diff options
Diffstat (limited to 'ext')
| -rw-r--r-- | ext/faad/gstfaad.c | 353 | ||||
| -rw-r--r-- | ext/faad/gstfaad.h | 18 |
2 files changed, 242 insertions, 129 deletions
diff --git a/ext/faad/gstfaad.c b/ext/faad/gstfaad.c index b2e2b56e..ba3423a0 100644 --- a/ext/faad/gstfaad.c +++ b/ext/faad/gstfaad.c @@ -22,11 +22,16 @@ #endif #include <string.h> - #include <gst/audio/multichannel.h> - #include "gstfaad.h" +static GstElementDetails faad_details = { + "Free AAC Decoder (FAAD)", + "Codec/Decoder/Audio", + "Free MPEG-2/4 AAC decoder", + "Ronald Bultje <rbultje@ronald.bitfreak.net>" +}; + static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, @@ -42,12 +47,14 @@ static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", "rate = (int) [ 8000, 96000 ], " \ "channels = (int) [ 1, 8 ]" +#if 0 #define STATIC_FLOAT_CAPS(bpp) \ "audio/x-raw-float, " \ "endianness = (int) BYTE_ORDER, " \ "depth = (int) " G_STRINGIFY (bpp) ", " \ "rate = (int) [ 8000, 96000 ], " \ "channels = (int) [ 1, 8 ]" +#endif /* * All except 16-bit integer are disabled until someone fixes FAAD. @@ -69,6 +76,7 @@ STATIC_FLOAT_CAPS (32) \ "; " \ STATIC_FLOAT_CAPS (64) #endif + static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, @@ -79,20 +87,13 @@ static void gst_faad_base_init (GstFaadClass * klass); static void gst_faad_class_init (GstFaadClass * klass); static void gst_faad_init (GstFaad * faad); -/* -static GstPadLinkReturn -gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps); -static GstPadLinkReturn -gst_faad_srcconnect (GstPad * pad, const GstCaps * caps);*/ static gboolean gst_faad_setcaps (GstPad * pad, GstCaps * caps); static GstCaps *gst_faad_srcgetcaps (GstPad * pad); static gboolean gst_faad_event (GstPad * pad, GstEvent * event); static GstFlowReturn gst_faad_chain (GstPad * pad, GstBuffer * buffer); static GstElementStateReturn gst_faad_change_state (GstElement * element); -static GstElementClass *parent_class = NULL; - -/* static guint gst_faad_signals[LAST_SIGNAL] = { 0 }; */ +static GstElementClass *parent_class; /* NULL */ GType gst_faad_get_type (void) @@ -122,11 +123,6 @@ gst_faad_get_type (void) static void gst_faad_base_init (GstFaadClass * klass) { - static GstElementDetails gst_faad_details = - GST_ELEMENT_DETAILS ("Free AAC Decoder (FAAD)", - "Codec/Decoder/Audio", - "Free MPEG-2/4 AAC decoder", - "Ronald Bultje <rbultje@ronald.bitfreak.net>"); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, @@ -134,7 +130,7 @@ gst_faad_base_init (GstFaadClass * klass) gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&sink_template)); - gst_element_class_set_details (element_class, &gst_faad_details); + gst_element_class_set_details (element_class, &faad_details); } static void @@ -142,7 +138,7 @@ gst_faad_class_init (GstFaadClass * klass) { GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); - parent_class = g_type_class_ref (GST_TYPE_ELEMENT); + parent_class = g_type_class_peek_parent (klass); gstelement_class->change_state = gst_faad_change_state; } @@ -157,8 +153,10 @@ gst_faad_init (GstFaad * faad) faad->need_channel_setup = TRUE; faad->channel_positions = NULL; faad->init = FALSE; - - /* GST_FLAG_SET (faad, GST_ELEMENT_EVENT_AWARE); */ + faad->next_ts = 0; + faad->bytes_in = 0; + faad->sum_dur_out = 0; + faad->packetised = FALSE; faad->sinkpad = gst_pad_new_from_template (gst_static_pad_template_get (&sink_template), @@ -167,41 +165,65 @@ gst_faad_init (GstFaad * faad) gst_pad_set_event_function (faad->sinkpad, gst_faad_event); gst_pad_set_setcaps_function (faad->sinkpad, gst_faad_setcaps); gst_pad_set_chain_function (faad->sinkpad, gst_faad_chain); - /*gst_pad_set_link_function (faad->sinkpad, gst_faad_sinkconnect); */ faad->srcpad = gst_pad_new_from_template (gst_static_pad_template_get (&src_template), "src"); gst_element_add_pad (GST_ELEMENT (faad), faad->srcpad); gst_pad_use_fixed_caps (faad->srcpad); - /*gst_pad_set_link_function (faad->srcpad, gst_faad_srcconnect); */ gst_pad_set_getcaps_function (faad->srcpad, gst_faad_srcgetcaps); } static gboolean gst_faad_setcaps (GstPad * pad, GstCaps * caps) { - GstStructure *structure; - GstFaad *faad; - GstCaps *copy; + GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad)); + GstStructure *str = gst_caps_get_structure (caps, 0); + GstBuffer *buf; + const GValue *value; - faad = GST_FAAD (GST_PAD_PARENT (pad)); + /* Assume raw stream */ + faad->packetised = FALSE; - structure = gst_caps_get_structure (caps, 0); + if ((value = gst_structure_get_value (str, "codec_data"))) { + gulong samplerate; + guchar channels; + + /* We have codec data, means packetised stream */ + faad->packetised = TRUE; + buf = g_value_get_boxed (value); - /* get channel count */ - gst_structure_get_int (structure, "channels", &faad->channels); - gst_structure_get_int (structure, "rate", &faad->samplerate); + if (faad->handle) { + GST_DEBUG ("faad handle already open; closing before re-initing"); + faacDecClose (faad->handle); + } - /* create reverse caps */ - copy = gst_caps_new_simple ("audio/x-raw-float", - "channels", G_TYPE_INT, faad->channels, - "depth", G_TYPE_INT, G_STRINGIFY (bpp), - "endianness", G_TYPE_INT, G_BYTE_ORDER, - "rate", G_TYPE_INT, faad->samplerate); + /* someone forgot that char can be unsigned when writing the API */ + if ((gint8) faacDecInit2 (faad->handle, GST_BUFFER_DATA (buf), + GST_BUFFER_SIZE (buf), &samplerate, &channels) < 0) { + GST_DEBUG ("faacDecInit2() failed"); + return FALSE; + } +#if 0 + faad->samplerate = samplerate; + faad->channels = channels; +#endif + /* not updating these here, so they are updated in the + * chain function, and new caps are created etc. */ + faad->samplerate = 0; + faad->channels = 0; + + faad->init = TRUE; + + if (faad->tempbuf) { + gst_buffer_unref (faad->tempbuf); + faad->tempbuf = NULL; + } + } else { + faad->init = FALSE; + } - gst_pad_set_caps (faad->srcpad, copy); - gst_caps_unref (copy); + faad->need_channel_setup = TRUE; return TRUE; } @@ -258,6 +280,7 @@ gst_faad_chanpos_from_gst (GstAudioChannelPosition * pos, guint num) return fpos; } */ + static GstAudioChannelPosition * gst_faad_chanpos_to_gst (guchar * fpos, guint num) { @@ -352,10 +375,11 @@ gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps) return GST_PAD_LINK_OK; } */ + static GstCaps * gst_faad_srcgetcaps (GstPad * pad) { - GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad)); + GstFaad *faad = GST_FAAD (GST_OBJECT_PARENT (pad)); static GstAudioChannelPosition *supported_positions = NULL; static gint num_supported_positions = LFE_CHANNEL - FRONT_CHANNEL_CENTER + 1; GstCaps *templ; @@ -430,7 +454,7 @@ gst_faad_srcgetcaps (GstPad * pad) if (faad->channels != -1) { gst_structure_set (str, "channels", G_TYPE_INT, faad->channels, NULL); - // put channel information here */ + /* put channel information here */ if (faad->channel_positions) { GstAudioChannelPosition *pos; @@ -578,31 +602,124 @@ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps) return GST_PAD_LINK_REFUSED; }*/ -/* - * Data reading. - */ static gboolean gst_faad_event (GstPad * pad, GstEvent * event) { GstFaad *faad; - gboolean res; + gboolean res = TRUE; faad = GST_FAAD (gst_pad_get_parent (pad)); GST_LOG ("handling event %d", GST_EVENT_TYPE (event)); + /* FIXME: we should probably handle FLUSH and also + * SEEK in the case where we are not in a container + * (when our newsegment was in BYTES) */ switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: + if (faad->tempbuf != NULL) { + gst_buffer_unref (faad->tempbuf); + faad->tempbuf = NULL; + } + GST_STREAM_LOCK (pad); + res = gst_pad_push_event (faad->srcpad, event); + GST_STREAM_UNLOCK (pad); + break; case GST_EVENT_NEWSEGMENT: + { + GstFormat fmt; + guint64 start, end, base; + gdouble rate; + + gst_event_parse_newsegment (event, &rate, &fmt, &start, &end, &base); + if (fmt == GST_FORMAT_TIME) { + GST_DEBUG ("Got NEWSEGMENT event in GST_FORMAT_TIME, passing on (" + GST_TIME_FORMAT " - " GST_TIME_FORMAT ")", GST_TIME_ARGS (start), + GST_TIME_ARGS (end)); + } else if (fmt == GST_FORMAT_BYTES) { + GstEvent *new_ev; + guint64 new_start = 0; + guint64 new_end = GST_CLOCK_TIME_NONE; + + GST_DEBUG ("Got NEWSEGMENT event in GST_FORMAT_BYTES (%" + G_GUINT64_FORMAT " - %" G_GUINT64_FORMAT ")", start, end); + + if (faad->bytes_in > 0 && faad->sum_dur_out > 0) { + /* try to convert based on the average bitrate so far */ + new_start = (faad->sum_dur_out * start) / faad->bytes_in; + if (new_end != (guint64) - 1) { + new_end = (faad->sum_dur_out * end) / faad->bytes_in; + } + } else { + GST_DEBUG + ("no average bitrate yet, sending newsegment with start at 0"); + } + new_ev = + gst_event_new_newsegment (rate, GST_FORMAT_TIME, new_start, new_end, + base); + gst_event_unref (event); + event = new_ev; + GST_DEBUG ("Sending new NEWSEGMENT event, time " GST_TIME_FORMAT " - " + GST_TIME_FORMAT, GST_TIME_ARGS (new_start), + GST_TIME_ARGS (new_end)); + } + + GST_STREAM_LOCK (pad); + res = gst_pad_push_event (faad->srcpad, event); + GST_STREAM_UNLOCK (pad); + break; + } default: + GST_STREAM_LOCK (pad); + res = gst_pad_push_event (faad->srcpad, event); + GST_STREAM_UNLOCK (pad); break; } - res = gst_pad_event_default (faad->sinkpad, event); +/* res = gst_pad_event_default (faad->sinkpad, event); */ return res; } +static gboolean +gst_faad_update_caps (GstFaad * faad, faacDecFrameInfo * info, + GstCaps ** p_caps) +{ + GstAudioChannelPosition *pos; + GstCaps *caps; + + /* store new negotiation information */ + faad->samplerate = info->samplerate; + faad->channels = info->channels; + g_free (faad->channel_positions); + faad->channel_positions = g_memdup (info->channel_position, faad->channels); + + caps = gst_caps_new_simple ("audio/x-raw-int", + "endianness", G_TYPE_INT, G_BYTE_ORDER, + "signed", G_TYPE_BOOLEAN, TRUE, + "width", G_TYPE_INT, 16, + "depth", G_TYPE_INT, 16, + "rate", G_TYPE_INT, faad->samplerate, + "channels", G_TYPE_INT, faad->channels, NULL); + + faad->bps = 16 / 8; + + pos = gst_faad_chanpos_to_gst (faad->channel_positions, faad->channels); + gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos); + g_free (pos); + + GST_DEBUG ("New output caps: %" GST_PTR_FORMAT, caps); + + if (!gst_pad_set_caps (faad->srcpad, caps)) { + gst_caps_unref (caps); + return FALSE; + } + + *p_caps = caps; + + return TRUE; +} + static GstFlowReturn gst_faad_chain (GstPad * pad, GstBuffer * buffer) { @@ -610,37 +727,22 @@ gst_faad_chain (GstPad * pad, GstBuffer * buffer) guint input_size; guint skip_bytes = 0; guchar *input_data; - GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad)); + GstFaad *faad; GstBuffer *outbuf; - faacDecFrameInfo *info; - guint64 next_ts; + GstCaps *caps = NULL; + faacDecFrameInfo info; void *out; gboolean run_loop = TRUE; -/* - if (GST_IS_EVENT (data)) { - GstEvent *event = GST_EVENT (data); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_EOS: - if (faad->tempbuf != NULL) { - gst_buffer_unref (faad->tempbuf); - faad->tempbuf = NULL; - } - gst_element_set_eos (GST_ELEMENT (faad)); - gst_pad_push (faad->srcpad, data); - return; - default: - gst_pad_event_default (pad, event); - return; - } + faad = GST_FAAD (GST_OBJECT_PARENT (pad)); + + if (GST_BUFFER_TIMESTAMP (buffer) != GST_CLOCK_TIME_NONE) { + faad->next_ts = GST_BUFFER_TIMESTAMP (buffer); + GST_DEBUG ("Timestamp on incoming buffer: %" GST_TIME_FORMAT, + GST_TIME_ARGS (faad->next_ts)); } -*/ - info = g_new0 (faacDecFrameInfo, 1); /* buffer + remaining data */ - /* buf = GST_BUFFER (data); */ - next_ts = GST_BUFFER_TIMESTAMP (buffer); if (faad->tempbuf) { buffer = gst_buffer_join (faad->tempbuf, buffer); faad->tempbuf = NULL; @@ -663,19 +765,15 @@ gst_faad_chain (GstPad * pad, GstBuffer * buffer) skip_bytes = 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; - } else { - info->samplerate = 0; - info->channels = 0; + /* make sure we create new caps below */ + faad->samplerate = 0; + faad->channels = 0; } /* decode cycle */ input_data = GST_BUFFER_DATA (buffer); input_size = GST_BUFFER_SIZE (buffer); - info->bytesconsumed = input_size - skip_bytes; + info.bytesconsumed = input_size - skip_bytes; if (!faad->packetised) { /* We must check that ourselves for raw stream */ @@ -688,72 +786,75 @@ gst_faad_chain (GstPad * pad, GstBuffer * buffer) /* 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) { + if (info.error) { GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL), - ("Failed to decode buffer: %s", - faacDecGetErrorMessage (info->error))); - break; + ("Failed to decode buffer: %s", faacDecGetErrorMessage (info.error))); + ret = GST_FLOW_ERROR; + goto out; } - 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; } } if (fmt_change) { - /*GstPadLinkReturn ret; */ - - /* store new negotiation information */ - 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); - - /* and negotiate - ret = gst_pad_renegotiate (faad->srcpad); - if (GST_PAD_LINK_FAILED (ret)) { - GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL), (NULL)); - break; - } */ + if (!gst_faad_update_caps (faad, &info, &caps)) { + GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL), + ("Setting caps on source pad failed")); + ret = GST_FLOW_ERROR; + goto out; + } } /* play decoded data */ - 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; - if (GST_CLOCK_TIME_IS_VALID (next_ts)) { - next_ts += GST_BUFFER_DURATION (outbuf); + if (info.samples > 0 && GST_PAD_PEER (faad->srcpad)) { + GstFlowReturn r; + guint bufsize = info.samples * faad->bps; + + /* note: info.samples is total samples, not per channel */ + r = gst_pad_alloc_buffer (faad->srcpad, 0, bufsize, caps, &outbuf); + if (r != GST_FLOW_OK) { + GST_DEBUG ("Failed to allocate buffer"); + ret = GST_FLOW_OK; /* CHECK: or return something else? */ + goto out; } + + memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf)); + GST_BUFFER_OFFSET (outbuf) = + (faad->next_ts * faad->samplerate) / GST_SECOND; + GST_BUFFER_TIMESTAMP (outbuf) = faad->next_ts; + GST_BUFFER_DURATION (outbuf) = (guint64) GST_SECOND *info.samples / (faad->samplerate * 2); ///////// over 2? + + faad->next_ts += GST_BUFFER_DURATION (outbuf); + faad->sum_dur_out += GST_BUFFER_DURATION (outbuf); + + GST_DEBUG ("pushing buffer, off=%" G_GUINT64_FORMAT ", ts=%" + GST_TIME_FORMAT, GST_BUFFER_OFFSET (outbuf), + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf))); gst_pad_push (faad->srcpad, outbuf); } } @@ -770,9 +871,14 @@ gst_faad_chain (GstPad * pad, GstBuffer * buffer) } } - gst_buffer_unref (buffer); + faad->bytes_in += input_size; + +out: - g_free (info); + if (caps) + gst_caps_unref (caps); + + gst_buffer_unref (buffer); return ret; } @@ -784,6 +890,7 @@ gst_faad_change_state (GstElement * element) switch (GST_STATE_TRANSITION (element)) { case GST_STATE_NULL_TO_READY: + { if (!(faad->handle = faacDecOpen ())) return GST_STATE_FAILURE; else { @@ -791,10 +898,13 @@ gst_faad_change_state (GstElement * element) conf = faacDecGetCurrentConfiguration (faad->handle); conf->defObjectType = LC; - //conf->dontUpSampleImplicitSBR = 1; - faacDecSetConfiguration (faad->handle, conf); + /* conf->dontUpSampleImplicitSBR = 1; */ + conf->outputFormat = FAAD_FMT_16BIT; + if (faacDecSetConfiguration (faad->handle, conf) == 0) + return GST_STATE_FAILURE; } break; + } case GST_STATE_PAUSED_TO_READY: faad->samplerate = -1; faad->channels = -1; @@ -802,6 +912,7 @@ gst_faad_change_state (GstElement * element) faad->init = FALSE; g_free (faad->channel_positions); faad->channel_positions = NULL; + faad->next_ts = 0; break; case GST_STATE_READY_TO_NULL: faacDecClose (faad->handle); diff --git a/ext/faad/gstfaad.h b/ext/faad/gstfaad.h index fa083aa4..e9f0a3cb 100644 --- a/ext/faad/gstfaad.h +++ b/ext/faad/gstfaad.h @@ -39,16 +39,14 @@ G_BEGIN_DECLS typedef struct _GstFaad { GstElement element; - /* pads */ - GstPad *srcpad, *sinkpad; + GstPad *srcpad; + GstPad *sinkpad; - /* cache for latest MPEG-frame */ - gint samplerate, - channels, - bps; + guint samplerate; /* sample rate of the last MPEG frame */ + guint channels; /* number of channels of the last frame */ + guint bps; /* bytes per sample */ - /* used to keep input leftovers */ - GstBuffer *tempbuf; + GstBuffer *tempbuf; /* used to keep input leftovers */ /* FAAD object */ faacDecHandle handle; @@ -58,6 +56,10 @@ typedef struct _GstFaad { guchar *channel_positions; gboolean need_channel_setup; gboolean packetised; /* We must differentiate between raw and packetised streams */ + + guint64 next_ts; /* timestamp of next buffer */ + guint64 bytes_in; /* bytes received */ + guint64 sum_dur_out; /* sum of durations of decoded buffers we sent out */ } GstFaad; typedef struct _GstFaadClass { |