static gboolean
gst_audio_aggregator_src_query (GstAggregator * agg, GstQuery * query)
{
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_DURATION:
res = gst_audio_aggregator_query_duration (aagg, query);
break;
case GST_QUERY_POSITION:
{
GstFormat format;
gst_query_parse_position (query, &format, NULL);
GST_OBJECT_LOCK (aagg);
switch (format) {
case GST_FORMAT_TIME:
gst_query_set_position (query, format,
gst_segment_to_stream_time (&agg->segment, GST_FORMAT_TIME,
agg->segment.position));
res = TRUE;
break;
case GST_FORMAT_BYTES:
if (GST_AUDIO_INFO_BPF (&aagg->info)) {
gst_query_set_position (query, format, aagg->priv->offset *
GST_AUDIO_INFO_BPF (&aagg->info));
res = TRUE;
}
break;
case GST_FORMAT_DEFAULT:
gst_query_set_position (query, format, aagg->priv->offset);
res = TRUE;
break;
default:
break;
}
GST_OBJECT_UNLOCK (aagg);
break;
}
default:
res =
GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->src_query
(agg, query);
break;
}
return res;
}
static void
gst_ladspa_source_type_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstLADSPASource *ladspa = GST_LADSPA_SOURCE (object);
switch (prop_id) {
case GST_LADSPA_SOURCE_PROP_SAMPLES_PER_BUFFER:
ladspa->samples_per_buffer = g_value_get_int (value);
gst_base_src_set_blocksize (GST_BASE_SRC (ladspa),
GST_AUDIO_INFO_BPF (&ladspa->info) * ladspa->samples_per_buffer);
break;
case GST_LADSPA_SOURCE_PROP_IS_LIVE:
gst_base_src_set_live (GST_BASE_SRC (ladspa),
g_value_get_boolean (value));
break;
case GST_LADSPA_SOURCE_PROP_TIMESTAMP_OFFSET:
ladspa->timestamp_offset = g_value_get_int64 (value);
break;
case GST_LADSPA_SOURCE_PROP_CAN_ACTIVATE_PUSH:
GST_BASE_SRC (ladspa)->can_activate_push = g_value_get_boolean (value);
break;
case GST_LADSPA_SOURCE_PROP_CAN_ACTIVATE_PULL:
ladspa->can_activate_pull = g_value_get_boolean (value);
break;
default:
gst_ladspa_object_set_property (&ladspa->ladspa, object, prop_id, value,
pspec);
break;
}
}
static gboolean
gst_lv2_source_set_caps (GstBaseSrc * base, GstCaps * caps)
{
GstLV2Source *lv2 = (GstLV2Source *) base;
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps)) {
GST_ERROR_OBJECT (base, "received invalid caps");
return FALSE;
}
GST_DEBUG_OBJECT (lv2, "negotiated to caps %" GST_PTR_FORMAT, caps);
lv2->info = info;
gst_base_src_set_blocksize (base,
GST_AUDIO_INFO_BPF (&info) * lv2->samples_per_buffer);
if (!gst_lv2_setup (&lv2->lv2, GST_AUDIO_INFO_RATE (&info)))
goto no_instance;
return TRUE;
no_instance:
{
GST_ERROR_OBJECT (lv2, "could not create instance");
return FALSE;
}
}
/* GObject vmethods implementation */
static void
gst_lv2_source_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstLV2Source *self = (GstLV2Source *) object;
switch (prop_id) {
case GST_LV2_SOURCE_PROP_SAMPLES_PER_BUFFER:
self->samples_per_buffer = g_value_get_int (value);
gst_base_src_set_blocksize (GST_BASE_SRC (self),
GST_AUDIO_INFO_BPF (&self->info) * self->samples_per_buffer);
break;
case GST_LV2_SOURCE_PROP_IS_LIVE:
gst_base_src_set_live (GST_BASE_SRC (self), g_value_get_boolean (value));
break;
case GST_LV2_SOURCE_PROP_TIMESTAMP_OFFSET:
self->timestamp_offset = g_value_get_int64 (value);
break;
case GST_LV2_SOURCE_PROP_CAN_ACTIVATE_PUSH:
GST_BASE_SRC (self)->can_activate_push = g_value_get_boolean (value);
break;
case GST_LV2_SOURCE_PROP_CAN_ACTIVATE_PULL:
self->can_activate_pull = g_value_get_boolean (value);
break;
default:
gst_lv2_object_set_property (&self->lv2, object, prop_id, value, pspec);
break;
}
}
static gboolean
gst_audio_test_src_setcaps (GstBaseSrc * basesrc, GstCaps * caps)
{
GstAudioTestSrc *src = GST_AUDIO_TEST_SRC (basesrc);
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps))
goto invalid_caps;
GST_DEBUG_OBJECT (src, "negotiated to caps %" GST_PTR_FORMAT, caps);
src->info = info;
gst_base_src_set_blocksize (basesrc,
GST_AUDIO_INFO_BPF (&info) * src->samples_per_buffer);
gst_audio_test_src_change_wave (src);
return TRUE;
/* ERROR */
invalid_caps:
{
GST_ERROR_OBJECT (basesrc, "received invalid caps");
return FALSE;
}
}
static gboolean
gst_audio_fx_base_fir_filter_transform_size (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, gsize size, GstCaps * othercaps,
gsize * othersize)
{
GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
guint blocklen;
GstAudioInfo info;
gint bpf;
if (!self->fft || self->low_latency || direction == GST_PAD_SRC) {
*othersize = size;
return TRUE;
}
if (!gst_audio_info_from_caps (&info, caps))
return FALSE;
bpf = GST_AUDIO_INFO_BPF (&info);
size /= bpf;
blocklen = self->block_length - self->kernel_length + 1;
*othersize = ((size + blocklen - 1) / blocklen) * blocklen;
*othersize *= bpf;
return TRUE;
}
/* seek to time, will be called when we operate in push mode. In pull mode we
* get the requested byte offset. */
static gboolean
gst_lv2_source_do_seek (GstBaseSrc * base, GstSegment * segment)
{
GstLV2Source *lv2 = (GstLV2Source *) base;
GstClockTime time;
gint samplerate, bpf;
gint64 next_sample;
GST_DEBUG_OBJECT (lv2, "seeking %" GST_SEGMENT_FORMAT, segment);
time = segment->position;
lv2->reverse = (segment->rate < 0.0);
samplerate = GST_AUDIO_INFO_RATE (&lv2->info);
bpf = GST_AUDIO_INFO_BPF (&lv2->info);
/* now move to the time indicated, don't seek to the sample *after* the time */
next_sample = gst_util_uint64_scale_int (time, samplerate, GST_SECOND);
lv2->next_byte = next_sample * bpf;
if (samplerate == 0)
lv2->next_time = 0;
else
lv2->next_time =
gst_util_uint64_scale_round (next_sample, GST_SECOND, samplerate);
GST_DEBUG_OBJECT (lv2, "seeking next_sample=%" G_GINT64_FORMAT
" next_time=%" GST_TIME_FORMAT, next_sample,
GST_TIME_ARGS (lv2->next_time));
g_assert (lv2->next_time <= time);
lv2->next_sample = next_sample;
if (!lv2->reverse) {
if (GST_CLOCK_TIME_IS_VALID (segment->start)) {
segment->time = segment->start;
}
} else {
if (GST_CLOCK_TIME_IS_VALID (segment->stop)) {
segment->time = segment->stop;
}
}
if (GST_CLOCK_TIME_IS_VALID (segment->stop)) {
time = segment->stop;
lv2->sample_stop =
gst_util_uint64_scale_round (time, samplerate, GST_SECOND);
lv2->check_seek_stop = TRUE;
} else {
lv2->check_seek_stop = FALSE;
}
lv2->eos_reached = FALSE;
return TRUE;
}
static gboolean
gst_alsasrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
if (!alsasrc_parse_spec (alsa, spec))
goto spec_parse;
CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
CHECK (set_hwparams (alsa), hw_params_failed);
CHECK (set_swparams (alsa), sw_params_failed);
CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
spec->segsize = alsa->period_size * alsa->bpf;
spec->segtotal = alsa->buffer_size / alsa->period_size;
return TRUE;
/* ERRORS */
spec_parse:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Error parsing spec"));
return FALSE;
}
non_block:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Could not set device to blocking: %s", snd_strerror (err)));
return FALSE;
}
hw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Setting of hwparams failed: %s", snd_strerror (err)));
return FALSE;
}
sw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Setting of swparams failed: %s", snd_strerror (err)));
return FALSE;
}
prepare_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Prepare failed: %s", snd_strerror (err)));
return FALSE;
}
}
static gboolean
gst_decklink_audio_sink_ringbuffer_acquire (GstAudioRingBuffer * rb,
GstAudioRingBufferSpec * spec)
{
GstDecklinkAudioSinkRingBuffer *self =
GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST (rb);
HRESULT ret;
BMDAudioSampleType sample_depth;
GST_DEBUG_OBJECT (self->sink, "Acquire");
if (spec->info.finfo->format == GST_AUDIO_FORMAT_S16LE) {
sample_depth = bmdAudioSampleType16bitInteger;
} else {
sample_depth = bmdAudioSampleType32bitInteger;
}
ret = self->output->output->EnableAudioOutput (bmdAudioSampleRate48kHz,
sample_depth, 2, bmdAudioOutputStreamContinuous);
if (ret != S_OK) {
GST_WARNING_OBJECT (self->sink, "Failed to enable audio output 0x%08x",
ret);
return FALSE;
}
g_mutex_lock (&self->output->lock);
self->output->audio_enabled = TRUE;
if (self->output->start_scheduled_playback)
self->output->start_scheduled_playback (self->output->videosink);
g_mutex_unlock (&self->output->lock);
ret =
self->output->
output->SetAudioCallback (new GStreamerAudioOutputCallback (self));
if (ret != S_OK) {
GST_WARNING_OBJECT (self->sink,
"Failed to set audio output callback 0x%08x", ret);
return FALSE;
}
spec->segsize =
(spec->latency_time * GST_AUDIO_INFO_RATE (&spec->info) /
G_USEC_PER_SEC) * GST_AUDIO_INFO_BPF (&spec->info);
spec->segtotal = spec->buffer_time / spec->latency_time;
// set latency to one more segment as we need some headroom
spec->seglatency = spec->segtotal + 1;
rb->size = spec->segtotal * spec->segsize;
rb->memory = (guint8 *) g_malloc0 (rb->size);
return TRUE;
}
static GstBuffer *
gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg,
guint num_frames)
{
GstBuffer *outbuf = gst_buffer_new_allocate (NULL, num_frames *
GST_AUDIO_INFO_BPF (&aagg->info), NULL);
GstMapInfo outmap;
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
gst_audio_format_fill_silence (aagg->info.finfo, outmap.data, outmap.size);
gst_buffer_unmap (outbuf, &outmap);
return outbuf;
}
static gboolean
gst_bml_transform_get_unit_size (GstBaseTransform * base, GstCaps * caps,
gsize * size)
{
GstAudioInfo info;
g_assert (size);
if (!gst_audio_info_from_caps (&info, caps))
return FALSE;
*size = GST_AUDIO_INFO_BPF (&info);
return TRUE;
}
static GstBuffer *
gst_audio_aggregator_do_clip (GstAggregator * agg,
GstAggregatorPad * bpad, GstBuffer * buffer)
{
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (bpad);
gint rate, bpf;
rate = GST_AUDIO_INFO_RATE (&pad->info);
bpf = GST_AUDIO_INFO_BPF (&pad->info);
GST_OBJECT_LOCK (bpad);
buffer = gst_audio_buffer_clip (buffer, &bpad->segment, rate, bpf);
GST_OBJECT_UNLOCK (bpad);
return buffer;
}
static gboolean
gst_freeverb_get_unit_size (GstBaseTransform * base, GstCaps * caps,
gsize * size)
{
GstAudioInfo info;
g_assert (size);
if (!gst_audio_info_from_caps (&info, caps))
return FALSE;
*size = GST_AUDIO_INFO_BPF (&info);
GST_INFO_OBJECT (base, "unit size: %" G_GSIZE_FORMAT, *size);
return TRUE;
}
virtual HRESULT RenderAudioSamples (bool preroll)
{
guint8 *ptr;
gint seg;
gint len;
gint bpf;
guint written, written_sum;
HRESULT res;
GST_LOG_OBJECT (m_ringbuffer->sink, "Writing audio samples (preroll: %d)",
preroll);
if (!gst_audio_ring_buffer_prepare_read (GST_AUDIO_RING_BUFFER_CAST
(m_ringbuffer), &seg, &ptr, &len)) {
GST_WARNING_OBJECT (m_ringbuffer->sink, "No segment available");
return E_FAIL;
}
bpf =
GST_AUDIO_INFO_BPF (&GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer)->
spec.info);
len /= bpf;
GST_LOG_OBJECT (m_ringbuffer->sink,
"Write audio samples: %p size %d segment: %d", ptr, len, seg);
written_sum = 0;
do {
res =
m_ringbuffer->output->output->ScheduleAudioSamples (ptr, len,
0, 0, &written);
len -= written;
ptr += written * bpf;
written_sum += written;
} while (len > 0 && res == S_OK);
GST_LOG_OBJECT (m_ringbuffer->sink, "Wrote %u samples: 0x%08x", written_sum,
res);
gst_audio_ring_buffer_clear (GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer),
seg);
gst_audio_ring_buffer_advance (GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer),
1);
return res;
}
static gboolean
gst_ladspa_source_type_set_caps (GstBaseSrc * base, GstCaps * caps)
{
GstLADSPASource *ladspa = GST_LADSPA_SOURCE (base);
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps)) {
GST_ERROR_OBJECT (base, "received invalid caps");
return FALSE;
}
GST_DEBUG_OBJECT (ladspa, "negotiated to caps %" GST_PTR_FORMAT, caps);
ladspa->info = info;
gst_base_src_set_blocksize (base,
GST_AUDIO_INFO_BPF (&info) * ladspa->samples_per_buffer);
return gst_ladspa_setup (&ladspa->ladspa, GST_AUDIO_INFO_RATE (&info));
}
static GstBuffer *
gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg,
guint num_frames)
{
GstAllocator *allocator;
GstAllocationParams params;
GstBuffer *outbuf;
GstMapInfo outmap;
gst_aggregator_get_allocator (GST_AGGREGATOR (aagg), &allocator, ¶ms);
outbuf = gst_buffer_new_allocate (allocator, num_frames *
GST_AUDIO_INFO_BPF (&aagg->info), ¶ms);
if (allocator)
gst_object_unref (allocator);
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
gst_audio_format_fill_silence (aagg->info.finfo, outmap.data, outmap.size);
gst_buffer_unmap (outbuf, &outmap);
return outbuf;
}
static gboolean
gst_directsound_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstDirectSoundSrc *dsoundsrc;
WAVEFORMATEX wfx; /* Wave format structure */
HRESULT hRes; /* Result for windows functions */
DSCBUFFERDESC descSecondary; /* Capturebuffer description */
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DEBUG_OBJECT (asrc, "preparing directsoundsrc");
/* Define buffer */
memset (&wfx, 0, sizeof (WAVEFORMATEX));
wfx.wFormatTag = WAVE_FORMAT_PCM;
wfx.nChannels = GST_AUDIO_INFO_CHANNELS (&spec->info);
wfx.nSamplesPerSec = GST_AUDIO_INFO_RATE (&spec->info);
wfx.wBitsPerSample = GST_AUDIO_INFO_BPF (&spec->info) * 8 / wfx.nChannels;
wfx.nBlockAlign = GST_AUDIO_INFO_BPF (&spec->info);
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
/* Ignored for WAVE_FORMAT_PCM. */
wfx.cbSize = 0;
if (wfx.wBitsPerSample != 16 && wfx.wBitsPerSample != 8)
goto dodgy_width;
/* Set the buffer size to two seconds.
This should never reached.
*/
dsoundsrc->buffer_size = wfx.nAvgBytesPerSec * 2;
GST_DEBUG_OBJECT (asrc, "Buffer size: %d", dsoundsrc->buffer_size);
/* Init secondary buffer desciption */
memset (&descSecondary, 0, sizeof (DSCBUFFERDESC));
descSecondary.dwSize = sizeof (DSCBUFFERDESC);
descSecondary.dwFlags = 0;
descSecondary.dwReserved = 0;
/* This is not primary buffer so have to set size */
descSecondary.dwBufferBytes = dsoundsrc->buffer_size;
descSecondary.lpwfxFormat = &wfx;
/* Create buffer */
hRes = IDirectSoundCapture_CreateCaptureBuffer (dsoundsrc->pDSC,
&descSecondary, &dsoundsrc->pDSBSecondary, NULL);
if (hRes != DS_OK)
goto capture_buffer;
dsoundsrc->bytes_per_sample = GST_AUDIO_INFO_BPF (&spec->info);
GST_DEBUG ("latency time: %" G_GUINT64_FORMAT " - buffer time: %"
G_GUINT64_FORMAT, spec->latency_time, spec->buffer_time);
/* Buffer-time should be always more than 2*latency */
if (spec->buffer_time < spec->latency_time * 2) {
spec->buffer_time = spec->latency_time * 2;
GST_WARNING ("buffer-time was less than latency");
}
/* Save the times */
dsoundsrc->buffer_time = spec->buffer_time;
dsoundsrc->latency_time = spec->latency_time;
dsoundsrc->latency_size = (gint) wfx.nAvgBytesPerSec *
dsoundsrc->latency_time / 1000000.0;
spec->segsize = (guint) (((double) spec->buffer_time / 1000000.0) *
wfx.nAvgBytesPerSec);
/* just in case */
if (spec->segsize < 1)
spec->segsize = 1;
spec->segtotal = GST_AUDIO_INFO_BPF (&spec->info) * 8 *
(wfx.nAvgBytesPerSec / spec->segsize);
GST_DEBUG_OBJECT (asrc,
"bytes/sec: %lu, buffer size: %d, segsize: %d, segtotal: %d",
wfx.nAvgBytesPerSec, dsoundsrc->buffer_size, spec->segsize,
spec->segtotal);
/* Not read anything yet */
dsoundsrc->current_circular_offset = 0;
GST_DEBUG_OBJECT (asrc, "channels: %d, rate: %d, bytes_per_sample: %d"
" WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d,"
" WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld",
GST_AUDIO_INFO_CHANNELS (&spec->info), GST_AUDIO_INFO_RATE (&spec->info),
GST_AUDIO_INFO_BPF (&spec->info), wfx.nSamplesPerSec, wfx.wBitsPerSample,
wfx.nBlockAlign, wfx.nAvgBytesPerSec);
return TRUE;
capture_buffer:
{
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to create capturebuffer"), (NULL));
return FALSE;
}
dodgy_width:
{
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unexpected width %d", wfx.wBitsPerSample), (NULL));
return FALSE;
}
}
static gboolean
gst_audiomixer_aggregate_one_buffer (GstAudioAggregator * aagg,
GstAudioAggregatorPad * aaggpad, GstBuffer * inbuf, guint in_offset,
GstBuffer * outbuf, guint out_offset, guint num_frames)
{
GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (aaggpad);
GstMapInfo inmap;
GstMapInfo outmap;
gint bpf;
GST_OBJECT_LOCK (aagg);
GST_OBJECT_LOCK (aaggpad);
if (pad->mute || pad->volume < G_MINDOUBLE) {
GST_DEBUG_OBJECT (pad, "Skipping muted pad");
GST_OBJECT_UNLOCK (aaggpad);
GST_OBJECT_UNLOCK (aagg);
return FALSE;
}
bpf = GST_AUDIO_INFO_BPF (&aagg->info);
gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE);
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
GST_LOG_OBJECT (pad, "mixing %u bytes at offset %u from offset %u",
num_frames * bpf, out_offset * bpf, in_offset * bpf);
/* further buffers, need to add them */
if (pad->volume == 1.0) {
switch (aagg->info.finfo->format) {
case GST_AUDIO_FORMAT_U8:
audiomixer_orc_add_u8 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_S8:
audiomixer_orc_add_s8 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_U16:
audiomixer_orc_add_u16 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_S16:
audiomixer_orc_add_s16 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_U32:
audiomixer_orc_add_u32 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_S32:
audiomixer_orc_add_s32 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_F32:
audiomixer_orc_add_f32 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_F64:
audiomixer_orc_add_f64 ((gpointer) (outmap.data + out_offset * bpf),
(gpointer) (inmap.data + in_offset * bpf),
num_frames * aagg->info.channels);
break;
default:
g_assert_not_reached ();
break;
}
} else {
switch (aagg->info.finfo->format) {
case GST_AUDIO_FORMAT_U8:
audiomixer_orc_add_volume_u8 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume_i8, num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_S8:
audiomixer_orc_add_volume_s8 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume_i8, num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_U16:
audiomixer_orc_add_volume_u16 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume_i16, num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_S16:
audiomixer_orc_add_volume_s16 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume_i16, num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_U32:
audiomixer_orc_add_volume_u32 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume_i32, num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_S32:
audiomixer_orc_add_volume_s32 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume_i32, num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_F32:
audiomixer_orc_add_volume_f32 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume, num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_F64:
audiomixer_orc_add_volume_f64 ((gpointer) (outmap.data +
out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
pad->volume, num_frames * aagg->info.channels);
break;
default:
g_assert_not_reached ();
break;
}
}
gst_buffer_unmap (inbuf, &inmap);
gst_buffer_unmap (outbuf, &outmap);
GST_OBJECT_UNLOCK (aaggpad);
GST_OBJECT_UNLOCK (aagg);
return TRUE;
}
/* allocate a buffer and setup resources to process the audio samples of
* the format as specified in @spec.
*
* We allocate N jack ports, one for each channel. If we are asked to
* automatically make a connection with physical ports, we connect as many
* ports as there are physical ports, leaving leftover ports unconnected.
*
* It is assumed that samplerate and number of channels are acceptable since our
* getcaps method will always provide correct values. If unacceptable caps are
* received for some reason, we fail here.
*/
static gboolean
gst_jack_ring_buffer_acquire (GstAudioRingBuffer * buf,
GstAudioRingBufferSpec * spec)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
const char **ports;
gint sample_rate, buffer_size;
gint i, rate, bpf, channels, res;
jack_client_t *client;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
abuf = GST_JACK_RING_BUFFER_CAST (buf);
GST_DEBUG_OBJECT (sink, "acquire");
client = gst_jack_audio_client_get_client (sink->client);
rate = GST_AUDIO_INFO_RATE (&spec->info);
/* sample rate must be that of the server */
sample_rate = jack_get_sample_rate (client);
if (sample_rate != rate)
goto wrong_samplerate;
channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
bpf = GST_AUDIO_INFO_BPF (&spec->info);
if (!gst_jack_audio_sink_allocate_channels (sink, channels))
goto out_of_ports;
buffer_size = jack_get_buffer_size (client);
/* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
* for all channels */
spec->segsize = buffer_size * sizeof (gfloat) * channels;
spec->latency_time = gst_util_uint64_scale (spec->segsize,
(GST_SECOND / GST_USECOND), rate * bpf);
/* segtotal based on buffer-time latency */
spec->segtotal = spec->buffer_time / spec->latency_time;
if (spec->segtotal < 2) {
spec->segtotal = 2;
spec->buffer_time = spec->latency_time * spec->segtotal;
}
GST_DEBUG_OBJECT (sink, "buffer time: %" G_GINT64_FORMAT " usec",
spec->buffer_time);
GST_DEBUG_OBJECT (sink, "latency time: %" G_GINT64_FORMAT " usec",
spec->latency_time);
GST_DEBUG_OBJECT (sink, "buffer_size %d, segsize %d, segtotal %d",
buffer_size, spec->segsize, spec->segtotal);
/* allocate the ringbuffer memory now */
buf->size = spec->segtotal * spec->segsize;
buf->memory = g_malloc0 (buf->size);
if ((res = gst_jack_audio_client_set_active (sink->client, TRUE)))
goto could_not_activate;
/* if we need to automatically connect the ports, do so now. We must do this
* after activating the client. */
if (sink->connect == GST_JACK_CONNECT_AUTO
|| sink->connect == GST_JACK_CONNECT_AUTO_FORCED) {
/* find all the physical input ports. A physical input port is a port
* associated with a hardware device. Someone needs connect to a physical
* port in order to hear something. */
ports = jack_get_ports (client, NULL, NULL,
JackPortIsPhysical | JackPortIsInput);
if (ports == NULL) {
/* no ports? fine then we don't do anything except for posting a warning
* message. */
GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
("No physical input ports found, leaving ports unconnected"));
goto done;
}
for (i = 0; i < channels; i++) {
/* stop when all input ports are exhausted */
if (ports[i] == NULL) {
/* post a warning that we could not connect all ports */
GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
("No more physical ports, leaving some ports unconnected"));
break;
}
GST_DEBUG_OBJECT (sink, "try connecting to %s",
jack_port_name (sink->ports[i]));
/* connect the port to a physical port */
res = jack_connect (client, jack_port_name (sink->ports[i]), ports[i]);
if (res != 0 && res != EEXIST)
goto cannot_connect;
}
free (ports);
}
done:
abuf->sample_rate = sample_rate;
abuf->buffer_size = buffer_size;
abuf->channels = channels;
return TRUE;
/* ERRORS */
wrong_samplerate:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Wrong samplerate, server is running at %d and we received %d",
sample_rate, rate));
return FALSE;
}
out_of_ports:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Cannot allocate more Jack ports"));
return FALSE;
}
could_not_activate:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not activate client (%d:%s)", res, g_strerror (res)));
return FALSE;
}
cannot_connect:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not connect output ports to physical ports (%d:%s)",
res, g_strerror (res)));
free (ports);
return FALSE;
}
}
static gboolean
gst_alsasink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
{
GstAlsaSink *alsa;
gint err;
alsa = GST_ALSA_SINK (asink);
if (alsa->iec958) {
snd_pcm_close (alsa->handle);
alsa->handle = gst_alsa_open_iec958_pcm (GST_OBJECT (alsa), alsa->device);
if (G_UNLIKELY (!alsa->handle)) {
goto no_iec958;
}
}
if (!alsasink_parse_spec (alsa, spec))
goto spec_parse;
CHECK (set_hwparams (alsa), hw_params_failed);
CHECK (set_swparams (alsa), sw_params_failed);
alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
spec->segsize = alsa->period_size * alsa->bpf;
spec->segtotal = alsa->buffer_size / alsa->period_size;
{
snd_output_t *out_buf = NULL;
char *msg = NULL;
snd_output_buffer_open (&out_buf);
snd_pcm_dump_hw_setup (alsa->handle, out_buf);
snd_output_buffer_string (out_buf, &msg);
GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg);
snd_output_close (out_buf);
snd_output_buffer_open (&out_buf);
snd_pcm_dump_sw_setup (alsa->handle, out_buf);
snd_output_buffer_string (out_buf, &msg);
GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg);
snd_output_close (out_buf);
}
#ifdef SND_CHMAP_API_VERSION
alsa_detect_channels_mapping (GST_OBJECT (alsa), alsa->handle, spec,
alsa->channels, GST_AUDIO_BASE_SINK (alsa)->ringbuffer);
#endif /* SND_CHMAP_API_VERSION */
return TRUE;
/* ERRORS */
no_iec958:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_WRITE, (NULL),
("Could not open IEC958 (SPDIF) device for playback"));
return FALSE;
}
spec_parse:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Error parsing spec"));
return FALSE;
}
hw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Setting of hwparams failed: %s", snd_strerror (err)));
return FALSE;
}
sw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Setting of swparams failed: %s", snd_strerror (err)));
return FALSE;
}
}
static GstFlowReturn
gst_cutter_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstCutter *filter;
GstMapInfo map;
gint16 *in_data;
gint bpf, rate;
gsize in_size;
guint num_samples;
gdouble NCS = 0.0; /* Normalized Cumulative Square of buffer */
gdouble RMS = 0.0; /* RMS of signal in buffer */
gdouble NMS = 0.0; /* Normalized Mean Square of buffer */
GstBuffer *prebuf; /* pointer to a prebuffer element */
GstClockTime duration;
filter = GST_CUTTER (parent);
if (GST_AUDIO_INFO_FORMAT (&filter->info) == GST_AUDIO_FORMAT_UNKNOWN)
goto not_negotiated;
bpf = GST_AUDIO_INFO_BPF (&filter->info);
rate = GST_AUDIO_INFO_RATE (&filter->info);
gst_buffer_map (buf, &map, GST_MAP_READ);
in_data = (gint16 *) map.data;
in_size = map.size;
GST_LOG_OBJECT (filter, "length of prerec buffer: %" GST_TIME_FORMAT,
GST_TIME_ARGS (filter->pre_run_length));
/* calculate mean square value on buffer */
switch (GST_AUDIO_INFO_FORMAT (&filter->info)) {
case GST_AUDIO_FORMAT_S16:
num_samples = in_size / 2;
gst_cutter_calculate_gint16 (in_data, num_samples, &NCS);
NMS = NCS / num_samples;
break;
case GST_AUDIO_FORMAT_S8:
num_samples = in_size;
gst_cutter_calculate_gint8 ((gint8 *) in_data, num_samples, &NCS);
NMS = NCS / num_samples;
break;
default:
/* this shouldn't happen */
g_warning ("no mean square function for format");
break;
}
gst_buffer_unmap (buf, &map);
filter->silent_prev = filter->silent;
duration = gst_util_uint64_scale (in_size / bpf, GST_SECOND, rate);
RMS = sqrt (NMS);
/* if RMS below threshold, add buffer length to silent run length count
* if not, reset
*/
GST_LOG_OBJECT (filter, "buffer stats: NMS %f, RMS %f, audio length %f", NMS,
RMS, gst_guint64_to_gdouble (duration));
if (RMS < filter->threshold_level)
filter->silent_run_length += gst_guint64_to_gdouble (duration);
else {
filter->silent_run_length = 0 * GST_SECOND;
filter->silent = FALSE;
}
if (filter->silent_run_length > filter->threshold_length)
/* it has been silent long enough, flag it */
filter->silent = TRUE;
/* has the silent status changed ? if so, send right signal
* and, if from silent -> not silent, flush pre_record buffer
*/
if (filter->silent != filter->silent_prev) {
if (filter->silent) {
GstMessage *m =
gst_cutter_message_new (filter, FALSE, GST_BUFFER_TIMESTAMP (buf));
GST_DEBUG_OBJECT (filter, "signaling CUT_STOP");
gst_element_post_message (GST_ELEMENT (filter), m);
} else {
gint count = 0;
GstMessage *m =
gst_cutter_message_new (filter, TRUE, GST_BUFFER_TIMESTAMP (buf));
GST_DEBUG_OBJECT (filter, "signaling CUT_START");
gst_element_post_message (GST_ELEMENT (filter), m);
/* first of all, flush current buffer */
GST_DEBUG_OBJECT (filter, "flushing buffer of length %" GST_TIME_FORMAT,
GST_TIME_ARGS (filter->pre_run_length));
while (filter->pre_buffer) {
prebuf = (g_list_first (filter->pre_buffer))->data;
filter->pre_buffer = g_list_remove (filter->pre_buffer, prebuf);
gst_pad_push (filter->srcpad, prebuf);
++count;
}
GST_DEBUG_OBJECT (filter, "flushed %d buffers", count);
filter->pre_run_length = 0 * GST_SECOND;
}
}
/* now check if we have to send the new buffer to the internal buffer cache
* or to the srcpad */
if (filter->silent) {
filter->pre_buffer = g_list_append (filter->pre_buffer, buf);
filter->pre_run_length += gst_guint64_to_gdouble (duration);
while (filter->pre_run_length > filter->pre_length) {
GstClockTime pduration;
gsize psize;
prebuf = (g_list_first (filter->pre_buffer))->data;
g_assert (GST_IS_BUFFER (prebuf));
psize = gst_buffer_get_size (prebuf);
pduration = gst_util_uint64_scale (psize / bpf, GST_SECOND, rate);
filter->pre_buffer = g_list_remove (filter->pre_buffer, prebuf);
filter->pre_run_length -= gst_guint64_to_gdouble (pduration);
/* only pass buffers if we don't leak */
if (!filter->leaky)
ret = gst_pad_push (filter->srcpad, prebuf);
else
gst_buffer_unref (prebuf);
}
} else
ret = gst_pad_push (filter->srcpad, buf);
return ret;
/* ERRORS */
not_negotiated:
{
return GST_FLOW_NOT_NEGOTIATED;
}
}
static void
run_decoding_test (GstElement * mpg123audiodec, gchar const *filename)
{
GstBus *bus;
unsigned int num_input_buffers, num_decoded_buffers;
gint expected_size;
GstCaps *out_caps, *caps;
GstAudioInfo audioinfo;
GstElement *input_pipeline, *input_appsink;
int i;
GstBuffer *outbuffer;
/* 440 Hz = frequency of sine wave in audio data
* 44100 Hz = sample rate
* (44100 / 2) Hz = Nyquist frequency */
static double const expected_frequency_spot = 440.0 / (44100.0 / 2.0);
fail_unless (gst_element_set_state (mpg123audiodec,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
bus = gst_bus_new ();
gst_element_set_bus (mpg123audiodec, bus);
setup_input_pipeline (filename, &input_pipeline, &input_appsink);
num_input_buffers = 0;
while (TRUE) {
GstSample *sample;
GstBuffer *input_buffer;
sample = gst_app_sink_pull_sample (GST_APP_SINK (input_appsink));
if (sample == NULL)
break;
fail_unless (GST_IS_SAMPLE (sample));
input_buffer = gst_sample_get_buffer (sample);
fail_if (input_buffer == NULL);
/* This is done to be on the safe side - docs say lifetime of the input buffer
* depends *solely* on the sample */
input_buffer = gst_buffer_copy (input_buffer);
fail_unless_equals_int (gst_pad_push (mysrcpad, input_buffer), GST_FLOW_OK);
++num_input_buffers;
gst_sample_unref (sample);
}
num_decoded_buffers = g_list_length (buffers);
/* check number of decoded buffers */
fail_unless_equals_int (num_decoded_buffers, num_input_buffers - 2);
caps = gst_pad_get_current_caps (mysinkpad);
GST_LOG ("output caps %" GST_PTR_FORMAT, caps);
fail_unless (gst_audio_info_from_caps (&audioinfo, caps),
"Getting audio info from caps failed");
/* check caps */
out_caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, "S32LE",
"layout", G_TYPE_STRING, "interleaved",
"rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1, NULL);
fail_unless (gst_caps_is_equal_fixed (caps, out_caps), "Incorrect out caps");
gst_caps_unref (out_caps);
gst_caps_unref (caps);
/* here, test if decoded data is a sine tone, and if the sine frequency is at the
* right spot in the spectrum */
for (i = 0; i < num_decoded_buffers; ++i) {
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL, "Invalid buffer retrieved");
/* MPEG 1 layer 2 uses 1152 samples per frame */
expected_size = 1152 * GST_AUDIO_INFO_BPF (&audioinfo);
fail_unless_equals_int (gst_buffer_get_size (outbuffer), expected_size);
check_main_frequency_spot_S32 (outbuffer, expected_frequency_spot);
buffers = g_list_remove (buffers, outbuffer);
gst_buffer_unref (outbuffer);
outbuffer = NULL;
}
g_list_free (buffers);
buffers = NULL;
cleanup_input_pipeline (input_pipeline);
gst_bus_set_flushing (bus, TRUE);
gst_element_set_bus (mpg123audiodec, NULL);
gst_object_unref (GST_OBJECT (bus));
}
static GstFlowReturn
gst_lv2_source_fill (GstBaseSrc * base, guint64 offset,
guint length, GstBuffer * buffer)
{
GstLV2Source *lv2 = (GstLV2Source *) base;
GstLV2SourceClass *klass = (GstLV2SourceClass *) GST_BASE_SRC_GET_CLASS (lv2);
GstLV2Class *lv2_class = &klass->lv2;
GstLV2Group *lv2_group;
GstLV2Port *lv2_port;
GstClockTime next_time;
gint64 next_sample, next_byte;
guint bytes, samples;
GstElementClass *eclass;
GstMapInfo map;
gint samplerate, bpf;
guint j, k, l;
gfloat *out = NULL, *cv = NULL, *mem;
gfloat val;
/* example for tagging generated data */
if (!lv2->tags_pushed) {
GstTagList *taglist;
taglist = gst_tag_list_new (GST_TAG_DESCRIPTION, "lv2 wave", NULL);
eclass = GST_ELEMENT_CLASS (parent_class);
if (eclass->send_event)
eclass->send_event (GST_ELEMENT (base), gst_event_new_tag (taglist));
else
gst_tag_list_unref (taglist);
lv2->tags_pushed = TRUE;
}
if (lv2->eos_reached) {
GST_INFO_OBJECT (lv2, "eos");
return GST_FLOW_EOS;
}
samplerate = GST_AUDIO_INFO_RATE (&lv2->info);
bpf = GST_AUDIO_INFO_BPF (&lv2->info);
/* if no length was given, use our default length in samples otherwise convert
* the length in bytes to samples. */
if (length == -1)
samples = lv2->samples_per_buffer;
else
samples = length / bpf;
/* if no offset was given, use our next logical byte */
if (offset == -1)
offset = lv2->next_byte;
/* now see if we are at the byteoffset we think we are */
if (offset != lv2->next_byte) {
GST_DEBUG_OBJECT (lv2, "seek to new offset %" G_GUINT64_FORMAT, offset);
/* we have a discont in the expected sample offset, do a 'seek' */
lv2->next_sample = offset / bpf;
lv2->next_time =
gst_util_uint64_scale_int (lv2->next_sample, GST_SECOND, samplerate);
lv2->next_byte = offset;
}
/* check for eos */
if (lv2->check_seek_stop &&
(lv2->sample_stop > lv2->next_sample) &&
(lv2->sample_stop < lv2->next_sample + samples)
) {
/* calculate only partial buffer */
lv2->generate_samples_per_buffer = lv2->sample_stop - lv2->next_sample;
next_sample = lv2->sample_stop;
lv2->eos_reached = TRUE;
GST_INFO_OBJECT (lv2, "eos reached");
} else {
/* calculate full buffer */
lv2->generate_samples_per_buffer = samples;
next_sample = lv2->next_sample + (lv2->reverse ? (-samples) : samples);
}
bytes = lv2->generate_samples_per_buffer * bpf;
next_byte = lv2->next_byte + (lv2->reverse ? (-bytes) : bytes);
next_time = gst_util_uint64_scale_int (next_sample, GST_SECOND, samplerate);
GST_LOG_OBJECT (lv2, "samplerate %d", samplerate);
GST_LOG_OBJECT (lv2,
"next_sample %" G_GINT64_FORMAT ", ts %" GST_TIME_FORMAT, next_sample,
GST_TIME_ARGS (next_time));
gst_buffer_set_size (buffer, bytes);
GST_BUFFER_OFFSET (buffer) = lv2->next_sample;
GST_BUFFER_OFFSET_END (buffer) = next_sample;
if (!lv2->reverse) {
GST_BUFFER_TIMESTAMP (buffer) = lv2->timestamp_offset + lv2->next_time;
GST_BUFFER_DURATION (buffer) = next_time - lv2->next_time;
} else {
GST_BUFFER_TIMESTAMP (buffer) = lv2->timestamp_offset + next_time;
GST_BUFFER_DURATION (buffer) = lv2->next_time - next_time;
}
gst_object_sync_values (GST_OBJECT (lv2), GST_BUFFER_TIMESTAMP (buffer));
lv2->next_time = next_time;
lv2->next_sample = next_sample;
lv2->next_byte = next_byte;
GST_LOG_OBJECT (lv2, "generating %u samples at ts %" GST_TIME_FORMAT,
samples, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
/* multi channel outputs */
lv2_group = &lv2_class->out_group;
if (lv2_group->ports->len > 1) {
out = g_new0 (gfloat, samples * lv2_group->ports->len);
for (j = 0; j < lv2_group->ports->len; ++j) {
lv2_port = &g_array_index (lv2_group->ports, GstLV2Port, j);
lilv_instance_connect_port (lv2->lv2.instance, lv2_port->index,
out + (j * samples));
GST_LOG_OBJECT (lv2, "connected port %d/%d", j, lv2_group->ports->len);
}
} else {
lv2_port = &g_array_index (lv2_group->ports, GstLV2Port, 0);
lilv_instance_connect_port (lv2->lv2.instance, lv2_port->index,
(gfloat *) map.data);
GST_LOG_OBJECT (lv2, "connected port 0");
}
/* cv ports */
cv = g_new (gfloat, samples * lv2_class->num_cv_in);
for (j = k = 0; j < lv2_class->control_in_ports->len; j++) {
lv2_port = &g_array_index (lv2_class->control_in_ports, GstLV2Port, j);
if (lv2_port->type != GST_LV2_PORT_CV)
continue;
mem = cv + (k * samples);
val = lv2->lv2.ports.control.in[j];
/* FIXME: use gst_control_binding_get_value_array */
for (l = 0; l < samples; l++)
mem[l] = val;
lilv_instance_connect_port (lv2->lv2.instance, lv2_port->index, mem);
k++;
}
lilv_instance_run (lv2->lv2.instance, samples);
if (lv2_group->ports->len > 1) {
gst_lv2_source_interleave_data (lv2_group->ports->len,
(gfloat *) map.data, samples, out);
g_free (out);
}
g_free (cv);
gst_buffer_unmap (buffer, &map);
return GST_FLOW_OK;
}
static gboolean
gst_directsound_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstDirectSoundSrc *dsoundsrc;
WAVEFORMATEX wfx; /* Wave format structure */
HRESULT hRes; /* Result for windows functions */
DSCBUFFERDESC descSecondary; /* Capturebuffer description */
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DEBUG_OBJECT (asrc, "preparing directsoundsrc");
/* Define buffer */
memset (&wfx, 0, sizeof (WAVEFORMATEX));
wfx.wFormatTag = WAVE_FORMAT_PCM;
wfx.nChannels = GST_AUDIO_INFO_CHANNELS (&spec->info);
wfx.nSamplesPerSec = GST_AUDIO_INFO_RATE (&spec->info);
wfx.wBitsPerSample = GST_AUDIO_INFO_BPF (&spec->info) * 8 / wfx.nChannels;
wfx.nBlockAlign = GST_AUDIO_INFO_BPF (&spec->info);
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
/* Ignored for WAVE_FORMAT_PCM. */
wfx.cbSize = 0;
if (wfx.wBitsPerSample != 16 && wfx.wBitsPerSample != 8)
goto dodgy_width;
GST_INFO_OBJECT (asrc, "latency time: %" G_GUINT64_FORMAT " - buffer time: %"
G_GUINT64_FORMAT, spec->latency_time, spec->buffer_time);
/* Buffer-time should always be >= 2*latency */
if (spec->buffer_time < spec->latency_time * 2) {
spec->buffer_time = spec->latency_time * 2;
GST_WARNING ("buffer-time was less than 2*latency-time, clamping");
}
/* Set the buffer size from our configured buffer time (in microsecs) */
dsoundsrc->buffer_size =
gst_util_uint64_scale_int (spec->buffer_time, wfx.nAvgBytesPerSec,
GST_SECOND / GST_USECOND);
GST_INFO_OBJECT (asrc, "Buffer size: %d", dsoundsrc->buffer_size);
spec->segsize =
gst_util_uint64_scale (spec->latency_time, wfx.nAvgBytesPerSec,
GST_SECOND / GST_USECOND);
/* Sanitized segsize */
if (spec->segsize < GST_AUDIO_INFO_BPF (&spec->info))
spec->segsize = GST_AUDIO_INFO_BPF (&spec->info);
else if (spec->segsize % GST_AUDIO_INFO_BPF (&spec->info) != 0)
spec->segsize =
((spec->segsize + GST_AUDIO_INFO_BPF (&spec->info) -
1) / GST_AUDIO_INFO_BPF (&spec->info)) *
GST_AUDIO_INFO_BPF (&spec->info);
spec->segtotal = dsoundsrc->buffer_size / spec->segsize;
/* The device usually takes time = 1-2 segments to start producing buffers */
spec->seglatency = spec->segtotal + 2;
/* Fetch and set the actual latency time that will be used */
dsoundsrc->latency_time =
gst_util_uint64_scale (spec->segsize, GST_SECOND / GST_USECOND,
GST_AUDIO_INFO_BPF (&spec->info) * GST_AUDIO_INFO_RATE (&spec->info));
GST_INFO_OBJECT (asrc, "actual latency time: %" G_GUINT64_FORMAT,
spec->latency_time);
/* Init secondary buffer desciption */
memset (&descSecondary, 0, sizeof (DSCBUFFERDESC));
descSecondary.dwSize = sizeof (DSCBUFFERDESC);
descSecondary.dwFlags = 0;
descSecondary.dwReserved = 0;
/* This is not primary buffer so have to set size */
descSecondary.dwBufferBytes = dsoundsrc->buffer_size;
descSecondary.lpwfxFormat = &wfx;
/* Create buffer */
hRes = IDirectSoundCapture_CreateCaptureBuffer (dsoundsrc->pDSC,
&descSecondary, &dsoundsrc->pDSBSecondary, NULL);
if (hRes != DS_OK)
goto capture_buffer;
dsoundsrc->bytes_per_sample = GST_AUDIO_INFO_BPF (&spec->info);
GST_INFO_OBJECT (asrc,
"bytes/sec: %lu, buffer size: %d, segsize: %d, segtotal: %d",
wfx.nAvgBytesPerSec, dsoundsrc->buffer_size, spec->segsize,
spec->segtotal);
/* Not read anything yet */
dsoundsrc->current_circular_offset = 0;
GST_INFO_OBJECT (asrc, "channels: %d, rate: %d, bytes_per_sample: %d"
" WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d,"
" WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld",
GST_AUDIO_INFO_CHANNELS (&spec->info), GST_AUDIO_INFO_RATE (&spec->info),
GST_AUDIO_INFO_BPF (&spec->info), wfx.nSamplesPerSec, wfx.wBitsPerSample,
wfx.nBlockAlign, wfx.nAvgBytesPerSec);
return TRUE;
capture_buffer:
{
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to create capturebuffer"), (NULL));
return FALSE;
}
dodgy_width:
{
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unexpected width %d", wfx.wBitsPerSample), (NULL));
return FALSE;
}
}
static gboolean
gst_osx_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
GstAudioRingBufferSpec * spec)
{
gboolean ret = FALSE, is_passthrough = FALSE;
GstOsxAudioRingBuffer *osxbuf;
AudioStreamBasicDescription format;
osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);
if (RINGBUFFER_IS_SPDIF (spec->type)) {
format.mFormatID = kAudioFormat60958AC3;
format.mSampleRate = (double) GST_AUDIO_INFO_RATE (&spec->info);
format.mChannelsPerFrame = 2;
format.mFormatFlags = kAudioFormatFlagIsSignedInteger |
kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonMixable;
format.mBytesPerFrame = 0;
format.mBitsPerChannel = 16;
format.mBytesPerPacket = 6144;
format.mFramesPerPacket = 1536;
format.mReserved = 0;
spec->segsize = 6144;
spec->segtotal = 10;
is_passthrough = TRUE;
} else {
int width, depth;
/* Fill out the audio description we're going to be using */
format.mFormatID = kAudioFormatLinearPCM;
format.mSampleRate = (double) GST_AUDIO_INFO_RATE (&spec->info);
format.mChannelsPerFrame = GST_AUDIO_INFO_CHANNELS (&spec->info);
if (GST_AUDIO_INFO_IS_FLOAT (&spec->info)) {
format.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;
width = depth = GST_AUDIO_INFO_WIDTH (&spec->info);
} else {
format.mFormatFlags = kAudioFormatFlagIsSignedInteger;
width = GST_AUDIO_INFO_WIDTH (&spec->info);
depth = GST_AUDIO_INFO_DEPTH (&spec->info);
if (width == depth) {
format.mFormatFlags |= kAudioFormatFlagIsPacked;
} else {
format.mFormatFlags |= kAudioFormatFlagIsAlignedHigh;
}
}
if (GST_AUDIO_INFO_IS_BIG_ENDIAN (&spec->info)) {
format.mFormatFlags |= kAudioFormatFlagIsBigEndian;
}
format.mBytesPerFrame = GST_AUDIO_INFO_BPF (&spec->info);
format.mBitsPerChannel = depth;
format.mBytesPerPacket = GST_AUDIO_INFO_BPF (&spec->info);
format.mFramesPerPacket = 1;
format.mReserved = 0;
spec->segsize =
(spec->latency_time * GST_AUDIO_INFO_RATE (&spec->info) /
G_USEC_PER_SEC) * GST_AUDIO_INFO_BPF (&spec->info);
spec->segtotal = spec->buffer_time / spec->latency_time;
is_passthrough = FALSE;
}
GST_DEBUG_OBJECT (osxbuf, "Format: " CORE_AUDIO_FORMAT,
CORE_AUDIO_FORMAT_ARGS (format));
/* gst_audio_ring_buffer_set_channel_positions is not called
* since the AUs perform channel reordering themselves.
* (see gst_core_audio_set_channel_layout) */
buf->size = spec->segtotal * spec->segsize;
buf->memory = g_malloc0 (buf->size);
ret = gst_core_audio_initialize (osxbuf->core_audio, format, spec->caps,
is_passthrough);
if (!ret) {
g_free (buf->memory);
buf->memory = NULL;
buf->size = 0;
}
osxbuf->segoffset = 0;
return ret;
}
static gboolean
gst_alsasrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
if (!alsasrc_parse_spec (alsa, spec))
goto spec_parse;
CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
CHECK (set_hwparams (alsa), hw_params_failed);
CHECK (set_swparams (alsa), sw_params_failed);
CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
spec->segsize = alsa->period_size * alsa->bpf;
spec->segtotal = alsa->buffer_size / alsa->period_size;
{
snd_output_t *out_buf = NULL;
char *msg = NULL;
snd_output_buffer_open (&out_buf);
snd_pcm_dump_hw_setup (alsa->handle, out_buf);
snd_output_buffer_string (out_buf, &msg);
GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg);
snd_output_close (out_buf);
snd_output_buffer_open (&out_buf);
snd_pcm_dump_sw_setup (alsa->handle, out_buf);
snd_output_buffer_string (out_buf, &msg);
GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg);
snd_output_close (out_buf);
}
#ifdef SND_CHMAP_API_VERSION
alsa_detect_channels_mapping (GST_OBJECT (alsa), alsa->handle, spec,
alsa->channels, GST_AUDIO_BASE_SRC (alsa)->ringbuffer);
#endif /* SND_CHMAP_API_VERSION */
return TRUE;
/* ERRORS */
spec_parse:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Error parsing spec"));
return FALSE;
}
non_block:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Could not set device to blocking: %s", snd_strerror (err)));
return FALSE;
}
hw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Setting of hwparams failed: %s", snd_strerror (err)));
return FALSE;
}
sw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Setting of swparams failed: %s", snd_strerror (err)));
return FALSE;
}
prepare_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Prepare failed: %s", snd_strerror (err)));
return FALSE;
}
}
static GstFlowReturn
gst_videoframe_audiolevel_asink_chain (GstPad * pad, GstObject * parent,
GstBuffer * inbuf)
{
GstClockTime timestamp, cur_time;
GstVideoFrameAudioLevel *self = GST_VIDEOFRAME_AUDIOLEVEL (parent);
GstBuffer *buf;
gsize inbuf_size;
guint64 start_offset, end_offset;
GstClockTime running_time;
gint rate, bpf;
gboolean discont = FALSE;
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
running_time =
gst_segment_to_running_time (&self->asegment, GST_FORMAT_TIME, timestamp);
rate = GST_AUDIO_INFO_RATE (&self->ainfo);
bpf = GST_AUDIO_INFO_BPF (&self->ainfo);
start_offset = gst_util_uint64_scale (timestamp, rate, GST_SECOND);
inbuf_size = gst_buffer_get_size (inbuf);
end_offset = start_offset + inbuf_size / bpf;
g_mutex_lock (&self->mutex);
if (GST_BUFFER_IS_DISCONT (inbuf)
|| GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_RESYNC)
|| self->first_time == GST_CLOCK_TIME_NONE) {
discont = TRUE;
} else {
guint64 diff, max_sample_diff;
/* Check discont, based on audiobasesink */
if (start_offset <= self->next_offset)
diff = self->next_offset - start_offset;
else
diff = start_offset - self->next_offset;
max_sample_diff =
gst_util_uint64_scale_int (self->alignment_threshold, rate, GST_SECOND);
/* Discont! */
if (G_UNLIKELY (diff >= max_sample_diff)) {
if (self->discont_wait > 0) {
if (self->discont_time == GST_CLOCK_TIME_NONE) {
self->discont_time = timestamp;
} else if (timestamp - self->discont_time >= self->discont_wait) {
discont = TRUE;
self->discont_time = GST_CLOCK_TIME_NONE;
}
} else {
discont = TRUE;
}
} else if (G_UNLIKELY (self->discont_time != GST_CLOCK_TIME_NONE)) {
/* we have had a discont, but are now back on track! */
self->discont_time = GST_CLOCK_TIME_NONE;
}
}
if (discont) {
/* Have discont, need resync */
if (self->next_offset != -1)
GST_INFO_OBJECT (pad, "Have discont. Expected %"
G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
self->next_offset, start_offset);
self->total_frames = 0;
self->first_time = running_time;
self->next_offset = end_offset;
} else {
self->next_offset += inbuf_size / bpf;
}
gst_adapter_push (self->adapter, gst_buffer_ref (inbuf));
GST_DEBUG_OBJECT (self, "Queue length %i",
g_queue_get_length (&self->vtimeq));
while (TRUE) {
GstClockTime *vt0, *vt1;
GstClockTime vtemp;
GstMessage *msg;
gsize bytes, available_bytes;
vtemp = GST_CLOCK_TIME_NONE;
while (!(g_queue_get_length (&self->vtimeq) >= 2 || self->video_eos_flag
|| self->audio_flush_flag || self->shutdown_flag))
g_cond_wait (&self->cond, &self->mutex);
if (self->audio_flush_flag || self->shutdown_flag) {
g_mutex_unlock (&self->mutex);
gst_buffer_unref (inbuf);
return GST_FLOW_FLUSHING;
} else if (self->video_eos_flag) {
GST_DEBUG_OBJECT (self, "Video EOS flag alert");
/* nothing to do here if queue is empty */
if (g_queue_get_length (&self->vtimeq) == 0)
break;
if (g_queue_get_length (&self->vtimeq) < 2) {
vtemp = self->vsegment.position;
} else if (self->vsegment.position == GST_CLOCK_TIME_NONE) {
/* g_queue_get_length is surely >= 2 at this point
* so the adapter isn't empty */
buf =
gst_adapter_take_buffer (self->adapter,
gst_adapter_available (self->adapter));
if (buf != NULL) {
GstMessage *msg;
msg = update_rms_from_buffer (self, buf);
g_mutex_unlock (&self->mutex);
gst_element_post_message (GST_ELEMENT (self), msg);
gst_buffer_unref (buf);
g_mutex_lock (&self->mutex); /* we unlock again later */
}
break;
}
} else if (g_queue_get_length (&self->vtimeq) < 2) {
continue;
}
vt0 = g_queue_pop_head (&self->vtimeq);
if (vtemp == GST_CLOCK_TIME_NONE)
vt1 = g_queue_peek_head (&self->vtimeq);
else
vt1 = &vtemp;
cur_time =
self->first_time + gst_util_uint64_scale (self->total_frames,
GST_SECOND, rate);
GST_DEBUG_OBJECT (self,
"Processing: current time is %" GST_TIME_FORMAT,
GST_TIME_ARGS (cur_time));
GST_DEBUG_OBJECT (self, "Total frames is %i with a rate of %d",
self->total_frames, rate);
GST_DEBUG_OBJECT (self, "Start time is %" GST_TIME_FORMAT,
GST_TIME_ARGS (self->first_time));
GST_DEBUG_OBJECT (self, "Time on top is %" GST_TIME_FORMAT,
GST_TIME_ARGS (*vt0));
if (cur_time < *vt0) {
guint num_frames =
gst_util_uint64_scale (*vt0 - cur_time, rate, GST_SECOND);
bytes = num_frames * GST_AUDIO_INFO_BPF (&self->ainfo);
available_bytes = gst_adapter_available (self->adapter);
if (available_bytes == 0) {
g_queue_push_head (&self->vtimeq, vt0);
break;
}
if (bytes == 0) {
cur_time = *vt0;
} else {
GST_DEBUG_OBJECT (self,
"Flushed %" G_GSIZE_FORMAT " out of %" G_GSIZE_FORMAT " bytes",
bytes, available_bytes);
gst_adapter_flush (self->adapter, MIN (bytes, available_bytes));
self->total_frames += num_frames;
if (available_bytes <= bytes) {
g_queue_push_head (&self->vtimeq, vt0);
break;
}
cur_time =
self->first_time + gst_util_uint64_scale (self->total_frames,
GST_SECOND, rate);
}
}
if (*vt1 > cur_time) {
bytes =
GST_AUDIO_INFO_BPF (&self->ainfo) * gst_util_uint64_scale (*vt1 -
cur_time, rate, GST_SECOND);
} else {
bytes = 0; /* We just need to discard vt0 */
}
available_bytes = gst_adapter_available (self->adapter);
GST_DEBUG_OBJECT (self,
"Adapter contains %" G_GSIZE_FORMAT " out of %" G_GSIZE_FORMAT " bytes",
available_bytes, bytes);
if (available_bytes < bytes) {
g_queue_push_head (&self->vtimeq, vt0);
goto done;
}
if (bytes > 0) {
buf = gst_adapter_take_buffer (self->adapter, bytes);
g_assert (buf != NULL);
} else {
/* Just an empty buffer */
buf = gst_buffer_new ();
}
msg = update_rms_from_buffer (self, buf);
g_mutex_unlock (&self->mutex);
gst_element_post_message (GST_ELEMENT (self), msg);
g_mutex_lock (&self->mutex);
gst_buffer_unref (buf);
g_free (vt0);
if (available_bytes == bytes)
break;
}
done:
g_mutex_unlock (&self->mutex);
return gst_pad_push (self->asrcpad, inbuf);
}
/**
* gst_audio_info_convert:
* @info: a #GstAudioInfo
* @src_fmt: #GstFormat of the @src_val
* @src_val: value to convert
* @dest_fmt: #GstFormat of the @dest_val
* @dest_val: pointer to destination value
*
* Converts among various #GstFormat types. This function handles
* GST_FORMAT_BYTES, GST_FORMAT_TIME, and GST_FORMAT_DEFAULT. For
* raw audio, GST_FORMAT_DEFAULT corresponds to audio frames. This
* function can be used to handle pad queries of the type GST_QUERY_CONVERT.
*
* Returns: TRUE if the conversion was successful.
*/
gboolean
gst_audio_info_convert (const GstAudioInfo * info,
GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val)
{
gboolean res = TRUE;
gint bpf, rate;
GST_DEBUG ("converting value %" G_GINT64_FORMAT " from %s (%d) to %s (%d)",
src_val, gst_format_get_name (src_fmt), src_fmt,
gst_format_get_name (dest_fmt), dest_fmt);
if (src_fmt == dest_fmt || src_val == -1) {
*dest_val = src_val;
goto done;
}
/* get important info */
bpf = GST_AUDIO_INFO_BPF (info);
rate = GST_AUDIO_INFO_RATE (info);
if (bpf == 0 || rate == 0) {
GST_DEBUG ("no rate or bpf configured");
res = FALSE;
goto done;
}
switch (src_fmt) {
case GST_FORMAT_BYTES:
switch (dest_fmt) {
case GST_FORMAT_TIME:
*dest_val = GST_FRAMES_TO_CLOCK_TIME (src_val / bpf, rate);
break;
case GST_FORMAT_DEFAULT:
*dest_val = src_val / bpf;
break;
default:
res = FALSE;
break;
}
break;
case GST_FORMAT_DEFAULT:
switch (dest_fmt) {
case GST_FORMAT_TIME:
*dest_val = GST_FRAMES_TO_CLOCK_TIME (src_val, rate);
break;
case GST_FORMAT_BYTES:
*dest_val = src_val * bpf;
break;
default:
res = FALSE;
break;
}
break;
case GST_FORMAT_TIME:
switch (dest_fmt) {
case GST_FORMAT_DEFAULT:
*dest_val = GST_CLOCK_TIME_TO_FRAMES (src_val, rate);
break;
case GST_FORMAT_BYTES:
*dest_val = GST_CLOCK_TIME_TO_FRAMES (src_val, rate);
*dest_val *= bpf;
break;
default:
res = FALSE;
break;
}
break;
default:
res = FALSE;
break;
}
done:
GST_DEBUG ("ret=%d result %" G_GINT64_FORMAT, res, res ? *dest_val : -1);
return res;
}
static GstFlowReturn
gst_ladspa_source_type_fill (GstBaseSrc * base, guint64 offset,
guint length, GstBuffer * buffer)
{
GstLADSPASource *ladspa = GST_LADSPA_SOURCE (base);
GstClockTime next_time;
gint64 next_sample, next_byte;
gint bytes, samples;
GstElementClass *eclass;
GstMapInfo map;
gint samplerate, bpf;
/* example for tagging generated data */
if (!ladspa->tags_pushed) {
GstTagList *taglist;
taglist = gst_tag_list_new (GST_TAG_DESCRIPTION, "ladspa wave", NULL);
eclass = GST_ELEMENT_CLASS (gst_ladspa_source_type_parent_class);
if (eclass->send_event)
eclass->send_event (GST_ELEMENT (base), gst_event_new_tag (taglist));
else
gst_tag_list_unref (taglist);
ladspa->tags_pushed = TRUE;
}
if (ladspa->eos_reached) {
GST_INFO_OBJECT (ladspa, "eos");
return GST_FLOW_EOS;
}
samplerate = GST_AUDIO_INFO_RATE (&ladspa->info);
bpf = GST_AUDIO_INFO_BPF (&ladspa->info);
/* if no length was given, use our default length in samples otherwise convert
* the length in bytes to samples. */
if (length == -1)
samples = ladspa->samples_per_buffer;
else
samples = length / bpf;
/* if no offset was given, use our next logical byte */
if (offset == -1)
offset = ladspa->next_byte;
/* now see if we are at the byteoffset we think we are */
if (offset != ladspa->next_byte) {
GST_DEBUG_OBJECT (ladspa, "seek to new offset %" G_GUINT64_FORMAT, offset);
/* we have a discont in the expected sample offset, do a 'seek' */
ladspa->next_sample = offset / bpf;
ladspa->next_time =
gst_util_uint64_scale_int (ladspa->next_sample, GST_SECOND, samplerate);
ladspa->next_byte = offset;
}
/* check for eos */
if (ladspa->check_seek_stop &&
(ladspa->sample_stop > ladspa->next_sample) &&
(ladspa->sample_stop < ladspa->next_sample + samples)
) {
/* calculate only partial buffer */
ladspa->generate_samples_per_buffer =
ladspa->sample_stop - ladspa->next_sample;
next_sample = ladspa->sample_stop;
ladspa->eos_reached = TRUE;
} else {
/* calculate full buffer */
ladspa->generate_samples_per_buffer = samples;
next_sample =
ladspa->next_sample + (ladspa->reverse ? (-samples) : samples);
}
bytes = ladspa->generate_samples_per_buffer * bpf;
next_byte = ladspa->next_byte + (ladspa->reverse ? (-bytes) : bytes);
next_time = gst_util_uint64_scale_int (next_sample, GST_SECOND, samplerate);
GST_LOG_OBJECT (ladspa, "samplerate %d", samplerate);
GST_LOG_OBJECT (ladspa,
"next_sample %" G_GINT64_FORMAT ", ts %" GST_TIME_FORMAT, next_sample,
GST_TIME_ARGS (next_time));
gst_buffer_set_size (buffer, bytes);
GST_BUFFER_OFFSET (buffer) = ladspa->next_sample;
GST_BUFFER_OFFSET_END (buffer) = next_sample;
if (!ladspa->reverse) {
GST_BUFFER_TIMESTAMP (buffer) =
ladspa->timestamp_offset + ladspa->next_time;
GST_BUFFER_DURATION (buffer) = next_time - ladspa->next_time;
} else {
GST_BUFFER_TIMESTAMP (buffer) = ladspa->timestamp_offset + next_time;
GST_BUFFER_DURATION (buffer) = ladspa->next_time - next_time;
}
gst_object_sync_values (GST_OBJECT (ladspa), GST_BUFFER_TIMESTAMP (buffer));
ladspa->next_time = next_time;
ladspa->next_sample = next_sample;
ladspa->next_byte = next_byte;
GST_LOG_OBJECT (ladspa, "generating %u samples at ts %" GST_TIME_FORMAT,
ladspa->generate_samples_per_buffer,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
gst_ladspa_transform (&ladspa->ladspa, map.data,
ladspa->generate_samples_per_buffer, NULL);
gst_buffer_unmap (buffer, &map);
return GST_FLOW_OK;
}
static gboolean
gst_alsasrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstAlsaSrc *alsa;
gint err;
alsa = GST_ALSA_SRC (asrc);
if (!alsasrc_parse_spec (alsa, spec))
goto spec_parse;
CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block);
CHECK (set_hwparams (alsa), hw_params_failed);
CHECK (set_swparams (alsa), sw_params_failed);
CHECK (snd_pcm_prepare (alsa->handle), prepare_failed);
alsa->bpf = GST_AUDIO_INFO_BPF (&spec->info);
spec->segsize = alsa->period_size * alsa->bpf;
spec->segtotal = alsa->buffer_size / alsa->period_size;
{
snd_output_t *out_buf = NULL;
char *msg = NULL;
snd_output_buffer_open (&out_buf);
snd_pcm_dump_hw_setup (alsa->handle, out_buf);
snd_output_buffer_string (out_buf, &msg);
GST_DEBUG_OBJECT (alsa, "Hardware setup: \n%s", msg);
snd_output_close (out_buf);
snd_output_buffer_open (&out_buf);
snd_pcm_dump_sw_setup (alsa->handle, out_buf);
snd_output_buffer_string (out_buf, &msg);
GST_DEBUG_OBJECT (alsa, "Software setup: \n%s", msg);
snd_output_close (out_buf);
}
#ifdef SND_CHMAP_API_VERSION
if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW && alsa->channels < 9) {
snd_pcm_chmap_t *chmap = snd_pcm_get_chmap (alsa->handle);
if (chmap && chmap->channels == alsa->channels) {
GstAudioChannelPosition pos[8];
if (alsa_chmap_to_channel_positions (chmap, pos))
gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
(alsa)->ringbuffer, pos);
}
free (chmap);
}
#endif /* SND_CHMAP_API_VERSION */
return TRUE;
/* ERRORS */
spec_parse:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Error parsing spec"));
return FALSE;
}
non_block:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Could not set device to blocking: %s", snd_strerror (err)));
return FALSE;
}
hw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Setting of hwparams failed: %s", snd_strerror (err)));
return FALSE;
}
sw_params_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Setting of swparams failed: %s", snd_strerror (err)));
return FALSE;
}
prepare_failed:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Prepare failed: %s", snd_strerror (err)));
return FALSE;
}
}
