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voaacenc: port to audioencoder

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This commit is contained in:
Mark Nauwelaerts 2011-11-17 23:02:39 +01:00
parent 56701570e9
commit ea92249ee2
3 changed files with 120 additions and 195 deletions
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2
ext/voaacenc/Makefile.am
View file

@ -4,7 +4,7 @@ libgstvoaacenc_la_SOURCES = \
gstvoaac.c \
gstvoaacenc.c
libgstvoaacenc_la_CFLAGS = \
libgstvoaacenc_la_CFLAGS = -DGST_USE_UNSTABLE_API \
$(GST_PLUGINS_BASE_CFLAGS) \
$(GST_BASE_CFLAGS) \
$(GST_CFLAGS) \

300
ext/voaacenc/gstvoaacenc.c
View file

@ -91,19 +91,20 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_DEBUG_CATEGORY_STATIC (gst_voaacenc_debug);
#define GST_CAT_DEFAULT gst_voaacenc_debug
static void gst_voaacenc_finalize (GObject * object);
static GstFlowReturn gst_voaacenc_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_voaacenc_setcaps (GstPad * pad, GstCaps * caps);
static GstStateChangeReturn gst_voaacenc_state_change (GstElement * element,
GstStateChange transition);
static gboolean voaacenc_core_init (GstVoAacEnc * voaacenc);
static gboolean voaacenc_core_set_parameter (GstVoAacEnc * voaacenc);
static void voaacenc_core_uninit (GstVoAacEnc * voaacenc);
static GstCaps *gst_voaacenc_getcaps (GstPad * pad);
static GstCaps *gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc);
static gint voaacenc_get_rate_index (gint rate);
static gboolean gst_voaacenc_start (GstAudioEncoder * enc);
static gboolean gst_voaacenc_stop (GstAudioEncoder * enc);
static gboolean gst_voaacenc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_voaacenc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
static GstCaps *gst_voaacenc_getcaps (GstAudioEncoder * enc);
static gpointer
gst_voaacenc_generate_sink_caps (gpointer data)
{
@ -200,27 +201,11 @@ gst_voaacenc_get_sink_caps (void)
g_once (&g_once, gst_voaacenc_generate_sink_caps, NULL);
caps = g_once.retval;
return gst_caps_ref (caps);
return caps;
}
static void
_do_init (GType object_type)
{
const GInterfaceInfo preset_interface_info = {
NULL, /* interface init */
NULL, /* interface finalize */
NULL /* interface_data */
};
g_type_add_interface_static (object_type, GST_TYPE_PRESET,
&preset_interface_info);
GST_DEBUG_CATEGORY_INIT (gst_voaacenc_debug, "voaacenc", 0,
"AAC audio encoder");
}
GST_BOILERPLATE_FULL (GstVoAacEnc, gst_voaacenc, GstElement, GST_TYPE_ELEMENT,
_do_init);
GST_BOILERPLATE (GstVoAacEnc, gst_voaacenc, GstAudioEncoder,
GST_TYPE_AUDIO_ENCODER);
static void
gst_voaacenc_set_property (GObject * object, guint prop_id,
@ -274,11 +259,16 @@ static void
gst_voaacenc_class_init (GstVoAacEncClass * klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
object_class->set_property = GST_DEBUG_FUNCPTR (gst_voaacenc_set_property);
object_class->get_property = GST_DEBUG_FUNCPTR (gst_voaacenc_get_property);
object_class->finalize = GST_DEBUG_FUNCPTR (gst_voaacenc_finalize);
base_class->start = GST_DEBUG_FUNCPTR (gst_voaacenc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_voaacenc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_voaacenc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_voaacenc_handle_frame);
base_class->getcaps = GST_DEBUG_FUNCPTR (gst_voaacenc_getcaps);
g_object_class_install_property (object_class, PROP_BITRATE,
g_param_spec_int ("bitrate",
@ -286,30 +276,11 @@ gst_voaacenc_class_init (GstVoAacEncClass * klass)
"Target Audio Bitrate",
0, G_MAXINT, VOAAC_ENC_DEFAULT_BITRATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
element_class->change_state = GST_DEBUG_FUNCPTR (gst_voaacenc_state_change);
}
static void
gst_voaacenc_init (GstVoAacEnc * voaacenc, GstVoAacEncClass * klass)
{
/* create the sink pad */
voaacenc->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_setcaps_function (voaacenc->sinkpad,
GST_DEBUG_FUNCPTR (gst_voaacenc_setcaps));
gst_pad_set_getcaps_function (voaacenc->sinkpad,
GST_DEBUG_FUNCPTR (gst_voaacenc_getcaps));
gst_pad_set_chain_function (voaacenc->sinkpad,
GST_DEBUG_FUNCPTR (gst_voaacenc_chain));
gst_element_add_pad (GST_ELEMENT (voaacenc), voaacenc->sinkpad);
/* create the src pad */
voaacenc->srcpad = gst_pad_new_from_static_template (&src_template, "src");
gst_pad_use_fixed_caps (voaacenc->srcpad);
gst_element_add_pad (GST_ELEMENT (voaacenc), voaacenc->srcpad);
voaacenc->adapter = gst_adapter_new ();
voaacenc->bitrate = VOAAC_ENC_DEFAULT_BITRATE;
voaacenc->output_format = VOAAC_ENC_DEFAULT_OUTPUTFORMAT;
@ -317,17 +288,31 @@ gst_voaacenc_init (GstVoAacEnc * voaacenc, GstVoAacEncClass * klass)
voaacenc->handle = NULL;
}
static void
gst_voaacenc_finalize (GObject * object)
static gboolean
gst_voaacenc_start (GstAudioEncoder * enc)
{
GstVoAacEnc *voaacenc;
GstVoAacEnc *voaacenc = GST_VOAACENC (enc);
voaacenc = GST_VOAACENC (object);
GST_DEBUG_OBJECT (enc, "start");
g_object_unref (G_OBJECT (voaacenc->adapter));
voaacenc->adapter = NULL;
if (voaacenc_core_init (voaacenc) == FALSE)
return FALSE;
G_OBJECT_CLASS (parent_class)->finalize (object);
voaacenc->rate = 0;
voaacenc->channels = 0;
return TRUE;
}
static gboolean
gst_voaacenc_stop (GstAudioEncoder * enc)
{
GstVoAacEnc *voaacenc = GST_VOAACENC (enc);
GST_DEBUG_OBJECT (enc, "stop");
voaacenc_core_uninit (voaacenc);
return TRUE;
}
/* check downstream caps to configure format */
@ -336,7 +321,7 @@ gst_voaacenc_negotiate (GstVoAacEnc * voaacenc)
{
GstCaps *caps;
caps = gst_pad_get_allowed_caps (voaacenc->srcpad);
caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (voaacenc));
GST_DEBUG_OBJECT (voaacenc, "allowed caps: %" GST_PTR_FORMAT, caps);
@ -363,31 +348,25 @@ gst_voaacenc_negotiate (GstVoAacEnc * voaacenc)
}
static GstCaps *
gst_voaacenc_getcaps (GstPad * pad)
gst_voaacenc_getcaps (GstAudioEncoder * benc)
{
return gst_voaacenc_get_sink_caps ();
return gst_audio_encoder_proxy_getcaps (benc, gst_voaacenc_get_sink_caps ());
}
static gboolean
gst_voaacenc_setcaps (GstPad * pad, GstCaps * caps)
gst_voaacenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
gboolean ret = FALSE;
GstStructure *structure;
GstVoAacEnc *voaacenc;
GstCaps *src_caps;
voaacenc = GST_VOAACENC (GST_PAD_PARENT (pad));
structure = gst_caps_get_structure (caps, 0);
voaacenc = GST_VOAACENC (benc);
/* get channel count */
gst_structure_get_int (structure, "channels", &voaacenc->channels);
gst_structure_get_int (structure, "rate", &voaacenc->rate);
voaacenc->channels = GST_AUDIO_INFO_CHANNELS (info);
voaacenc->rate = GST_AUDIO_INFO_RATE (info);
/* precalc duration as it's constant now */
voaacenc->duration =
gst_util_uint64_scale_int (1024, GST_SECOND, voaacenc->rate);
/* precalc buffer size as it's constant now */
voaacenc->inbuf_size = voaacenc->channels * 2 * 1024;
gst_voaacenc_negotiate (voaacenc);
@ -396,105 +375,84 @@ gst_voaacenc_setcaps (GstPad * pad, GstCaps * caps)
src_caps = gst_voaacenc_create_source_pad_caps (voaacenc);
if (src_caps) {
gst_pad_set_caps (voaacenc->srcpad, src_caps);
gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (voaacenc), src_caps);
gst_caps_unref (src_caps);
ret = voaacenc_core_set_parameter (voaacenc);
}
/* report needs to base class */
gst_audio_encoder_set_frame_samples_min (benc, 1024);
gst_audio_encoder_set_frame_samples_max (benc, 1024);
gst_audio_encoder_set_frame_max (benc, 1);
return ret;
}
static GstFlowReturn
gst_voaacenc_chain (GstPad * pad, GstBuffer * buffer)
gst_voaacenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstVoAacEnc *voaacenc;
GstFlowReturn ret;
guint64 timestamp, distance = 0;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out;
VO_AUDIO_OUTPUTINFO output_info = { {0} };
VO_CODECBUFFER input = { 0 };
VO_CODECBUFFER output = { 0 };
voaacenc = GST_VOAACENC (GST_PAD_PARENT (pad));
voaacenc = GST_VOAACENC (benc);
g_return_val_if_fail (voaacenc->handle, GST_FLOW_WRONG_STATE);
g_return_val_if_fail (voaacenc->handle, GST_FLOW_NOT_NEGOTIATED);
if (voaacenc->rate == 0 || voaacenc->channels == 0)
goto not_negotiated;
/* discontinuity clears adapter, FIXME, maybe we can set some
* encoder flag to mask the discont. */
if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
gst_adapter_clear (voaacenc->adapter);
voaacenc->discont = TRUE;
/* we don't deal with squeezing remnants, so simply discard those */
if (G_UNLIKELY (buf == NULL)) {
GST_DEBUG_OBJECT (benc, "no data");
goto exit;
}
ret = GST_FLOW_OK;
gst_adapter_push (voaacenc->adapter, buffer);
/* Collect samples until we have enough for an output frame */
while (gst_adapter_available (voaacenc->adapter) >= voaacenc->inbuf_size) {
GstBuffer *out;
guint8 *data;
VO_CODECBUFFER input = { 0 }
, output = {
0};
VO_AUDIO_OUTPUTINFO output_info = { {0}
};
/* max size */
if ((ret =
gst_pad_alloc_buffer_and_set_caps (voaacenc->srcpad, 0,
voaacenc->inbuf_size, GST_PAD_CAPS (voaacenc->srcpad),
&out)) != GST_FLOW_OK) {
return ret;
}
output.Buffer = GST_BUFFER_DATA (out);
output.Length = voaacenc->inbuf_size;
if (voaacenc->discont) {
GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DISCONT);
voaacenc->discont = FALSE;
}
data =
(guint8 *) gst_adapter_peek (voaacenc->adapter, voaacenc->inbuf_size);
input.Buffer = data;
input.Length = voaacenc->inbuf_size;
voaacenc->codec_api.SetInputData (voaacenc->handle, &input);
/* encode */
if (voaacenc->codec_api.GetOutputData (voaacenc->handle, &output,
&output_info) != VO_ERR_NONE) {
gst_buffer_unref (out);
return GST_FLOW_ERROR;
}
/* get timestamp from adapter */
timestamp = gst_adapter_prev_timestamp (voaacenc->adapter, &distance);
if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (timestamp))) {
GST_BUFFER_TIMESTAMP (out) =
timestamp +
GST_FRAMES_TO_CLOCK_TIME (distance / voaacenc->channels /
VOAAC_ENC_BITS_PER_SAMPLE, voaacenc->rate);
}
GST_BUFFER_DURATION (out) =
GST_FRAMES_TO_CLOCK_TIME (voaacenc->inbuf_size / voaacenc->channels /
VOAAC_ENC_BITS_PER_SAMPLE, voaacenc->rate);
GST_LOG_OBJECT (voaacenc, "Pushing out buffer time: %" GST_TIME_FORMAT
" duration: %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out)),
GST_TIME_ARGS (GST_BUFFER_DURATION (out)));
GST_BUFFER_SIZE (out) = output.Length;
/* flush the among of data we have peek */
gst_adapter_flush (voaacenc->adapter, voaacenc->inbuf_size);
/* play */
if ((ret = gst_pad_push (voaacenc->srcpad, out)) != GST_FLOW_OK)
break;
if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < voaacenc->inbuf_size)) {
GST_DEBUG_OBJECT (voaacenc, "discarding trailing data %d",
buf ? GST_BUFFER_SIZE (buf) : 0);
ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
goto exit;
}
/* max size */
if ((ret =
gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD
(voaacenc), 0, voaacenc->inbuf_size,
GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (voaacenc)),
&out)) != GST_FLOW_OK) {
goto exit;
}
output.Buffer = GST_BUFFER_DATA (out);
output.Length = voaacenc->inbuf_size;
g_assert (GST_BUFFER_SIZE (buf) == voaacenc->inbuf_size);
input.Buffer = GST_BUFFER_DATA (buf);
input.Length = voaacenc->inbuf_size;
voaacenc->codec_api.SetInputData (voaacenc->handle, &input);
/* encode */
if (voaacenc->codec_api.GetOutputData (voaacenc->handle, &output,
&output_info) != VO_ERR_NONE) {
gst_buffer_unref (out);
goto encode_failed;
}
GST_LOG_OBJECT (voaacenc, "encoded to %d bytes", output.Length);
GST_BUFFER_SIZE (out) = output.Length;
GST_LOG_OBJECT (voaacenc, "Pushing out buffer time: %" GST_TIME_FORMAT
" duration: %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out)),
GST_TIME_ARGS (GST_BUFFER_DURATION (out)));
ret = gst_audio_encoder_finish_frame (benc, out, 1024);
exit:
return ret;
/* ERRORS */
@ -502,45 +460,17 @@ not_negotiated:
{
GST_ELEMENT_ERROR (voaacenc, STREAM, TYPE_NOT_FOUND,
(NULL), ("unknown type"));
return GST_FLOW_NOT_NEGOTIATED;
ret = GST_FLOW_NOT_NEGOTIATED;
goto exit;
}
encode_failed:
{
GST_ELEMENT_ERROR (voaacenc, STREAM, ENCODE, (NULL), ("encode failed"));
ret = GST_FLOW_ERROR;
goto exit;
}
}
static GstStateChangeReturn
gst_voaacenc_state_change (GstElement * element, GstStateChange transition)
{
GstVoAacEnc *voaacenc;
GstStateChangeReturn ret;
voaacenc = GST_VOAACENC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (voaacenc_core_init (voaacenc) == FALSE)
return GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
voaacenc->rate = 0;
voaacenc->channels = 0;
voaacenc->discont = FALSE;
gst_adapter_clear (voaacenc->adapter);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
voaacenc_core_uninit (voaacenc);
gst_adapter_clear (voaacenc->adapter);
break;
default:
break;
}
return ret;
}
static GstCaps *
gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc)

13
ext/voaacenc/gstvoaacenc.h
View file

@ -21,11 +21,11 @@
#define __GST_VOAACENC_H__
#include <gst/gst.h>
#include <gst/audio/gstaudioencoder.h>
#include <vo-aacenc/voAAC.h>
#include <vo-aacenc/cmnMemory.h>
#include <gst/base/gstadapter.h>
G_BEGIN_DECLS
#define GST_TYPE_VOAACENC \
@ -43,14 +43,10 @@ typedef struct _GstVoAacEnc GstVoAacEnc;
typedef struct _GstVoAacEncClass GstVoAacEncClass;
struct _GstVoAacEnc {
GstElement element;
GstAudioEncoder element;
/* pads */
GstPad *sinkpad, *srcpad;
gboolean discont;
GstAdapter *adapter;
/* desired bitrate */
gint bitrate;
@ -58,7 +54,6 @@ struct _GstVoAacEnc {
gint channels;
gint rate;
gint output_format;
gint duration;
gint inbuf_size;
@ -70,7 +65,7 @@ struct _GstVoAacEnc {
};
struct _GstVoAacEncClass {
GstElementClass parent_class;
GstAudioEncoderClass parent_class;
};
GType gst_voaacenc_get_type (void);