static int
gst_opus_enc_find_channel_position (GstOpusEnc * enc, const GstAudioInfo * info,
GstAudioChannelPosition position)
{
int n;
for (n = 0; n < enc->n_channels; ++n) {
if (GST_AUDIO_INFO_POSITION (info, n) == position) {
return n;
}
}
return -1;
}
static void
gst_deinterleave_set_pads_caps (GstDeinterleave * self, GstCaps * caps)
{
GList *l;
gint i;
for (l = self->srcpads, i = 0; l; l = l->next, i++) {
GstPad *pad = GST_PAD (l->data);
GstCaps *srccaps;
GstAudioInfo info;
gst_audio_info_from_caps (&info, caps);
if (self->keep_positions)
GST_AUDIO_INFO_POSITION (&info, 0) =
GST_AUDIO_INFO_POSITION (&self->audio_info, i);
srccaps = gst_audio_info_to_caps (&info);
gst_pad_set_caps (pad, srccaps);
gst_caps_unref (srccaps);
}
}
static void
gst_deinterleave_add_new_pads (GstDeinterleave * self, GstCaps * caps)
{
GstPad *pad;
guint i;
for (i = 0; i < GST_AUDIO_INFO_CHANNELS (&self->audio_info); i++) {
gchar *name = g_strdup_printf ("src_%u", i);
GstCaps *srccaps;
GstAudioInfo info;
GstAudioFormat format = GST_AUDIO_INFO_FORMAT (&self->audio_info);
gint rate = GST_AUDIO_INFO_RATE (&self->audio_info);
GstAudioChannelPosition position = GST_AUDIO_CHANNEL_POSITION_MONO;
CopyStickyEventsData data;
/* Set channel position if we know it */
if (self->keep_positions)
position = GST_AUDIO_INFO_POSITION (&self->audio_info, i);
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, format, rate, 1, &position);
srccaps = gst_audio_info_to_caps (&info);
pad = gst_pad_new_from_static_template (&src_template, name);
g_free (name);
gst_pad_use_fixed_caps (pad);
gst_pad_set_query_function (pad,
GST_DEBUG_FUNCPTR (gst_deinterleave_src_query));
gst_pad_set_active (pad, TRUE);
data.pad = pad;
data.caps = srccaps;
gst_pad_sticky_events_foreach (self->sink, copy_sticky_events, &data);
if (data.caps)
gst_pad_set_caps (pad, data.caps);
gst_element_add_pad (GST_ELEMENT (self), pad);
self->srcpads = g_list_prepend (self->srcpads, gst_object_ref (pad));
gst_caps_unref (srccaps);
}
gst_element_no_more_pads (GST_ELEMENT (self));
self->srcpads = g_list_reverse (self->srcpads);
}
GstFlowReturn AudioSourceProviderGStreamer::handleAudioBuffer(GstAppSink* sink)
{
if (!m_client)
return GST_FLOW_OK;
// Pull a buffer from appsink and store it the appropriate buffer
// list for the audio channel it represents.
GRefPtr<GstSample> sample = adoptGRef(gst_app_sink_pull_sample(sink));
if (!sample)
return gst_app_sink_is_eos(sink) ? GST_FLOW_EOS : GST_FLOW_ERROR;
GstBuffer* buffer = gst_sample_get_buffer(sample.get());
if (!buffer)
return GST_FLOW_ERROR;
GstCaps* caps = gst_sample_get_caps(sample.get());
if (!caps)
return GST_FLOW_ERROR;
GstAudioInfo info;
gst_audio_info_from_caps(&info, caps);
WTF::GMutexLocker<GMutex> lock(m_adapterMutex);
// Check the first audio channel. The buffer is supposed to store
// data of a single channel anyway.
switch (GST_AUDIO_INFO_POSITION(&info, 0)) {
case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
case GST_AUDIO_CHANNEL_POSITION_MONO:
gst_adapter_push(m_frontLeftAdapter, gst_buffer_ref(buffer));
break;
case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
gst_adapter_push(m_frontRightAdapter, gst_buffer_ref(buffer));
break;
default:
break;
}
return GST_FLOW_OK;
}
GstFlowReturn AudioFileReader::handleSample(GstAppSink* sink)
{
GstSample* sample = gst_app_sink_pull_sample(sink);
if (!sample)
return GST_FLOW_ERROR;
GstBuffer* buffer = gst_sample_get_buffer(sample);
if (!buffer) {
gst_sample_unref(sample);
return GST_FLOW_ERROR;
}
GstCaps* caps = gst_sample_get_caps(sample);
if (!caps) {
gst_sample_unref(sample);
return GST_FLOW_ERROR;
}
GstAudioInfo info;
gst_audio_info_from_caps(&info, caps);
int frames = gst_buffer_get_size(buffer) / info.bpf;
// Check the first audio channel. The buffer is supposed to store
// data of a single channel anyway.
switch (GST_AUDIO_INFO_POSITION(&info, 0)) {
case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
case GST_AUDIO_CHANNEL_POSITION_MONO:
gst_buffer_list_add(m_frontLeftBuffers, gst_buffer_ref(buffer));
m_channelSize += frames;
break;
case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
gst_buffer_list_add(m_frontRightBuffers, gst_buffer_ref(buffer));
break;
default:
break;
}
gst_sample_unref(sample);
return GST_FLOW_OK;
}
GstBuffer* AudioLiveInputPipeline::pullNewBuffer(GstAppSink* sink)
{
GstSample* sample = gst_app_sink_pull_sample(sink);
if (!sample)
return nullptr;
GstBuffer* buffer = gst_sample_get_buffer(sample);
if (!buffer) {
gst_sample_unref(sample);
return nullptr;
}
GstCaps* caps = gst_sample_get_caps(sample);
if (!caps) {
gst_sample_unref(sample);
return nullptr;
}
GstAudioInfo info;
gst_audio_info_from_caps(&info, caps);
// Check the first audio channel.
// The buffer is supposed to store data of a single channel anyway.
switch (GST_AUDIO_INFO_POSITION(&info, 0)) {
case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
// Transferring ownership of buffer to the calling object.
gst_buffer_ref(buffer);
break;
default:
buffer = nullptr;
break;
}
gst_sample_unref(sample);
return buffer;
}
static void
gst_opus_enc_setup_channel_mappings (GstOpusEnc * enc,
const GstAudioInfo * info)
{
#define MAPS(idx,pos) (GST_AUDIO_INFO_POSITION (info, (idx)) == GST_AUDIO_CHANNEL_POSITION_##pos)
int n;
GST_DEBUG_OBJECT (enc, "Setting up channel mapping for %d channels",
enc->n_channels);
/* Start by setting up a default trivial mapping */
enc->n_stereo_streams = 0;
gst_opus_enc_setup_trivial_mapping (enc, enc->encoding_channel_mapping);
gst_opus_enc_setup_trivial_mapping (enc, enc->decoding_channel_mapping);
/* For one channel, use the basic RTP mapping */
if (enc->n_channels == 1) {
GST_INFO_OBJECT (enc, "Mono, trivial RTP mapping");
enc->channel_mapping_family = 0;
/* implicit mapping for family 0 */
return;
}
/* For two channels, use the basic RTP mapping if the channels are
mapped as left/right. */
if (enc->n_channels == 2) {
if (MAPS (0, FRONT_LEFT) && MAPS (1, FRONT_RIGHT)) {
GST_INFO_OBJECT (enc, "Stereo, canonical mapping");
enc->channel_mapping_family = 0;
enc->n_stereo_streams = 1;
/* The channel mapping is implicit for family 0, that's why we do not
attempt to create one for right/left - this will be mapped to the
Vorbis mapping below. */
return;
} else {
GST_DEBUG_OBJECT (enc, "Stereo, but not canonical mapping, continuing");
}
}
/* For channels between 1 and 8, we use the Vorbis mapping if we can
find a permutation that matches it. Mono will have been taken care
of earlier, but this code also handles it. Same for left/right stereo.
There are two mappings. One maps the input channels to an ordering
which has the natural pairs first so they can benefit from the Opus
stereo channel coupling, and the other maps this ordering to the
Vorbis ordering. */
if (enc->n_channels >= 1 && enc->n_channels <= 8) {
int c0, c1, c0v, c1v;
int mapped;
gboolean positions_done[256];
static const GstAudioChannelPosition pairs[][2] = {
{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_FRONT_LEFT_OF_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER},
{GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER},
{GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT},
};
size_t pair;
GST_DEBUG_OBJECT (enc,
"In range for the Vorbis mapping, building channel mapping tables");
enc->n_stereo_streams = 0;
mapped = 0;
for (n = 0; n < 256; ++n)
positions_done[n] = FALSE;
/* First, find any natural pairs, and move them to the front */
for (pair = 0; pair < G_N_ELEMENTS (pairs); ++pair) {
GstAudioChannelPosition p0 = pairs[pair][0];
GstAudioChannelPosition p1 = pairs[pair][1];
c0 = gst_opus_enc_find_channel_position (enc, info, p0);
c1 = gst_opus_enc_find_channel_position (enc, info, p1);
if (c0 >= 0 && c1 >= 0) {
/* We found a natural pair */
GST_DEBUG_OBJECT (enc, "Natural pair '%s/%s' found at %d %d",
gst_opus_channel_names[p0], gst_opus_channel_names[p1], c0, c1);
/* Find where they map in Vorbis order */
c0v = gst_opus_enc_find_channel_position_in_vorbis_order (enc, p0);
c1v = gst_opus_enc_find_channel_position_in_vorbis_order (enc, p1);
if (c0v < 0 || c1v < 0) {
GST_WARNING_OBJECT (enc,
"Cannot map channel positions to Vorbis order, using unknown mapping");
enc->channel_mapping_family = 255;
enc->n_stereo_streams = 0;
return;
}
enc->encoding_channel_mapping[mapped] = c0;
enc->encoding_channel_mapping[mapped + 1] = c1;
enc->decoding_channel_mapping[c0v] = mapped;
enc->decoding_channel_mapping[c1v] = mapped + 1;
enc->n_stereo_streams++;
mapped += 2;
positions_done[p0] = positions_done[p1] = TRUE;
}
}
/* Now add all other input channels as mono streams */
for (n = 0; n < enc->n_channels; ++n) {
GstAudioChannelPosition position = GST_AUDIO_INFO_POSITION (info, n);
/* if we already mapped it while searching for pairs, nothing else
needs to be done */
if (!positions_done[position]) {
int cv;
GST_DEBUG_OBJECT (enc, "Channel position %s is not mapped yet, adding",
gst_opus_channel_names[position]);
cv = gst_opus_enc_find_channel_position_in_vorbis_order (enc, position);
if (cv < 0) {
GST_WARNING_OBJECT (enc,
"Cannot map channel positions to Vorbis order, using unknown mapping");
enc->channel_mapping_family = 255;
enc->n_stereo_streams = 0;
return;
}
enc->encoding_channel_mapping[mapped] = n;
enc->decoding_channel_mapping[cv] = mapped;
mapped++;
}
}
#ifndef GST_DISABLE_DEBUG
GST_INFO_OBJECT (enc,
"Mapping tables built: %d channels, %d stereo streams", enc->n_channels,
enc->n_stereo_streams);
gst_opus_common_log_channel_mapping_table (GST_ELEMENT (enc), opusenc_debug,
"Encoding mapping table", enc->n_channels,
enc->encoding_channel_mapping);
gst_opus_common_log_channel_mapping_table (GST_ELEMENT (enc), opusenc_debug,
"Decoding mapping table", enc->n_channels,
enc->decoding_channel_mapping);
#endif
enc->channel_mapping_family = 1;
return;
}
/* More than 8 channels, if future mappings are added for those */
/* For other cases, we use undefined, with the default trivial mapping
and all mono streams */
GST_WARNING_OBJECT (enc, "Unknown mapping");
enc->channel_mapping_family = 255;
enc->n_stereo_streams = 0;
#undef MAPS
}
static gboolean
gst_interleave_sink_event (GstCollectPads * pads, GstCollectData * data,
GstEvent * event, gpointer user_data)
{
GstInterleave *self = GST_INTERLEAVE (user_data);
gboolean ret = TRUE;
GST_DEBUG ("Got %s event on pad %s:%s", GST_EVENT_TYPE_NAME (event),
GST_DEBUG_PAD_NAME (data->pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
GST_OBJECT_LOCK (self);
gst_event_replace (&self->pending_segment, NULL);
GST_OBJECT_UNLOCK (self);
break;
case GST_EVENT_SEGMENT:
{
GST_OBJECT_LOCK (self);
gst_event_replace (&self->pending_segment, event);
GST_OBJECT_UNLOCK (self);
break;
}
case GST_EVENT_CAPS:
{
GstCaps *caps;
GstAudioInfo info;
GValue *val;
guint channel;
gst_event_parse_caps (event, &caps);
if (!gst_audio_info_from_caps (&info, caps)) {
GST_WARNING_OBJECT (self, "invalid sink caps");
gst_event_unref (event);
event = NULL;
ret = FALSE;
break;
}
if (self->channel_positions_from_input
&& GST_AUDIO_INFO_CHANNELS (&info) == 1) {
channel = GST_INTERLEAVE_PAD_CAST (data->pad)->channel;
val = g_value_array_get_nth (self->input_channel_positions, channel);
g_value_set_enum (val, GST_AUDIO_INFO_POSITION (&info, 0));
}
if (!gst_pad_has_current_caps (data->pad))
g_atomic_int_add (&self->configured_sinkpads_counter, 1);
/* Last caps that are set on a sink pad are used as output caps */
if (g_atomic_int_get (&self->configured_sinkpads_counter) ==
self->channels) {
ret = gst_interleave_sink_setcaps (self, data->pad, caps, &info);
gst_event_unref (event);
event = NULL;
}
break;
}
case GST_EVENT_TAG:
GST_FIXME_OBJECT (self, "FIXME: merge tags and send after stream-start");
break;
default:
break;
}
/* now GstCollectPads can take care of the rest, e.g. EOS */
if (event != NULL)
return gst_collect_pads_event_default (pads, data, event, FALSE);
return ret;
}
static GstFlowReturn
gst_fdkaacenc_handle_frame (GstAudioEncoder * enc, GstBuffer * inbuf)
{
GstFdkAacEnc *self = GST_FDKAACENC (enc);
GstFlowReturn ret = GST_FLOW_OK;
GstAudioInfo *info;
GstMapInfo imap, omap;
GstBuffer *outbuf;
AACENC_BufDesc in_desc = { 0 };
AACENC_BufDesc out_desc = { 0 };
AACENC_InArgs in_args = { 0 };
AACENC_OutArgs out_args = { 0 };
gint in_id = IN_AUDIO_DATA, out_id = OUT_BITSTREAM_DATA;
gint in_sizes, out_sizes;
gint in_el_sizes, out_el_sizes;
AACENC_ERROR err;
info = gst_audio_encoder_get_audio_info (enc);
if (inbuf) {
if (self->need_reorder) {
inbuf = gst_buffer_copy (inbuf);
gst_buffer_map (inbuf, &imap, GST_MAP_READWRITE);
gst_audio_reorder_channels (imap.data, imap.size,
GST_AUDIO_INFO_FORMAT (info), GST_AUDIO_INFO_CHANNELS (info),
&GST_AUDIO_INFO_POSITION (info, 0), self->aac_positions);
} else {
gst_buffer_map (inbuf, &imap, GST_MAP_READ);
}
in_args.numInSamples = imap.size / GST_AUDIO_INFO_BPS (info);
in_sizes = imap.size;
in_el_sizes = GST_AUDIO_INFO_BPS (info);
in_desc.numBufs = 1;
} else {
in_args.numInSamples = -1;
in_sizes = 0;
in_el_sizes = 0;
in_desc.numBufs = 0;
}
in_desc.bufferIdentifiers = &in_id;
in_desc.bufs = (void *) &imap.data;
in_desc.bufSizes = &in_sizes;
in_desc.bufElSizes = &in_el_sizes;
outbuf = gst_audio_encoder_allocate_output_buffer (enc, self->outbuf_size);
if (!outbuf) {
ret = GST_FLOW_ERROR;
goto out;
}
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_sizes = omap.size;
out_el_sizes = 1;
out_desc.bufferIdentifiers = &out_id;
out_desc.numBufs = 1;
out_desc.bufs = (void *) &omap.data;
out_desc.bufSizes = &out_sizes;
out_desc.bufElSizes = &out_el_sizes;
err = aacEncEncode (self->enc, &in_desc, &out_desc, &in_args, &out_args);
if (err == AACENC_ENCODE_EOF && !inbuf)
goto out;
else if (err != AACENC_OK) {
GST_ERROR_OBJECT (self, "Failed to encode data: %d", err);
ret = GST_FLOW_ERROR;
goto out;
}
if (inbuf) {
gst_buffer_unmap (inbuf, &imap);
if (self->need_reorder)
gst_buffer_unref (inbuf);
inbuf = NULL;
}
if (!out_args.numOutBytes)
goto out;
gst_buffer_unmap (outbuf, &omap);
gst_buffer_set_size (outbuf, out_args.numOutBytes);
ret = gst_audio_encoder_finish_frame (enc, outbuf, self->samples_per_frame);
outbuf = NULL;
out:
if (outbuf) {
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unref (outbuf);
}
if (inbuf) {
gst_buffer_unmap (inbuf, &imap);
if (self->need_reorder)
gst_buffer_unref (inbuf);
}
return ret;
}
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_raw_audio_parse_caps_to_config (GstRawAudioParse * raw_audio_parse,
GstCaps * caps, GstRawAudioParseConfig * config)
{
gboolean ret = FALSE;
GstStructure *structure;
/* Caps might get copied, and the copy needs to be unref'd.
* Also, the caller retains ownership over the original caps.
* So, to make this mechanism also work with cases where the
* caps are *not* copied, ref the original caps here first. */
gst_caps_ref (caps);
structure = gst_caps_get_structure (caps, 0);
/* For unaligned raw data, the output caps stay the same,
* except that audio/x-unaligned-raw becomes audio/x-raw,
* since the parser aligns the sample data */
if (gst_structure_has_name (structure, "audio/x-unaligned-raw")) {
/* Copy the caps to be able to modify them */
GstCaps *new_caps = gst_caps_copy (caps);
gst_caps_unref (caps);
caps = new_caps;
/* Change the media type to audio/x-raw , otherwise
* gst_audio_info_from_caps() won't work */
structure = gst_caps_get_structure (caps, 0);
gst_structure_set_name (structure, "audio/x-raw");
}
if (gst_structure_has_name (structure, "audio/x-raw")) {
guint num_channels;
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps)) {
GST_ERROR_OBJECT (raw_audio_parse,
"failed to parse caps %" GST_PTR_FORMAT, (gpointer) caps);
goto done;
}
num_channels = GST_AUDIO_INFO_CHANNELS (&info);
config->format = GST_RAW_AUDIO_PARSE_FORMAT_PCM;
config->pcm_format = GST_AUDIO_INFO_FORMAT (&info);
config->bpf = GST_AUDIO_INFO_BPF (&info);
config->sample_rate = GST_AUDIO_INFO_RATE (&info);
config->interleaved =
(GST_AUDIO_INFO_LAYOUT (&info) == GST_AUDIO_LAYOUT_INTERLEAVED);
gst_raw_audio_parse_set_config_channels (config, num_channels, 0, FALSE);
memcpy (config->channel_positions, &(GST_AUDIO_INFO_POSITION (&info, 0)),
sizeof (GstAudioChannelPosition) * num_channels);
} else if (gst_structure_has_name (structure, "audio/x-alaw")
|| gst_structure_has_name (structure, "audio/x-mulaw")) {
gint i;
guint64 channel_mask;
guint num_channels;
config->format =
gst_structure_has_name (structure,
"audio/x-alaw") ? GST_RAW_AUDIO_PARSE_FORMAT_ALAW :
GST_RAW_AUDIO_PARSE_FORMAT_MULAW;
if (!gst_structure_get_int (structure, "rate", &i)) {
GST_ERROR_OBJECT (raw_audio_parse,
"missing rate value in caps %" GST_PTR_FORMAT, (gpointer) caps);
goto done;
}
config->sample_rate = i;
if (!gst_structure_get_int (structure, "channels", &i)) {
GST_ERROR_OBJECT (raw_audio_parse,
"missing channels value in caps %" GST_PTR_FORMAT, (gpointer) caps);
goto done;
}
num_channels = i;
if (!gst_structure_get (structure, "channel-mask", GST_TYPE_BITMASK,
&channel_mask, NULL)) {
channel_mask = gst_audio_channel_get_fallback_mask (num_channels);
GST_DEBUG_OBJECT (raw_audio_parse,
"input caps have no channel mask - using fallback mask %#"
G_GINT64_MODIFIER "x for %u channels", channel_mask, num_channels);
}
if (!gst_raw_audio_parse_set_config_channels (config, num_channels,
channel_mask, TRUE)) {
GST_ERROR_OBJECT (raw_audio_parse,
"could not use channel mask %#" G_GINT64_MODIFIER
"x for channel positions", channel_mask);
goto done;
}
/* A-law and mu-law both use 1 byte per sample */
config->bpf = 1 * num_channels;
} else {
GST_ERROR_OBJECT (raw_audio_parse,
"caps %" GST_PTR_FORMAT " have an unsupported media type",
(gpointer) caps);
goto done;
}
ret = TRUE;
done:
gst_caps_unref (caps);
if (ret)
config->ready = TRUE;
return ret;
}
