/* this function does the actual processing
*/
static GstFlowReturn
gst_freeverb_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstFreeverb *filter = GST_FREEVERB (base);
guint num_samples;
GstClockTime timestamp;
GstMapInfo inmap, outmap;
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
timestamp =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
num_samples = outmap.size / (2 * GST_AUDIO_INFO_BPS (&filter->info));
GST_DEBUG_OBJECT (filter, "processing %u samples at %" GST_TIME_FORMAT,
num_samples, GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (GST_OBJECT (filter), timestamp);
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DISCONT))) {
filter->drained = FALSE;
}
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))) {
if (filter->drained) {
memset (outmap.data, 0, outmap.size);
}
} else {
filter->drained = FALSE;
}
if (!filter->drained) {
filter->drained =
filter->process (filter, inmap.data, outmap.data, num_samples);
}
if (filter->drained) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
}
gst_buffer_unmap (inbuf, &inmap);
gst_buffer_unmap (outbuf, &outmap);
return GST_FLOW_OK;
}
/* this function does the actual processing
*/
static GstFlowReturn
gst_audio_panorama_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstAudioPanorama *filter = GST_AUDIO_PANORAMA (base);
GstClockTime timestamp, stream_time;
GstMapInfo inmap, outmap;
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (filter), stream_time);
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
memset (outmap.data, 0, outmap.size);
} else {
/* output always stereo, input mono or stereo,
* and info describes input format */
guint num_samples = outmap.size / (2 * GST_AUDIO_INFO_BPS (&filter->info));
filter->process (filter, inmap.data, outmap.data, num_samples);
}
gst_buffer_unmap (inbuf, &inmap);
gst_buffer_unmap (outbuf, &outmap);
return GST_FLOW_OK;
}
static void
gst_openal_sink_parse_spec (GstOpenALSink * sink,
const GstAudioRingBufferSpec * spec)
{
ALuint format = AL_NONE;
GST_DEBUG_OBJECT (sink,
"looking up format for type %d, gst-format %d, and %d channels",
spec->type, GST_AUDIO_INFO_FORMAT (&spec->info),
GST_AUDIO_INFO_CHANNELS (&spec->info));
/* Don't need to verify supported formats, since the probed caps will only
* report what was detected and we shouldn't get anything different */
switch (spec->type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
case GST_AUDIO_FORMAT_U8:
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
case 1:
format = AL_FORMAT_MONO8;
break;
case 2:
format = AL_FORMAT_STEREO8;
break;
case 4:
format = AL_FORMAT_QUAD8;
break;
case 6:
format = AL_FORMAT_51CHN8;
break;
case 7:
format = AL_FORMAT_61CHN8;
break;
case 8:
format = AL_FORMAT_71CHN8;
break;
default:
break;
}
break;
case GST_AUDIO_FORMAT_S16:
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
case 1:
format = AL_FORMAT_MONO16;
break;
case 2:
format = AL_FORMAT_STEREO16;
break;
case 4:
format = AL_FORMAT_QUAD16;
break;
case 6:
format = AL_FORMAT_51CHN16;
break;
case 7:
format = AL_FORMAT_61CHN16;
break;
case 8:
format = AL_FORMAT_71CHN16;
break;
default:
break;
}
break;
case GST_AUDIO_FORMAT_F32:
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
case 1:
format = AL_FORMAT_MONO_FLOAT32;
break;
case 2:
format = AL_FORMAT_STEREO_FLOAT32;
break;
case 4:
format = AL_FORMAT_QUAD32;
break;
case 6:
format = AL_FORMAT_51CHN32;
break;
case 7:
format = AL_FORMAT_61CHN32;
break;
case 8:
format = AL_FORMAT_71CHN32;
break;
default:
break;
}
break;
case GST_AUDIO_FORMAT_F64:
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
case 1:
format = AL_FORMAT_MONO_DOUBLE_EXT;
break;
case 2:
format = AL_FORMAT_STEREO_DOUBLE_EXT;
break;
default:
break;
}
break;
default:
break;
}
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM:
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
case 1:
format = AL_FORMAT_MONO_IMA4;
break;
case 2:
format = AL_FORMAT_STEREO_IMA4;
break;
default:
break;
}
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
case 1:
format = AL_FORMAT_MONO_ALAW_EXT;
break;
case 2:
format = AL_FORMAT_STEREO_ALAW_EXT;
break;
default:
break;
}
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
case 1:
format = AL_FORMAT_MONO_MULAW;
break;
case 2:
format = AL_FORMAT_STEREO_MULAW;
break;
case 4:
format = AL_FORMAT_QUAD_MULAW;
break;
case 6:
format = AL_FORMAT_51CHN_MULAW;
break;
case 7:
format = AL_FORMAT_61CHN_MULAW;
break;
case 8:
format = AL_FORMAT_71CHN_MULAW;
break;
default:
break;
}
break;
default:
break;
}
sink->bytes_per_sample = GST_AUDIO_INFO_BPS (&spec->info);
sink->rate = GST_AUDIO_INFO_RATE (&spec->info);
sink->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
sink->format = format;
sink->buffer_count = spec->segtotal;
sink->buffer_length = spec->segsize;
}
static GstFlowReturn
gst_fdkaacenc_handle_frame (GstAudioEncoder * enc, GstBuffer * inbuf)
{
GstFdkAacEnc *self = GST_FDKAACENC (enc);
GstFlowReturn ret = GST_FLOW_OK;
GstAudioInfo *info;
GstMapInfo imap, omap;
GstBuffer *outbuf;
AACENC_BufDesc in_desc = { 0 };
AACENC_BufDesc out_desc = { 0 };
AACENC_InArgs in_args = { 0 };
AACENC_OutArgs out_args = { 0 };
gint in_id = IN_AUDIO_DATA, out_id = OUT_BITSTREAM_DATA;
gint in_sizes, out_sizes;
gint in_el_sizes, out_el_sizes;
AACENC_ERROR err;
info = gst_audio_encoder_get_audio_info (enc);
if (inbuf) {
if (self->need_reorder) {
inbuf = gst_buffer_copy (inbuf);
gst_buffer_map (inbuf, &imap, GST_MAP_READWRITE);
gst_audio_reorder_channels (imap.data, imap.size,
GST_AUDIO_INFO_FORMAT (info), GST_AUDIO_INFO_CHANNELS (info),
&GST_AUDIO_INFO_POSITION (info, 0), self->aac_positions);
} else {
gst_buffer_map (inbuf, &imap, GST_MAP_READ);
}
in_args.numInSamples = imap.size / GST_AUDIO_INFO_BPS (info);
in_sizes = imap.size;
in_el_sizes = GST_AUDIO_INFO_BPS (info);
in_desc.numBufs = 1;
} else {
in_args.numInSamples = -1;
in_sizes = 0;
in_el_sizes = 0;
in_desc.numBufs = 0;
}
in_desc.bufferIdentifiers = &in_id;
in_desc.bufs = (void *) &imap.data;
in_desc.bufSizes = &in_sizes;
in_desc.bufElSizes = &in_el_sizes;
outbuf = gst_audio_encoder_allocate_output_buffer (enc, self->outbuf_size);
if (!outbuf) {
ret = GST_FLOW_ERROR;
goto out;
}
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_sizes = omap.size;
out_el_sizes = 1;
out_desc.bufferIdentifiers = &out_id;
out_desc.numBufs = 1;
out_desc.bufs = (void *) &omap.data;
out_desc.bufSizes = &out_sizes;
out_desc.bufElSizes = &out_el_sizes;
err = aacEncEncode (self->enc, &in_desc, &out_desc, &in_args, &out_args);
if (err == AACENC_ENCODE_EOF && !inbuf)
goto out;
else if (err != AACENC_OK) {
GST_ERROR_OBJECT (self, "Failed to encode data: %d", err);
ret = GST_FLOW_ERROR;
goto out;
}
if (inbuf) {
gst_buffer_unmap (inbuf, &imap);
if (self->need_reorder)
gst_buffer_unref (inbuf);
inbuf = NULL;
}
if (!out_args.numOutBytes)
goto out;
gst_buffer_unmap (outbuf, &omap);
gst_buffer_set_size (outbuf, out_args.numOutBytes);
ret = gst_audio_encoder_finish_frame (enc, outbuf, self->samples_per_frame);
outbuf = NULL;
out:
if (outbuf) {
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unref (outbuf);
}
if (inbuf) {
gst_buffer_unmap (inbuf, &imap);
if (self->need_reorder)
gst_buffer_unref (inbuf);
}
return ret;
}
static void
gst_openal_src_parse_spec (GstOpenalSrc * openalsrc,
const GstAudioRingBufferSpec * spec)
{
ALuint format = AL_NONE;
GST_DEBUG_OBJECT (openalsrc,
"looking up format for type %d, gst-format %d, and %d channels",
spec->type, GST_AUDIO_INFO_FORMAT (&spec->info),
GST_AUDIO_INFO_CHANNELS (&spec->info));
switch (spec->type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:
switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
case GST_AUDIO_FORMAT_U8:
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
case 1:
format = AL_FORMAT_MONO8;
break;
default:
break;
}
break;
case GST_AUDIO_FORMAT_U16:
case GST_AUDIO_FORMAT_S16:
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
case 1:
format = AL_FORMAT_MONO16;
break;
default:
break;
}
break;
case GST_AUDIO_FORMAT_F32:
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
case 1:
format = AL_FORMAT_MONO_FLOAT32;
break;
default:
break;
}
break;
case GST_AUDIO_FORMAT_F64:
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
case 1:
format = AL_FORMAT_MONO_DOUBLE_EXT;
break;
default:
break;
}
break;
default:
break;
}
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM:
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
case 1:
format = AL_FORMAT_MONO_IMA4;
break;
default:
break;
}
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
case 1:
format = AL_FORMAT_MONO_ALAW_EXT;
break;
default:
break;
}
break;
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:
switch (GST_AUDIO_INFO_CHANNELS (&spec->info)) {
case 1:
format = AL_FORMAT_MONO_MULAW;
break;
default:
break;
}
break;
default:
break;
}
openalsrc->bytes_per_sample = GST_AUDIO_INFO_BPS (&spec->info);
openalsrc->rate = GST_AUDIO_INFO_RATE (&spec->info);
openalsrc->buffer_length = spec->segsize;
openalsrc->format = format;
}
static GstFlowReturn
gst_level_transform_ip (GstBaseTransform * trans, GstBuffer * in)
{
GstLevel *filter;
GstMapInfo map;
guint8 *in_data;
gsize in_size;
gdouble CS;
guint i;
guint num_frames;
guint num_int_samples = 0; /* number of interleaved samples
* ie. total count for all channels combined */
guint block_size, block_int_size; /* we subdivide buffers to not skip message
* intervals */
GstClockTimeDiff falloff_time;
gint channels, rate, bps;
filter = GST_LEVEL (trans);
channels = GST_AUDIO_INFO_CHANNELS (&filter->info);
bps = GST_AUDIO_INFO_BPS (&filter->info);
rate = GST_AUDIO_INFO_RATE (&filter->info);
gst_buffer_map (in, &map, GST_MAP_READ);
in_data = map.data;
in_size = map.size;
num_int_samples = in_size / bps;
GST_LOG_OBJECT (filter, "analyzing %u sample frames at ts %" GST_TIME_FORMAT,
num_int_samples, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (in)));
g_return_val_if_fail (num_int_samples % channels == 0, GST_FLOW_ERROR);
if (GST_BUFFER_FLAG_IS_SET (in, GST_BUFFER_FLAG_DISCONT)) {
filter->message_ts = GST_BUFFER_TIMESTAMP (in);
filter->num_frames = 0;
}
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (filter->message_ts))) {
filter->message_ts = GST_BUFFER_TIMESTAMP (in);
}
num_frames = num_int_samples / channels;
while (num_frames > 0) {
block_size = filter->interval_frames - filter->num_frames;
block_size = MIN (block_size, num_frames);
block_int_size = block_size * channels;
for (i = 0; i < channels; ++i) {
if (!GST_BUFFER_FLAG_IS_SET (in, GST_BUFFER_FLAG_GAP)) {
filter->process (in_data + (bps * i), block_int_size, channels, &CS,
&filter->peak[i]);
GST_LOG_OBJECT (filter,
"[%d]: cumulative squares %lf, over %d samples/%d channels",
i, CS, block_int_size, channels);
filter->CS[i] += CS;
} else {
filter->peak[i] = 0.0;
}
filter->decay_peak_age[i] += GST_FRAMES_TO_CLOCK_TIME (num_frames, rate);
GST_LOG_OBJECT (filter,
"[%d]: peak %f, last peak %f, decay peak %f, age %" GST_TIME_FORMAT,
i, filter->peak[i], filter->last_peak[i], filter->decay_peak[i],
GST_TIME_ARGS (filter->decay_peak_age[i]));
/* update running peak */
if (filter->peak[i] > filter->last_peak[i])
filter->last_peak[i] = filter->peak[i];
/* make decay peak fall off if too old */
falloff_time =
GST_CLOCK_DIFF (gst_gdouble_to_guint64 (filter->decay_peak_ttl),
filter->decay_peak_age[i]);
if (falloff_time > 0) {
gdouble falloff_dB;
gdouble falloff;
gdouble length; /* length of falloff time in seconds */
length = (gdouble) falloff_time / (gdouble) GST_SECOND;
falloff_dB = filter->decay_peak_falloff * length;
falloff = pow (10, falloff_dB / -20.0);
GST_LOG_OBJECT (filter,
"falloff: current %f, base %f, interval %" GST_TIME_FORMAT
", dB falloff %f, factor %e",
filter->decay_peak[i], filter->decay_peak_base[i],
GST_TIME_ARGS (falloff_time), falloff_dB, falloff);
filter->decay_peak[i] = filter->decay_peak_base[i] * falloff;
GST_LOG_OBJECT (filter,
"peak is %" GST_TIME_FORMAT " old, decayed with factor %e to %f",
GST_TIME_ARGS (filter->decay_peak_age[i]), falloff,
filter->decay_peak[i]);
} else {
GST_LOG_OBJECT (filter, "peak not old enough, not decaying");
}
/* if the peak of this run is higher, the decay peak gets reset */
if (filter->peak[i] >= filter->decay_peak[i]) {
GST_LOG_OBJECT (filter, "new peak, %f", filter->peak[i]);
filter->decay_peak[i] = filter->peak[i];
filter->decay_peak_base[i] = filter->peak[i];
filter->decay_peak_age[i] = G_GINT64_CONSTANT (0);
}
}
in_data += block_size * bps * channels;
filter->num_frames += block_size;
num_frames -= block_size;
/* do we need to message ? */
if (filter->num_frames >= filter->interval_frames) {
gst_level_post_message (filter);
}
}
gst_buffer_unmap (in, &map);
return GST_FLOW_OK;
}
static GstMessage *
update_rms_from_buffer (GstVideoFrameAudioLevel * self, GstBuffer * inbuf)
{
GstMapInfo map;
guint8 *in_data;
gsize in_size;
gdouble CS;
guint i;
guint num_frames, frames;
guint num_int_samples = 0; /* number of interleaved samples
* ie. total count for all channels combined */
gint channels, rate, bps;
GValue v = G_VALUE_INIT;
GValue va = G_VALUE_INIT;
GValueArray *a;
GstStructure *s;
GstMessage *msg;
GstClockTime duration, running_time;
channels = GST_AUDIO_INFO_CHANNELS (&self->ainfo);
bps = GST_AUDIO_INFO_BPS (&self->ainfo);
rate = GST_AUDIO_INFO_RATE (&self->ainfo);
gst_buffer_map (inbuf, &map, GST_MAP_READ);
in_data = map.data;
in_size = map.size;
num_int_samples = in_size / bps;
GST_LOG_OBJECT (self, "analyzing %u sample frames at ts %" GST_TIME_FORMAT,
num_int_samples, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)));
g_return_val_if_fail (num_int_samples % channels == 0, NULL);
num_frames = num_int_samples / channels;
frames = num_frames;
duration = GST_FRAMES_TO_CLOCK_TIME (frames, rate);
if (num_frames > 0) {
for (i = 0; i < channels; ++i) {
self->process (in_data + (bps * i), num_int_samples, channels, &CS);
GST_LOG_OBJECT (self,
"[%d]: cumulative squares %lf, over %d samples/%d channels",
i, CS, num_int_samples, channels);
self->CS[i] += CS;
}
in_data += num_frames * bps;
self->total_frames += num_frames;
}
running_time =
self->first_time + gst_util_uint64_scale (self->total_frames, GST_SECOND,
rate);
a = g_value_array_new (channels);
s = gst_structure_new ("videoframe-audiolevel", "running-time", G_TYPE_UINT64,
running_time, "duration", G_TYPE_UINT64, duration, NULL);
g_value_init (&v, G_TYPE_DOUBLE);
g_value_init (&va, G_TYPE_VALUE_ARRAY);
for (i = 0; i < channels; i++) {
gdouble rms;
if (frames == 0 || self->CS[i] == 0) {
rms = 0; /* empty buffer */
} else {
rms = sqrt (self->CS[i] / frames);
}
self->CS[i] = 0.0;
g_value_set_double (&v, rms);
g_value_array_append (a, &v);
}
g_value_take_boxed (&va, a);
gst_structure_take_value (s, "rms", &va);
msg = gst_message_new_element (GST_OBJECT (self), s);
gst_buffer_unmap (inbuf, &map);
return msg;
}
