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fdkaacenc: add support for HE-AACv1 and HE-AACv2

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
This commit is contained in:
Piotrek Brzeziński 2022-02-23 17:17:25 +01:00 committed by GStreamer Marge Bot
parent 2a59e8af97
commit 8cda666cb0
2 changed files with 62 additions and 8 deletions
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2
subprojects/gst-plugins-bad/docs/plugins/gst_plugins_cache.json
View file

@ -16708,7 +16708,7 @@
"presence": "always"
},
"src": {
"caps": "audio/mpeg:\n mpegversion: 4\n rate: { (int)8000, (int)11025, (int)12000, (int)16000, (int)22050, (int)24000, (int)32000, (int)44100, (int)48000, (int)64000, (int)88200, (int)96000 }\n channels: { (int)1, (int)2, (int)3, (int)4, (int)5, (int)6, (int)8 }\n stream-format: { (string)adts, (string)adif, (string)raw }\n base-profile: lc\n framed: true\n",
"caps": "audio/mpeg:\n mpegversion: 4\n rate: { (int)8000, (int)11025, (int)12000, (int)16000, (int)22050, (int)24000, (int)32000, (int)44100, (int)48000, (int)64000, (int)88200, (int)96000 }\n channels: { (int)1, (int)2, (int)3, (int)4, (int)5, (int)6, (int)8 }\n stream-format: { (string)adts, (string)adif, (string)raw }\n profile: { (string)lc, (string)sbr, (string)ps }\n framed: true\n",
"direction": "src",
"presence": "always"
}

68
subprojects/gst-plugins-bad/ext/fdkaac/gstfdkaacenc.c
View file

@ -74,7 +74,7 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
"rate = (int) { " SAMPLE_RATES " }, "
"channels = (int) {1, 2, 3, 4, 5, 6, 8}, "
"stream-format = (string) { adts, adif, raw }, "
"base-profile = (string) lc, " "framed = (boolean) true")
"profile = (string) { lc, sbr, ps }, " "framed = (boolean) true")
);
GST_DEBUG_CATEGORY_STATIC (gst_fdkaacenc_debug);
@ -162,14 +162,34 @@ static GstCaps *
gst_fdkaacenc_get_caps (GstAudioEncoder * enc, GstCaps * filter)
{
const GstFdkAacChannelLayout *layout;
GstCaps *res, *caps;
GstCaps *res, *caps, *allowed_caps;
gboolean allow_mono = TRUE;
allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (enc));
GST_DEBUG_OBJECT (enc, "allowed caps %" GST_PTR_FORMAT, allowed_caps);
/* We need at least 2 channels if Parametric Stereo is in use. */
if (allowed_caps && gst_caps_get_size (allowed_caps) > 0) {
GstStructure *s = gst_caps_get_structure (allowed_caps, 0);
const gchar *profile = NULL;
if ((profile = gst_structure_get_string (s, "profile"))
&& strcmp (profile, "ps") == 0) {
allow_mono = FALSE;
}
}
gst_clear_caps (&allowed_caps);
caps = gst_caps_new_empty ();
for (layout = channel_layouts; layout->channels; layout++) {
GstCaps *tmp;
gint channels = layout->channels;
GstCaps *tmp =
gst_caps_make_writable (gst_pad_get_pad_template_caps
if (channels == 1 && !allow_mono)
continue;
tmp = gst_caps_make_writable (gst_pad_get_pad_template_caps
(GST_AUDIO_ENCODER_SINK_PAD (enc)));
if (channels == 1) {
@ -203,7 +223,8 @@ gst_fdkaacenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
gint mpegversion = 4;
CHANNEL_MODE channel_mode;
AACENC_InfoStruct enc_info = { 0 };
gint bitrate;
gint bitrate, signaling_mode;
const gchar *ext_profile;
if (self->enc && !self->is_drained) {
/* drain */
@ -233,6 +254,19 @@ gst_fdkaacenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
}
}
if ((str = gst_structure_get_string (s, "profile"))) {
if (strcmp (str, "lc") == 0) {
GST_DEBUG_OBJECT (self, "using LC profile for output");
aot = AOT_AAC_LC;
} else if (strcmp (str, "sbr") == 0) {
GST_DEBUG_OBJECT (self, "using SBR (HE-AAC) profile for output");
aot = AOT_SBR;
} else if (strcmp (str, "ps") == 0) {
GST_DEBUG_OBJECT (self, "using PS (HE-AACv2) profile for output");
aot = AOT_PS;
}
}
gst_structure_get_int (s, "mpegversion", &mpegversion);
}
if (allowed_caps)
@ -244,13 +278,25 @@ gst_fdkaacenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
return FALSE;
}
aot = AOT_AAC_LC;
if ((err = aacEncoder_SetParam (self->enc, AACENC_AOT, aot)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set AOT %d: %d", aot, err);
return FALSE;
}
/* Use explicit hierarchical signaling (2) with raw output stream-format
* and implicit signaling (0) with ADTS/ADIF */
if (transmux == 0)
signaling_mode = 2;
else
signaling_mode = 0;
if ((err = aacEncoder_SetParam (self->enc, AACENC_SIGNALING_MODE,
signaling_mode)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set signaling mode %d: %d",
signaling_mode, err);
return FALSE;
}
if ((err = aacEncoder_SetParam (self->enc, AACENC_SAMPLERATE,
GST_AUDIO_INFO_RATE (info))) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set sample rate %d: %d",
@ -415,6 +461,14 @@ gst_fdkaacenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
gst_codec_utils_aac_caps_set_level_and_profile (src_caps, enc_info.confBuf,
enc_info.confSize);
/* The above only parses the "base" profile, which is always going to be LC.
* Let's retrieve the extension AOT and set it as our profile in the caps. */
ext_profile = gst_codec_utils_aac_get_extension_profile (enc_info.confBuf,
enc_info.confSize);
if (ext_profile)
gst_caps_set_simple (src_caps, "profile", G_TYPE_STRING, ext_profile, NULL);
ret = gst_audio_encoder_set_output_format (enc, src_caps);
gst_caps_unref (src_caps);