pa_channel_map *
gst_pulse_gst_to_channel_map (pa_channel_map * map,
const GstRingBufferSpec * spec)
{
int i;
GstAudioChannelPosition *pos;
pa_channel_map_init (map);
if (!(pos =
gst_audio_get_channel_positions (gst_caps_get_structure (spec->caps,
0)))) {
return NULL;
}
for (i = 0; i < spec->channels; i++) {
if (pos[i] == GST_AUDIO_CHANNEL_POSITION_NONE) {
/* no valid mappings for these channels */
g_free (pos);
return NULL;
} else if (pos[i] < GST_AUDIO_CHANNEL_POSITION_NUM)
map->map[i] = gst_pos_to_pa[pos[i]];
else
map->map[i] = PA_CHANNEL_POSITION_INVALID;
}
g_free (pos);
map->channels = spec->channels;
if (!pa_channel_map_valid (map)) {
return NULL;
}
return map;
}
static gboolean
gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
{
GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
GstStructure *structure = gst_caps_get_structure (caps, 0);
GstAudioChannelPosition *pos;
if (!gst_structure_get_int (structure, "channels", &enc->channels) ||
!gst_structure_get_int (structure, "rate", &enc->samplerate) ||
!gst_structure_get_int (structure, "depth", &enc->depth)) {
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("got invalid caps: %" GST_PTR_FORMAT, caps));
gst_object_unref (enc);
return FALSE;
}
pos = gst_audio_get_channel_positions (structure);
/* If one channel is NONE they'll be all undefined */
if (pos != NULL && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE) {
g_free (pos);
pos = NULL;
}
if (pos == NULL) {
GST_ELEMENT_ERROR (enc, STREAM, FORMAT, (NULL),
("input has no valid channel layout"));
gst_object_unref (enc);
return FALSE;
}
enc->channel_mask =
gst_wavpack_get_channel_mask_from_positions (pos, enc->channels);
enc->need_channel_remap =
gst_wavpack_set_channel_mapping (pos, enc->channels,
enc->channel_mapping);
g_free (pos);
/* set fixed src pad caps now that we know what we will get */
caps = gst_caps_new_simple ("audio/x-wavpack",
"channels", G_TYPE_INT, enc->channels,
"rate", G_TYPE_INT, enc->samplerate,
"width", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
if (!gst_wavpack_set_channel_layout (caps, enc->channel_mask))
GST_WARNING_OBJECT (enc, "setting channel layout failed");
if (!gst_pad_set_caps (enc->srcpad, caps)) {
GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
("setting caps failed: %" GST_PTR_FORMAT, caps));
gst_caps_unref (caps);
gst_object_unref (enc);
return FALSE;
}
gst_pad_use_fixed_caps (enc->srcpad);
gst_caps_unref (caps);
gst_object_unref (enc);
return TRUE;
}
GstFlowReturn AudioFileReader::handleBuffer(GstAppSink* sink)
{
GstBuffer* buffer = gst_app_sink_pull_buffer(sink);
if (!buffer)
return GST_FLOW_ERROR;
GstCaps* caps = gst_buffer_get_caps(buffer);
GstStructure* structure = gst_caps_get_structure(caps, 0);
gint channels = 0;
if (!gst_structure_get_int(structure, "channels", &channels) || !channels) {
gst_caps_unref(caps);
gst_buffer_unref(buffer);
return GST_FLOW_ERROR;
}
gint sampleRate = 0;
if (!gst_structure_get_int(structure, "rate", &sampleRate) || !sampleRate) {
gst_caps_unref(caps);
gst_buffer_unref(buffer);
return GST_FLOW_ERROR;
}
gint width = 0;
if (!gst_structure_get_int(structure, "width", &width) || !width) {
gst_caps_unref(caps);
gst_buffer_unref(buffer);
return GST_FLOW_ERROR;
}
GstClockTime duration = (static_cast<guint64>(GST_BUFFER_SIZE(buffer)) * 8 * GST_SECOND) / (sampleRate * channels * width);
int frames = GST_CLOCK_TIME_TO_FRAMES(duration, sampleRate);
// Check the first audio channel. The buffer is supposed to store
// data of a single channel anyway.
GstAudioChannelPosition* positions = gst_audio_get_channel_positions(structure);
switch (positions[0]) {
case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
gst_buffer_list_iterator_add(m_frontLeftBuffersIterator, buffer);
m_channelSize += frames;
break;
case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
gst_buffer_list_iterator_add(m_frontRightBuffersIterator, buffer);
break;
default:
gst_buffer_unref(buffer);
break;
}
g_free(positions);
gst_caps_unref(caps);
return GST_FLOW_OK;
}
static gboolean
gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps)
{
GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
GstStructure *structure = gst_caps_get_structure (caps, 0);
/* Check if we can set the caps here already */
if (gst_structure_get_int (structure, "channels", &dec->channels) &&
gst_structure_get_int (structure, "rate", &dec->sample_rate) &&
gst_structure_get_int (structure, "width", &dec->depth)) {
GstCaps *caps;
GstAudioChannelPosition *pos;
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->sample_rate,
"channels", G_TYPE_INT, dec->channels,
"depth", G_TYPE_INT, dec->depth,
"width", G_TYPE_INT, 32,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
/* If we already have the channel layout set from upstream
* take this */
if (gst_structure_has_field (structure, "channel-positions")) {
pos = gst_audio_get_channel_positions (structure);
if (pos != NULL && dec->channels > 2) {
GstStructure *new_str = gst_caps_get_structure (caps, 0);
gst_audio_set_channel_positions (new_str, pos);
dec->channel_mask =
gst_wavpack_get_channel_mask_from_positions (pos, dec->channels);
}
if (pos != NULL)
g_free (pos);
}
GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
/* should always succeed */
gst_pad_set_caps (dec->srcpad, caps);
gst_caps_unref (caps);
/* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
* is decoded or after the format has changed */
gst_wavpack_dec_post_tags (dec);
}
gst_object_unref (dec);
return TRUE;
}
static gboolean query_data(struct input_handle* ih) {
GstBuffer *preroll;
GstCaps *src_caps;
GstStructure *s;
int i;
ih->n_channels = 0;
ih->sample_rate = 0;
ih->channel_positions = NULL;
preroll = gst_app_sink_pull_preroll(GST_APP_SINK(ih->appsink));
src_caps = gst_buffer_get_caps(preroll);
s = gst_caps_get_structure(src_caps, 0);
gst_structure_get_int(s, "rate", &(ih->sample_rate));
gst_structure_get_int(s, "channels", &(ih->n_channels));
if (!ih->sample_rate || !ih->n_channels) {
gst_caps_unref(src_caps);
gst_buffer_unref(preroll);
return FALSE;
}
ih->channel_positions = gst_audio_get_channel_positions(s);
if (verbose) {
if (ih->channel_positions) {
for (i = 0; i < ih->n_channels; ++i) {
printf("Channel %d: %d\n", i, ih->channel_positions[i]);
}
}
g_print ("%d channels @ %d Hz\n", ih->n_channels, ih->sample_rate);
}
gst_caps_unref(src_caps);
gst_buffer_unref(preroll);
return TRUE;
}
static gboolean
gst_deinterleave_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstDeinterleave *self;
GstCaps *srccaps;
GstStructure *s;
self = GST_DEINTERLEAVE (gst_pad_get_parent (pad));
GST_DEBUG_OBJECT (self, "got caps: %" GST_PTR_FORMAT, caps);
if (self->sinkcaps && !gst_caps_is_equal (caps, self->sinkcaps)) {
gint new_channels, i;
GstAudioChannelPosition *pos;
gboolean same_layout = TRUE;
s = gst_caps_get_structure (caps, 0);
/* We allow caps changes as long as the number of channels doesn't change
* and the channel positions stay the same. _getcaps() should've cared
* for this already but better be safe.
*/
if (!gst_structure_get_int (s, "channels", &new_channels) ||
new_channels != self->channels ||
!gst_deinterleave_set_process_function (self, caps))
goto cannot_change_caps;
/* Now check the channel positions. If we had no channel positions
* and get them or the other way around things have changed.
* If we had channel positions and get different ones things have
* changed too of course
*/
pos = gst_audio_get_channel_positions (s);
if ((pos && !self->pos) || (!pos && self->pos))
goto cannot_change_caps;
if (pos) {
for (i = 0; i < self->channels; i++) {
if (self->pos[i] != pos[i]) {
same_layout = FALSE;
break;
}
}
g_free (pos);
if (!same_layout)
goto cannot_change_caps;
}
} else {
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "channels", &self->channels))
goto no_channels;
if (!gst_deinterleave_set_process_function (self, caps))
goto unsupported_caps;
self->pos = gst_audio_get_channel_positions (s);
}
gst_caps_replace (&self->sinkcaps, caps);
/* Get srcpad caps */
srccaps = gst_caps_copy (caps);
s = gst_caps_get_structure (srccaps, 0);
gst_structure_set (s, "channels", G_TYPE_INT, 1, NULL);
gst_structure_remove_field (s, "channel-positions");
/* If we already have pads, update the caps otherwise
* add new pads */
if (self->srcpads) {
gst_deinterleave_set_pads_caps (self, srccaps);
} else {
gst_deinterleave_add_new_pads (self, srccaps);
}
gst_caps_unref (srccaps);
gst_object_unref (self);
return TRUE;
cannot_change_caps:
{
GST_ERROR_OBJECT (self, "can't set new caps: %" GST_PTR_FORMAT, caps);
gst_object_unref (self);
return FALSE;
}
unsupported_caps:
{
GST_ERROR_OBJECT (self, "caps not supported: %" GST_PTR_FORMAT, caps);
gst_object_unref (self);
return FALSE;
}
no_channels:
{
GST_ERROR_OBJECT (self, "invalid caps");
gst_object_unref (self);
return FALSE;
}
}
static gboolean
gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
{
GstRtpG722Pay *rtpg722pay;
GstStructure *structure;
gint rate, channels, clock_rate;
gboolean res;
gchar *params;
#if 0
GstAudioChannelPosition *pos;
const GstRTPChannelOrder *order;
#endif
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
rtpg722pay = GST_RTP_G722_PAY (basepayload);
structure = gst_caps_get_structure (caps, 0);
/* first parse input caps */
if (!gst_structure_get_int (structure, "rate", &rate))
goto no_rate;
if (!gst_structure_get_int (structure, "channels", &channels))
goto no_channels;
/* FIXME: Do something with the channel positions */
#if 0
/* get the channel order */
pos = gst_audio_get_channel_positions (structure);
if (pos)
order = gst_rtp_channels_get_by_pos (channels, pos);
else
order = NULL;
#endif
/* Clock rate is always 8000 Hz for G722 according to
* RFC 3551 although the sampling rate is 16000 Hz */
clock_rate = 8000;
gst_rtp_base_payload_set_options (basepayload, "audio",
basepayload->pt != GST_RTP_PAYLOAD_G722, "G722", clock_rate);
params = g_strdup_printf ("%d", channels);
#if 0
if (!order && channels > 2) {
GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE,
(NULL), ("Unknown channel order for %d channels", channels));
}
if (order && order->name) {
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, "channel-order", G_TYPE_STRING, order->name, NULL);
} else {
#endif
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, NULL);
#if 0
}
#endif
g_free (params);
#if 0
g_free (pos);
#endif
rtpg722pay->rate = rate;
rtpg722pay->channels = channels;
/* bits-per-sample is 4 * channels for G722, but as the RTP clock runs at
* half speed (8 instead of 16 khz), pretend it's 8 bits per sample
* channels. */
gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
8 * rtpg722pay->channels);
return res;
/* ERRORS */
no_rate:
{
GST_DEBUG_OBJECT (rtpg722pay, "no rate given");
return FALSE;
}
no_channels:
{
GST_DEBUG_OBJECT (rtpg722pay, "no channels given");
return FALSE;
}
}
static gboolean
gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
{
GstRtpG722Pay *rtpg722pay;
GstStructure *structure;
gint rate, channels, clock_rate;
gboolean res;
gchar *params;
GstAudioChannelPosition *pos;
const GstRTPChannelOrder *order;
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
rtpg722pay = GST_RTP_G722_PAY (basepayload);
structure = gst_caps_get_structure (caps, 0);
/* first parse input caps */
if (!gst_structure_get_int (structure, "rate", &rate))
goto no_rate;
if (!gst_structure_get_int (structure, "channels", &channels))
goto no_channels;
/* get the channel order */
pos = gst_audio_get_channel_positions (structure);
if (pos)
order = gst_rtp_channels_get_by_pos (channels, pos);
else
order = NULL;
/* Clock rate is always 8000 Hz for G722 according to
* RFC 3551 although the sampling rate is 16000 Hz */
clock_rate = 8000;
gst_basertppayload_set_options (basepayload, "audio", TRUE, "G722",
clock_rate);
params = g_strdup_printf ("%d", channels);
if (!order && channels > 2) {
GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE,
(NULL), ("Unknown channel order for %d channels", channels));
}
if (order && order->name) {
res = gst_basertppayload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, "channel-order", G_TYPE_STRING, order->name, NULL);
} else {
res = gst_basertppayload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, NULL);
}
g_free (params);
g_free (pos);
rtpg722pay->rate = rate;
rtpg722pay->channels = channels;
/* octet-per-sample is 1 * channels for G722 */
gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
4 * rtpg722pay->channels);
return res;
/* ERRORS */
no_rate:
{
GST_DEBUG_OBJECT (rtpg722pay, "no rate given");
return FALSE;
}
no_channels:
{
GST_DEBUG_OBJECT (rtpg722pay, "no channels given");
return FALSE;
}
}
gboolean
gst_core_audio_set_channels_layout (GstCoreAudio * core_audio,
gint channels, GstCaps * caps)
{
/* Configure the output stream and allocate ringbuffer memory */
AudioChannelLayout *layout = NULL;
OSStatus status;
int layoutSize, element, i;
AudioUnitScope scope;
GstStructure *structure;
GstAudioChannelPosition *positions;
/* Describe channels */
layoutSize = sizeof (AudioChannelLayout) +
channels * sizeof (AudioChannelDescription);
layout = g_malloc (layoutSize);
structure = gst_caps_get_structure (caps, 0);
positions = gst_audio_get_channel_positions (structure);
layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions;
layout->mChannelBitmap = 0; /* Not used */
layout->mNumberChannelDescriptions = channels;
for (i = 0; i < channels; i++) {
if (positions) {
layout->mChannelDescriptions[i].mChannelLabel =
gst_audio_channel_position_to_coreaudio_channel_label (positions[i],
i);
} else {
/* Discrete channel numbers are ORed into this */
layout->mChannelDescriptions[i].mChannelLabel =
kAudioChannelLabel_Discrete_0 | i;
}
/* Others unused */
layout->mChannelDescriptions[i].mChannelFlags = 0;
layout->mChannelDescriptions[i].mCoordinates[0] = 0.f;
layout->mChannelDescriptions[i].mCoordinates[1] = 0.f;
layout->mChannelDescriptions[i].mCoordinates[2] = 0.f;
}
if (positions) {
g_free (positions);
positions = NULL;
}
scope = core_audio->is_src ? kAudioUnitScope_Output : kAudioUnitScope_Input;
element = core_audio->is_src ? 1 : 0;
if (layoutSize) {
status = AudioUnitSetProperty (core_audio->audiounit,
kAudioUnitProperty_AudioChannelLayout,
scope, element, layout, layoutSize);
if (status) {
GST_WARNING_OBJECT (core_audio->osxbuf,
"Failed to set output channel layout: %" GST_FOURCC_FORMAT,
GST_FOURCC_ARGS (status));
return FALSE;
}
}
g_free (layout);
return TRUE;
}
