#include <string.h>
-#include <gst/audio/multichannel.h>
#include <gst/pbutils/codec-utils.h>
#include "gstvoaacenc.h"
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "width = (int) 16, "
- "depth = (int) 16, "
- "signed = (boolean) TRUE, "
- "endianness = (int) BYTE_ORDER, "
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " GST_AUDIO_NE (S16) ", "
+ "layout = (string) interleaved, "
"rate = (int) { " SAMPLE_RATES " }, " "channels = (int) [1, 2]")
);
GstAudioInfo * info);
static GstFlowReturn gst_voaacenc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
-static GstCaps *gst_voaacenc_getcaps (GstAudioEncoder * enc);
+static GstCaps *gst_voaacenc_getcaps (GstAudioEncoder * enc, GstCaps * filter);
-GST_BOILERPLATE (GstVoAacEnc, gst_voaacenc, GstAudioEncoder,
- GST_TYPE_AUDIO_ENCODER);
+G_DEFINE_TYPE (GstVoAacEnc, gst_voaacenc, GST_TYPE_AUDIO_ENCODER);
static void
gst_voaacenc_set_property (GObject * object, guint prop_id,
}
static void
-gst_voaacenc_base_init (gpointer klass)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&sink_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&src_template));
-
- gst_element_class_set_details_simple (element_class, "AAC audio encoder",
- "Codec/Encoder/Audio", "AAC audio encoder", "Kan Hu <kan.hu@linaro.org>");
-}
-
-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);
"Target Audio Bitrate",
0, G_MAXINT, VOAAC_ENC_DEFAULT_BITRATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&sink_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&src_template));
+
+ gst_element_class_set_details_simple (element_class, "AAC audio encoder",
+ "Codec/Encoder/Audio", "AAC audio encoder", "Kan Hu <kan.hu@linaro.org>");
}
static void
-gst_voaacenc_init (GstVoAacEnc * voaacenc, GstVoAacEncClass * klass)
+gst_voaacenc_init (GstVoAacEnc * voaacenc)
{
voaacenc->bitrate = VOAAC_ENC_DEFAULT_BITRATE;
voaacenc->output_format = VOAAC_ENC_DEFAULT_OUTPUTFORMAT;
return TRUE;
}
+#define VOAAC_ENC_MAX_CHANNELS 6
+/* describe the channels position */
+static const GstAudioChannelPosition
+ aac_channel_positions[][VOAAC_ENC_MAX_CHANNELS] = {
+ { /* 1 ch: Mono */
+ GST_AUDIO_CHANNEL_POSITION_MONO},
+ { /* 2 ch: front left + front right (front stereo) */
+ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
+ { /* 3 ch: front center + front stereo */
+ GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
+ { /* 4 ch: front center + front stereo + back center */
+ GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+ GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+ GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
+ { /* 5 ch: front center + front stereo + back stereo */
+ 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},
+ { /* 6ch: front center + front stereo + back stereo + LFE */
+ 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_LFE1}
+};
+
static gpointer
gst_voaacenc_generate_sink_caps (gpointer data)
{
-#define VOAAC_ENC_MAX_CHANNELS 6
-/* describe the channels position */
- static const GstAudioChannelPosition
- gst_voaacenc_channel_position[][VOAAC_ENC_MAX_CHANNELS] = {
- { /* 1 ch: Mono */
- GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
- { /* 2 ch: front left + front right (front stereo) */
- GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
- GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
- { /* 3 ch: front center + front stereo */
- GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
- GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
- GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
- { /* 4 ch: front center + front stereo + back center */
- GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
- GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
- GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
- GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
- { /* 5 ch: front center + front stereo + back stereo */
- 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},
- { /* 6ch: front center + front stereo + back stereo + LFE */
- 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_LFE}
- };
GstCaps *caps = gst_caps_new_empty ();
gint i, c;
static const int rates[] = {
};
GValue rates_arr = { 0, };
GValue tmp = { 0, };
+ GstStructure *s, *t;
g_value_init (&rates_arr, GST_TYPE_LIST);
g_value_init (&tmp, G_TYPE_INT);
}
g_value_unset (&tmp);
- for (i = 0; i < 2 /*VOAAC_ENC_MAX_CHANNELS */ ; i++) {
- GValue chanpos = { 0 };
- GValue pos = { 0 };
- GstStructure *structure;
+ s = gst_structure_new ("audio/x-raw",
+ "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
+ "layout", G_TYPE_STRING, "interleaved", NULL);
+ gst_structure_set_value (s, "rate", &rates_arr);
- g_value_init (&chanpos, GST_TYPE_ARRAY);
- g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
+ caps = gst_caps_new_empty ();
- for (c = 0; c <= i; c++) {
- g_value_set_enum (&pos, gst_voaacenc_channel_position[i][c]);
- gst_value_array_append_value (&chanpos, &pos);
- }
+ for (i = 1; i <= 2 /* VOAAC_ENC_MAX_CHANNELS */ ; i++) {
+ guint64 channel_mask = 0;
+ t = gst_structure_copy (s);
- g_value_unset (&pos);
+ gst_structure_set (t, "channels", G_TYPE_INT, i, NULL);
+ if (i == 1)
+ continue;
- structure = gst_structure_new ("audio/x-raw-int",
- "width", G_TYPE_INT, 16,
- "depth", G_TYPE_INT, 16,
- "signed", G_TYPE_BOOLEAN, TRUE,
- "endianness", G_TYPE_INT, G_BYTE_ORDER,
- "channels", G_TYPE_INT, i + 1, NULL);
+ for (c = 0; c < i; c++)
+ channel_mask |= G_GUINT64_CONSTANT (1) << aac_channel_positions[i - 1][c];
- gst_structure_set_value (structure, "rate", &rates_arr);
- gst_structure_set_value (structure, "channel-positions", &chanpos);
- g_value_unset (&chanpos);
-
- gst_caps_append_structure (caps, structure);
+ gst_structure_set (t, "channel-mask", GST_TYPE_BITMASK, channel_mask, NULL);
+ gst_caps_append_structure (caps, t);
}
+ gst_structure_free (s);
g_value_unset (&rates_arr);
GST_DEBUG ("generated sink caps: %" GST_PTR_FORMAT, caps);
}
static GstCaps *
-gst_voaacenc_getcaps (GstAudioEncoder * benc)
+gst_voaacenc_getcaps (GstAudioEncoder * benc, GstCaps * filter)
{
return gst_audio_encoder_proxy_getcaps (benc, gst_voaacenc_get_sink_caps ());
}
gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc)
{
GstCaps *caps = NULL;
- GstBuffer *codec_data;
gint index;
- guint8 data[VOAAC_ENC_CODECDATA_LEN];
+ GstBuffer *codec_data;
+ guint8 *data;
if ((index = gst_voaacenc_get_rate_index (voaacenc->rate)) >= 0) {
+ codec_data = gst_buffer_new_and_alloc (VOAAC_ENC_CODECDATA_LEN);
+ data = gst_buffer_map (codec_data, NULL, NULL, GST_MAP_WRITE);
/* LC profile only */
data[0] = ((0x02 << 3) | (index >> 1));
data[1] = ((index & 0x01) << 7) | (voaacenc->channels << 3);
(voaacenc->output_format ? "adts" : "raw")
, NULL);
- gst_codec_utils_aac_caps_set_level_and_profile (caps, data, sizeof (data));
+ gst_codec_utils_aac_caps_set_level_and_profile (caps, data,
+ VOAAC_ENC_CODECDATA_LEN);
+ gst_buffer_unmap (codec_data, data, -1);
if (!voaacenc->output_format) {
- codec_data = gst_buffer_new_and_alloc (VOAAC_ENC_CODECDATA_LEN);
-
- memcpy (GST_BUFFER_DATA (codec_data), data, sizeof (data));
-
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data,
NULL);
-
- gst_buffer_unref (codec_data);
}
+ gst_buffer_unref (codec_data);
}
return caps;
VO_AUDIO_OUTPUTINFO output_info = { {0} };
VO_CODECBUFFER input = { 0 };
VO_CODECBUFFER output = { 0 };
+ gsize size;
+ guint8 *data, *out_data;
+ GstAudioInfo *info = gst_audio_encoder_get_audio_info (benc);
voaacenc = GST_VOAACENC (benc);
goto exit;
}
- if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < voaacenc->inbuf_size)) {
- GST_DEBUG_OBJECT (voaacenc, "discarding trailing data %d",
- buf ? GST_BUFFER_SIZE (buf) : 0);
+ if (memcmp (info->position, aac_channel_positions[info->channels - 1],
+ sizeof (GstAudioChannelPosition) * info->channels) != 0) {
+ buf = gst_buffer_make_writable (buf);
+ gst_audio_buffer_reorder_channels (buf, info->finfo->format,
+ info->channels, info->position,
+ aac_channel_positions[info->channels - 1]);
+ }
+
+ data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
+
+ if (G_UNLIKELY (size < voaacenc->inbuf_size)) {
+ gst_buffer_unmap (buf, data, -1);
+ GST_DEBUG_OBJECT (voaacenc, "discarding trailing data %d", (gint) size);
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;
- }
+ out = gst_buffer_new_and_alloc (voaacenc->inbuf_size);
+ out_data = gst_buffer_map (out, NULL, NULL, GST_MAP_WRITE);
- output.Buffer = GST_BUFFER_DATA (out);
+ output.Buffer = out_data;
output.Length = voaacenc->inbuf_size;
- g_assert (GST_BUFFER_SIZE (buf) == voaacenc->inbuf_size);
- input.Buffer = GST_BUFFER_DATA (buf);
+ g_assert (size == 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_unmap (buf, data, -1);
+ gst_buffer_unmap (out, out_data, -1);
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)));
+ gst_buffer_unmap (out, out_data, output.Length);
+ gst_buffer_unmap (buf, data, -1);
ret = gst_audio_encoder_finish_frame (benc, out, 1024);
projects / platform / upstream / gstreamer.git / commitdiff