From 1c37934861f2dd7c3ca38fe770734e2b3b30cfdb Mon Sep 17 00:00:00 2001 From: "Ronald S. Bultje" Date: Wed, 9 Mar 2005 12:06:56 +0000 Subject: configure.ac: Fix FAAD detection problems against FAAD-CVS. Original commit message from CVS: * configure.ac: Fix FAAD detection problems against FAAD-CVS. * ext/faad/gstfaad.c: (gst_faad_class_init), (gst_faad_chanpos_to_gst), (gst_faad_srcconnect), (gst_faad_sync), (gst_faad_chain): Fix FAAD channel positions for mono/stereo against FAAD CVS. Implement raw stream sync support for AAC+ radio support. Embed info structure in our function to prevent unneeded excessive allocations. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_populate), (gst_ogg_demux_push): Only set first/last positions when we search for them. Fixes invalid length reporting for some video files. * gst/playback/gstdecodebin.c: (remove_element_chain): Always remove only our own kids. * gst/qtdemux/qtdemux.c: (qtdemux_parse), (qtdemux_parse_trak): Fix ESDS atom finding bug. * gst/typefind/gsttypefindfunctions.c: (aac_type_find): Implement frame-finding (similar to MP3) to support AAC+ radio. --- ext/faad/gstfaad.c | 158 ++++++++++++++++++++++++++++++++++++++++------------- 1 file changed, 120 insertions(+), 38 deletions(-) (limited to 'ext/faad/gstfaad.c') diff --git a/ext/faad/gstfaad.c b/ext/faad/gstfaad.c index 14fda859..e08ebcda 100644 --- a/ext/faad/gstfaad.c +++ b/ext/faad/gstfaad.c @@ -27,6 +27,9 @@ #include "gstfaad.h" +GST_DEBUG_CATEGORY_STATIC (faad_debug); +#define GST_CAT_DEFAULT faad_debug + static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, @@ -142,6 +145,8 @@ gst_faad_class_init (GstFaadClass * klass) parent_class = g_type_class_ref (GST_TYPE_ELEMENT); gstelement_class->change_state = gst_faad_change_state; + + GST_DEBUG_CATEGORY_INIT (faad_debug, "faad", 0, "faad MPEG-AAC decoding"); } static void @@ -263,6 +268,16 @@ gst_faad_chanpos_to_gst (guchar * fpos, guint num) case LFE_CHANNEL: pos[n] = GST_AUDIO_CHANNEL_POSITION_LFE; break; + case UNKNOWN_CHANNEL: + if (num == 1) { + pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO; + return pos; + } else if (num == 2) { + pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; + pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; + return pos; + } + /* fall-through */ default: GST_WARNING ("Unsupported FAAD channel position 0x%x encountered", fpos[n]); @@ -464,7 +479,7 @@ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps) } /* Another internal checkup. */ - if (faad->need_channel_setup) { + if (faad->need_channel_setup && 0) { GstAudioChannelPosition *pos; guchar *fpos; guint i; @@ -542,6 +557,68 @@ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps) return GST_PAD_LINK_REFUSED; } +/* + * Find syncpoint in ADTS/ADIF stream. Doesn't work for raw, + * packetized streams. Be careful when calling. + * Returns FALSE on no-sync, fills offset/length if one/two + * syncpoints are found, only returns TRUE when it finds two + * subsequent syncpoints (similar to mp3 typefinding in + * gst/typefind/) for ADTS because 12 bits isn't very reliable. + */ + +static gboolean +gst_faad_sync (GstBuffer * buf, guint * off) +{ + guint8 *data = GST_BUFFER_DATA (buf); + guint size = GST_BUFFER_SIZE (buf), n; + gint snc; + + GST_DEBUG ("Finding syncpoint"); + + /* FIXME: for no-sync, we go over the same data for every new buffer. + * We should save the information somewhere. */ + for (n = 0; n < size - 3; n++) { + snc = GST_READ_UINT16_BE (&data[n]); + if ((snc & 0xfff6) == 0xfff0) { + /* we have an ADTS syncpoint. Parse length and find + * next syncpoint. */ + guint len; + + GST_DEBUG ("Found one ADTS syncpoint at offset 0x%x, tracing next...", n); + + if (size - n < 5) { + GST_DEBUG ("Not enough data to parse ADTS header"); + return FALSE; + } + + *off = n; + len = ((data[n + 3] & 0x03) << 11) | + (data[n + 4] << 3) | ((data[n + 5] & 0xe0) >> 5); + if (n + len + 2 >= size) { + GST_DEBUG ("Next frame is not within reach"); + return FALSE; + } + + snc = GST_READ_UINT16_BE (&data[n + len]); + if ((snc & 0xfff6) == 0xfff0) { + GST_DEBUG ("Found ADTS syncpoint at offset 0x%x (framelen %u)", n, len); + return TRUE; + } + + GST_DEBUG ("No next frame found... (should be at 0x%x)", n + len); + } else if (!memcmp (&data[n], "ADIF", 4)) { + /* we have an ADIF syncpoint. 4 bytes is enough. */ + *off = n; + GST_DEBUG ("Found ADIF syncpoint at offset 0x%x", n); + return TRUE; + } + } + + GST_DEBUG ("Found no syncpoint"); + + return FALSE; +} + static void gst_faad_chain (GstPad * pad, GstData * data) { @@ -550,10 +627,11 @@ gst_faad_chain (GstPad * pad, GstData * data) guchar *input_data; GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad)); GstBuffer *buf, *outbuf; - faacDecFrameInfo *info; + faacDecFrameInfo info; guint64 next_ts; void *out; gboolean run_loop = TRUE; + guint sync_off; if (GST_IS_EVENT (data)) { GstEvent *event = GST_EVENT (data); @@ -573,8 +651,6 @@ gst_faad_chain (GstPad * pad, GstData * data) } } - info = g_new0 (faacDecFrameInfo, 1); - /* buffer + remaining data */ buf = GST_BUFFER (data); next_ts = GST_BUFFER_TIMESTAMP (buf); @@ -582,6 +658,16 @@ gst_faad_chain (GstPad * pad, GstData * data) buf = gst_buffer_join (faad->tempbuf, buf); faad->tempbuf = NULL; } + input_data = GST_BUFFER_DATA (buf); + input_size = GST_BUFFER_SIZE (buf); + if (!faad->packetised) { + if (!gst_faad_sync (buf, &sync_off)) + goto next; + else { + input_data += sync_off; + input_size -= sync_off; + } + } /* init if not already done during capsnego */ if (!faad->init) { @@ -589,29 +675,28 @@ gst_faad_chain (GstPad * pad, GstData * data) guchar channels; glong init_res; - init_res = faacDecInit (faad->handle, - GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf), &samplerate, &channels); + init_res = faacDecInit (faad->handle, input_data, input_size, + &samplerate, &channels); if (init_res < 0) { GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL), ("Failed to init decoder from stream")); return; } - skip_bytes = init_res; + skip_bytes = 0; //init_res; faad->init = TRUE; /* store for renegotiation later on */ /* FIXME: that's moot, info will get zeroed in DecDecode() */ - info->samplerate = samplerate; - info->channels = channels; + info.samplerate = samplerate; + info.channels = channels; } else { - info->samplerate = 0; - info->channels = 0; + info.samplerate = 0; + info.channels = 0; } /* decode cycle */ - input_data = GST_BUFFER_DATA (buf); - input_size = GST_BUFFER_SIZE (buf); - info->bytesconsumed = input_size - skip_bytes; + info.bytesconsumed = input_size - skip_bytes; + info.error = 0; if (!faad->packetised) { /* We must check that ourselves for raw stream */ @@ -624,37 +709,36 @@ gst_faad_chain (GstPad * pad, GstData * data) /* Only one packet per buffer, no matter how much is really consumed */ run_loop = FALSE; } else { - if (input_size < FAAD_MIN_STREAMSIZE || info->bytesconsumed <= 0) { + if (input_size < FAAD_MIN_STREAMSIZE || info.bytesconsumed <= 0) { break; } } - out = faacDecDecode (faad->handle, info, input_data + skip_bytes, + out = faacDecDecode (faad->handle, &info, input_data + skip_bytes, input_size - skip_bytes); - if (info->error) { - GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL), - ("Failed to decode buffer: %s", - faacDecGetErrorMessage (info->error))); + if (info.error) { + GST_ERROR_OBJECT (faad, "Failed to decode buffer: %s", + faacDecGetErrorMessage (info.error)); break; } - if (info->bytesconsumed > input_size) - info->bytesconsumed = input_size; - input_size -= info->bytesconsumed; - input_data += info->bytesconsumed; + if (info.bytesconsumed > input_size) + info.bytesconsumed = input_size; + input_size -= info.bytesconsumed; + input_data += info.bytesconsumed; - if (out && info->samples > 0) { + if (out && info.samples > 0) { gboolean fmt_change = FALSE; /* see if we need to renegotiate */ - if (info->samplerate != faad->samplerate || - info->channels != faad->channels || !faad->channel_positions) { + if (info.samplerate != faad->samplerate || + info.channels != faad->channels || !faad->channel_positions) { fmt_change = TRUE; } else { gint i; - for (i = 0; i < info->channels; i++) { - if (info->channel_position[i] != faad->channel_positions[i]) + for (i = 0; i < info.channels; i++) { + if (info.channel_position[i] != faad->channel_positions[i]) fmt_change = TRUE; } } @@ -663,13 +747,12 @@ gst_faad_chain (GstPad * pad, GstData * data) GstPadLinkReturn ret; /* store new negotiation information */ - faad->samplerate = info->samplerate; - faad->channels = info->channels; + faad->samplerate = info.samplerate; + faad->channels = info.channels; if (faad->channel_positions) g_free (faad->channel_positions); faad->channel_positions = g_new (guint8, faad->channels); - memcpy (faad->channel_positions, info->channel_position, - faad->channels); + memcpy (faad->channel_positions, info.channel_position, faad->channels); /* and negotiate */ ret = gst_pad_renegotiate (faad->srcpad); @@ -680,13 +763,13 @@ gst_faad_chain (GstPad * pad, GstData * data) } /* play decoded data */ - if (info->samples > 0) { - outbuf = gst_buffer_new_and_alloc (info->samples * faad->bps); + if (info.samples > 0) { + outbuf = gst_buffer_new_and_alloc (info.samples * faad->bps); /* ugh */ memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf)); GST_BUFFER_TIMESTAMP (outbuf) = next_ts; GST_BUFFER_DURATION (outbuf) = - (guint64) GST_SECOND *info->samples / faad->samplerate; + (guint64) GST_SECOND *info.samples / faad->samplerate; if (GST_CLOCK_TIME_IS_VALID (next_ts)) { next_ts += GST_BUFFER_DURATION (outbuf); } @@ -695,6 +778,7 @@ gst_faad_chain (GstPad * pad, GstData * data) } } +next: /* Keep the leftovers in raw stream */ if (input_size > 0 && !faad->packetised) { if (input_size < GST_BUFFER_SIZE (buf)) { @@ -707,8 +791,6 @@ gst_faad_chain (GstPad * pad, GstData * data) } gst_buffer_unref (buf); - - g_free (info); } static GstElementStateReturn -- cgit v1.2.1