static void
gst_opus_enc_setup_base_class (GstOpusEnc * enc, GstAudioEncoder * benc)
{
gst_audio_encoder_set_latency (benc,
gst_opus_enc_get_latency (enc), gst_opus_enc_get_latency (enc));
gst_audio_encoder_set_frame_samples_min (benc, enc->frame_samples);
gst_audio_encoder_set_frame_samples_max (benc, enc->frame_samples);
gst_audio_encoder_set_frame_max (benc, 1);
}
static gboolean
gst_speex_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstSpeexEnc *enc;
enc = GST_SPEEX_ENC (benc);
enc->channels = GST_AUDIO_INFO_CHANNELS (info);
enc->rate = GST_AUDIO_INFO_RATE (info);
/* handle reconfigure */
if (enc->state) {
speex_encoder_destroy (enc->state);
enc->state = NULL;
}
if (!gst_speex_enc_setup (enc))
return FALSE;
/* feedback to base class */
gst_audio_encoder_set_latency (benc,
gst_speex_enc_get_latency (enc), gst_speex_enc_get_latency (enc));
gst_audio_encoder_set_lookahead (benc, enc->lookahead);
if (enc->nframes == 0) {
/* as many frames as available input allows */
gst_audio_encoder_set_frame_samples_min (benc, enc->frame_size);
gst_audio_encoder_set_frame_samples_max (benc, enc->frame_size);
gst_audio_encoder_set_frame_max (benc, 0);
} else {
/* exactly as many frames as configured */
gst_audio_encoder_set_frame_samples_min (benc,
enc->frame_size * enc->nframes);
gst_audio_encoder_set_frame_samples_max (benc,
enc->frame_size * enc->nframes);
gst_audio_encoder_set_frame_max (benc, 1);
}
return TRUE;
}
static gboolean
gst_celt_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstCeltEnc *enc;
GstCaps *otherpadcaps;
enc = GST_CELT_ENC (benc);
enc->channels = GST_AUDIO_INFO_CHANNELS (info);
enc->rate = GST_AUDIO_INFO_RATE (info);
/* handle reconfigure */
if (enc->state) {
celt_encoder_destroy (enc->state);
enc->state = NULL;
}
if (enc->mode) {
celt_mode_destroy (enc->mode);
enc->mode = NULL;
}
memset (&enc->header, 0, sizeof (enc->header));
otherpadcaps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (enc));
if (otherpadcaps) {
if (!gst_caps_is_empty (otherpadcaps)) {
GstStructure *ps = gst_caps_get_structure (otherpadcaps, 0);
gst_structure_get_int (ps, "frame-size", &enc->frame_size);
}
gst_caps_unref (otherpadcaps);
}
if (enc->requested_frame_size > 0)
enc->frame_size = enc->requested_frame_size;
GST_DEBUG_OBJECT (enc, "channels=%d rate=%d frame-size=%d",
enc->channels, enc->rate, enc->frame_size);
if (!gst_celt_enc_setup (enc))
return FALSE;
/* feedback to base class */
gst_audio_encoder_set_latency (benc,
gst_celt_enc_get_latency (enc), gst_celt_enc_get_latency (enc));
gst_audio_encoder_set_frame_samples_min (benc, enc->frame_size);
gst_audio_encoder_set_frame_samples_max (benc, enc->frame_size);
gst_audio_encoder_set_frame_max (benc, 1);
return TRUE;
}
static gboolean
gst_gsmenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstCaps *srccaps;
srccaps = gst_static_pad_template_get_caps (&gsmenc_src_template);
gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (benc), srccaps);
/* report needs to base class */
gst_audio_encoder_set_frame_samples_min (benc, 160);
gst_audio_encoder_set_frame_samples_max (benc, 160);
gst_audio_encoder_set_frame_max (benc, 1);
return TRUE;
}
static gboolean
gst_amrnbenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstAmrnbEnc *amrnbenc;
GstCaps *copy;
amrnbenc = GST_AMRNBENC (enc);
/* parameters already parsed for us */
amrnbenc->rate = GST_AUDIO_INFO_RATE (info);
amrnbenc->channels = GST_AUDIO_INFO_CHANNELS (info);
/* we do not really accept other input, but anyway ... */
/* this is not wrong but will sound bad */
if (amrnbenc->channels != 1) {
g_warning ("amrnbdec is only optimized for mono channels");
}
if (amrnbenc->rate != 8000) {
g_warning ("amrnbdec is only optimized for 8000 Hz samplerate");
}
/* create reverse caps */
copy = gst_caps_new_simple ("audio/AMR",
"channels", G_TYPE_INT, amrnbenc->channels,
"rate", G_TYPE_INT, amrnbenc->rate, NULL);
gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (amrnbenc), copy);
gst_caps_unref (copy);
/* report needs to base class: hand one frame at a time */
gst_audio_encoder_set_frame_samples_min (enc, 160);
gst_audio_encoder_set_frame_samples_max (enc, 160);
gst_audio_encoder_set_frame_max (enc, 1);
return TRUE;
}
static gboolean
gst_ffmpegaudenc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
{
GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
GstCaps *other_caps;
GstCaps *allowed_caps;
GstCaps *icaps;
gsize frame_size;
GstFFMpegAudEncClass *oclass =
(GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc);
/* close old session */
if (ffmpegaudenc->opened) {
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
ffmpegaudenc->opened = FALSE;
}
/* if we set it in _getcaps we should set it also in _link */
ffmpegaudenc->context->strict_std_compliance = -1;
/* user defined properties */
if (ffmpegaudenc->bitrate > 0) {
GST_INFO_OBJECT (ffmpegaudenc, "Setting avcontext to bitrate %d",
ffmpegaudenc->bitrate);
ffmpegaudenc->context->bit_rate = ffmpegaudenc->bitrate;
ffmpegaudenc->context->bit_rate_tolerance = ffmpegaudenc->bitrate;
} else {
GST_INFO_OBJECT (ffmpegaudenc, "Using avcontext default bitrate %d",
ffmpegaudenc->context->bit_rate);
}
/* RTP payload used for GOB production (for Asterisk) */
if (ffmpegaudenc->rtp_payload_size) {
ffmpegaudenc->context->rtp_payload_size = ffmpegaudenc->rtp_payload_size;
}
/* some other defaults */
ffmpegaudenc->context->rc_strategy = 2;
ffmpegaudenc->context->b_frame_strategy = 0;
ffmpegaudenc->context->coder_type = 0;
ffmpegaudenc->context->context_model = 0;
ffmpegaudenc->context->scenechange_threshold = 0;
ffmpegaudenc->context->inter_threshold = 0;
/* fetch pix_fmt and so on */
gst_ffmpeg_audioinfo_to_context (info, ffmpegaudenc->context);
if (!ffmpegaudenc->context->time_base.den) {
ffmpegaudenc->context->time_base.den = GST_AUDIO_INFO_RATE (info);
ffmpegaudenc->context->time_base.num = 1;
ffmpegaudenc->context->ticks_per_frame = 1;
}
if (ffmpegaudenc->context->channel_layout) {
gst_ffmpeg_channel_layout_to_gst (ffmpegaudenc->context->channel_layout,
ffmpegaudenc->context->channels, ffmpegaudenc->ffmpeg_layout);
ffmpegaudenc->needs_reorder =
(memcmp (ffmpegaudenc->ffmpeg_layout, info->position,
sizeof (GstAudioChannelPosition) *
ffmpegaudenc->context->channels) != 0);
}
/* open codec */
if (gst_ffmpeg_avcodec_open (ffmpegaudenc->context, oclass->in_plugin) < 0) {
if (ffmpegaudenc->context->priv_data)
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
GST_DEBUG_OBJECT (ffmpegaudenc, "avenc_%s: Failed to open FFMPEG codec",
oclass->in_plugin->name);
return FALSE;
}
/* some codecs support more than one format, first auto-choose one */
GST_DEBUG_OBJECT (ffmpegaudenc, "picking an output format ...");
allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (encoder));
if (!allowed_caps) {
GST_DEBUG_OBJECT (ffmpegaudenc, "... but no peer, using template caps");
/* we need to copy because get_allowed_caps returns a ref, and
* get_pad_template_caps doesn't */
allowed_caps =
gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SRC_PAD (encoder));
}
GST_DEBUG_OBJECT (ffmpegaudenc, "chose caps %" GST_PTR_FORMAT, allowed_caps);
gst_ffmpeg_caps_with_codecid (oclass->in_plugin->id,
oclass->in_plugin->type, allowed_caps, ffmpegaudenc->context);
/* try to set this caps on the other side */
other_caps = gst_ffmpeg_codecid_to_caps (oclass->in_plugin->id,
ffmpegaudenc->context, TRUE);
if (!other_caps) {
gst_caps_unref (allowed_caps);
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
GST_DEBUG ("Unsupported codec - no caps found");
return FALSE;
}
icaps = gst_caps_intersect (allowed_caps, other_caps);
gst_caps_unref (allowed_caps);
gst_caps_unref (other_caps);
if (gst_caps_is_empty (icaps)) {
gst_caps_unref (icaps);
return FALSE;
}
icaps = gst_caps_truncate (icaps);
if (!gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (ffmpegaudenc),
icaps)) {
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
gst_caps_unref (icaps);
return FALSE;
}
gst_caps_unref (icaps);
frame_size = ffmpegaudenc->context->frame_size;
if (frame_size > 1) {
gst_audio_encoder_set_frame_samples_min (GST_AUDIO_ENCODER (ffmpegaudenc),
frame_size);
gst_audio_encoder_set_frame_samples_max (GST_AUDIO_ENCODER (ffmpegaudenc),
frame_size);
gst_audio_encoder_set_frame_max (GST_AUDIO_ENCODER (ffmpegaudenc), 1);
} else {
gst_audio_encoder_set_frame_samples_min (GST_AUDIO_ENCODER (ffmpegaudenc),
0);
gst_audio_encoder_set_frame_samples_max (GST_AUDIO_ENCODER (ffmpegaudenc),
0);
gst_audio_encoder_set_frame_max (GST_AUDIO_ENCODER (ffmpegaudenc), 0);
}
/* success! */
ffmpegaudenc->opened = TRUE;
return TRUE;
}
static gboolean
gst_two_lame_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstTwoLame *twolame;
gint out_samplerate;
gint version;
GstCaps *othercaps;
twolame = GST_TWO_LAME (enc);
/* parameters already parsed for us */
twolame->samplerate = GST_AUDIO_INFO_RATE (info);
twolame->num_channels = GST_AUDIO_INFO_CHANNELS (info);
twolame->float_input = !GST_AUDIO_INFO_IS_INTEGER (info);
/* but we might be asked to reconfigure, so reset */
gst_two_lame_release_memory (twolame);
GST_DEBUG_OBJECT (twolame, "setting up twolame");
if (!gst_two_lame_setup (twolame))
goto setup_failed;
out_samplerate = twolame_get_out_samplerate (twolame->glopts);
if (out_samplerate == 0)
goto zero_output_rate;
if (out_samplerate != twolame->samplerate) {
GST_WARNING_OBJECT (twolame,
"output samplerate %d is different from incoming samplerate %d",
out_samplerate, twolame->samplerate);
}
version = twolame_get_version (twolame->glopts);
if (version == TWOLAME_MPEG2)
version = 2;
else
version = 1;
othercaps =
gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 1,
"mpegaudioversion", G_TYPE_INT, version,
"layer", G_TYPE_INT, 2,
"channels", G_TYPE_INT,
twolame->mode == TWOLAME_MONO ? 1 : twolame->num_channels, "rate",
G_TYPE_INT, out_samplerate, NULL);
/* and use these caps */
gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (twolame), othercaps);
gst_caps_unref (othercaps);
/* report needs to base class:
* hand one frame at a time, if we are pretty sure what a frame is */
if (out_samplerate == twolame->samplerate) {
gst_audio_encoder_set_frame_samples_min (enc, 1152);
gst_audio_encoder_set_frame_samples_max (enc, 1152);
gst_audio_encoder_set_frame_max (enc, 1);
}
return TRUE;
zero_output_rate:
{
GST_ELEMENT_ERROR (twolame, LIBRARY, SETTINGS, (NULL),
("TwoLAME decided on a zero sample rate"));
return FALSE;
}
setup_failed:
{
GST_ELEMENT_ERROR (twolame, LIBRARY, SETTINGS,
(_("Failed to configure TwoLAME encoder. Check your encoding parameters.")), (NULL));
return FALSE;
}
}
static gboolean
gst_fdkaacenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstFdkAacEnc *self = GST_FDKAACENC (enc);
gboolean ret = FALSE;
GstCaps *allowed_caps;
GstCaps *src_caps;
AACENC_ERROR err;
gint transmux = 0, aot = AOT_AAC_LC;
gint mpegversion = 4;
CHANNEL_MODE channel_mode;
AACENC_InfoStruct enc_info = { 0 };
gint bitrate;
if (self->enc) {
/* drain */
gst_fdkaacenc_handle_frame (enc, NULL);
aacEncClose (&self->enc);
}
allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (self));
GST_DEBUG_OBJECT (self, "allowed caps: %" GST_PTR_FORMAT, allowed_caps);
if (allowed_caps && gst_caps_get_size (allowed_caps) > 0) {
GstStructure *s = gst_caps_get_structure (allowed_caps, 0);
const gchar *str = NULL;
if ((str = gst_structure_get_string (s, "stream-format"))) {
if (strcmp (str, "adts") == 0) {
GST_DEBUG_OBJECT (self, "use ADTS format for output");
transmux = 2;
} else if (strcmp (str, "adif") == 0) {
GST_DEBUG_OBJECT (self, "use ADIF format for output");
transmux = 1;
} else if (strcmp (str, "raw") == 0) {
GST_DEBUG_OBJECT (self, "use RAW format for output");
transmux = 0;
}
}
gst_structure_get_int (s, "mpegversion", &mpegversion);
}
if (allowed_caps)
gst_caps_unref (allowed_caps);
err = aacEncOpen (&self->enc, 0, GST_AUDIO_INFO_CHANNELS (info));
if (err != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to open encoder: %d\n", err);
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\n", aot, 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\n",
GST_AUDIO_INFO_RATE (info), err);
return FALSE;
}
if (GST_AUDIO_INFO_CHANNELS (info) == 1) {
channel_mode = MODE_1;
self->need_reorder = FALSE;
self->aac_positions = NULL;
} else {
guint64 in_channel_mask, out_channel_mask;
gint i;
for (i = 0; i < G_N_ELEMENTS (channel_layouts); i++) {
if (channel_layouts[i].channels != GST_AUDIO_INFO_CHANNELS (info))
continue;
gst_audio_channel_positions_to_mask (&GST_AUDIO_INFO_POSITION (info, 0),
GST_AUDIO_INFO_CHANNELS (info), FALSE, &in_channel_mask);
gst_audio_channel_positions_to_mask (channel_layouts[i].positions,
channel_layouts[i].channels, FALSE, &out_channel_mask);
if (in_channel_mask == out_channel_mask) {
channel_mode = channel_layouts[i].mode;
self->need_reorder =
memcmp (channel_layouts[i].positions,
&GST_AUDIO_INFO_POSITION (info, 0),
GST_AUDIO_INFO_CHANNELS (info) *
sizeof (GstAudioChannelPosition)) != 0;
self->aac_positions = channel_layouts[i].positions;
break;
}
}
if (i == G_N_ELEMENTS (channel_layouts)) {
GST_ERROR_OBJECT (self, "Couldn't find a valid channel layout");
return FALSE;
}
}
if ((err = aacEncoder_SetParam (self->enc, AACENC_CHANNELMODE,
channel_mode)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set channel mode %d: %d", channel_mode,
err);
return FALSE;
}
/* MPEG channel order */
if ((err = aacEncoder_SetParam (self->enc, AACENC_CHANNELORDER,
0)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set channel order %d: %d", channel_mode,
err);
return FALSE;
}
bitrate = self->bitrate;
/* See
* http://wiki.hydrogenaud.io/index.php?title=Fraunhofer_FDK_AAC#Recommended_Sampling_Rate_and_Bitrate_Combinations
*/
if (bitrate == 0) {
if (GST_AUDIO_INFO_CHANNELS (info) == 1) {
if (GST_AUDIO_INFO_RATE (info) < 16000) {
bitrate = 8000;
} else if (GST_AUDIO_INFO_RATE (info) == 16000) {
bitrate = 16000;
} else if (GST_AUDIO_INFO_RATE (info) < 32000) {
bitrate = 24000;
} else if (GST_AUDIO_INFO_RATE (info) == 32000) {
bitrate = 32000;
} else if (GST_AUDIO_INFO_RATE (info) <= 44100) {
bitrate = 56000;
} else {
bitrate = 160000;
}
} else if (GST_AUDIO_INFO_CHANNELS (info) == 2) {
if (GST_AUDIO_INFO_RATE (info) < 16000) {
bitrate = 16000;
} else if (GST_AUDIO_INFO_RATE (info) == 16000) {
bitrate = 24000;
} else if (GST_AUDIO_INFO_RATE (info) < 22050) {
bitrate = 32000;
} else if (GST_AUDIO_INFO_RATE (info) < 32000) {
bitrate = 40000;
} else if (GST_AUDIO_INFO_RATE (info) == 32000) {
bitrate = 96000;
} else if (GST_AUDIO_INFO_RATE (info) <= 44100) {
bitrate = 112000;
} else {
bitrate = 320000;
}
} else {
/* 5, 5.1 */
if (GST_AUDIO_INFO_RATE (info) < 32000) {
bitrate = 160000;
} else if (GST_AUDIO_INFO_RATE (info) <= 44100) {
bitrate = 240000;
} else {
bitrate = 320000;
}
}
}
if ((err = aacEncoder_SetParam (self->enc, AACENC_TRANSMUX,
transmux)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set transmux %d: %d", transmux, err);
return FALSE;
}
if ((err = aacEncoder_SetParam (self->enc, AACENC_BITRATE,
bitrate)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to set bitrate %d: %d", bitrate, err);
return FALSE;
}
if ((err = aacEncEncode (self->enc, NULL, NULL, NULL, NULL)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to initialize encoder: %d", err);
return FALSE;
}
if ((err = aacEncInfo (self->enc, &enc_info)) != AACENC_OK) {
GST_ERROR_OBJECT (self, "Unable to get encoder info: %d", err);
return FALSE;
}
gst_audio_encoder_set_frame_max (enc, 1);
gst_audio_encoder_set_frame_samples_min (enc, enc_info.frameLength);
gst_audio_encoder_set_frame_samples_max (enc, enc_info.frameLength);
gst_audio_encoder_set_hard_min (enc, FALSE);
self->outbuf_size = enc_info.maxOutBufBytes;
self->samples_per_frame = enc_info.frameLength;
src_caps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, mpegversion,
"channels", G_TYPE_INT, GST_AUDIO_INFO_CHANNELS (info),
"framed", G_TYPE_BOOLEAN, TRUE,
"rate", G_TYPE_INT, GST_AUDIO_INFO_RATE (info), NULL);
/* raw */
if (transmux == 0) {
GstBuffer *codec_data =
gst_buffer_new_wrapped (g_memdup (enc_info.confBuf, enc_info.confSize),
enc_info.confSize);
gst_caps_set_simple (src_caps, "codec_data", GST_TYPE_BUFFER, codec_data,
"stream-format", G_TYPE_STRING, "raw", NULL);
gst_buffer_unref (codec_data);
} else if (transmux == 1) {
gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "adif",
NULL);
} else if (transmux == 2) {
gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "adts",
NULL);
} else {
g_assert_not_reached ();
}
gst_codec_utils_aac_caps_set_level_and_profile (src_caps, enc_info.confBuf,
enc_info.confSize);
ret = gst_audio_encoder_set_output_format (enc, src_caps);
gst_caps_unref (src_caps);
return ret;
}
static gboolean
gst_sbc_enc_set_format (GstAudioEncoder * audio_enc, GstAudioInfo * info)
{
const gchar *allocation_method, *channel_mode;
GstSbcEnc *enc = GST_SBC_ENC (audio_enc);
GstStructure *s;
GstCaps *caps, *filter_caps;
GstCaps *output_caps = NULL;
guint sampleframes_per_frame;
enc->rate = GST_AUDIO_INFO_RATE (info);
enc->channels = GST_AUDIO_INFO_CHANNELS (info);
/* negotiate output format based on downstream caps restrictions */
caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (enc));
if (caps == GST_CAPS_NONE || gst_caps_is_empty (caps))
goto failure;
if (caps == NULL)
caps = gst_static_pad_template_get_caps (&sbc_enc_src_factory);
/* fixate output caps */
filter_caps = gst_caps_new_simple ("audio/x-sbc", "rate", G_TYPE_INT,
enc->rate, "channels", G_TYPE_INT, enc->channels, NULL);
output_caps = gst_caps_intersect (caps, filter_caps);
gst_caps_unref (filter_caps);
if (output_caps == NULL || gst_caps_is_empty (output_caps)) {
GST_WARNING_OBJECT (enc, "Couldn't negotiate output caps with input rate "
"%d and input channels %d and allowed output caps %" GST_PTR_FORMAT,
enc->rate, enc->channels, caps);
goto failure;
}
gst_caps_unref (caps);
caps = NULL;
GST_DEBUG_OBJECT (enc, "fixating caps %" GST_PTR_FORMAT, output_caps);
output_caps = gst_caps_truncate (output_caps);
s = gst_caps_get_structure (output_caps, 0);
if (enc->channels == 1)
gst_structure_fixate_field_string (s, "channel-mode", "mono");
else
gst_structure_fixate_field_string (s, "channel-mode", "joint");
gst_structure_fixate_field_nearest_int (s, "bitpool", 64);
gst_structure_fixate_field_nearest_int (s, "blocks", 16);
gst_structure_fixate_field_nearest_int (s, "subbands", 8);
gst_structure_fixate_field_string (s, "allocation-method", "loudness");
s = NULL;
/* in case there's anything else left to fixate */
output_caps = gst_caps_fixate (output_caps);
gst_caps_set_simple (output_caps, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
GST_INFO_OBJECT (enc, "output caps %" GST_PTR_FORMAT, output_caps);
/* let's see what we fixated to */
s = gst_caps_get_structure (output_caps, 0);
gst_structure_get_int (s, "blocks", &enc->blocks);
gst_structure_get_int (s, "subbands", &enc->subbands);
gst_structure_get_int (s, "bitpool", &enc->bitpool);
allocation_method = gst_structure_get_string (s, "allocation-method");
channel_mode = gst_structure_get_string (s, "channel-mode");
/* We want channel-mode and channels coherent */
if (enc->channels == 1) {
if (g_strcmp0 (channel_mode, "mono") != 0) {
GST_ERROR_OBJECT (enc, "Can't have channel-mode '%s' for 1 channel",
channel_mode);
goto failure;
}
} else {
if (g_strcmp0 (channel_mode, "joint") != 0 &&
g_strcmp0 (channel_mode, "stereo") != 0 &&
g_strcmp0 (channel_mode, "dual") != 0) {
GST_ERROR_OBJECT (enc, "Can't have channel-mode '%s' for 2 channels",
channel_mode);
goto failure;
}
}
/* we want to be handed all available samples in handle_frame, but always
* enough to encode a frame */
sampleframes_per_frame = enc->blocks * enc->subbands;
gst_audio_encoder_set_frame_samples_min (audio_enc, sampleframes_per_frame);
gst_audio_encoder_set_frame_samples_max (audio_enc, sampleframes_per_frame);
gst_audio_encoder_set_frame_max (audio_enc, 0);
/* FIXME: what to do with left-over samples at the end? can we encode them? */
gst_audio_encoder_set_hard_min (audio_enc, TRUE);
/* and configure encoder based on the output caps we negotiated */
if (enc->rate == 16000)
enc->sbc.frequency = SBC_FREQ_16000;
else if (enc->rate == 32000)
enc->sbc.frequency = SBC_FREQ_32000;
else if (enc->rate == 44100)
enc->sbc.frequency = SBC_FREQ_44100;
else if (enc->rate == 48000)
enc->sbc.frequency = SBC_FREQ_48000;
else
goto failure;
if (enc->blocks == 4)
enc->sbc.blocks = SBC_BLK_4;
else if (enc->blocks == 8)
enc->sbc.blocks = SBC_BLK_8;
else if (enc->blocks == 12)
enc->sbc.blocks = SBC_BLK_12;
else if (enc->blocks == 16)
enc->sbc.blocks = SBC_BLK_16;
else
goto failure;
enc->sbc.subbands = (enc->subbands == 4) ? SBC_SB_4 : SBC_SB_8;
enc->sbc.bitpool = enc->bitpool;
if (channel_mode == NULL || allocation_method == NULL)
goto failure;
if (strcmp (channel_mode, "joint") == 0)
enc->sbc.mode = SBC_MODE_JOINT_STEREO;
else if (strcmp (channel_mode, "stereo") == 0)
enc->sbc.mode = SBC_MODE_STEREO;
else if (strcmp (channel_mode, "dual") == 0)
enc->sbc.mode = SBC_MODE_DUAL_CHANNEL;
else if (strcmp (channel_mode, "mono") == 0)
enc->sbc.mode = SBC_MODE_MONO;
else if (strcmp (channel_mode, "auto") == 0)
enc->sbc.mode = SBC_MODE_JOINT_STEREO;
else
goto failure;
if (strcmp (allocation_method, "loudness") == 0)
enc->sbc.allocation = SBC_AM_LOUDNESS;
else if (strcmp (allocation_method, "snr") == 0)
enc->sbc.allocation = SBC_AM_SNR;
else
goto failure;
if (!gst_audio_encoder_set_output_format (audio_enc, output_caps))
goto failure;
return gst_audio_encoder_negotiate (audio_enc);
failure:
if (output_caps)
gst_caps_unref (output_caps);
if (caps)
gst_caps_unref (caps);
return FALSE;
}
static gboolean
gst_ffmpegaudenc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
{
GstFFMpegAudEnc *ffmpegaudenc = (GstFFMpegAudEnc *) encoder;
GstCaps *other_caps;
GstCaps *allowed_caps;
GstCaps *icaps;
gsize frame_size;
GstFFMpegAudEncClass *oclass =
(GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc);
/* close old session */
if (ffmpegaudenc->opened) {
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
ffmpegaudenc->opened = FALSE;
if (avcodec_get_context_defaults3 (ffmpegaudenc->context,
oclass->in_plugin) < 0) {
GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to set context defaults");
return FALSE;
}
}
/* if we set it in _getcaps we should set it also in _link */
ffmpegaudenc->context->strict_std_compliance = ffmpegaudenc->compliance;
/* user defined properties */
if (ffmpegaudenc->bitrate > 0) {
GST_INFO_OBJECT (ffmpegaudenc, "Setting avcontext to bitrate %d",
ffmpegaudenc->bitrate);
ffmpegaudenc->context->bit_rate = ffmpegaudenc->bitrate;
ffmpegaudenc->context->bit_rate_tolerance = ffmpegaudenc->bitrate;
} else {
GST_INFO_OBJECT (ffmpegaudenc,
"Using avcontext default bitrate %" G_GINT64_FORMAT,
(gint64) ffmpegaudenc->context->bit_rate);
}
/* RTP payload used for GOB production (for Asterisk) */
if (ffmpegaudenc->rtp_payload_size) {
ffmpegaudenc->context->rtp_payload_size = ffmpegaudenc->rtp_payload_size;
}
/* some other defaults */
ffmpegaudenc->context->rc_strategy = 2;
ffmpegaudenc->context->b_frame_strategy = 0;
ffmpegaudenc->context->coder_type = 0;
ffmpegaudenc->context->context_model = 0;
ffmpegaudenc->context->scenechange_threshold = 0;
/* fetch pix_fmt and so on */
gst_ffmpeg_audioinfo_to_context (info, ffmpegaudenc->context);
if (!ffmpegaudenc->context->time_base.den) {
ffmpegaudenc->context->time_base.den = GST_AUDIO_INFO_RATE (info);
ffmpegaudenc->context->time_base.num = 1;
ffmpegaudenc->context->ticks_per_frame = 1;
}
if (ffmpegaudenc->context->channel_layout) {
gst_ffmpeg_channel_layout_to_gst (ffmpegaudenc->context->channel_layout,
ffmpegaudenc->context->channels, ffmpegaudenc->ffmpeg_layout);
ffmpegaudenc->needs_reorder =
(memcmp (ffmpegaudenc->ffmpeg_layout, info->position,
sizeof (GstAudioChannelPosition) *
ffmpegaudenc->context->channels) != 0);
}
/* some codecs support more than one format, first auto-choose one */
GST_DEBUG_OBJECT (ffmpegaudenc, "picking an output format ...");
allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (encoder));
if (!allowed_caps) {
GST_DEBUG_OBJECT (ffmpegaudenc, "... but no peer, using template caps");
/* we need to copy because get_allowed_caps returns a ref, and
* get_pad_template_caps doesn't */
allowed_caps =
gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SRC_PAD (encoder));
}
GST_DEBUG_OBJECT (ffmpegaudenc, "chose caps %" GST_PTR_FORMAT, allowed_caps);
gst_ffmpeg_caps_with_codecid (oclass->in_plugin->id,
oclass->in_plugin->type, allowed_caps, ffmpegaudenc->context);
/* open codec */
if (gst_ffmpeg_avcodec_open (ffmpegaudenc->context, oclass->in_plugin) < 0) {
gst_caps_unref (allowed_caps);
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
GST_DEBUG_OBJECT (ffmpegaudenc, "avenc_%s: Failed to open FFMPEG codec",
oclass->in_plugin->name);
if (avcodec_get_context_defaults3 (ffmpegaudenc->context,
oclass->in_plugin) < 0)
GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to set context defaults");
if ((oclass->in_plugin->capabilities & CODEC_CAP_EXPERIMENTAL) &&
ffmpegaudenc->compliance != GST_FFMPEG_EXPERIMENTAL) {
GST_ELEMENT_ERROR (ffmpegaudenc, LIBRARY, SETTINGS,
("Codec is experimental, but settings don't allow encoders to "
"produce output of experimental quality"),
("This codec may not create output that is conformant to the specs "
"or of good quality. If you must use it anyway, set the "
"compliance property to experimental"));
}
return FALSE;
}
/* try to set this caps on the other side */
other_caps = gst_ffmpeg_codecid_to_caps (oclass->in_plugin->id,
ffmpegaudenc->context, TRUE);
if (!other_caps) {
gst_caps_unref (allowed_caps);
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
GST_DEBUG ("Unsupported codec - no caps found");
if (avcodec_get_context_defaults3 (ffmpegaudenc->context,
oclass->in_plugin) < 0)
GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to set context defaults");
return FALSE;
}
icaps = gst_caps_intersect (allowed_caps, other_caps);
gst_caps_unref (allowed_caps);
gst_caps_unref (other_caps);
if (gst_caps_is_empty (icaps)) {
gst_caps_unref (icaps);
return FALSE;
}
icaps = gst_caps_fixate (icaps);
if (!gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (ffmpegaudenc),
icaps)) {
gst_ffmpeg_avcodec_close (ffmpegaudenc->context);
gst_caps_unref (icaps);
if (avcodec_get_context_defaults3 (ffmpegaudenc->context,
oclass->in_plugin) < 0)
GST_DEBUG_OBJECT (ffmpegaudenc, "Failed to set context defaults");
return FALSE;
}
gst_caps_unref (icaps);
frame_size = ffmpegaudenc->context->frame_size;
if (frame_size > 1) {
gst_audio_encoder_set_frame_samples_min (GST_AUDIO_ENCODER (ffmpegaudenc),
frame_size);
gst_audio_encoder_set_frame_samples_max (GST_AUDIO_ENCODER (ffmpegaudenc),
frame_size);
gst_audio_encoder_set_frame_max (GST_AUDIO_ENCODER (ffmpegaudenc), 1);
} else {
gst_audio_encoder_set_frame_samples_min (GST_AUDIO_ENCODER (ffmpegaudenc),
0);
gst_audio_encoder_set_frame_samples_max (GST_AUDIO_ENCODER (ffmpegaudenc),
0);
gst_audio_encoder_set_frame_max (GST_AUDIO_ENCODER (ffmpegaudenc), 0);
}
/* Store some tags */
{
GstTagList *tags = gst_tag_list_new_empty ();
const gchar *codec;
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE, GST_TAG_NOMINAL_BITRATE,
(guint) ffmpegaudenc->context->bit_rate, NULL);
if ((codec =
gst_ffmpeg_get_codecid_longname (ffmpegaudenc->context->codec_id)))
gst_tag_list_add (tags, GST_TAG_MERGE_REPLACE, GST_TAG_AUDIO_CODEC, codec,
NULL);
gst_audio_encoder_merge_tags (encoder, tags, GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (tags);
}
/* success! */
ffmpegaudenc->opened = TRUE;
return TRUE;
}
