static GstCaps *
gst_inter_audio_src_get_caps (GstBaseSrc * src, GstCaps * filter)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GstCaps *caps;
GST_DEBUG_OBJECT (interaudiosrc, "get_caps");
if (!interaudiosrc->surface)
return GST_BASE_SRC_CLASS (parent_class)->get_caps (src, filter);
g_mutex_lock (&interaudiosrc->surface->mutex);
if (interaudiosrc->surface->audio_info.finfo) {
caps = gst_audio_info_to_caps (&interaudiosrc->surface->audio_info);
if (filter) {
GstCaps *tmp;
tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = tmp;
}
} else {
caps = NULL;
}
g_mutex_unlock (&interaudiosrc->surface->mutex);
if (caps)
return caps;
else
return GST_BASE_SRC_CLASS (parent_class)->get_caps (src, filter);
}
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);
}
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);
}
}
/* Converts an AudioStreamBasicDescription to preferred caps.
*
* These caps will indicate the AU element's canonical format, which won't
* make Core Audio resample nor convert.
*
* NOTE ON MULTI-CHANNEL AUDIO:
*
* If layout is not NULL, resulting caps will only include the subset
* of channels supported by GStreamer. If the Core Audio layout contained
* ANY positioned channels, then ONLY positioned channels will be included
* in the resulting caps. Otherwise, resulting caps will be unpositioned,
* and include only unpositioned channels.
* (Channels with unsupported AudioChannelLabel will be skipped either way.)
*
* Naturally, the number of channels indicated by 'channels' can be lower
* than the AU element's total number of channels.
*/
GstCaps *
gst_core_audio_asbd_to_caps (AudioStreamBasicDescription * asbd,
AudioChannelLayout * layout)
{
GstAudioInfo info;
GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN;
guint rate, channels, bps, endianness;
guint64 channel_mask;
gboolean sign, interleaved;
GstAudioChannelPosition pos[GST_OSX_AUDIO_MAX_CHANNEL];
if (asbd->mFormatID != kAudioFormatLinearPCM) {
GST_WARNING ("Only linear PCM is supported");
goto error;
}
if (!(asbd->mFormatFlags & kAudioFormatFlagIsPacked)) {
GST_WARNING ("Only packed formats supported");
goto error;
}
if (asbd->mFormatFlags & kLinearPCMFormatFlagsSampleFractionMask) {
GST_WARNING ("Fixed point audio is unsupported");
goto error;
}
rate = asbd->mSampleRate;
if (rate == kAudioStreamAnyRate) {
GST_WARNING ("No sample rate");
goto error;
}
bps = asbd->mBitsPerChannel;
endianness = asbd->mFormatFlags & kAudioFormatFlagIsBigEndian ?
G_BIG_ENDIAN : G_LITTLE_ENDIAN;
sign = asbd->mFormatID & kAudioFormatFlagIsSignedInteger ? TRUE : FALSE;
interleaved = asbd->mFormatFlags & kAudioFormatFlagIsNonInterleaved ?
TRUE : FALSE;
if (asbd->mFormatFlags & kAudioFormatFlagIsFloat) {
if (bps == 32) {
if (endianness == G_LITTLE_ENDIAN)
format = GST_AUDIO_FORMAT_F32LE;
else
format = GST_AUDIO_FORMAT_F32BE;
} else if (bps == 64) {
if (endianness == G_LITTLE_ENDIAN)
format = GST_AUDIO_FORMAT_F64LE;
else
format = GST_AUDIO_FORMAT_F64BE;
}
} else {
format = gst_audio_format_build_integer (sign, endianness, bps, bps);
}
if (format == GST_AUDIO_FORMAT_UNKNOWN) {
GST_WARNING ("Unsupported sample format");
goto error;
}
if (layout) {
if (!gst_core_audio_parse_channel_layout (layout, &channels, &channel_mask,
pos)) {
GST_WARNING
("Failed to parse channel layout, best effort channels layout mapping will be used");
layout = NULL;
}
}
if (layout) {
/* The AU can have arbitrary channel order, but we're using GstAudioInfo
* which supports only the GStreamer channel order.
* Also, we're eventually producing caps, which only have channel-mask
* (whose implied order is the GStreamer channel order). */
gst_audio_channel_positions_to_valid_order (pos, channels);
gst_audio_info_set_format (&info, format, rate, channels, pos);
} else {
channels = MIN (asbd->mChannelsPerFrame, GST_OSX_AUDIO_MAX_CHANNEL);
gst_audio_info_set_format (&info, format, rate, channels, NULL);
}
return gst_audio_info_to_caps (&info);
error:
return NULL;
}
static GstCaps *
gst_audio_parse_get_caps (GstRawParse * rp)
{
GstAudioParse *ap = GST_AUDIO_PARSE (rp);
GstCaps *caps, *ncaps;
GstAudioInfo info;
gint fps_n, fps_d;
const GValue *val;
if (ap->use_sink_caps) {
gint rate;
GstCaps *caps = gst_pad_get_current_caps (rp->sinkpad);
gst_audio_info_from_caps (&info, caps);
rate = GST_AUDIO_INFO_RATE (&info);
gst_raw_parse_set_fps (GST_RAW_PARSE (ap), rate, 1);
return caps;
}
gst_raw_parse_get_fps (rp, &fps_n, &fps_d);
gst_audio_parse_setup_channel_positions (ap);
/* yes, even when format not raw */
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, ap->raw_format, fps_n, ap->channels,
ap->channel_order);
info.layout = ap->interleaved ? GST_AUDIO_LAYOUT_INTERLEAVED :
GST_AUDIO_LAYOUT_NON_INTERLEAVED;
caps = gst_audio_info_to_caps (&info);
switch (ap->format) {
case GST_AUDIO_PARSE_FORMAT_RAW:
break;
case GST_AUDIO_PARSE_FORMAT_ALAW:
ncaps = gst_caps_new_simple ("audio/x-alaw",
"rate", G_TYPE_INT, fps_n,
"channels", G_TYPE_INT, ap->channels, NULL);
/* pick mask stuff from faked raw format */
val = gst_structure_get_value (gst_caps_get_structure (caps, 0),
"channel-mask");
if (val)
gst_caps_set_value (ncaps, "channel-mask", val);
gst_caps_unref (caps);
caps = ncaps;
break;
case GST_AUDIO_PARSE_FORMAT_MULAW:
ncaps = gst_caps_new_simple ("audio/x-mulaw",
"rate", G_TYPE_INT, fps_n,
"channels", G_TYPE_INT, ap->channels, NULL);
/* pick mask stuff from faked raw format */
val = gst_structure_get_value (gst_caps_get_structure (caps, 0),
"channel-mask");
if (val)
gst_caps_set_value (ncaps, "channel-mask", val);
gst_caps_unref (caps);
caps = ncaps;
break;
default:
caps = gst_caps_new_empty ();
GST_ERROR_OBJECT (rp, "unexpected format %d", ap->format);
break;
}
return caps;
}
static gboolean
gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpL16Depay *rtpL16depay;
gint clock_rate, payload;
gint channels;
GstCaps *srccaps;
gboolean res;
const gchar *channel_order;
const GstRTPChannelOrder *order;
GstAudioInfo *info;
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
payload = 96;
gst_structure_get_int (structure, "payload", &payload);
switch (payload) {
case GST_RTP_PAYLOAD_L16_STEREO:
channels = 2;
clock_rate = 44100;
break;
case GST_RTP_PAYLOAD_L16_MONO:
channels = 1;
clock_rate = 44100;
break;
default:
/* no fixed mapping, we need clock-rate */
channels = 0;
clock_rate = 0;
break;
}
/* caps can overwrite defaults */
clock_rate =
gst_rtp_L16_depay_parse_int (structure, "clock-rate", clock_rate);
if (clock_rate == 0)
goto no_clockrate;
channels =
gst_rtp_L16_depay_parse_int (structure, "encoding-params", channels);
if (channels == 0) {
channels = gst_rtp_L16_depay_parse_int (structure, "channels", channels);
if (channels == 0) {
/* channels defaults to 1 otherwise */
channels = 1;
}
}
depayload->clock_rate = clock_rate;
info = &rtpL16depay->info;
gst_audio_info_init (info);
info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S16BE);
info->rate = clock_rate;
info->channels = channels;
info->bpf = (info->finfo->width / 8) * channels;
/* add channel positions */
channel_order = gst_structure_get_string (structure, "channel-order");
order = gst_rtp_channels_get_by_order (channels, channel_order);
rtpL16depay->order = order;
if (order) {
memcpy (info->position, order->pos,
sizeof (GstAudioChannelPosition) * channels);
gst_audio_channel_positions_to_valid_order (info->position, info->channels);
} else {
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
(NULL), ("Unknown channel order '%s' for %d channels",
GST_STR_NULL (channel_order), channels));
/* create default NONE layout */
gst_rtp_channels_create_default (channels, info->position);
}
srccaps = gst_audio_info_to_caps (info);
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
return res;
/* ERRORS */
no_clockrate:
{
GST_ERROR_OBJECT (depayload, "no clock-rate specified");
return FALSE;
}
}
static GstCaps *
gst_dshowaudiosrc_getcaps_from_streamcaps (GstDshowAudioSrc * src, IPin * pin,
IAMStreamConfig * streamcaps)
{
GstCaps *caps = NULL;
HRESULT hres = S_OK;
int icount = 0;
int isize = 0;
AUDIO_STREAM_CONFIG_CAPS ascc;
int i = 0;
if (!streamcaps)
return NULL;
streamcaps->GetNumberOfCapabilities (&icount, &isize);
if (isize != sizeof (ascc))
return NULL;
for (; i < icount; i++) {
GstCapturePinMediaType *pin_mediatype = g_new0 (GstCapturePinMediaType, 1);
pin->AddRef ();
pin_mediatype->capture_pin = pin;
hres = streamcaps->GetStreamCaps (i, &pin_mediatype->mediatype,
(BYTE *) & ascc);
if (hres == S_OK && pin_mediatype->mediatype) {
GstCaps *mediacaps = NULL;
if (!caps)
caps = gst_caps_new_empty ();
if (gst_dshow_check_mediatype (pin_mediatype->mediatype, MEDIASUBTYPE_PCM,
FORMAT_WaveFormatEx)) {
GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN;
WAVEFORMATEX *wavformat =
(WAVEFORMATEX *) pin_mediatype->mediatype->pbFormat;
switch (wavformat->wFormatTag) {
case WAVE_FORMAT_PCM:
format = gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, wavformat->wBitsPerSample, wavformat->wBitsPerSample);
break;
default:
break;
}
if (format != GST_AUDIO_FORMAT_UNKNOWN) {
GstAudioInfo info;
gst_audio_info_init(&info);
gst_audio_info_set_format(&info,
format,
wavformat->nSamplesPerSec,
wavformat->nChannels,
NULL);
mediacaps = gst_audio_info_to_caps(&info);
}
if (mediacaps) {
src->pins_mediatypes =
g_list_append (src->pins_mediatypes, pin_mediatype);
gst_caps_append (caps, mediacaps);
} else {
gst_dshow_free_pin_mediatype (pin_mediatype);
}
} else {
gst_dshow_free_pin_mediatype (pin_mediatype);
}
} else {
gst_dshow_free_pin_mediatype (pin_mediatype);
}
}
if (caps && gst_caps_is_empty (caps)) {
gst_caps_unref (caps);
caps = NULL;
}
return caps;
}
static gboolean
gst_pulsesrc_create_stream (GstPulseSrc * pulsesrc, GstCaps ** caps)
{
pa_channel_map channel_map;
const pa_channel_map *m;
GstStructure *s;
gboolean need_channel_layout = FALSE;
GstAudioRingBufferSpec spec;
const gchar *name;
s = gst_caps_get_structure (*caps, 0);
gst_structure_get_int (s, "channels", &spec.info.channels);
if (!gst_structure_has_field (s, "channel-mask")) {
if (spec.info.channels == 1) {
pa_channel_map_init_mono (&channel_map);
} else if (spec.info.channels == 2) {
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK,
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |
GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT), NULL);
pa_channel_map_init_stereo (&channel_map);
} else {
need_channel_layout = TRUE;
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK,
G_GUINT64_CONSTANT (0), NULL);
}
}
memset (&spec, 0, sizeof (GstAudioRingBufferSpec));
spec.latency_time = GST_SECOND;
if (!gst_audio_ring_buffer_parse_caps (&spec, *caps))
goto invalid_caps;
/* Keep the refcount of the caps at 1 to make them writable */
gst_caps_unref (spec.caps);
if (!need_channel_layout
&& !gst_pulse_gst_to_channel_map (&channel_map, &spec)) {
need_channel_layout = TRUE;
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK,
G_GUINT64_CONSTANT (0), NULL);
memset (spec.info.position, 0xff, sizeof (spec.info.position));
}
if (!gst_pulse_fill_sample_spec (&spec, &pulsesrc->sample_spec))
goto invalid_spec;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (!pulsesrc->context)
goto bad_context;
name = "Record Stream";
if (pulsesrc->proplist) {
if (!(pulsesrc->stream = pa_stream_new_with_proplist (pulsesrc->context,
name, &pulsesrc->sample_spec,
(need_channel_layout) ? NULL : &channel_map,
pulsesrc->proplist)))
goto create_failed;
} else if (!(pulsesrc->stream = pa_stream_new (pulsesrc->context,
name, &pulsesrc->sample_spec,
(need_channel_layout) ? NULL : &channel_map)))
goto create_failed;
m = pa_stream_get_channel_map (pulsesrc->stream);
gst_pulse_channel_map_to_gst (m, &spec);
gst_audio_channel_positions_to_valid_order (spec.info.position,
spec.info.channels);
gst_caps_unref (*caps);
*caps = gst_audio_info_to_caps (&spec.info);
GST_DEBUG_OBJECT (pulsesrc, "Caps are %" GST_PTR_FORMAT, *caps);
pa_stream_set_state_callback (pulsesrc->stream, gst_pulsesrc_stream_state_cb,
pulsesrc);
pa_stream_set_read_callback (pulsesrc->stream, gst_pulsesrc_stream_request_cb,
pulsesrc);
pa_stream_set_underflow_callback (pulsesrc->stream,
gst_pulsesrc_stream_underflow_cb, pulsesrc);
pa_stream_set_overflow_callback (pulsesrc->stream,
gst_pulsesrc_stream_overflow_cb, pulsesrc);
pa_stream_set_latency_update_callback (pulsesrc->stream,
gst_pulsesrc_stream_latency_update_cb, pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return TRUE;
/* ERRORS */
invalid_caps:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
("Can't parse caps."), (NULL));
goto fail;
}
invalid_spec:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
("Invalid sample specification."), (NULL));
goto fail;
}
bad_context:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Bad context"), (NULL));
goto unlock_and_fail;
}
create_failed:
{
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to create stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
unlock_and_fail:
{
gst_pulsesrc_destroy_stream (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
fail:
return FALSE;
}
}
static gboolean
gst_raw_audio_parse_config_to_caps (GstRawAudioParse * raw_audio_parse,
GstCaps ** caps, GstRawAudioParseConfig * config)
{
gboolean ret = TRUE;
GstAudioChannelPosition *channel_positions;
g_assert (caps != NULL);
if (config->bpf == 0) {
GST_ERROR_OBJECT (raw_audio_parse,
"cannot convert config to caps - config not filled with valid values");
*caps = NULL;
return FALSE;
}
channel_positions =
config->needs_channel_reordering ? &(config->
reordered_channel_positions[0]) : &(config->channel_positions[0]);
switch (config->format) {
case GST_RAW_AUDIO_PARSE_FORMAT_PCM:
{
GstAudioInfo info;
gst_audio_info_init (&info);
gst_audio_info_set_format (&info,
config->pcm_format,
config->sample_rate, config->num_channels, channel_positions);
*caps = gst_audio_info_to_caps (&info);
break;
}
case GST_RAW_AUDIO_PARSE_FORMAT_ALAW:
case GST_RAW_AUDIO_PARSE_FORMAT_MULAW:
{
guint64 channel_mask;
if (!gst_audio_channel_positions_to_mask (channel_positions,
config->num_channels, TRUE, &channel_mask)) {
GST_ERROR_OBJECT (raw_audio_parse, "invalid channel positions");
ret = FALSE;
break;
}
*caps = gst_caps_new_simple (
(config->format ==
GST_RAW_AUDIO_PARSE_FORMAT_ALAW) ? "audio/x-alaw" :
"audio/x-mulaw", "rate", G_TYPE_INT, config->sample_rate, "channels",
G_TYPE_INT, config->num_channels, "channel-mask", GST_TYPE_BITMASK,
channel_mask, NULL);
break;
}
default:
g_assert_not_reached ();
ret = FALSE;
}
if (!ret)
*caps = NULL;
return ret;
}
static GstFlowReturn
gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
GstBuffer ** buf)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GstCaps *caps;
GstBuffer *buffer;
guint n, bpf;
guint64 period_time;
guint64 period_samples;
GST_DEBUG_OBJECT (interaudiosrc, "create");
buffer = NULL;
caps = NULL;
g_mutex_lock (&interaudiosrc->surface->mutex);
if (interaudiosrc->surface->audio_info.finfo) {
if (!gst_audio_info_is_equal (&interaudiosrc->surface->audio_info,
&interaudiosrc->info)) {
caps = gst_audio_info_to_caps (&interaudiosrc->surface->audio_info);
interaudiosrc->timestamp_offset +=
gst_util_uint64_scale (interaudiosrc->n_samples, GST_SECOND,
interaudiosrc->info.rate);
interaudiosrc->n_samples = 0;
}
}
bpf = interaudiosrc->surface->audio_info.bpf;
period_time = interaudiosrc->surface->audio_period_time;
period_samples =
gst_util_uint64_scale (period_time, interaudiosrc->info.rate, GST_SECOND);
if (bpf > 0)
n = gst_adapter_available (interaudiosrc->surface->audio_adapter) / bpf;
else
n = 0;
if (n > period_samples)
n = period_samples;
if (n > 0) {
buffer = gst_adapter_take_buffer (interaudiosrc->surface->audio_adapter,
n * bpf);
} else {
buffer = gst_buffer_new ();
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_GAP);
}
g_mutex_unlock (&interaudiosrc->surface->mutex);
if (caps) {
gboolean ret = gst_base_src_set_caps (src, caps);
gst_caps_unref (caps);
if (!ret) {
GST_ERROR_OBJECT (src, "Failed to set caps %" GST_PTR_FORMAT, caps);
if (buffer)
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
}
}
buffer = gst_buffer_make_writable (buffer);
bpf = interaudiosrc->info.bpf;
if (n < period_samples) {
GstMapInfo map;
GstMemory *mem;
GST_DEBUG_OBJECT (interaudiosrc,
"creating %" G_GUINT64_FORMAT " samples of silence",
period_samples - n);
mem = gst_allocator_alloc (NULL, (period_samples - n) * bpf, NULL);
if (gst_memory_map (mem, &map, GST_MAP_WRITE)) {
gst_audio_format_fill_silence (interaudiosrc->info.finfo, map.data,
map.size);
gst_memory_unmap (mem, &map);
}
gst_buffer_prepend_memory (buffer, mem);
}
n = period_samples;
GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
GST_BUFFER_OFFSET_END (buffer) = interaudiosrc->n_samples + n;
GST_BUFFER_TIMESTAMP (buffer) = interaudiosrc->timestamp_offset +
gst_util_uint64_scale (interaudiosrc->n_samples, GST_SECOND,
interaudiosrc->info.rate);
GST_DEBUG_OBJECT (interaudiosrc, "create ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
GST_BUFFER_DURATION (buffer) = interaudiosrc->timestamp_offset +
gst_util_uint64_scale (interaudiosrc->n_samples + n, GST_SECOND,
interaudiosrc->info.rate) - GST_BUFFER_TIMESTAMP (buffer);
GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
if (interaudiosrc->n_samples == 0) {
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
}
interaudiosrc->n_samples += n;
*buf = buffer;
return GST_FLOW_OK;
}
static gpointer
push_abuffers (gpointer data)
{
GstSegment segment;
GstPad *pad = data;
gint i, j, k;
GstClockTime timestamp = 0;
GstAudioInfo info;
GstCaps *caps;
guint buf_size = 1000;
if (audiodelay)
g_usleep (2000);
if (early_video)
timestamp = 50 * GST_MSECOND;
gst_pad_send_event (pad, gst_event_new_stream_start ("test"));
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S8, buf_size, channels,
NULL);
caps = gst_audio_info_to_caps (&info);
gst_pad_send_event (pad, gst_event_new_caps (caps));
gst_caps_unref (caps);
gst_segment_init (&segment, GST_FORMAT_TIME);
gst_pad_send_event (pad, gst_event_new_segment (&segment));
for (i = 0; i < n_abuffers; i++) {
GstBuffer *buf = gst_buffer_new_and_alloc (channels * buf_size);
if (per_channel) {
GstMapInfo map;
guint8 *in_data;
gst_buffer_map (buf, &map, GST_MAP_WRITE);
in_data = map.data;
for (j = 0; j < buf_size; j++) {
for (k = 0; k < channels; k++) {
in_data[j * channels + k] = fill_value_per_channel[k];
}
}
gst_buffer_unmap (buf, &map);
} else {
gst_buffer_memset (buf, 0, fill_value, channels * buf_size);
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
timestamp += 1 * GST_SECOND;
if (audio_drift)
timestamp += 50 * GST_MSECOND;
else if (i == 4 && audio_nondiscont)
timestamp += 30 * GST_MSECOND;
GST_BUFFER_DURATION (buf) = timestamp - GST_BUFFER_TIMESTAMP (buf);
fail_unless (gst_pad_chain (pad, buf) == GST_FLOW_OK);
}
gst_pad_send_event (pad, gst_event_new_eos ());
return NULL;
}
