolean got_audio_frame; GstMapInfo map; /* check if this is a flac audio frame (rather than a header or junk) */ gst_buffer_map (buf, &map, GST_MAP_READ); got_audio_frame = gst_flac_dec_scan_got_frame (dec, map.data, map.size); gst_buffer_unmap (buf, &map); if (!got_audio_frame) { GST_INFO_OBJECT (dec, "dropping in-stream header, %" G_GSIZE_FORMAT " " "bytes", map.size); gst_audio_decoder_finish_frame (audio_dec, NULL, 1); return GST_FLOW_OK; } GST_INFO_OBJECT (dec, "first audio frame, got all in-stream headers now"); dec->got_headers = TRUE; } gst_adapter_push (dec->adapter, gst_buffer_ref (buf)); buf = NULL; dec->last_flow = GST_FLOW_OK; /* framed - there should always be enough data to decode something */ GST_LOG_OBJECT (dec, "%" G_GSIZE_FORMAT " bytes available", gst_adapter_available (dec->adapter)); if (!FLAC__stream_decoder_process_single (dec->decoder)) { GST_INFO_OBJECT (dec, "process_single failed"); if (FLAC__stream_decoder_get_state (dec->decoder) == FLAC__STREAM_DECODER_ABORTED) { GST_WARNING_OBJECT (dec, "Read callback caused internal abort"); /* allow recovery */ gst_adapter_clear (dec->adapter); FLAC__stream_decoder_flush (dec->decoder); gst_flac_dec_handle_decoder_error (dec, TRUE); } } return dec->last_flow; }