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; } }