static void
gst_msdkdec_set_latency (GstMsdkDec * thiz)
{
GstVideoInfo *info = &thiz->input_state->info;
gint min_delayed_frames;
GstClockTime latency;
min_delayed_frames = thiz->async_depth;
if (info->fps_n) {
latency = gst_util_uint64_scale_ceil (GST_SECOND * info->fps_d,
min_delayed_frames, info->fps_n);
} else {
/* FIXME: Assume 25fps. This is better than reporting no latency at
* all and then later failing in live pipelines
*/
latency = gst_util_uint64_scale_ceil (GST_SECOND * 1,
min_delayed_frames, 25);
}
GST_INFO_OBJECT (thiz,
"Updating latency to %" GST_TIME_FORMAT " (%d frames)",
GST_TIME_ARGS (latency), min_delayed_frames);
gst_video_decoder_set_latency (GST_VIDEO_DECODER (thiz), latency, latency);
}
static void
gst_x265_enc_set_latency (GstX265Enc * encoder)
{
GstVideoInfo *info = &encoder->input_state->info;
gint max_delayed_frames;
GstClockTime latency;
/* FIXME get a real value from the encoder, this is currently not exposed */
if (encoder->tune > 0 && encoder->tune <= G_N_ELEMENTS (x265_tune_names) &&
strcmp (x265_tune_names[encoder->tune + 1], "zerolatency") == 0)
max_delayed_frames = 0;
else
max_delayed_frames = 5;
if (info->fps_n) {
latency = gst_util_uint64_scale_ceil (GST_SECOND * info->fps_d,
max_delayed_frames, info->fps_n);
} else {
/* FIXME: Assume 25fps. This is better than reporting no latency at
* all and then later failing in live pipelines
*/
latency = gst_util_uint64_scale_ceil (GST_SECOND * 1,
max_delayed_frames, 25);
}
GST_INFO_OBJECT (encoder,
"Updating latency to %" GST_TIME_FORMAT " (%d frames)",
GST_TIME_ARGS (latency), max_delayed_frames);
gst_video_encoder_set_latency (GST_VIDEO_ENCODER (encoder), latency, latency);
}
static gboolean
gst_decklink_video_src_query (GstBaseSrc * bsrc, GstQuery * query)
{
GstDecklinkVideoSrc *self = GST_DECKLINK_VIDEO_SRC_CAST (bsrc);
gboolean ret = TRUE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:{
if (self->input) {
GstClockTime min, max;
const GstDecklinkMode *mode;
g_mutex_lock (&self->lock);
mode = gst_decklink_get_mode (self->caps_mode);
g_mutex_unlock (&self->lock);
min = gst_util_uint64_scale_ceil (GST_SECOND, mode->fps_d, mode->fps_n);
max = self->buffer_size * min;
gst_query_set_latency (query, TRUE, min, max);
ret = TRUE;
} else {
ret = FALSE;
}
break;
}
default:
ret = GST_BASE_SRC_CLASS (parent_class)->query (bsrc, query);
break;
}
return ret;
}
static void
gst_x265_enc_set_latency (GstX265Enc * encoder)
{
GstVideoInfo *info = &encoder->input_state->info;
if (info->fps_n) {
GstClockTime latency;
gint max_delayed_frames;
// FIXME get a real value from the encoder, this is currently not exposed
max_delayed_frames = 5;
latency = gst_util_uint64_scale_ceil (GST_SECOND * info->fps_d,
max_delayed_frames, info->fps_n);
GST_INFO_OBJECT (encoder,
"Updating latency to %" GST_TIME_FORMAT " (%d frames)",
GST_TIME_ARGS (latency), max_delayed_frames);
gst_video_encoder_set_latency (GST_VIDEO_ENCODER (encoder), latency,
latency);
} else {
/* We can't do live as we don't know our latency */
gst_video_encoder_set_latency (GST_VIDEO_ENCODER (encoder),
0, GST_CLOCK_TIME_NONE);
}
}
/*
* gst_aiff_parse_calculate_duration:
* @aiff: aiffparse object
*
* Calculate duration on demand and store in @aiff.
*
* Returns: %TRUE if duration is available.
*/
static gboolean
gst_aiff_parse_calculate_duration (GstAiffParse * aiff)
{
if (aiff->duration > 0)
return TRUE;
if (aiff->datasize > 0 && aiff->bps > 0) {
aiff->duration =
gst_util_uint64_scale_ceil (aiff->datasize, GST_SECOND,
(guint64) aiff->bps);
GST_INFO_OBJECT (aiff, "Got duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (aiff->duration));
return TRUE;
}
return FALSE;
}
static gboolean
gst_omx_audio_enc_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
{
GstOMXAudioEnc *self;
GstOMXAudioEncClass *klass;
gboolean needs_disable = FALSE;
OMX_PARAM_PORTDEFINITIONTYPE port_def;
OMX_AUDIO_PARAM_PCMMODETYPE pcm_param;
gint i;
OMX_ERRORTYPE err;
self = GST_OMX_AUDIO_ENC (encoder);
klass = GST_OMX_AUDIO_ENC_GET_CLASS (encoder);
GST_DEBUG_OBJECT (self, "Setting new caps");
/* Set audio encoder base class properties */
gst_audio_encoder_set_frame_samples_min (encoder,
gst_util_uint64_scale_ceil (OMX_MIN_PCMPAYLOAD_MSEC,
GST_MSECOND * info->rate, GST_SECOND));
gst_audio_encoder_set_frame_samples_max (encoder, 0);
gst_omx_port_get_port_definition (self->enc_in_port, &port_def);
needs_disable =
gst_omx_component_get_state (self->enc,
GST_CLOCK_TIME_NONE) != OMX_StateLoaded;
/* If the component is not in Loaded state and a real format change happens
* we have to disable the port and re-allocate all buffers. If no real
* format change happened we can just exit here.
*/
if (needs_disable) {
GST_DEBUG_OBJECT (self, "Need to disable and drain encoder");
gst_omx_audio_enc_drain (self);
gst_omx_port_set_flushing (self->enc_out_port, 5 * GST_SECOND, TRUE);
/* Wait until the srcpad loop is finished,
* unlock GST_AUDIO_ENCODER_STREAM_LOCK to prevent deadlocks
* caused by using this lock from inside the loop function */
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
gst_pad_stop_task (GST_AUDIO_ENCODER_SRC_PAD (encoder));
GST_AUDIO_ENCODER_STREAM_LOCK (self);
if (gst_omx_port_set_enabled (self->enc_in_port, FALSE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_set_enabled (self->enc_out_port, FALSE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_buffers_released (self->enc_in_port,
5 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_buffers_released (self->enc_out_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_deallocate_buffers (self->enc_in_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_deallocate_buffers (self->enc_out_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->enc_in_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->enc_out_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
GST_DEBUG_OBJECT (self, "Encoder drained and disabled");
}
port_def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
GST_DEBUG_OBJECT (self, "Setting inport port definition");
if (gst_omx_port_update_port_definition (self->enc_in_port,
&port_def) != OMX_ErrorNone)
return FALSE;
GST_OMX_INIT_STRUCT (&pcm_param);
pcm_param.nPortIndex = self->enc_in_port->index;
pcm_param.nChannels = info->channels;
pcm_param.eNumData =
((info->finfo->flags & GST_AUDIO_FORMAT_FLAG_SIGNED) ?
OMX_NumericalDataSigned : OMX_NumericalDataUnsigned);
pcm_param.eEndian =
((info->finfo->endianness == G_LITTLE_ENDIAN) ?
OMX_EndianLittle : OMX_EndianBig);
pcm_param.bInterleaved = OMX_TRUE;
pcm_param.nBitPerSample = info->finfo->width;
pcm_param.nSamplingRate = info->rate;
pcm_param.ePCMMode = OMX_AUDIO_PCMModeLinear;
for (i = 0; i < pcm_param.nChannels; i++) {
OMX_AUDIO_CHANNELTYPE pos;
switch (info->position[i]) {
case GST_AUDIO_CHANNEL_POSITION_MONO:
case GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER:
pos = OMX_AUDIO_ChannelCF;
break;
case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
pos = OMX_AUDIO_ChannelLF;
break;
case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
pos = OMX_AUDIO_ChannelRF;
break;
case GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT:
pos = OMX_AUDIO_ChannelLS;
break;
case GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT:
pos = OMX_AUDIO_ChannelRS;
break;
case GST_AUDIO_CHANNEL_POSITION_LFE1:
pos = OMX_AUDIO_ChannelLFE;
break;
case GST_AUDIO_CHANNEL_POSITION_REAR_CENTER:
pos = OMX_AUDIO_ChannelCS;
break;
case GST_AUDIO_CHANNEL_POSITION_REAR_LEFT:
pos = OMX_AUDIO_ChannelLR;
break;
case GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT:
pos = OMX_AUDIO_ChannelRR;
break;
default:
pos = OMX_AUDIO_ChannelNone;
break;
}
pcm_param.eChannelMapping[i] = pos;
}
GST_DEBUG_OBJECT (self, "Setting PCM parameters");
err =
gst_omx_component_set_parameter (self->enc, OMX_IndexParamAudioPcm,
&pcm_param);
if (err != OMX_ErrorNone) {
GST_ERROR_OBJECT (self, "Failed to set PCM parameters: %s (0x%08x)",
gst_omx_error_to_string (err), err);
return FALSE;
}
if (klass->set_format) {
if (!klass->set_format (self, self->enc_in_port, info)) {
GST_ERROR_OBJECT (self, "Subclass failed to set the new format");
return FALSE;
}
}
GST_DEBUG_OBJECT (self, "Updating outport port definition");
if (gst_omx_port_update_port_definition (self->enc_out_port,
NULL) != OMX_ErrorNone)
return FALSE;
GST_DEBUG_OBJECT (self, "Enabling component");
if (needs_disable) {
if (gst_omx_port_set_enabled (self->enc_in_port, TRUE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_allocate_buffers (self->enc_in_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->enc_in_port,
5 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_mark_reconfigured (self->enc_in_port) != OMX_ErrorNone)
return FALSE;
} else {
/* Disable output port */
if (gst_omx_port_set_enabled (self->enc_out_port, FALSE) != OMX_ErrorNone)
return FALSE;
if (gst_omx_port_wait_enabled (self->enc_out_port,
1 * GST_SECOND) != OMX_ErrorNone)
return FALSE;
if (gst_omx_component_set_state (self->enc, OMX_StateIdle) != OMX_ErrorNone)
return FALSE;
/* Need to allocate buffers to reach Idle state */
if (gst_omx_port_allocate_buffers (self->enc_in_port) != OMX_ErrorNone)
return FALSE;
if (gst_omx_component_get_state (self->enc,
GST_CLOCK_TIME_NONE) != OMX_StateIdle)
return FALSE;
if (gst_omx_component_set_state (self->enc,
OMX_StateExecuting) != OMX_ErrorNone)
return FALSE;
if (gst_omx_component_get_state (self->enc,
GST_CLOCK_TIME_NONE) != OMX_StateExecuting)
return FALSE;
}
/* Unset flushing to allow ports to accept data again */
gst_omx_port_set_flushing (self->enc_in_port, 5 * GST_SECOND, FALSE);
gst_omx_port_set_flushing (self->enc_out_port, 5 * GST_SECOND, FALSE);
if (gst_omx_component_get_last_error (self->enc) != OMX_ErrorNone) {
GST_ERROR_OBJECT (self, "Component in error state: %s (0x%08x)",
gst_omx_component_get_last_error_string (self->enc),
gst_omx_component_get_last_error (self->enc));
return FALSE;
}
/* Start the srcpad loop again */
GST_DEBUG_OBJECT (self, "Starting task again");
self->downstream_flow_ret = GST_FLOW_OK;
gst_pad_start_task (GST_AUDIO_ENCODER_SRC_PAD (self),
(GstTaskFunction) gst_omx_audio_enc_loop, encoder, NULL);
return TRUE;
}
/* Called with the object lock for both the element and pad held,
* as well as the aagg lock
*/
static gboolean
gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
GstAudioAggregatorPad * pad, GstBuffer * inbuf)
{
GstClockTime start_time, end_time;
gboolean discont = FALSE;
guint64 start_offset, end_offset;
gint rate, bpf;
GstAggregator *agg = GST_AGGREGATOR (aagg);
GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
g_assert (pad->priv->buffer == NULL);
rate = GST_AUDIO_INFO_RATE (&pad->info);
bpf = GST_AUDIO_INFO_BPF (&pad->info);
pad->priv->position = 0;
pad->priv->size = gst_buffer_get_size (inbuf) / bpf;
if (!GST_BUFFER_PTS_IS_VALID (inbuf)) {
if (pad->priv->output_offset == -1)
pad->priv->output_offset = aagg->priv->offset;
if (pad->priv->next_offset == -1)
pad->priv->next_offset = pad->priv->size;
else
pad->priv->next_offset += pad->priv->size;
goto done;
}
start_time = GST_BUFFER_PTS (inbuf);
end_time =
start_time + gst_util_uint64_scale_ceil (pad->priv->size, GST_SECOND,
rate);
/* Clipping should've ensured this */
g_assert (start_time >= aggpad->segment.start);
start_offset =
gst_util_uint64_scale (start_time - aggpad->segment.start, rate,
GST_SECOND);
end_offset = start_offset + pad->priv->size;
if (GST_BUFFER_IS_DISCONT (inbuf)
|| GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_RESYNC)
|| pad->priv->new_segment || pad->priv->next_offset == -1) {
discont = TRUE;
pad->priv->new_segment = FALSE;
} else {
guint64 diff, max_sample_diff;
/* Check discont, based on audiobasesink */
if (start_offset <= pad->priv->next_offset)
diff = pad->priv->next_offset - start_offset;
else
diff = start_offset - pad->priv->next_offset;
max_sample_diff =
gst_util_uint64_scale_int (aagg->priv->alignment_threshold, rate,
GST_SECOND);
/* Discont! */
if (G_UNLIKELY (diff >= max_sample_diff)) {
if (aagg->priv->discont_wait > 0) {
if (pad->priv->discont_time == GST_CLOCK_TIME_NONE) {
pad->priv->discont_time = start_time;
} else if (start_time - pad->priv->discont_time >=
aagg->priv->discont_wait) {
discont = TRUE;
pad->priv->discont_time = GST_CLOCK_TIME_NONE;
}
} else {
discont = TRUE;
}
} else if (G_UNLIKELY (pad->priv->discont_time != GST_CLOCK_TIME_NONE)) {
/* we have had a discont, but are now back on track! */
pad->priv->discont_time = GST_CLOCK_TIME_NONE;
}
}
if (discont) {
/* Have discont, need resync */
if (pad->priv->next_offset != -1)
GST_INFO_OBJECT (pad, "Have discont. Expected %"
G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
pad->priv->next_offset, start_offset);
pad->priv->output_offset = -1;
pad->priv->next_offset = end_offset;
} else {
pad->priv->next_offset += pad->priv->size;
}
if (pad->priv->output_offset == -1) {
GstClockTime start_running_time;
GstClockTime end_running_time;
guint64 start_output_offset;
guint64 end_output_offset;
start_running_time =
gst_segment_to_running_time (&aggpad->segment,
GST_FORMAT_TIME, start_time);
end_running_time =
gst_segment_to_running_time (&aggpad->segment,
GST_FORMAT_TIME, end_time);
/* Convert to position in the output segment */
start_output_offset =
gst_segment_to_position (&agg->segment, GST_FORMAT_TIME,
start_running_time);
if (start_output_offset != -1)
start_output_offset =
gst_util_uint64_scale (start_output_offset - agg->segment.start, rate,
GST_SECOND);
end_output_offset =
gst_segment_to_position (&agg->segment, GST_FORMAT_TIME,
end_running_time);
if (end_output_offset != -1)
end_output_offset =
gst_util_uint64_scale (end_output_offset - agg->segment.start, rate,
GST_SECOND);
if (start_output_offset == -1 && end_output_offset == -1) {
/* Outside output segment, drop */
gst_buffer_unref (inbuf);
pad->priv->buffer = NULL;
pad->priv->position = 0;
pad->priv->size = 0;
pad->priv->output_offset = -1;
GST_DEBUG_OBJECT (pad, "Buffer outside output segment");
return FALSE;
}
/* Calculate end_output_offset if it was outside the output segment */
if (end_output_offset == -1)
end_output_offset = start_output_offset + pad->priv->size;
if (end_output_offset < aagg->priv->offset) {
/* Before output segment, drop */
gst_buffer_unref (inbuf);
pad->priv->buffer = NULL;
pad->priv->position = 0;
pad->priv->size = 0;
pad->priv->output_offset = -1;
GST_DEBUG_OBJECT (pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT " < %"
G_GINT64_FORMAT, end_output_offset, aagg->priv->offset);
return FALSE;
}
if (start_output_offset == -1 || start_output_offset < aagg->priv->offset) {
guint diff;
if (start_output_offset == -1 && end_output_offset < pad->priv->size) {
diff = pad->priv->size - end_output_offset + aagg->priv->offset;
} else if (start_output_offset == -1) {
start_output_offset = end_output_offset - pad->priv->size;
if (start_output_offset < aagg->priv->offset)
diff = aagg->priv->offset - start_output_offset;
else
diff = 0;
} else {
diff = aagg->priv->offset - start_output_offset;
}
pad->priv->position += diff;
if (pad->priv->position >= pad->priv->size) {
/* Empty buffer, drop */
gst_buffer_unref (inbuf);
pad->priv->buffer = NULL;
pad->priv->position = 0;
pad->priv->size = 0;
pad->priv->output_offset = -1;
GST_DEBUG_OBJECT (pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT
" < %" G_GINT64_FORMAT, end_output_offset, aagg->priv->offset);
return FALSE;
}
}
if (start_output_offset == -1 || start_output_offset < aagg->priv->offset)
pad->priv->output_offset = aagg->priv->offset;
else
pad->priv->output_offset = start_output_offset;
GST_DEBUG_OBJECT (pad,
"Buffer resynced: Pad offset %" G_GUINT64_FORMAT
", current audio aggregator offset %" G_GINT64_FORMAT,
pad->priv->output_offset, aagg->priv->offset);
}
done:
GST_LOG_OBJECT (pad,
"Queued new buffer at offset %" G_GUINT64_FORMAT,
pad->priv->output_offset);
pad->priv->buffer = inbuf;
return TRUE;
}
/* This function is used to perform seeks on the element in
* pull mode.
*
* It also works when event is NULL, in which case it will just
* start from the last configured segment. This technique is
* used when activating the element and to perform the seek in
* READY.
*/
static gboolean
gst_aiff_parse_perform_seek (GstAiffParse * aiff, GstEvent * event)
{
gboolean res;
gdouble rate;
GstFormat format, bformat;
GstSeekFlags flags;
GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
gint64 cur, stop, upstream_size;
gboolean flush;
gboolean update;
GstSegment seeksegment = { 0, };
gint64 last_stop;
if (event) {
GST_DEBUG_OBJECT (aiff, "doing seek with event");
gst_event_parse_seek (event, &rate, &format, &flags,
&cur_type, &cur, &stop_type, &stop);
/* no negative rates yet */
if (rate < 0.0)
goto negative_rate;
if (format != aiff->segment.format) {
GST_INFO_OBJECT (aiff, "converting seek-event from %s to %s",
gst_format_get_name (format),
gst_format_get_name (aiff->segment.format));
res = TRUE;
if (cur_type != GST_SEEK_TYPE_NONE)
res =
gst_pad_query_convert (aiff->srcpad, format, cur,
&aiff->segment.format, &cur);
if (res && stop_type != GST_SEEK_TYPE_NONE)
res =
gst_pad_query_convert (aiff->srcpad, format, stop,
&aiff->segment.format, &stop);
if (!res)
goto no_format;
format = aiff->segment.format;
}
} else {
GST_DEBUG_OBJECT (aiff, "doing seek without event");
flags = 0;
rate = 1.0;
cur_type = GST_SEEK_TYPE_SET;
stop_type = GST_SEEK_TYPE_SET;
}
/* get flush flag */
flush = flags & GST_SEEK_FLAG_FLUSH;
/* now we need to make sure the streaming thread is stopped. We do this by
* either sending a FLUSH_START event downstream which will cause the
* streaming thread to stop with a WRONG_STATE.
* For a non-flushing seek we simply pause the task, which will happen as soon
* as it completes one iteration (and thus might block when the sink is
* blocking in preroll). */
if (flush) {
GST_DEBUG_OBJECT (aiff, "sending flush start");
gst_pad_push_event (aiff->srcpad, gst_event_new_flush_start ());
} else {
gst_pad_pause_task (aiff->sinkpad);
}
/* we should now be able to grab the streaming thread because we stopped it
* with the above flush/pause code */
GST_PAD_STREAM_LOCK (aiff->sinkpad);
/* save current position */
last_stop = aiff->segment.last_stop;
GST_DEBUG_OBJECT (aiff, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
/* copy segment, we need this because we still need the old
* segment when we close the current segment. */
memcpy (&seeksegment, &aiff->segment, sizeof (GstSegment));
/* configure the seek parameters in the seeksegment. We will then have the
* right values in the segment to perform the seek */
if (event) {
GST_DEBUG_OBJECT (aiff, "configuring seek");
gst_segment_set_seek (&seeksegment, rate, format, flags,
cur_type, cur, stop_type, stop, &update);
}
/* figure out the last position we need to play. If it's configured (stop !=
* -1), use that, else we play until the total duration of the file */
if ((stop = seeksegment.stop) == -1)
stop = seeksegment.duration;
GST_DEBUG_OBJECT (aiff, "cur_type =%d", cur_type);
if ((cur_type != GST_SEEK_TYPE_NONE)) {
/* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
* we can just copy the last_stop. If not, we use the bps to convert TIME to
* bytes. */
if (aiff->bps > 0)
aiff->offset =
gst_util_uint64_scale_ceil (seeksegment.last_stop,
(guint64) aiff->bps, GST_SECOND);
else
aiff->offset = seeksegment.last_stop;
GST_LOG_OBJECT (aiff, "offset=%" G_GUINT64_FORMAT, aiff->offset);
aiff->offset -= (aiff->offset % aiff->bytes_per_sample);
GST_LOG_OBJECT (aiff, "offset=%" G_GUINT64_FORMAT, aiff->offset);
aiff->offset += aiff->datastart;
GST_LOG_OBJECT (aiff, "offset=%" G_GUINT64_FORMAT, aiff->offset);
} else {
GST_LOG_OBJECT (aiff, "continue from offset=%" G_GUINT64_FORMAT,
aiff->offset);
}
if (stop_type != GST_SEEK_TYPE_NONE) {
if (aiff->bps > 0)
aiff->end_offset =
gst_util_uint64_scale_ceil (stop, (guint64) aiff->bps, GST_SECOND);
else
aiff->end_offset = stop;
GST_LOG_OBJECT (aiff, "end_offset=%" G_GUINT64_FORMAT, aiff->end_offset);
aiff->end_offset -= (aiff->end_offset % aiff->bytes_per_sample);
GST_LOG_OBJECT (aiff, "end_offset=%" G_GUINT64_FORMAT, aiff->end_offset);
aiff->end_offset += aiff->datastart;
GST_LOG_OBJECT (aiff, "end_offset=%" G_GUINT64_FORMAT, aiff->end_offset);
} else {
GST_LOG_OBJECT (aiff, "continue to end_offset=%" G_GUINT64_FORMAT,
aiff->end_offset);
}
/* make sure filesize is not exceeded due to rounding errors or so,
* same precaution as in _stream_headers */
bformat = GST_FORMAT_BYTES;
if (gst_pad_query_peer_duration (aiff->sinkpad, &bformat, &upstream_size))
aiff->end_offset = MIN (aiff->end_offset, upstream_size);
/* this is the range of bytes we will use for playback */
aiff->offset = MIN (aiff->offset, aiff->end_offset);
aiff->dataleft = aiff->end_offset - aiff->offset;
GST_DEBUG_OBJECT (aiff,
"seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT
", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, aiff->offset,
aiff->end_offset, GST_TIME_ARGS (seeksegment.start),
GST_TIME_ARGS (stop));
/* prepare for streaming again */
if (flush) {
/* if we sent a FLUSH_START, we now send a FLUSH_STOP */
GST_DEBUG_OBJECT (aiff, "sending flush stop");
gst_pad_push_event (aiff->srcpad, gst_event_new_flush_stop ());
} else if (aiff->segment_running) {
/* we are running the current segment and doing a non-flushing seek,
* close the segment first based on the previous last_stop. */
GST_DEBUG_OBJECT (aiff, "closing running segment %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, aiff->segment.accum, aiff->segment.last_stop);
/* queue the segment for sending in the stream thread */
if (aiff->close_segment)
gst_event_unref (aiff->close_segment);
aiff->close_segment = gst_event_new_new_segment (TRUE,
aiff->segment.rate, aiff->segment.format,
aiff->segment.accum, aiff->segment.last_stop, aiff->segment.accum);
/* keep track of our last_stop */
seeksegment.accum = aiff->segment.last_stop;
}
/* now we did the seek and can activate the new segment values */
memcpy (&aiff->segment, &seeksegment, sizeof (GstSegment));
/* if we're doing a segment seek, post a SEGMENT_START message */
if (aiff->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (aiff),
gst_message_new_segment_start (GST_OBJECT_CAST (aiff),
aiff->segment.format, aiff->segment.last_stop));
}
/* now create the newsegment */
GST_DEBUG_OBJECT (aiff, "Creating newsegment from %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, aiff->segment.last_stop, stop);
/* store the newsegment event so it can be sent from the streaming thread. */
if (aiff->start_segment)
gst_event_unref (aiff->start_segment);
aiff->start_segment =
gst_event_new_new_segment (FALSE, aiff->segment.rate,
aiff->segment.format, aiff->segment.last_stop, stop,
aiff->segment.last_stop);
/* mark discont if we are going to stream from another position. */
if (last_stop != aiff->segment.last_stop) {
GST_DEBUG_OBJECT (aiff, "mark DISCONT, we did a seek to another position");
aiff->discont = TRUE;
}
/* and start the streaming task again */
aiff->segment_running = TRUE;
if (!aiff->streaming) {
gst_pad_start_task (aiff->sinkpad, (GstTaskFunction) gst_aiff_parse_loop,
aiff->sinkpad);
}
GST_PAD_STREAM_UNLOCK (aiff->sinkpad);
return TRUE;
/* ERRORS */
negative_rate:
{
GST_DEBUG_OBJECT (aiff, "negative playback rates are not supported yet.");
return FALSE;
}
no_format:
{
GST_DEBUG_OBJECT (aiff, "unsupported format given, seek aborted.");
return FALSE;
}
}
static gboolean
gst_aiff_parse_pad_convert (GstPad * pad,
GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value)
{
GstAiffParse *aiffparse;
gboolean res = TRUE;
aiffparse = GST_AIFF_PARSE (GST_PAD_PARENT (pad));
if (*dest_format == src_format) {
*dest_value = src_value;
return TRUE;
}
if (aiffparse->bytes_per_sample <= 0)
return FALSE;
GST_INFO_OBJECT (aiffparse, "converting value from %s to %s",
gst_format_get_name (src_format), gst_format_get_name (*dest_format));
switch (src_format) {
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_DEFAULT:
*dest_value = src_value / aiffparse->bytes_per_sample;
break;
case GST_FORMAT_TIME:
if (aiffparse->bps > 0) {
*dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND,
(guint64) aiffparse->bps);
break;
}
/* Else fallthrough */
default:
res = FALSE;
goto done;
}
break;
case GST_FORMAT_DEFAULT:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = src_value * aiffparse->bytes_per_sample;
break;
case GST_FORMAT_TIME:
*dest_value = gst_util_uint64_scale (src_value, GST_SECOND,
(guint64) aiffparse->rate);
break;
default:
res = FALSE;
goto done;
}
break;
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
if (aiffparse->bps > 0) {
*dest_value = gst_util_uint64_scale (src_value,
(guint64) aiffparse->bps, GST_SECOND);
break;
}
/* Else fallthrough */
break;
case GST_FORMAT_DEFAULT:
*dest_value = gst_util_uint64_scale (src_value,
(guint64) aiffparse->rate, GST_SECOND);
break;
default:
res = FALSE;
goto done;
}
break;
default:
res = FALSE;
goto done;
}
done:
return res;
}
static GstFlowReturn
gst_aiff_parse_stream_data (GstAiffParse * aiff)
{
GstBuffer *buf = NULL;
GstFlowReturn res = GST_FLOW_OK;
guint64 desired, obtained;
GstClockTime timestamp, next_timestamp, duration;
guint64 pos, nextpos;
iterate_adapter:
GST_LOG_OBJECT (aiff,
"offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
G_GINT64_FORMAT, aiff->offset, aiff->end_offset, aiff->dataleft);
/* Get the next n bytes and output them */
if (aiff->dataleft == 0 || aiff->dataleft < aiff->bytes_per_sample)
goto found_eos;
/* scale the amount of data by the segment rate so we get equal
* amounts of data regardless of the playback rate */
desired =
MIN (gst_guint64_to_gdouble (aiff->dataleft),
MAX_BUFFER_SIZE * aiff->segment.abs_rate);
if (desired >= aiff->bytes_per_sample && aiff->bytes_per_sample > 0)
desired -= (desired % aiff->bytes_per_sample);
GST_LOG_OBJECT (aiff, "Fetching %" G_GINT64_FORMAT " bytes of data "
"from the sinkpad", desired);
if (aiff->streaming) {
guint avail = gst_adapter_available (aiff->adapter);
if (avail < desired) {
GST_LOG_OBJECT (aiff, "Got only %d bytes of data from the sinkpad",
avail);
return GST_FLOW_OK;
}
buf = gst_adapter_take_buffer (aiff->adapter, desired);
} else {
if ((res = gst_pad_pull_range (aiff->sinkpad, aiff->offset,
desired, &buf)) != GST_FLOW_OK)
goto pull_error;
}
/* If we have a pending close/start segment, send it now. */
if (G_UNLIKELY (aiff->close_segment != NULL)) {
gst_pad_push_event (aiff->srcpad, aiff->close_segment);
aiff->close_segment = NULL;
}
if (G_UNLIKELY (aiff->start_segment != NULL)) {
gst_pad_push_event (aiff->srcpad, aiff->start_segment);
aiff->start_segment = NULL;
}
if (G_UNLIKELY (aiff->tags != NULL)) {
gst_element_found_tags_for_pad (GST_ELEMENT_CAST (aiff), aiff->srcpad,
aiff->tags);
aiff->tags = NULL;
}
obtained = GST_BUFFER_SIZE (buf);
/* our positions in bytes */
pos = aiff->offset - aiff->datastart;
nextpos = pos + obtained;
/* update offsets, does not overflow. */
GST_BUFFER_OFFSET (buf) = pos / aiff->bytes_per_sample;
GST_BUFFER_OFFSET_END (buf) = nextpos / aiff->bytes_per_sample;
if (aiff->bps > 0) {
/* and timestamps if we have a bitrate, be careful for overflows */
timestamp =
gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) aiff->bps);
next_timestamp =
gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) aiff->bps);
duration = next_timestamp - timestamp;
/* update current running segment position */
gst_segment_set_last_stop (&aiff->segment, GST_FORMAT_TIME, next_timestamp);
} else {
/* no bitrate, all we know is that the first sample has timestamp 0, all
* other positions and durations have unknown timestamp. */
if (pos == 0)
timestamp = 0;
else
timestamp = GST_CLOCK_TIME_NONE;
duration = GST_CLOCK_TIME_NONE;
/* update current running segment position with byte offset */
gst_segment_set_last_stop (&aiff->segment, GST_FORMAT_BYTES, nextpos);
}
if (aiff->discont) {
GST_DEBUG_OBJECT (aiff, "marking DISCONT");
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
aiff->discont = FALSE;
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
gst_buffer_set_caps (buf, aiff->caps);
GST_LOG_OBJECT (aiff,
"Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
", size:%u", GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration),
GST_BUFFER_SIZE (buf));
if ((res = gst_pad_push (aiff->srcpad, buf)) != GST_FLOW_OK)
goto push_error;
if (obtained < aiff->dataleft) {
aiff->offset += obtained;
aiff->dataleft -= obtained;
} else {
aiff->offset += aiff->dataleft;
aiff->dataleft = 0;
}
/* Iterate until need more data, so adapter size won't grow */
if (aiff->streaming) {
GST_LOG_OBJECT (aiff,
"offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, aiff->offset,
aiff->end_offset);
goto iterate_adapter;
}
return res;
/* ERROR */
found_eos:
{
GST_DEBUG_OBJECT (aiff, "found EOS");
return GST_FLOW_UNEXPECTED;
}
pull_error:
{
/* check if we got EOS */
if (res == GST_FLOW_UNEXPECTED)
goto found_eos;
GST_WARNING_OBJECT (aiff,
"Error getting %" G_GINT64_FORMAT " bytes from the "
"sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, aiff->dataleft);
return res;
}
push_error:
{
GST_INFO_OBJECT (aiff,
"Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
GST_DEBUG_PAD_NAME (aiff->srcpad), gst_flow_get_name (res),
gst_pad_is_linked (aiff->srcpad));
return res;
}
}
