Commit 76f3f25e authored by Tim-Philipp Müller's avatar Tim-Philipp Müller 🐠
Browse files

ext/faad/gstfaad.*: Fix negotiation (#310932) and miscellaneous other stuff....

ext/faad/gstfaad.*: Fix negotiation (#310932) and miscellaneous other stuff. Probably still needs some more work.

Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_base_init), (gst_faad_class_init),
(gst_faad_init), (gst_faad_setcaps), (gst_faad_srcgetcaps),
(gst_faad_event), (gst_faad_update_caps), (gst_faad_chain),
(gst_faad_change_state):
* ext/faad/gstfaad.h:
Fix negotiation (#310932) and miscellaneous other stuff. Probably
still needs some more work.
parent a649c14c
2005-08-08 Tim-Philipp Müller <tim at centricular dot net>
* ext/faad/gstfaad.c: (gst_faad_base_init), (gst_faad_class_init),
(gst_faad_init), (gst_faad_setcaps), (gst_faad_srcgetcaps),
(gst_faad_event), (gst_faad_update_caps), (gst_faad_chain),
(gst_faad_change_state):
* ext/faad/gstfaad.h:
Fix negotiation (#310932) and miscellaneous other stuff. Probably
still needs some more work.
2005-08-08 Tim-Philipp Müller <tim at centricular dot net>
 
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init),
......
......@@ -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);
......
......@@ -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 {
......