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voaacenc: port to 0.11
This commit is contained in:
Mark Nauwelaerts
parent
9f136a667d
commit
e0494dcaa5
2 changed files with 100 additions and 108 deletions
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@ -305,7 +305,7 @@ GST_PLUGINS_NONPORTED=" adpcmdec adpcmenc aiff asfmux \
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sdi segmentclip siren speed subenc stereo tta videofilters \
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videomaxrate videomeasure videosignal vmnc \
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decklink fbdev linsys shm vcd \
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voaacenc apexsink bz2 cdaudio celt cog curl dc1394 dirac directfb resindvd \
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apexsink bz2 cdaudio celt cog curl dc1394 dirac directfb resindvd \
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gsettings gsm jp2k ladspa modplug mpeg2enc mplex mimic \
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musepack musicbrainz nas neon ofa openal opencv rsvg schro sdl smooth sndfile soundtouch spandsp timidity \
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wildmidi xvid apple_media lv2 teletextdec"
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@ -37,7 +37,6 @@
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#include <string.h>
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#include <gst/audio/multichannel.h>
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#include <gst/pbutils/codec-utils.h>
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#include "gstvoaacenc.h"
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@ -70,11 +69,9 @@ enum
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"signed = (boolean) TRUE, "
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"endianness = (int) BYTE_ORDER, "
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, "
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"rate = (int) { " SAMPLE_RATES " }, " "channels = (int) [1, 2]")
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);
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@ -101,10 +98,9 @@ static gboolean gst_voaacenc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_voaacenc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static GstCaps *gst_voaacenc_getcaps (GstAudioEncoder * enc);
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static GstCaps *gst_voaacenc_getcaps (GstAudioEncoder * enc, GstCaps * filter);
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GST_BOILERPLATE (GstVoAacEnc, gst_voaacenc, GstAudioEncoder,
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GST_TYPE_AUDIO_ENCODER);
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G_DEFINE_TYPE (GstVoAacEnc, gst_voaacenc, GST_TYPE_AUDIO_ENCODER);
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static void
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gst_voaacenc_set_property (GObject * object, guint prop_id,
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@ -140,24 +136,11 @@ gst_voaacenc_get_property (GObject * object, guint prop_id,
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return;
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}
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static void
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gst_voaacenc_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_details_simple (element_class, "AAC audio encoder",
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"Codec/Encoder/Audio", "AAC audio encoder", "Kan Hu <kan.hu@linaro.org>");
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}
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static void
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gst_voaacenc_class_init (GstVoAacEncClass * klass)
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{
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
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object_class->set_property = GST_DEBUG_FUNCPTR (gst_voaacenc_set_property);
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@ -175,10 +158,18 @@ gst_voaacenc_class_init (GstVoAacEncClass * klass)
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"Target Audio Bitrate",
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0, G_MAXINT, VOAAC_ENC_DEFAULT_BITRATE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_details_simple (element_class, "AAC audio encoder",
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"Codec/Encoder/Audio", "AAC audio encoder", "Kan Hu <kan.hu@linaro.org>");
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}
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static void
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gst_voaacenc_init (GstVoAacEnc * voaacenc, GstVoAacEncClass * klass)
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gst_voaacenc_init (GstVoAacEnc * voaacenc)
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{
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voaacenc->bitrate = VOAAC_ENC_DEFAULT_BITRATE;
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voaacenc->output_format = VOAAC_ENC_DEFAULT_OUTPUTFORMAT;
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@ -214,15 +205,12 @@ gst_voaacenc_stop (GstAudioEncoder * enc)
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return TRUE;
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}
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static gpointer
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gst_voaacenc_generate_sink_caps (gpointer data)
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{
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#define VOAAC_ENC_MAX_CHANNELS 6
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/* describe the channels position */
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static const GstAudioChannelPosition
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gst_voaacenc_channel_position[][VOAAC_ENC_MAX_CHANNELS] = {
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static const GstAudioChannelPosition
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aac_channel_positions[][VOAAC_ENC_MAX_CHANNELS] = {
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{ /* 1 ch: Mono */
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GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
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GST_AUDIO_CHANNEL_POSITION_MONO},
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{ /* 2 ch: front left + front right (front stereo) */
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GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
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@ -247,8 +235,12 @@ gst_voaacenc_generate_sink_caps (gpointer data)
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GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
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GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
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GST_AUDIO_CHANNEL_POSITION_LFE}
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};
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GST_AUDIO_CHANNEL_POSITION_LFE1}
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};
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static gpointer
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gst_voaacenc_generate_sink_caps (gpointer data)
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{
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GstCaps *caps = gst_caps_new_empty ();
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gint i, c;
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static const int rates[] = {
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@ -257,6 +249,7 @@ gst_voaacenc_generate_sink_caps (gpointer data)
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};
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GValue rates_arr = { 0, };
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GValue tmp = { 0, };
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GstStructure *s, *t;
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g_value_init (&rates_arr, GST_TYPE_LIST);
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g_value_init (&tmp, G_TYPE_INT);
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@ -266,35 +259,29 @@ gst_voaacenc_generate_sink_caps (gpointer data)
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}
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g_value_unset (&tmp);
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for (i = 0; i < 2 /*VOAAC_ENC_MAX_CHANNELS */ ; i++) {
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GValue chanpos = { 0 };
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GValue pos = { 0 };
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GstStructure *structure;
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s = gst_structure_new ("audio/x-raw",
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"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
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"layout", G_TYPE_STRING, "interleaved", NULL);
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gst_structure_set_value (s, "rate", &rates_arr);
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g_value_init (&chanpos, GST_TYPE_ARRAY);
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g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
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caps = gst_caps_new_empty ();
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for (c = 0; c <= i; c++) {
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g_value_set_enum (&pos, gst_voaacenc_channel_position[i][c]);
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gst_value_array_append_value (&chanpos, &pos);
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}
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g_value_unset (&pos);
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structure = gst_structure_new ("audio/x-raw-int",
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"width", G_TYPE_INT, 16,
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"depth", G_TYPE_INT, 16,
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"signed", G_TYPE_BOOLEAN, TRUE,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"channels", G_TYPE_INT, i + 1, NULL);
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gst_structure_set_value (structure, "rate", &rates_arr);
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gst_structure_set_value (structure, "channel-positions", &chanpos);
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g_value_unset (&chanpos);
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gst_caps_append_structure (caps, structure);
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for (i = 1; i <= 2 /* VOAAC_ENC_MAX_CHANNELS */ ; i++) {
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guint64 channel_mask = 0;
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t = gst_structure_copy (s);
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gst_structure_set (t, "channels", G_TYPE_INT, i, NULL);
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if (i == 1)
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continue;
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for (c = 0; c < i; c++)
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channel_mask |= G_GUINT64_CONSTANT (1) << aac_channel_positions[i - 1][c];
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gst_structure_set (t, "channel-mask", GST_TYPE_BITMASK, channel_mask, NULL);
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gst_caps_append_structure (caps, t);
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}
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gst_structure_free (s);
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g_value_unset (&rates_arr);
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GST_DEBUG ("generated sink caps: %" GST_PTR_FORMAT, caps);
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@ -314,7 +301,7 @@ gst_voaacenc_get_sink_caps (void)
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}
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static GstCaps *
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gst_voaacenc_getcaps (GstAudioEncoder * benc)
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gst_voaacenc_getcaps (GstAudioEncoder * benc, GstCaps * filter)
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{
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return gst_audio_encoder_proxy_getcaps (benc, gst_voaacenc_get_sink_caps ());
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}
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@ -371,11 +358,13 @@ static GstCaps *
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gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc)
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{
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GstCaps *caps = NULL;
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GstBuffer *codec_data;
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gint index;
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guint8 data[VOAAC_ENC_CODECDATA_LEN];
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GstBuffer *codec_data;
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guint8 *data;
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if ((index = gst_voaacenc_get_rate_index (voaacenc->rate)) >= 0) {
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codec_data = gst_buffer_new_and_alloc (VOAAC_ENC_CODECDATA_LEN);
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data = gst_buffer_map (codec_data, NULL, NULL, GST_MAP_WRITE);
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/* LC profile only */
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data[0] = ((0x02 << 3) | (index >> 1));
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data[1] = ((index & 0x01) << 7) | (voaacenc->channels << 3);
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@ -388,18 +377,15 @@ gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc)
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(voaacenc->output_format ? "adts" : "raw")
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, NULL);
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gst_codec_utils_aac_caps_set_level_and_profile (caps, data, sizeof (data));
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gst_codec_utils_aac_caps_set_level_and_profile (caps, data,
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VOAAC_ENC_CODECDATA_LEN);
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gst_buffer_unmap (codec_data, data, -1);
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if (!voaacenc->output_format) {
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codec_data = gst_buffer_new_and_alloc (VOAAC_ENC_CODECDATA_LEN);
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memcpy (GST_BUFFER_DATA (codec_data), data, sizeof (data));
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gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data,
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NULL);
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gst_buffer_unref (codec_data);
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}
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gst_buffer_unref (codec_data);
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}
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return caps;
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@ -449,6 +435,9 @@ gst_voaacenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
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VO_AUDIO_OUTPUTINFO output_info = { {0} };
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VO_CODECBUFFER input = { 0 };
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VO_CODECBUFFER output = { 0 };
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gsize size;
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guint8 *data, *out_data;
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GstAudioInfo *info = gst_audio_encoder_get_audio_info (benc);
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voaacenc = GST_VOAACENC (benc);
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@ -460,44 +449,47 @@ gst_voaacenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
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goto exit;
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}
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if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < voaacenc->inbuf_size)) {
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GST_DEBUG_OBJECT (voaacenc, "discarding trailing data %d",
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buf ? GST_BUFFER_SIZE (buf) : 0);
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if (memcmp (info->position, aac_channel_positions[info->channels - 1],
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sizeof (GstAudioChannelPosition) * info->channels) != 0) {
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buf = gst_buffer_make_writable (buf);
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gst_audio_buffer_reorder_channels (buf, info->finfo->format,
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info->channels, info->position,
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aac_channel_positions[info->channels - 1]);
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}
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data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
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if (G_UNLIKELY (size < voaacenc->inbuf_size)) {
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gst_buffer_unmap (buf, data, -1);
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GST_DEBUG_OBJECT (voaacenc, "discarding trailing data %d", (gint) size);
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ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
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goto exit;
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}
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/* max size */
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if ((ret =
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gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD
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(voaacenc), 0, voaacenc->inbuf_size,
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GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (voaacenc)),
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&out)) != GST_FLOW_OK) {
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goto exit;
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}
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out = gst_buffer_new_and_alloc (voaacenc->inbuf_size);
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out_data = gst_buffer_map (out, NULL, NULL, GST_MAP_WRITE);
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output.Buffer = GST_BUFFER_DATA (out);
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output.Buffer = out_data;
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output.Length = voaacenc->inbuf_size;
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g_assert (GST_BUFFER_SIZE (buf) == voaacenc->inbuf_size);
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input.Buffer = GST_BUFFER_DATA (buf);
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g_assert (size == voaacenc->inbuf_size);
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input.Buffer = data;
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input.Length = voaacenc->inbuf_size;
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voaacenc->codec_api.SetInputData (voaacenc->handle, &input);
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/* encode */
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if (voaacenc->codec_api.GetOutputData (voaacenc->handle, &output,
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&output_info) != VO_ERR_NONE) {
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gst_buffer_unmap (buf, data, -1);
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gst_buffer_unmap (out, out_data, -1);
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gst_buffer_unref (out);
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goto encode_failed;
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}
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GST_LOG_OBJECT (voaacenc, "encoded to %d bytes", output.Length);
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GST_BUFFER_SIZE (out) = output.Length;
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GST_LOG_OBJECT (voaacenc, "Pushing out buffer time: %" GST_TIME_FORMAT
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" duration: %" GST_TIME_FORMAT,
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (out)));
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gst_buffer_unmap (out, out_data, output.Length);
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gst_buffer_unmap (buf, data, -1);
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ret = gst_audio_encoder_finish_frame (benc, out, 1024);
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