static GstFlowReturn gst_jpeg_parse_push_buffer (GstJpegParse * parse, guint len) { GstBuffer *outbuf; GstFlowReturn ret = GST_FLOW_OK; gboolean header_ok; /* reset the offset (only when we flushed) */ parse->priv->last_offset = 0; parse->priv->last_entropy_len = 0; outbuf = gst_adapter_take_buffer (parse->priv->adapter, len); if (outbuf == NULL) { GST_ELEMENT_ERROR (parse, STREAM, DECODE, ("Failed to take buffer of size %u", len), ("Failed to take buffer of size %u", len)); return GST_FLOW_ERROR; } header_ok = gst_jpeg_parse_read_header (parse, outbuf); if (parse->priv->new_segment == TRUE || parse->priv->width != parse->priv->caps_width || parse->priv->height != parse->priv->caps_height || parse->priv->framerate_numerator != parse->priv->caps_framerate_numerator || parse->priv->framerate_denominator != parse->priv->caps_framerate_denominator) { if (!gst_jpeg_parse_set_new_caps (parse, header_ok)) { GST_ELEMENT_ERROR (parse, CORE, NEGOTIATION, ("Can't set caps to the src pad"), ("Can't set caps to the src pad")); return GST_FLOW_ERROR; } if (parse->priv->tags) { GST_DEBUG_OBJECT (parse, "Pushing tags: %" GST_PTR_FORMAT, parse->priv->tags); gst_element_found_tags_for_pad (GST_ELEMENT_CAST (parse), parse->priv->srcpad, parse->priv->tags); parse->priv->tags = NULL; } parse->priv->new_segment = FALSE; parse->priv->caps_width = parse->priv->width; parse->priv->caps_height = parse->priv->height; parse->priv->caps_framerate_numerator = parse->priv->framerate_numerator; parse->priv->caps_framerate_denominator = parse->priv->framerate_denominator; } GST_BUFFER_TIMESTAMP (outbuf) = parse->priv->next_ts; if (parse->priv->has_fps && GST_CLOCK_TIME_IS_VALID (parse->priv->next_ts) && GST_CLOCK_TIME_IS_VALID (parse->priv->duration)) { parse->priv->next_ts += parse->priv->duration; } else { parse->priv->duration = GST_CLOCK_TIME_NONE; parse->priv->next_ts = GST_CLOCK_TIME_NONE; } GST_BUFFER_DURATION (outbuf) = parse->priv->duration; gst_buffer_set_caps (outbuf, GST_PAD_CAPS (parse->priv->srcpad)); GST_LOG_OBJECT (parse, "pushing buffer (ts=%" GST_TIME_FORMAT ", len=%u)", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), len); ret = gst_pad_push (parse->priv->srcpad, outbuf); return ret; }
/* chain function * this function does the actual processing */ static GstFlowReturn gst_ivf_parse_chain (GstPad * pad, GstBuffer * buf) { GstIvfParse *ivf = GST_IVF_PARSE (GST_OBJECT_PARENT (pad)); gboolean res; /* lazy creation of the adapter */ if (G_UNLIKELY (ivf->adapter == NULL)) { ivf->adapter = gst_adapter_new (); } GST_LOG_OBJECT (ivf, "Pushing buffer of size %u to adapter", GST_BUFFER_SIZE (buf)); gst_adapter_push (ivf->adapter, buf); /* adapter takes ownership of buf */ res = GST_FLOW_OK; switch (ivf->state) { case GST_IVF_PARSE_START: if (gst_adapter_available (ivf->adapter) >= 32) { GstCaps *caps; const guint8 *data = gst_adapter_peek (ivf->adapter, 32); guint32 magic = GST_READ_UINT32_LE (data); guint16 version = GST_READ_UINT16_LE (data + 4); guint16 header_size = GST_READ_UINT16_LE (data + 6); guint32 fourcc = GST_READ_UINT32_LE (data + 8); guint16 width = GST_READ_UINT16_LE (data + 12); guint16 height = GST_READ_UINT16_LE (data + 14); guint32 rate_num = GST_READ_UINT32_LE (data + 16); guint32 rate_den = GST_READ_UINT32_LE (data + 20); #ifndef GST_DISABLE_GST_DEBUG guint32 num_frames = GST_READ_UINT32_LE (data + 24); #endif /* last 4 bytes unused */ gst_adapter_flush (ivf->adapter, 32); if (magic != GST_MAKE_FOURCC ('D', 'K', 'I', 'F') || version != 0 || header_size != 32 || fourcc != GST_MAKE_FOURCC ('V', 'P', '8', '0')) { GST_ELEMENT_ERROR (ivf, STREAM, WRONG_TYPE, (NULL), (NULL)); return GST_FLOW_ERROR; } /* create src pad caps */ caps = gst_caps_new_simple ("video/x-vp8", "width", G_TYPE_INT, width, "height", G_TYPE_INT, height, "framerate", GST_TYPE_FRACTION, rate_num, rate_den, NULL); GST_INFO_OBJECT (ivf, "Found stream: %" GST_PTR_FORMAT, caps); GST_LOG_OBJECT (ivf, "Stream has %d frames", num_frames); gst_pad_set_caps (ivf->srcpad, caps); gst_caps_unref (caps); /* keep framerate in instance for convenience */ ivf->rate_num = rate_num; ivf->rate_den = rate_den; gst_pad_push_event (ivf->srcpad, gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0)); /* move along */ ivf->state = GST_IVF_PARSE_DATA; } else { GST_LOG_OBJECT (ivf, "Header data not yet available."); break; } /* fall through */ case GST_IVF_PARSE_DATA: while (gst_adapter_available (ivf->adapter) > 12) { const guint8 *data = gst_adapter_peek (ivf->adapter, 12); guint32 frame_size = GST_READ_UINT32_LE (data); guint64 frame_pts = GST_READ_UINT64_LE (data + 4); GST_LOG_OBJECT (ivf, "Read frame header: size %u, pts %" G_GUINT64_FORMAT, frame_size, frame_pts); if (gst_adapter_available (ivf->adapter) >= 12 + frame_size) { GstBuffer *frame; gst_adapter_flush (ivf->adapter, 12); frame = gst_adapter_take_buffer (ivf->adapter, frame_size); gst_buffer_set_caps (frame, GST_PAD_CAPS (ivf->srcpad)); GST_BUFFER_TIMESTAMP (frame) = gst_util_uint64_scale_int (GST_SECOND * frame_pts, ivf->rate_den, ivf->rate_num); GST_BUFFER_DURATION (frame) = gst_util_uint64_scale_int (GST_SECOND, ivf->rate_den, ivf->rate_num); GST_DEBUG_OBJECT (ivf, "Pushing frame of size %u, ts %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT ", off %" G_GUINT64_FORMAT ", off_end %" G_GUINT64_FORMAT, GST_BUFFER_SIZE (frame), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (frame)), GST_TIME_ARGS (GST_BUFFER_DURATION (frame)), GST_BUFFER_OFFSET (frame), GST_BUFFER_OFFSET_END (frame)); res = gst_pad_push (ivf->srcpad, frame); if (res != GST_FLOW_OK) break; } else { GST_LOG_OBJECT (ivf, "Frame data not yet available."); break; } } break; default: g_return_val_if_reached (GST_FLOW_ERROR); } return res; }
static GstFlowReturn gst_rtp_h263p_pay_flush (GstRtpH263PPay * rtph263ppay) { guint avail; GstBufferList *list = NULL; GstBuffer *outbuf = NULL; GstFlowReturn ret; gboolean fragmented = FALSE; avail = gst_adapter_available (rtph263ppay->adapter); if (avail == 0) return GST_FLOW_OK; fragmented = FALSE; /* This algorithm assumes the H263/+/++ encoder sends complete frames in each * buffer */ /* With Fragmentation Mode at GST_FRAGMENTATION_MODE_NORMAL: * This algorithm implements the Follow-on packets method for packetization. * This assumes low packet loss network. * With Fragmentation Mode at GST_FRAGMENTATION_MODE_SYNC: * This algorithm separates large frames at synchronisation points (Segments) * (See RFC 4629 section 6). It would be interesting to have a property such as network * quality to select between both packetization methods */ /* TODO Add VRC supprt (See RFC 4629 section 5.2) */ while (avail > 0) { guint towrite; guint8 *payload; gint header_len; guint next_gop = 0; gboolean found_gob = FALSE; GstRTPBuffer rtp = { NULL }; GstBuffer *payload_buf; if (rtph263ppay->fragmentation_mode == GST_FRAGMENTATION_MODE_SYNC) { /* start after 1st gop possible */ /* Check if we have a gob or eos , eossbs */ /* FIXME EOS and EOSSBS packets should never contain any gobs and vice-versa */ next_gop = gst_adapter_masked_scan_uint32 (rtph263ppay->adapter, 0xffff8000, 0x00008000, 0, avail); if (next_gop == 0) { GST_DEBUG_OBJECT (rtph263ppay, " Found GOB header"); found_gob = TRUE; } /* Find next and cut the packet accordingly */ /* TODO we should get as many gobs as possible until MTU is reached, this * code seems to just get one GOB per packet */ if (next_gop == 0 && avail > 3) next_gop = gst_adapter_masked_scan_uint32 (rtph263ppay->adapter, 0xffff8000, 0x00008000, 3, avail - 3); GST_DEBUG_OBJECT (rtph263ppay, " Next GOB Detected at : %d", next_gop); if (next_gop == -1) next_gop = 0; } /* for picture start frames (non-fragmented), we need to remove the first * two 0x00 bytes and set P=1 */ if (!fragmented || found_gob) { gst_adapter_flush (rtph263ppay->adapter, 2); avail -= 2; } header_len = 2; towrite = MIN (avail, gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU (rtph263ppay) - header_len, 0, 0)); if (next_gop > 0) towrite = MIN (next_gop, towrite); outbuf = gst_rtp_buffer_new_allocate (header_len, 0, 0); gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp); /* last fragment gets the marker bit set */ gst_rtp_buffer_set_marker (&rtp, avail > towrite ? 0 : 1); payload = gst_rtp_buffer_get_payload (&rtp); /* 0 1 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * | RR |P|V| PLEN |PEBIT| * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ */ /* if fragmented or gop header , write p bit =1 */ payload[0] = (fragmented && !found_gob) ? 0x00 : 0x04; payload[1] = 0; GST_BUFFER_PTS (outbuf) = rtph263ppay->first_timestamp; GST_BUFFER_DURATION (outbuf) = rtph263ppay->first_duration; gst_rtp_buffer_unmap (&rtp); payload_buf = gst_adapter_take_buffer_fast (rtph263ppay->adapter, towrite); gst_rtp_copy_meta (GST_ELEMENT_CAST (rtph263ppay), outbuf, payload_buf, g_quark_from_static_string (GST_META_TAG_VIDEO_STR)); outbuf = gst_buffer_append (outbuf, payload_buf); avail -= towrite; /* If more data is available and this is our first iteration, * we create a buffer list and remember that we're fragmented. * * If we're fragmented already, add buffers to the previously * created buffer list. * * Otherwise fragmented will be FALSE and we just push the single output * buffer, and no list is allocated. */ if (avail && !fragmented) { fragmented = TRUE; list = gst_buffer_list_new (); gst_buffer_list_add (list, outbuf); } else if (fragmented) { gst_buffer_list_add (list, outbuf); } } if (fragmented) { ret = gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtph263ppay), list); } else { ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtph263ppay), outbuf); } return ret; }
static GstFlowReturn speex_dec_chain_parse_data (GstSpeexDec * dec, GstBuffer * buf, GstClockTime timestamp, GstClockTime duration) { GstFlowReturn res = GST_FLOW_OK; gint i, fpp; guint size; guint8 *data; SpeexBits *bits; if (!dec->frame_duration) goto not_negotiated; if (timestamp != -1) { dec->segment.last_stop = timestamp; } else { timestamp = dec->segment.last_stop; } if (buf) { data = GST_BUFFER_DATA (buf); size = GST_BUFFER_SIZE (buf); /* send data to the bitstream */ speex_bits_read_from (&dec->bits, (char *) data, size); fpp = 0; bits = &dec->bits; GST_DEBUG_OBJECT (dec, "received buffer of size %u, fpp %d", size, fpp); } else { /* concealment data, pass NULL as the bits parameters */ GST_DEBUG_OBJECT (dec, "creating concealment data"); fpp = dec->header->frames_per_packet; bits = NULL; } /* now decode each frame, catering for unknown number of them (e.g. rtp) */ for (i = 0; (!fpp || i < fpp) && (!bits || speex_bits_remaining (bits) > 0); i++) { GstBuffer *outbuf; gint16 *out_data; gint ret; GST_LOG_OBJECT (dec, "decoding frame %d/%d", i, fpp); res = gst_pad_alloc_buffer_and_set_caps (dec->srcpad, GST_BUFFER_OFFSET_NONE, dec->frame_size * dec->header->nb_channels * 2, GST_PAD_CAPS (dec->srcpad), &outbuf); if (res != GST_FLOW_OK) { GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res)); return res; } out_data = (gint16 *) GST_BUFFER_DATA (outbuf); ret = speex_decode_int (dec->state, bits, out_data); if (ret == -1) { /* uh? end of stream */ GST_WARNING_OBJECT (dec, "Unexpected end of stream found"); gst_buffer_unref (outbuf); outbuf = NULL; break; } else if (ret == -2) { GST_WARNING_OBJECT (dec, "Decoding error: corrupted stream?"); gst_buffer_unref (outbuf); outbuf = NULL; break; } if (bits && speex_bits_remaining (bits) < 0) { GST_WARNING_OBJECT (dec, "Decoding overflow: corrupted stream?"); gst_buffer_unref (outbuf); outbuf = NULL; break; } if (dec->header->nb_channels == 2) speex_decode_stereo_int (out_data, dec->frame_size, dec->stereo); GST_BUFFER_TIMESTAMP (outbuf) = timestamp; GST_BUFFER_DURATION (outbuf) = dec->frame_duration; dec->segment.last_stop += dec->frame_duration; timestamp = dec->segment.last_stop; GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp), GST_TIME_ARGS (dec->frame_duration)); res = gst_pad_push (dec->srcpad, outbuf); if (res != GST_FLOW_OK) { GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res)); break; } } return res; /* ERRORS */ not_negotiated: { GST_ELEMENT_ERROR (dec, CORE, NEGOTIATION, (NULL), ("decoder not initialized")); return GST_FLOW_NOT_NEGOTIATED; } }
static GstFlowReturn gst_gdiscreencapsrc_create (GstPushSrc * push_src, GstBuffer ** buf) { GstGDIScreenCapSrc *src = GST_GDISCREENCAPSRC (push_src); GstBuffer *new_buf; gint new_buf_size; GstClock *clock; GstClockTime buf_time, buf_dur; guint64 frame_number; if (G_UNLIKELY (!src->info.bmiHeader.biWidth || !src->info.bmiHeader.biHeight)) { GST_ELEMENT_ERROR (src, CORE, NEGOTIATION, (NULL), ("format wasn't negotiated before create function")); return GST_FLOW_NOT_NEGOTIATED; } new_buf_size = GST_ROUND_UP_4 (src->info.bmiHeader.biWidth * 3) * (-src->info.bmiHeader.biHeight); GST_LOG_OBJECT (src, "creating buffer of %d bytes with %dx%d image", new_buf_size, (gint) src->info.bmiHeader.biWidth, (gint) (-src->info.bmiHeader.biHeight)); new_buf = gst_buffer_new_and_alloc (new_buf_size); clock = gst_element_get_clock (GST_ELEMENT (src)); if (clock != NULL) { GstClockTime time, base_time; /* Calculate sync time. */ time = gst_clock_get_time (clock); base_time = gst_element_get_base_time (GST_ELEMENT (src)); buf_time = time - base_time; if (src->rate_numerator) { frame_number = gst_util_uint64_scale (buf_time, src->rate_numerator, GST_SECOND * src->rate_denominator); } else { frame_number = -1; } } else { buf_time = GST_CLOCK_TIME_NONE; frame_number = -1; } if (frame_number != -1 && frame_number == src->frame_number) { GstClockID id; GstClockReturn ret; /* Need to wait for the next frame */ frame_number += 1; /* Figure out what the next frame time is */ buf_time = gst_util_uint64_scale (frame_number, src->rate_denominator * GST_SECOND, src->rate_numerator); id = gst_clock_new_single_shot_id (clock, buf_time + gst_element_get_base_time (GST_ELEMENT (src))); GST_OBJECT_LOCK (src); src->clock_id = id; GST_OBJECT_UNLOCK (src); GST_DEBUG_OBJECT (src, "Waiting for next frame time %" G_GUINT64_FORMAT, buf_time); ret = gst_clock_id_wait (id, NULL); GST_OBJECT_LOCK (src); gst_clock_id_unref (id); src->clock_id = NULL; if (ret == GST_CLOCK_UNSCHEDULED) { /* Got woken up by the unlock function */ GST_OBJECT_UNLOCK (src); return GST_FLOW_FLUSHING; } GST_OBJECT_UNLOCK (src); /* Duration is a complete 1/fps frame duration */ buf_dur = gst_util_uint64_scale_int (GST_SECOND, src->rate_denominator, src->rate_numerator); } else if (frame_number != -1) { GstClockTime next_buf_time; GST_DEBUG_OBJECT (src, "No need to wait for next frame time %" G_GUINT64_FORMAT " next frame = %" G_GINT64_FORMAT " prev = %" G_GINT64_FORMAT, buf_time, frame_number, src->frame_number); next_buf_time = gst_util_uint64_scale (frame_number + 1, src->rate_denominator * GST_SECOND, src->rate_numerator); /* Frame duration is from now until the next expected capture time */ buf_dur = next_buf_time - buf_time; } else { buf_dur = GST_CLOCK_TIME_NONE; } src->frame_number = frame_number; GST_BUFFER_TIMESTAMP (new_buf) = buf_time; GST_BUFFER_DURATION (new_buf) = buf_dur; /* Do screen capture and put it into buffer... */ gst_gdiscreencapsrc_screen_capture (src, new_buf); gst_object_unref (clock); *buf = new_buf; return GST_FLOW_OK; }
static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf) { GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad)); uint32_t sample_count = GST_BUFFER_SIZE (buf) / 4; GstFlowReturn ret; /* reset the last returns to GST_FLOW_OK. This is only set to something else * while WavpackPackSamples() or more specific gst_wavpack_enc_push_block() * so not valid anymore */ enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK; GST_DEBUG ("got %u raw samples", sample_count); /* check if we already have a valid WavpackContext, otherwise make one */ if (!enc->wp_context) { /* create raw context */ enc->wp_context = WavpackOpenFileOutput (gst_wavpack_enc_push_block, &enc->wv_id, (enc->correction_mode > 0) ? &enc->wvc_id : NULL); if (!enc->wp_context) { GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL), ("error creating Wavpack context")); gst_object_unref (enc); gst_buffer_unref (buf); return GST_FLOW_ERROR; } /* set the WavpackConfig according to our parameters */ gst_wavpack_enc_set_wp_config (enc); /* set the configuration to the context now that we know everything * and initialize the encoder */ if (!WavpackSetConfiguration (enc->wp_context, enc->wp_config, (uint32_t) (-1)) || !WavpackPackInit (enc->wp_context)) { GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL), ("error setting up wavpack encoding context")); WavpackCloseFile (enc->wp_context); gst_object_unref (enc); gst_buffer_unref (buf); return GST_FLOW_ERROR; } GST_DEBUG ("setup of encoding context successfull"); } /* Save the timestamp of the first buffer. This will be later * used as offset for all following buffers */ if (enc->timestamp_offset == GST_CLOCK_TIME_NONE) { if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { enc->timestamp_offset = GST_BUFFER_TIMESTAMP (buf); enc->next_ts = GST_BUFFER_TIMESTAMP (buf); } else { enc->timestamp_offset = 0; enc->next_ts = 0; } } /* Check if we have a continous stream, if not drop some samples or the buffer or * insert some silence samples */ if (enc->next_ts != GST_CLOCK_TIME_NONE && GST_BUFFER_TIMESTAMP (buf) < enc->next_ts) { guint64 diff = enc->next_ts - GST_BUFFER_TIMESTAMP (buf); guint64 diff_bytes; GST_WARNING_OBJECT (enc, "Buffer is older than previous " "timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT "), cannot handle. Clipping buffer.", GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), GST_TIME_ARGS (enc->next_ts)); diff_bytes = GST_CLOCK_TIME_TO_FRAMES (diff, enc->samplerate) * enc->channels * 2; if (diff_bytes >= GST_BUFFER_SIZE (buf)) { gst_buffer_unref (buf); return GST_FLOW_OK; } buf = gst_buffer_make_metadata_writable (buf); GST_BUFFER_DATA (buf) += diff_bytes; GST_BUFFER_SIZE (buf) -= diff_bytes; GST_BUFFER_TIMESTAMP (buf) += diff; if (GST_BUFFER_DURATION_IS_VALID (buf)) GST_BUFFER_DURATION (buf) -= diff; } /* Allow a diff of at most 5 ms */ if (enc->next_ts != GST_CLOCK_TIME_NONE && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { if (GST_BUFFER_TIMESTAMP (buf) != enc->next_ts && GST_BUFFER_TIMESTAMP (buf) - enc->next_ts > 5 * GST_MSECOND) { GST_WARNING_OBJECT (enc, "Discontinuity detected: %" G_GUINT64_FORMAT " > %" G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (buf) - enc->next_ts, 5 * GST_MSECOND); WavpackFlushSamples (enc->wp_context); enc->timestamp_offset += (GST_BUFFER_TIMESTAMP (buf) - enc->next_ts); } } if (GST_BUFFER_TIMESTAMP_IS_VALID (buf) && GST_BUFFER_DURATION_IS_VALID (buf)) enc->next_ts = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf); else enc->next_ts = GST_CLOCK_TIME_NONE; if (enc->need_channel_remap) { buf = gst_buffer_make_writable (buf); gst_wavpack_enc_fix_channel_order (enc, (gint32 *) GST_BUFFER_DATA (buf), sample_count); } /* if we want to append the MD5 sum to the stream update it here * with the current raw samples */ if (enc->md5) { MD5Update (enc->md5_context, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); } /* encode and handle return values from encoding */ if (WavpackPackSamples (enc->wp_context, (int32_t *) GST_BUFFER_DATA (buf), sample_count / enc->channels)) { GST_DEBUG ("encoding samples successful"); ret = GST_FLOW_OK; } else { if ((enc->srcpad_last_return == GST_FLOW_RESEND) || (enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) { ret = GST_FLOW_RESEND; } else if ((enc->srcpad_last_return == GST_FLOW_OK) || (enc->wvcsrcpad_last_return == GST_FLOW_OK)) { ret = GST_FLOW_OK; } else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) && (enc->wvcsrcpad_last_return == GST_FLOW_NOT_LINKED)) { ret = GST_FLOW_NOT_LINKED; } else if ((enc->srcpad_last_return == GST_FLOW_WRONG_STATE) && (enc->wvcsrcpad_last_return == GST_FLOW_WRONG_STATE)) { ret = GST_FLOW_WRONG_STATE; } else { GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL), ("encoding samples failed")); ret = GST_FLOW_ERROR; } } gst_buffer_unref (buf); gst_object_unref (enc); return ret; }
static GstFlowReturn gst_rtp_celt_pay_flush_queued (GstRtpCELTPay * rtpceltpay) { GstFlowReturn ret; GstBuffer *buf, *outbuf; guint8 *payload, *spayload; guint payload_len; GstClockTime duration; GstRTPBuffer rtp = { NULL, }; payload_len = rtpceltpay->bytes + rtpceltpay->sbytes; duration = rtpceltpay->qduration; GST_DEBUG_OBJECT (rtpceltpay, "flushing out %u, duration %" GST_TIME_FORMAT, payload_len, GST_TIME_ARGS (rtpceltpay->qduration)); /* get a big enough packet for the sizes + payloads */ outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); GST_BUFFER_DURATION (outbuf) = duration; gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp); /* point to the payload for size headers and data */ spayload = gst_rtp_buffer_get_payload (&rtp); payload = spayload + rtpceltpay->sbytes; while ((buf = g_queue_pop_head (rtpceltpay->queue))) { guint size; /* copy first timestamp to output */ if (GST_BUFFER_PTS (outbuf) == -1) GST_BUFFER_PTS (outbuf) = GST_BUFFER_PTS (buf); /* write the size to the header */ size = gst_buffer_get_size (buf); while (size > 0xff) { *spayload++ = 0xff; size -= 0xff; } *spayload++ = size; /* copy payload */ size = gst_buffer_get_size (buf); gst_buffer_extract (buf, 0, payload, size); payload += size; gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpceltpay), outbuf, buf, g_quark_from_static_string (GST_META_TAG_AUDIO_STR)); gst_buffer_unref (buf); } gst_rtp_buffer_unmap (&rtp); /* we consumed it all */ rtpceltpay->bytes = 0; rtpceltpay->sbytes = 0; rtpceltpay->qduration = 0; ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpceltpay), outbuf); return ret; }
static GstFlowReturn gst_rtp_mux_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstRTPMux *rtp_mux; GstFlowReturn ret; GstRTPMuxPadPrivate *padpriv; gboolean drop; gboolean changed = FALSE; GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT; rtp_mux = GST_RTP_MUX (parent); if (gst_pad_check_reconfigure (rtp_mux->srcpad)) { GstCaps *current_caps = gst_pad_get_current_caps (pad); if (!gst_rtp_mux_setcaps (pad, rtp_mux, current_caps)) { ret = GST_FLOW_NOT_NEGOTIATED; gst_buffer_unref (buffer); goto out; } gst_caps_unref (current_caps); } GST_OBJECT_LOCK (rtp_mux); padpriv = gst_pad_get_element_private (pad); if (!padpriv) { GST_OBJECT_UNLOCK (rtp_mux); gst_buffer_unref (buffer); return GST_FLOW_NOT_LINKED; } buffer = gst_buffer_make_writable (buffer); if (!gst_rtp_buffer_map (buffer, GST_MAP_READWRITE, &rtpbuffer)) { GST_OBJECT_UNLOCK (rtp_mux); gst_buffer_unref (buffer); GST_ERROR_OBJECT (rtp_mux, "Invalid RTP buffer"); return GST_FLOW_ERROR; } drop = !process_buffer_locked (rtp_mux, padpriv, &rtpbuffer); gst_rtp_buffer_unmap (&rtpbuffer); if (!drop) { if (pad != rtp_mux->last_pad) { changed = TRUE; g_clear_object (&rtp_mux->last_pad); rtp_mux->last_pad = g_object_ref (pad); } if (GST_BUFFER_DURATION_IS_VALID (buffer) && GST_BUFFER_PTS_IS_VALID (buffer)) rtp_mux->last_stop = GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer); else rtp_mux->last_stop = GST_CLOCK_TIME_NONE; } GST_OBJECT_UNLOCK (rtp_mux); if (changed) gst_pad_sticky_events_foreach (pad, resend_events, rtp_mux); if (drop) { gst_buffer_unref (buffer); ret = GST_FLOW_OK; } else { ret = gst_pad_push (rtp_mux->srcpad, buffer); } out: return ret; }
bool GStreamerReader::DecodeVideoFrame(bool &aKeyFrameSkip, int64_t aTimeThreshold) { NS_ASSERTION(mDecoder->OnDecodeThread(), "Should be on decode thread."); GstBuffer *buffer = nullptr; { ReentrantMonitorAutoEnter mon(mGstThreadsMonitor); if (mReachedVideoEos && !mVideoSinkBufferCount) { return false; } /* Wait something to be decoded before return or continue */ if (!mVideoSinkBufferCount) { if (!mAudioSinkBufferCount) { /* We have nothing decoded so it makes no sense to return to the state machine * as it will call us back immediately, we'll return again and so on, wasting * CPU cycles for no job done. So, block here until there is either video or * audio data available */ mon.Wait(); if (!mVideoSinkBufferCount) { /* There is still no video data available, so either there is audio data or * something else has happened (Eos, etc...). Return to the state machine * to process it */ return true; } } else { return true; } } mDecoder->NotifyDecodedFrames(0, 1); #if GST_VERSION_MAJOR >= 1 GstSample *sample = gst_app_sink_pull_sample(mVideoAppSink); buffer = gst_buffer_ref(gst_sample_get_buffer(sample)); gst_sample_unref(sample); #else buffer = gst_app_sink_pull_buffer(mVideoAppSink); #endif mVideoSinkBufferCount--; } bool isKeyframe = !GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DELTA_UNIT); if ((aKeyFrameSkip && !isKeyframe)) { gst_buffer_unref(buffer); return true; } int64_t timestamp = GST_BUFFER_TIMESTAMP(buffer); { ReentrantMonitorAutoEnter mon(mGstThreadsMonitor); timestamp = gst_segment_to_stream_time(&mVideoSegment, GST_FORMAT_TIME, timestamp); } NS_ASSERTION(GST_CLOCK_TIME_IS_VALID(timestamp), "frame has invalid timestamp"); timestamp = GST_TIME_AS_USECONDS(timestamp); int64_t duration = 0; if (GST_CLOCK_TIME_IS_VALID(GST_BUFFER_DURATION(buffer))) duration = GST_TIME_AS_USECONDS(GST_BUFFER_DURATION(buffer)); else if (fpsNum && fpsDen) /* add 1-frame duration */ duration = gst_util_uint64_scale(GST_USECOND, fpsDen, fpsNum); if (timestamp < aTimeThreshold) { LOG(PR_LOG_DEBUG, ("skipping frame %" GST_TIME_FORMAT " threshold %" GST_TIME_FORMAT, GST_TIME_ARGS(timestamp * 1000), GST_TIME_ARGS(aTimeThreshold * 1000))); gst_buffer_unref(buffer); return true; } if (!buffer) /* no more frames */ return true; #if GST_VERSION_MAJOR >= 1 if (mConfigureAlignment && buffer->pool) { GstStructure *config = gst_buffer_pool_get_config(buffer->pool); GstVideoAlignment align; if (gst_buffer_pool_config_get_video_alignment(config, &align)) gst_video_info_align(&mVideoInfo, &align); gst_structure_free(config); mConfigureAlignment = false; } #endif nsRefPtr<PlanarYCbCrImage> image = GetImageFromBuffer(buffer); if (!image) { /* Ugh, upstream is not calling gst_pad_alloc_buffer(). Fallback to * allocating a PlanarYCbCrImage backed GstBuffer here and memcpy. */ GstBuffer* tmp = nullptr; CopyIntoImageBuffer(buffer, &tmp, image); gst_buffer_unref(buffer); buffer = tmp; } int64_t offset = mDecoder->GetResource()->Tell(); // Estimate location in media. VideoData* video = VideoData::CreateFromImage(mInfo.mVideo, mDecoder->GetImageContainer(), offset, timestamp, duration, static_cast<Image*>(image.get()), isKeyframe, -1, mPicture); mVideoQueue.Push(video); gst_buffer_unref(buffer); return true; }
/* we expect buffers starting on startcodes. */ static GstFlowReturn gst_rtp_mp4v_pay_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstRtpMP4VPay *rtpmp4vpay; GstFlowReturn ret; guint size, avail; guint packet_len; guint8 *data; gboolean flush; gint strip; GstClockTime timestamp, duration; ret = GST_FLOW_OK; rtpmp4vpay = GST_RTP_MP4V_PAY (basepayload); size = GST_BUFFER_SIZE (buffer); data = GST_BUFFER_DATA (buffer); timestamp = GST_BUFFER_TIMESTAMP (buffer); duration = GST_BUFFER_DURATION (buffer); avail = gst_adapter_available (rtpmp4vpay->adapter); if (duration == -1) duration = 0; /* empty buffer, take timestamp */ if (avail == 0) { rtpmp4vpay->first_timestamp = timestamp; rtpmp4vpay->duration = 0; } /* depay incomming data and see if we need to start a new RTP * packet */ flush = gst_rtp_mp4v_pay_depay_data (rtpmp4vpay, data, size, &strip); if (strip) { /* strip off config if requested */ if (!rtpmp4vpay->send_config) { GstBuffer *subbuf; /* strip off header */ subbuf = gst_buffer_create_sub (buffer, strip, size - strip); GST_BUFFER_TIMESTAMP (subbuf) = timestamp; gst_buffer_unref (buffer); buffer = subbuf; size = GST_BUFFER_SIZE (buffer); data = GST_BUFFER_DATA (buffer); } } /* if we need to flush, do so now */ if (flush) { ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay); rtpmp4vpay->first_timestamp = timestamp; rtpmp4vpay->duration = 0; avail = 0; } /* get packet length of data and see if we exceeded MTU. */ packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0); if (gst_basertppayload_is_filled (basepayload, packet_len, rtpmp4vpay->duration + duration)) { ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay); rtpmp4vpay->first_timestamp = timestamp; rtpmp4vpay->duration = 0; } /* push new data */ gst_adapter_push (rtpmp4vpay->adapter, buffer); rtpmp4vpay->duration += duration; return ret; }
static GstFlowReturn gst_audio_segment_clip_clip_buffer (GstSegmentClip * base, GstBuffer * buffer, GstBuffer ** outbuf) { GstAudioSegmentClip *self = GST_AUDIO_SEGMENT_CLIP (base); GstSegment *segment = &base->segment; GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer); GstClockTime duration = GST_BUFFER_DURATION (buffer); guint64 offset = GST_BUFFER_OFFSET (buffer); guint64 offset_end = GST_BUFFER_OFFSET_END (buffer); guint size = gst_buffer_get_size (buffer); if (!self->rate || !self->framesize) { GST_ERROR_OBJECT (self, "Not negotiated yet"); gst_buffer_unref (buffer); return GST_FLOW_NOT_NEGOTIATED; } if (segment->format != GST_FORMAT_DEFAULT && segment->format != GST_FORMAT_TIME) { GST_DEBUG_OBJECT (self, "Unsupported segment format %s", gst_format_get_name (segment->format)); *outbuf = buffer; return GST_FLOW_OK; } if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) { GST_WARNING_OBJECT (self, "Buffer without valid timestamp"); *outbuf = buffer; return GST_FLOW_OK; } *outbuf = gst_audio_buffer_clip (buffer, segment, self->rate, self->framesize); if (!*outbuf) { GST_DEBUG_OBJECT (self, "Buffer outside the configured segment"); /* Now return unexpected if we're before/after the end */ if (segment->format == GST_FORMAT_TIME) { if (segment->rate >= 0) { if (segment->stop != -1 && timestamp >= segment->stop) return GST_FLOW_EOS; } else { if (!GST_CLOCK_TIME_IS_VALID (duration)) duration = gst_util_uint64_scale_int (size, GST_SECOND, self->framesize * self->rate); if (segment->start != -1 && timestamp + duration <= segment->start) return GST_FLOW_EOS; } } else { if (segment->rate >= 0) { if (segment->stop != -1 && offset != -1 && offset >= segment->stop) return GST_FLOW_EOS; } else if (offset != -1 || offset_end != -1) { if (offset_end == -1) offset_end = offset + size / self->framesize; if (segment->start != -1 && offset_end <= segment->start) return GST_FLOW_EOS; } } } return GST_FLOW_OK; }
static void gst_musepackdec_loop (GstPad * sinkpad) { GstMusepackDec *musepackdec; GstFlowReturn flow; GstBuffer *out; #ifdef MPC_IS_OLD_API guint32 update_acc, update_bits; #else mpc_frame_info frame; mpc_status err; #endif gint num_samples, samplerate, bitspersample; musepackdec = GST_MUSEPACK_DEC (GST_PAD_PARENT (sinkpad)); samplerate = g_atomic_int_get (&musepackdec->rate); if (samplerate == 0) { if (!gst_musepack_stream_init (musepackdec)) goto pause_task; gst_musepackdec_send_newsegment (musepackdec); samplerate = g_atomic_int_get (&musepackdec->rate); } bitspersample = g_atomic_int_get (&musepackdec->bps); flow = gst_pad_alloc_buffer_and_set_caps (musepackdec->srcpad, -1, MPC_DECODER_BUFFER_LENGTH * 4, GST_PAD_CAPS (musepackdec->srcpad), &out); if (flow != GST_FLOW_OK) { GST_DEBUG_OBJECT (musepackdec, "Flow: %s", gst_flow_get_name (flow)); goto pause_task; } #ifdef MPC_IS_OLD_API num_samples = mpc_decoder_decode (musepackdec->d, (MPC_SAMPLE_FORMAT *) GST_BUFFER_DATA (out), &update_acc, &update_bits); if (num_samples < 0) { GST_ERROR_OBJECT (musepackdec, "Failed to decode sample"); GST_ELEMENT_ERROR (musepackdec, STREAM, DECODE, (NULL), (NULL)); goto pause_task; } else if (num_samples == 0) { goto eos_and_pause; } #else frame.buffer = (MPC_SAMPLE_FORMAT *) GST_BUFFER_DATA (out); err = mpc_demux_decode (musepackdec->d, &frame); if (err != MPC_STATUS_OK) { GST_ERROR_OBJECT (musepackdec, "Failed to decode sample"); GST_ELEMENT_ERROR (musepackdec, STREAM, DECODE, (NULL), (NULL)); goto pause_task; } else if (frame.bits == -1) { goto eos_and_pause; } num_samples = frame.samples; #endif GST_BUFFER_SIZE (out) = num_samples * bitspersample; GST_BUFFER_OFFSET (out) = musepackdec->segment.last_stop; GST_BUFFER_TIMESTAMP (out) = gst_util_uint64_scale_int (musepackdec->segment.last_stop, GST_SECOND, samplerate); GST_BUFFER_DURATION (out) = gst_util_uint64_scale_int (num_samples, GST_SECOND, samplerate); musepackdec->segment.last_stop += num_samples; GST_LOG_OBJECT (musepackdec, "Pushing buffer, timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out))); flow = gst_pad_push (musepackdec->srcpad, out); if (flow != GST_FLOW_OK) { GST_DEBUG_OBJECT (musepackdec, "Flow: %s", gst_flow_get_name (flow)); goto pause_task; } /* check if we're at the end of a configured segment */ if (musepackdec->segment.stop != -1 && musepackdec->segment.last_stop >= musepackdec->segment.stop) { gint64 stop_time; GST_DEBUG_OBJECT (musepackdec, "Reached end of configured segment"); if ((musepackdec->segment.flags & GST_SEEK_FLAG_SEGMENT) == 0) goto eos_and_pause; GST_DEBUG_OBJECT (musepackdec, "Posting SEGMENT_DONE message"); stop_time = gst_util_uint64_scale_int (musepackdec->segment.stop, GST_SECOND, samplerate); gst_element_post_message (GST_ELEMENT (musepackdec), gst_message_new_segment_done (GST_OBJECT (musepackdec), GST_FORMAT_TIME, stop_time)); goto pause_task; } return; eos_and_pause: { GST_DEBUG_OBJECT (musepackdec, "sending EOS event"); gst_pad_push_event (musepackdec->srcpad, gst_event_new_eos ()); /* fall through to pause */ } pause_task: { GST_DEBUG_OBJECT (musepackdec, "Pausing task"); gst_pad_pause_task (sinkpad); return; } }
static GstFlowReturn gst_ac3_parse_handle_frame (GstBaseParse * parse, GstBaseParseFrame * frame, gint * skipsize) { GstAc3Parse *ac3parse = GST_AC3_PARSE (parse); GstBuffer *buf = frame->buffer; GstByteReader reader; gint off; gboolean lost_sync, draining, eac, more = FALSE; guint frmsiz, blocks, sid; guint rate, chans; gboolean update_rate = FALSE; gint framesize = 0; gint have_blocks = 0; GstMapInfo map; gboolean ret = FALSE; GstFlowReturn res = GST_FLOW_OK; gst_buffer_map (buf, &map, GST_MAP_READ); if (G_UNLIKELY (map.size < 8)) { *skipsize = 1; goto cleanup; } gst_byte_reader_init (&reader, map.data, map.size); off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffff0000, 0x0b770000, 0, map.size); GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off); /* didn't find anything that looks like a sync word, skip */ if (off < 0) { *skipsize = map.size - 3; goto cleanup; } /* possible frame header, but not at offset 0? skip bytes before sync */ if (off > 0) { *skipsize = off; goto cleanup; } /* make sure the values in the frame header look sane */ if (!gst_ac3_parse_frame_header (ac3parse, buf, 0, &frmsiz, &rate, &chans, &blocks, &sid, &eac)) { *skipsize = off + 2; goto cleanup; } GST_LOG_OBJECT (parse, "size: %u, blocks: %u, rate: %u, chans: %u", frmsiz, blocks, rate, chans); framesize = frmsiz; if (G_UNLIKELY (g_atomic_int_get (&ac3parse->align) == GST_AC3_PARSE_ALIGN_NONE)) gst_ac3_parse_set_alignment (ac3parse, eac); GST_LOG_OBJECT (parse, "got frame"); lost_sync = GST_BASE_PARSE_LOST_SYNC (parse); draining = GST_BASE_PARSE_DRAINING (parse); if (g_atomic_int_get (&ac3parse->align) == GST_AC3_PARSE_ALIGN_IEC61937) { /* We need 6 audio blocks from each substream, so we keep going forwards * till we have it */ g_assert (blocks > 0); GST_LOG_OBJECT (ac3parse, "Need %d frames before pushing", 6 / blocks); if (sid != 0) { /* We need the first substream to be the one with id 0 */ GST_LOG_OBJECT (ac3parse, "Skipping till we find sid 0"); *skipsize = off + 2; goto cleanup; } framesize = 0; /* Loop till we have 6 blocks per substream */ for (have_blocks = 0; !more && have_blocks < 6; have_blocks += blocks) { /* Loop till we get one frame from each substream */ do { framesize += frmsiz; if (!gst_byte_reader_skip (&reader, frmsiz) || map.size < (framesize + 6)) { more = TRUE; break; } if (!gst_ac3_parse_frame_header (ac3parse, buf, framesize, &frmsiz, NULL, NULL, NULL, &sid, &eac)) { *skipsize = off + 2; goto cleanup; } } while (sid); } /* We're now at the next frame, so no need to skip if resyncing */ frmsiz = 0; } if (lost_sync && !draining) { guint16 word = 0; GST_DEBUG_OBJECT (ac3parse, "resyncing; checking next frame syncword"); if (more || !gst_byte_reader_skip (&reader, frmsiz) || !gst_byte_reader_get_uint16_be (&reader, &word)) { GST_DEBUG_OBJECT (ac3parse, "... but not sufficient data"); gst_base_parse_set_min_frame_size (parse, framesize + 8); *skipsize = 0; goto cleanup; } else { if (word != 0x0b77) { GST_DEBUG_OBJECT (ac3parse, "0x%x not OK", word); *skipsize = off + 2; goto cleanup; } else { /* ok, got sync now, let's assume constant frame size */ gst_base_parse_set_min_frame_size (parse, framesize); } } } /* expect to have found a frame here */ g_assert (framesize); ret = TRUE; /* arrange for metadata setup */ if (G_UNLIKELY (sid)) { /* dependent frame, no need to (ac)count for or consider further */ GST_LOG_OBJECT (parse, "sid: %d", sid); frame->flags |= GST_BASE_PARSE_FRAME_FLAG_NO_FRAME; /* TODO maybe also mark as DELTA_UNIT, * if that does not surprise baseparse elsewhere */ /* occupies same time space as previous base frame */ if (G_LIKELY (GST_BUFFER_TIMESTAMP (buf) >= GST_BUFFER_DURATION (buf))) GST_BUFFER_TIMESTAMP (buf) -= GST_BUFFER_DURATION (buf); /* only shortcut if we already arranged for caps */ if (G_LIKELY (ac3parse->sample_rate > 0)) goto cleanup; } if (G_UNLIKELY (ac3parse->sample_rate != rate || ac3parse->channels != chans || ac3parse->eac != eac)) { GstCaps *caps = gst_caps_new_simple (eac ? "audio/x-eac3" : "audio/x-ac3", "framed", G_TYPE_BOOLEAN, TRUE, "rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, chans, NULL); gst_caps_set_simple (caps, "alignment", G_TYPE_STRING, g_atomic_int_get (&ac3parse->align) == GST_AC3_PARSE_ALIGN_IEC61937 ? "iec61937" : "frame", NULL); gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps); gst_caps_unref (caps); ac3parse->sample_rate = rate; ac3parse->channels = chans; ac3parse->eac = eac; update_rate = TRUE; } if (G_UNLIKELY (ac3parse->blocks != blocks)) { ac3parse->blocks = blocks; update_rate = TRUE; } if (G_UNLIKELY (update_rate)) gst_base_parse_set_frame_rate (parse, rate, 256 * blocks, 2, 2); cleanup: gst_buffer_unmap (buf, &map); if (ret && framesize <= map.size) { res = gst_base_parse_finish_frame (parse, frame, framesize); } return res; }
GstPadProbeReturn GstEnginePipeline::HandoffCallback(GstPad*, GstPadProbeInfo* info, gpointer self) { GstEnginePipeline* instance = reinterpret_cast<GstEnginePipeline*>(self); GstBuffer* buf = gst_pad_probe_info_get_buffer(info); QList<BufferConsumer*> consumers; { QMutexLocker l(&instance->buffer_consumers_mutex_); consumers = instance->buffer_consumers_; } for (BufferConsumer* consumer : consumers) { gst_buffer_ref(buf); consumer->ConsumeBuffer(buf, instance->id()); } // Calculate the end time of this buffer so we can stop playback if it's // after the end time of this song. if (instance->end_offset_nanosec_ > 0) { quint64 start_time = GST_BUFFER_TIMESTAMP(buf) - instance->segment_start_; quint64 duration = GST_BUFFER_DURATION(buf); quint64 end_time = start_time + duration; if (end_time > instance->end_offset_nanosec_) { if (instance->has_next_valid_url()) { if (instance->next_url_ == instance->url_ && instance->next_beginning_offset_nanosec_ == instance->end_offset_nanosec_) { // The "next" song is actually the next segment of this file - so // cheat and keep on playing, but just tell the Engine we've moved on. instance->end_offset_nanosec_ = instance->next_end_offset_nanosec_; instance->next_url_ = QUrl(); instance->next_beginning_offset_nanosec_ = 0; instance->next_end_offset_nanosec_ = 0; // GstEngine will try to seek to the start of the new section, but // we're already there so ignore it. instance->ignore_next_seek_ = true; emit instance->EndOfStreamReached(instance->id(), true); } else { // We have a next song but we can't cheat, so move to it normally. instance->TransitionToNext(); } } else { // There's no next song emit instance->EndOfStreamReached(instance->id(), false); } } } if (instance->emit_track_ended_on_time_discontinuity_) { if (GST_BUFFER_FLAG_IS_SET(buf, GST_BUFFER_FLAG_DISCONT) || GST_BUFFER_OFFSET(buf) < instance->last_buffer_offset_) { qLog(Debug) << "Buffer discontinuity - emitting EOS"; instance->emit_track_ended_on_time_discontinuity_ = false; emit instance->EndOfStreamReached(instance->id(), true); } } instance->last_buffer_offset_ = GST_BUFFER_OFFSET(buf); return GST_PAD_PROBE_OK; }
static GstFlowReturn gst_rtp_amr_pay_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstRtpAMRPay *rtpamrpay; GstFlowReturn ret; guint size, payload_len; GstBuffer *outbuf; guint8 *payload, *data, *payload_amr; GstClockTime timestamp, duration; guint packet_len, mtu; gint i, num_packets, num_nonempty_packets; gint amr_len; gint *frame_size; rtpamrpay = GST_RTP_AMR_PAY (basepayload); mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpamrpay); size = GST_BUFFER_SIZE (buffer); data = GST_BUFFER_DATA (buffer); timestamp = GST_BUFFER_TIMESTAMP (buffer); duration = GST_BUFFER_DURATION (buffer); /* setup frame size pointer */ if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB) frame_size = nb_frame_size; else frame_size = wb_frame_size; GST_DEBUG_OBJECT (basepayload, "got %d bytes", size); /* FIXME, only * octet aligned, no interleaving, single channel, no CRC, * no robust-sorting. To fix this you need to implement the downstream * negotiation function. */ /* first count number of packets and total amr frame size */ amr_len = num_packets = num_nonempty_packets = 0; for (i = 0; i < size; i++) { guint8 FT; gint fr_size; FT = (data[i] & 0x78) >> 3; fr_size = frame_size[FT]; GST_DEBUG_OBJECT (basepayload, "frame size %d", fr_size); /* FIXME, we don't handle this yet.. */ if (fr_size <= 0) goto wrong_size; amr_len += fr_size; num_nonempty_packets++; num_packets++; i += fr_size; } if (amr_len > size) goto incomplete_frame; /* we need one extra byte for the CMR, the ToC is in the input * data */ payload_len = size + 1; /* get packet len to check against MTU */ packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0); if (packet_len > mtu) goto too_big; /* now alloc output buffer */ outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); /* copy timestamp */ GST_BUFFER_TIMESTAMP (outbuf) = timestamp; /* FIXME: when we do more than one AMR frame per packet, fix this */ if (duration != GST_CLOCK_TIME_NONE) GST_BUFFER_DURATION (outbuf) = duration; else { GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND; } if (GST_BUFFER_IS_DISCONT (buffer)) { GST_DEBUG_OBJECT (basepayload, "discont, setting marker bit"); GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); gst_rtp_buffer_set_marker (outbuf, TRUE); } /* get payload, this is now writable */ payload = gst_rtp_buffer_get_payload (outbuf); /* 0 1 2 3 4 5 6 7 * +-+-+-+-+-+-+-+-+ * | CMR |R|R|R|R| * +-+-+-+-+-+-+-+-+ */ payload[0] = 0xF0; /* CMR, no specific mode requested */ /* this is where we copy the AMR data, after num_packets FTs and the * CMR. */ payload_amr = payload + num_packets + 1; /* copy data in payload, first we copy all the FTs then all * the AMR data. The last FT has to have the F flag cleared. */ for (i = 1; i <= num_packets; i++) { guint8 FT; gint fr_size; /* 0 1 2 3 4 5 6 7 * +-+-+-+-+-+-+-+-+ * |F| FT |Q|P|P| more FT... * +-+-+-+-+-+-+-+-+ */ FT = (*data & 0x78) >> 3; fr_size = frame_size[FT]; if (i == num_packets) /* last packet, clear F flag */ payload[i] = *data & 0x7f; else /* set F flag */ payload[i] = *data | 0x80; memcpy (payload_amr, &data[1], fr_size); /* all sizes are > 0 since we checked for that above */ data += fr_size + 1; payload_amr += fr_size; } gst_buffer_unref (buffer); ret = gst_basertppayload_push (basepayload, outbuf); return ret; /* ERRORS */ wrong_size: { GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT, (NULL), ("received AMR frame with size <= 0")); gst_buffer_unref (buffer); return GST_FLOW_ERROR; } incomplete_frame: { GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT, (NULL), ("received incomplete AMR frames")); gst_buffer_unref (buffer); return GST_FLOW_ERROR; } too_big: { GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT, (NULL), ("received too many AMR frames for MTU")); gst_buffer_unref (buffer); return GST_FLOW_ERROR; } }
static GstFlowReturn gst_amc_audio_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf) { GstAmcAudioDec *self; gint idx; GstAmcBuffer *buf; GstAmcBufferInfo buffer_info; guint offset = 0; GstClockTime timestamp, duration, timestamp_offset = 0; GstMapInfo minfo; memset (&minfo, 0, sizeof (minfo)); self = GST_AMC_AUDIO_DEC (decoder); GST_DEBUG_OBJECT (self, "Handling frame"); /* Make sure to keep a reference to the input here, * it can be unreffed from the other thread if * finish_frame() is called */ if (inbuf) inbuf = gst_buffer_ref (inbuf); if (!self->started) { GST_ERROR_OBJECT (self, "Codec not started yet"); if (inbuf) gst_buffer_unref (inbuf); return GST_FLOW_NOT_NEGOTIATED; } if (self->eos) { GST_WARNING_OBJECT (self, "Got frame after EOS"); if (inbuf) gst_buffer_unref (inbuf); return GST_FLOW_EOS; } if (self->flushing) goto flushing; if (self->downstream_flow_ret != GST_FLOW_OK) goto downstream_error; if (!inbuf) return gst_amc_audio_dec_drain (self); timestamp = GST_BUFFER_PTS (inbuf); duration = GST_BUFFER_DURATION (inbuf); gst_buffer_map (inbuf, &minfo, GST_MAP_READ); while (offset < minfo.size) { /* Make sure to release the base class stream lock, otherwise * _loop() can't call _finish_frame() and we might block forever * because no input buffers are released */ GST_AUDIO_DECODER_STREAM_UNLOCK (self); /* Wait at most 100ms here, some codecs don't fail dequeueing if * the codec is flushing, causing deadlocks during shutdown */ idx = gst_amc_codec_dequeue_input_buffer (self->codec, 100000); GST_AUDIO_DECODER_STREAM_LOCK (self); if (idx < 0) { if (self->flushing) goto flushing; switch (idx) { case INFO_TRY_AGAIN_LATER: GST_DEBUG_OBJECT (self, "Dequeueing input buffer timed out"); continue; /* next try */ break; case G_MININT: GST_ERROR_OBJECT (self, "Failed to dequeue input buffer"); goto dequeue_error; default: g_assert_not_reached (); break; } continue; } if (idx >= self->n_input_buffers) goto invalid_buffer_index; if (self->flushing) goto flushing; if (self->downstream_flow_ret != GST_FLOW_OK) { memset (&buffer_info, 0, sizeof (buffer_info)); gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info); goto downstream_error; } /* Now handle the frame */ /* Copy the buffer content in chunks of size as requested * by the port */ buf = &self->input_buffers[idx]; memset (&buffer_info, 0, sizeof (buffer_info)); buffer_info.offset = 0; buffer_info.size = MIN (minfo.size - offset, buf->size); orc_memcpy (buf->data, minfo.data + offset, buffer_info.size); /* Interpolate timestamps if we're passing the buffer * in multiple chunks */ if (offset != 0 && duration != GST_CLOCK_TIME_NONE) { timestamp_offset = gst_util_uint64_scale (offset, duration, minfo.size); } if (timestamp != GST_CLOCK_TIME_NONE) { buffer_info.presentation_time_us = gst_util_uint64_scale (timestamp + timestamp_offset, 1, GST_USECOND); self->last_upstream_ts = timestamp + timestamp_offset; } if (duration != GST_CLOCK_TIME_NONE) self->last_upstream_ts += duration; if (offset == 0) { if (!GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DELTA_UNIT)) buffer_info.flags |= BUFFER_FLAG_SYNC_FRAME; } offset += buffer_info.size; GST_DEBUG_OBJECT (self, "Queueing buffer %d: size %d time %" G_GINT64_FORMAT " flags 0x%08x", idx, buffer_info.size, buffer_info.presentation_time_us, buffer_info.flags); if (!gst_amc_codec_queue_input_buffer (self->codec, idx, &buffer_info)) goto queue_error; } gst_buffer_unmap (inbuf, &minfo); gst_buffer_unref (inbuf); return self->downstream_flow_ret; downstream_error: { GST_ERROR_OBJECT (self, "Downstream returned %s", gst_flow_get_name (self->downstream_flow_ret)); if (minfo.data) gst_buffer_unmap (inbuf, &minfo); if (inbuf) gst_buffer_unref (inbuf); return self->downstream_flow_ret; } invalid_buffer_index: { GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL), ("Invalid input buffer index %d of %d", idx, self->n_input_buffers)); if (minfo.data) gst_buffer_unmap (inbuf, &minfo); if (inbuf) gst_buffer_unref (inbuf); return GST_FLOW_ERROR; } dequeue_error: { GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL), ("Failed to dequeue input buffer")); if (minfo.data) gst_buffer_unmap (inbuf, &minfo); if (inbuf) gst_buffer_unref (inbuf); return GST_FLOW_ERROR; } queue_error: { GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL), ("Failed to queue input buffer")); if (minfo.data) gst_buffer_unmap (inbuf, &minfo); if (inbuf) gst_buffer_unref (inbuf); return GST_FLOW_ERROR; } flushing: { GST_DEBUG_OBJECT (self, "Flushing -- returning FLUSHING"); if (minfo.data) gst_buffer_unmap (inbuf, &minfo); if (inbuf) gst_buffer_unref (inbuf); return GST_FLOW_FLUSHING; } }
static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count) { GstWavpackEncWriteID *wid = (GstWavpackEncWriteID *) id; GstWavpackEnc *enc = GST_WAVPACK_ENC (wid->wavpack_enc); GstFlowReturn *flow; GstBuffer *buffer; GstPad *pad; guchar *block = (guchar *) data; pad = (wid->correction) ? enc->wvcsrcpad : enc->srcpad; flow = (wid->correction) ? &enc->wvcsrcpad_last_return : &enc-> srcpad_last_return; *flow = gst_pad_alloc_buffer_and_set_caps (pad, GST_BUFFER_OFFSET_NONE, count, GST_PAD_CAPS (pad), &buffer); if (*flow != GST_FLOW_OK) { GST_WARNING_OBJECT (enc, "flow on %s:%s = %s", GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow)); return FALSE; } g_memmove (GST_BUFFER_DATA (buffer), block, count); if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) { /* if it's a Wavpack block set buffer timestamp and duration, etc */ WavpackHeader wph; GST_LOG_OBJECT (enc, "got %d bytes of encoded wavpack %sdata", count, (wid->correction) ? "correction " : ""); gst_wavpack_read_header (&wph, block); /* Only set when pushing the first buffer again, in that case * we don't want to delay the buffer or push newsegment events */ if (!wid->passthrough) { /* Only push complete blocks */ if (enc->pending_buffer == NULL) { enc->pending_buffer = buffer; enc->pending_offset = wph.block_index; } else if (enc->pending_offset == wph.block_index) { enc->pending_buffer = gst_buffer_join (enc->pending_buffer, buffer); } else { GST_ERROR ("Got incomplete block, dropping"); gst_buffer_unref (enc->pending_buffer); enc->pending_buffer = buffer; enc->pending_offset = wph.block_index; } if (!(wph.flags & FINAL_BLOCK)) return TRUE; buffer = enc->pending_buffer; enc->pending_buffer = NULL; enc->pending_offset = 0; /* if it's the first wavpack block, send a NEW_SEGMENT event */ if (wph.block_index == 0) { gst_pad_push_event (pad, gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, GST_BUFFER_OFFSET_NONE, 0)); /* save header for later reference, so we can re-send it later on * EOS with fixed up values for total sample count etc. */ if (enc->first_block == NULL && !wid->correction) { enc->first_block = g_memdup (GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer)); enc->first_block_size = GST_BUFFER_SIZE (buffer); } } } /* set buffer timestamp, duration, offset, offset_end from * the wavpack header */ GST_BUFFER_TIMESTAMP (buffer) = enc->timestamp_offset + gst_util_uint64_scale_int (GST_SECOND, wph.block_index, enc->samplerate); GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (GST_SECOND, wph.block_samples, enc->samplerate); GST_BUFFER_OFFSET (buffer) = wph.block_index; GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples; } else { /* if it's something else set no timestamp and duration on the buffer */ GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count); GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE; GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE; } /* push the buffer and forward errors */ GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes", GST_BUFFER_SIZE (buffer)); *flow = gst_pad_push (pad, buffer); if (*flow != GST_FLOW_OK) { GST_WARNING_OBJECT (enc, "flow on %s:%s = %s", GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow)); return FALSE; } return TRUE; }
static GstFlowReturn pad_chain (GstPad *pad, GstBuffer *buf) { GOmxCore *gomx; GOmxPort *in_port; GstOmxBaseFilter *self; GstFlowReturn ret = GST_FLOW_OK; self = GST_OMX_BASE_FILTER (GST_OBJECT_PARENT (pad)); gomx = self->gomx; GST_LOG_OBJECT (self, "begin"); GST_LOG_OBJECT (self, "gst_buffer: size=%u", GST_BUFFER_SIZE (buf)); GST_LOG_OBJECT (self, "state: %d", gomx->omx_state); if (G_UNLIKELY (gomx->omx_state == OMX_StateLoaded)) { g_mutex_lock (self->ready_lock); GST_INFO_OBJECT (self, "omx: prepare"); /** @todo this should probably go after doing preparations. */ if (self->omx_setup) { self->omx_setup (self); } setup_ports (self); g_omx_core_prepare (self->gomx); if (gomx->omx_state == OMX_StateIdle) { self->ready = TRUE; gst_pad_start_task (self->srcpad, output_loop, self->srcpad); } g_mutex_unlock (self->ready_lock); if (gomx->omx_state != OMX_StateIdle) goto out_flushing; } in_port = self->in_port; if (G_LIKELY (in_port->enabled)) { guint buffer_offset = 0; if (G_UNLIKELY (gomx->omx_state == OMX_StateIdle)) { GST_INFO_OBJECT (self, "omx: play"); g_omx_core_start (gomx); if (gomx->omx_state != OMX_StateExecuting) goto out_flushing; /* send buffer with codec data flag */ /** @todo move to util */ if (self->codec_data) { OMX_BUFFERHEADERTYPE *omx_buffer; GST_LOG_OBJECT (self, "request buffer"); omx_buffer = g_omx_port_request_buffer (in_port); if (G_LIKELY (omx_buffer)) { omx_buffer->nFlags |= 0x00000080; /* codec data flag */ omx_buffer->nFilledLen = GST_BUFFER_SIZE (self->codec_data); memcpy (omx_buffer->pBuffer + omx_buffer->nOffset, GST_BUFFER_DATA (self->codec_data), omx_buffer->nFilledLen); GST_LOG_OBJECT (self, "release_buffer"); g_omx_port_release_buffer (in_port, omx_buffer); } } } if (G_UNLIKELY (gomx->omx_state != OMX_StateExecuting)) { GST_ERROR_OBJECT (self, "Whoa! very wrong"); } while (G_LIKELY (buffer_offset < GST_BUFFER_SIZE (buf))) { OMX_BUFFERHEADERTYPE *omx_buffer; if (self->last_pad_push_return != GST_FLOW_OK || !(gomx->omx_state == OMX_StateExecuting || gomx->omx_state == OMX_StatePause)) { goto out_flushing; } GST_LOG_OBJECT (self, "request buffer"); omx_buffer = g_omx_port_request_buffer (in_port); GST_LOG_OBJECT (self, "omx_buffer: %p", omx_buffer); if (G_LIKELY (omx_buffer)) { GST_DEBUG_OBJECT (self, "omx_buffer: size=%lu, len=%lu, flags=%lu, offset=%lu, timestamp=%lld", omx_buffer->nAllocLen, omx_buffer->nFilledLen, omx_buffer->nFlags, omx_buffer->nOffset, omx_buffer->nTimeStamp); if (omx_buffer->nOffset == 0 && self->share_input_buffer) { { GstBuffer *old_buf; old_buf = omx_buffer->pAppPrivate; if (old_buf) { gst_buffer_unref (old_buf); } else if (omx_buffer->pBuffer) { g_free (omx_buffer->pBuffer); } } omx_buffer->pBuffer = GST_BUFFER_DATA (buf); omx_buffer->nAllocLen = GST_BUFFER_SIZE (buf); omx_buffer->nFilledLen = GST_BUFFER_SIZE (buf); omx_buffer->pAppPrivate = buf; } else { omx_buffer->nFilledLen = MIN (GST_BUFFER_SIZE (buf) - buffer_offset, omx_buffer->nAllocLen - omx_buffer->nOffset); memcpy (omx_buffer->pBuffer + omx_buffer->nOffset, GST_BUFFER_DATA (buf) + buffer_offset, omx_buffer->nFilledLen); } if (self->use_timestamps) { GstClockTime timestamp_offset = 0; if (buffer_offset && GST_BUFFER_DURATION (buf) != GST_CLOCK_TIME_NONE) { timestamp_offset = gst_util_uint64_scale_int (buffer_offset, GST_BUFFER_DURATION (buf), GST_BUFFER_SIZE (buf)); } omx_buffer->nTimeStamp = gst_util_uint64_scale_int (GST_BUFFER_TIMESTAMP (buf) + timestamp_offset, OMX_TICKS_PER_SECOND, GST_SECOND); } buffer_offset += omx_buffer->nFilledLen; GST_LOG_OBJECT (self, "release_buffer"); /** @todo untaint buffer */ g_omx_port_release_buffer (in_port, omx_buffer); } else { GST_WARNING_OBJECT (self, "null buffer"); ret = GST_FLOW_WRONG_STATE; goto out_flushing; } } } else { GST_WARNING_OBJECT (self, "done"); ret = GST_FLOW_UNEXPECTED; } if (!self->share_input_buffer) { gst_buffer_unref (buf); } leave: GST_LOG_OBJECT (self, "end"); return ret; /* special conditions */ out_flushing: { const gchar *error_msg = NULL; if (gomx->omx_error) { error_msg = "Error from OpenMAX component"; } else if (gomx->omx_state != OMX_StateExecuting && gomx->omx_state != OMX_StatePause) { error_msg = "OpenMAX component in wrong state"; } if (error_msg) { GST_ELEMENT_ERROR (self, STREAM, FAILED, (NULL), (error_msg)); ret = GST_FLOW_ERROR; } gst_buffer_unref (buf); goto leave; } }
/* Pipeline Callbacks */ static gboolean probe_cb (InsanityGstTest * ptest, GstPad * pad, GstMiniObject * object, gpointer userdata) { InsanityTest *test = INSANITY_TEST (ptest); global_last_probe = g_get_monotonic_time (); DECODER_TEST_LOCK (); if (GST_IS_BUFFER (object)) { GstBuffer *buf; GstClockTime ts; buf = GST_BUFFER (object); ts = GST_BUFFER_PTS (buf); /* First check clipping */ if (glob_testing_parser == FALSE && GST_CLOCK_TIME_IS_VALID (ts) && glob_waiting_segment == FALSE) { GstClockTime ts_end, cstart, cstop; /* Check if buffer is completely outside the segment */ ts_end = ts; if (GST_BUFFER_DURATION_IS_VALID (buf)) ts_end += GST_BUFFER_DURATION (buf); /* Check if buffer is completely outside the segment */ ts_end = ts; if (!gst_segment_clip (&glob_last_segment, glob_last_segment.format, ts, ts_end, &cstart, &cstop)) { char *msg = g_strdup_printf ("Got timestamp %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT ", outside configured segment (%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT "), method %s", GST_TIME_ARGS (ts), GST_TIME_ARGS (ts_end), GST_TIME_ARGS (glob_last_segment.start), GST_TIME_ARGS (glob_last_segment.stop), test_get_name (glob_in_progress)); insanity_test_validate_checklist_item (INSANITY_TEST (ptest), "segment-clipping", FALSE, msg); g_free (msg); glob_bad_segment_clipping = TRUE; } } switch (glob_in_progress) { case TEST_NONE: if (glob_waiting_first_segment == TRUE) insanity_test_validate_checklist_item (test, "first-segment", FALSE, "Got a buffer before the first segment"); /* Got the first buffer, starting testing dance */ next_test (test); break; case TEST_POSITION: test_position (test, buf); break; case TEST_FAST_FORWARD: case TEST_BACKWARD_PLAYBACK: case TEST_FAST_BACKWARD: { gint64 stime_ts; if (GST_CLOCK_TIME_IS_VALID (ts) == FALSE || glob_waiting_segment == TRUE) { break; } stime_ts = gst_segment_to_stream_time (&glob_last_segment, glob_last_segment.format, ts); if (GST_CLOCK_TIME_IS_VALID (glob_seek_first_buf_ts) == FALSE) { GstClockTime expected_ts = gst_segment_to_stream_time (&glob_last_segment, glob_last_segment.format, glob_seek_rate < 0 ? glob_seek_stop_ts : glob_seek_segment_seektime); GstClockTimeDiff diff = ABS (GST_CLOCK_DIFF (stime_ts, expected_ts)); if (diff > SEEK_THRESHOLD) { gchar *valmsg = g_strdup_printf ("Received buffer timestamp %" GST_TIME_FORMAT " Seeek wanted %" GST_TIME_FORMAT "", GST_TIME_ARGS (stime_ts), GST_TIME_ARGS (expected_ts)); validate_current_test (test, FALSE, valmsg); next_test (test); g_free (valmsg); } else glob_seek_first_buf_ts = stime_ts; } else { GstClockTimeDiff diff = GST_CLOCK_DIFF (stime_ts, glob_seek_first_buf_ts); if (diff < 0) diff = -diff; if (diff >= glob_playback_duration * GST_SECOND) { validate_current_test (test, TRUE, NULL); next_test (test); } } break; } default: break; } } else if (GST_IS_EVENT (object)) { GstEvent *event = GST_EVENT (object); guint seqnum = gst_event_get_seqnum (event); if (G_LIKELY (glob_seqnum_found == FALSE) && seqnum == glob_seqnum) glob_seqnum_found = TRUE; if (glob_seqnum_found == TRUE && seqnum != glob_seqnum) { gchar *message = g_strdup_printf ("Current seqnum %i != " "received %i", glob_seqnum, seqnum); insanity_test_validate_checklist_item (test, "seqnum-management", FALSE, message); glob_wrong_seqnum = TRUE; g_free (message); } switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEGMENT: { gst_event_copy_segment (event, &glob_last_segment); if (glob_waiting_segment == FALSE) /* Cache the segment as it will be our reference but don't look * further */ goto done; glob_last_segment_start_time = glob_last_segment.start; if (glob_waiting_first_segment == TRUE) { insanity_test_validate_checklist_item (test, "first-segment", TRUE, NULL); glob_waiting_first_segment = FALSE; } else if (glob_in_progress >= TEST_FAST_FORWARD && glob_in_progress <= TEST_FAST_BACKWARD) { GstClockTimeDiff diff; gboolean valid_stop = TRUE; GstClockTimeDiff wdiff, rdiff; rdiff = ABS (GST_CLOCK_DIFF (glob_last_segment.stop, glob_last_segment.start)) * ABS (glob_last_segment.rate * glob_last_segment.applied_rate); wdiff = ABS (GST_CLOCK_DIFF (glob_seek_stop_ts, glob_seek_segment_seektime)); diff = GST_CLOCK_DIFF (glob_last_segment.position, glob_seek_segment_seektime); if (diff < 0) diff = -diff; /* Now compare with the expected segment */ if ((glob_last_segment.rate * glob_last_segment.applied_rate) == glob_seek_rate && diff <= SEEK_THRESHOLD && valid_stop) { glob_seek_got_segment = TRUE; } else { GstClockTime stopdiff = ABS (GST_CLOCK_DIFF (rdiff, wdiff)); gchar *validate_msg = g_strdup_printf ("Wrong segment received, Rate %f expected " "%f, start time diff %" GST_TIME_FORMAT " stop diff %" GST_TIME_FORMAT, (glob_last_segment.rate * glob_last_segment.applied_rate), glob_seek_rate, GST_TIME_ARGS (diff), GST_TIME_ARGS (stopdiff)); validate_current_test (test, FALSE, validate_msg); next_test (test); g_free (validate_msg); } } glob_waiting_segment = FALSE; break; } default: break; } } done: DECODER_TEST_UNLOCK (); return TRUE; }
static GstFlowReturn gst_vaapidecode_step(GstVaapiDecode *decode) { GstVaapiSurfaceProxy *proxy; GstVaapiDecoderStatus status; GstBuffer *buffer; GstFlowReturn ret; GstClockTime timestamp; gint64 end_time; for (;;) { end_time = decode->render_time_base; if (!end_time) end_time = g_get_monotonic_time(); end_time += GST_TIME_AS_USECONDS(decode->last_buffer_time); end_time += G_TIME_SPAN_SECOND; proxy = gst_vaapi_decoder_get_surface(decode->decoder, &status); if (!proxy) { if (status == GST_VAAPI_DECODER_STATUS_ERROR_NO_SURFACE) { gboolean was_signalled; g_mutex_lock(decode->decoder_mutex); was_signalled = g_cond_wait_until( decode->decoder_ready, decode->decoder_mutex, end_time ); g_mutex_unlock(decode->decoder_mutex); if (was_signalled) continue; goto error_decode_timeout; } if (status != GST_VAAPI_DECODER_STATUS_ERROR_NO_DATA) goto error_decode; /* More data is needed */ break; } g_object_weak_ref( G_OBJECT(proxy), (GWeakNotify)gst_vaapidecode_release, decode ); buffer = gst_vaapi_video_buffer_new(decode->display); if (!buffer) goto error_create_buffer; timestamp = GST_VAAPI_SURFACE_PROXY_TIMESTAMP(proxy); if (!decode->render_time_base) decode->render_time_base = g_get_monotonic_time(); decode->last_buffer_time = timestamp; GST_BUFFER_TIMESTAMP(buffer) = timestamp; GST_BUFFER_DURATION(buffer) = GST_VAAPI_SURFACE_PROXY_DURATION(proxy); gst_buffer_set_caps(buffer, GST_PAD_CAPS(decode->srcpad)); if (GST_VAAPI_SURFACE_PROXY_TFF(proxy)) GST_BUFFER_FLAG_SET(buffer, GST_VIDEO_BUFFER_TFF); gst_vaapi_video_buffer_set_surface_proxy( GST_VAAPI_VIDEO_BUFFER(buffer), proxy ); ret = gst_pad_push(decode->srcpad, buffer); if (ret != GST_FLOW_OK) goto error_commit_buffer; g_object_unref(proxy); } return GST_FLOW_OK; /* ERRORS */ error_decode_timeout: { GST_DEBUG("decode timeout. Decoder required a VA surface but none " "got available within one second"); return GST_FLOW_UNEXPECTED; } error_decode: { GST_DEBUG("decode error %d", status); switch (status) { case GST_VAAPI_DECODER_STATUS_ERROR_UNSUPPORTED_CODEC: case GST_VAAPI_DECODER_STATUS_ERROR_UNSUPPORTED_PROFILE: case GST_VAAPI_DECODER_STATUS_ERROR_UNSUPPORTED_CHROMA_FORMAT: ret = GST_FLOW_NOT_SUPPORTED; break; default: ret = GST_FLOW_UNEXPECTED; break; } return ret; } error_create_buffer: { const GstVaapiID surface_id = gst_vaapi_surface_get_id(GST_VAAPI_SURFACE_PROXY_SURFACE(proxy)); GST_DEBUG("video sink failed to create video buffer for proxy'ed " "surface %" GST_VAAPI_ID_FORMAT " (error %d)", GST_VAAPI_ID_ARGS(surface_id), ret); g_object_unref(proxy); return GST_FLOW_UNEXPECTED; } error_commit_buffer: { GST_DEBUG("video sink rejected the video buffer (error %d)", ret); g_object_unref(proxy); return GST_FLOW_UNEXPECTED; } }
static GstFlowReturn gst_rtp_celt_pay_handle_buffer (GstRTPBasePayload * basepayload, GstBuffer * buffer) { GstFlowReturn ret; GstRtpCELTPay *rtpceltpay; gsize payload_len; GstMapInfo map; GstClockTime duration, packet_dur; guint i, ssize, packet_len; rtpceltpay = GST_RTP_CELT_PAY (basepayload); ret = GST_FLOW_OK; gst_buffer_map (buffer, &map, GST_MAP_READ); switch (rtpceltpay->packet) { case 0: /* ident packet. We need to parse the headers to construct the RTP * properties. */ if (!gst_rtp_celt_pay_parse_ident (rtpceltpay, map.data, map.size)) goto parse_error; goto cleanup; case 1: /* comment packet, we ignore it */ goto cleanup; default: /* other packets go in the payload */ break; } gst_buffer_unmap (buffer, &map); duration = GST_BUFFER_DURATION (buffer); GST_LOG_OBJECT (rtpceltpay, "got buffer of duration %" GST_TIME_FORMAT ", size %" G_GSIZE_FORMAT, GST_TIME_ARGS (duration), map.size); /* calculate the size of the size field and the payload */ ssize = 1; for (i = map.size; i > 0xff; i -= 0xff) ssize++; GST_DEBUG_OBJECT (rtpceltpay, "bytes for size %u", ssize); /* calculate what the new size and duration would be of the packet */ payload_len = ssize + map.size + rtpceltpay->bytes + rtpceltpay->sbytes; if (rtpceltpay->qduration != -1 && duration != -1) packet_dur = rtpceltpay->qduration + duration; else packet_dur = 0; packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0); if (gst_rtp_base_payload_is_filled (basepayload, packet_len, packet_dur)) { /* size or duration would overflow the packet, flush the queued data */ ret = gst_rtp_celt_pay_flush_queued (rtpceltpay); } /* queue the packet */ gst_rtp_celt_pay_add_queued (rtpceltpay, buffer, ssize, map.size, duration); done: rtpceltpay->packet++; return ret; /* ERRORS */ cleanup: { gst_buffer_unmap (buffer, &map); goto done; } parse_error: { GST_ELEMENT_ERROR (rtpceltpay, STREAM, DECODE, (NULL), ("Error parsing first identification packet.")); gst_buffer_unmap (buffer, &map); return GST_FLOW_ERROR; } }
static GstFlowReturn gst_visual_chain (GstPad * pad, GstBuffer * buffer) { GstBuffer *outbuf = NULL; guint i; GstVisual *visual = GST_VISUAL (gst_pad_get_parent (pad)); GstFlowReturn ret = GST_FLOW_OK; guint avail; GST_DEBUG_OBJECT (visual, "chain function called"); /* If we don't have an output format yet, preallocate a buffer to try and * set one */ if (GST_PAD_CAPS (visual->srcpad) == NULL) { ret = get_buffer (visual, &outbuf); if (ret != GST_FLOW_OK) { gst_buffer_unref (buffer); goto beach; } } /* resync on DISCONT */ if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) { gst_adapter_clear (visual->adapter); } GST_DEBUG_OBJECT (visual, "Input buffer has %d samples, time=%" G_GUINT64_FORMAT, GST_BUFFER_SIZE (buffer) / visual->bps, GST_BUFFER_TIMESTAMP (buffer)); gst_adapter_push (visual->adapter, buffer); while (TRUE) { gboolean need_skip; const guint16 *data; guint64 dist, timestamp; GST_DEBUG_OBJECT (visual, "processing buffer"); avail = gst_adapter_available (visual->adapter); GST_DEBUG_OBJECT (visual, "avail now %u", avail); /* we need at least 512 samples */ if (avail < 512 * visual->bps) break; /* we need at least enough samples to make one frame */ if (avail < visual->spf * visual->bps) break; /* get timestamp of the current adapter byte */ timestamp = gst_adapter_prev_timestamp (visual->adapter, &dist); if (GST_CLOCK_TIME_IS_VALID (timestamp)) { /* convert bytes to time */ dist /= visual->bps; timestamp += gst_util_uint64_scale_int (dist, GST_SECOND, visual->rate); } if (timestamp != -1) { gint64 qostime; /* QoS is done on running time */ qostime = gst_segment_to_running_time (&visual->segment, GST_FORMAT_TIME, timestamp); GST_OBJECT_LOCK (visual); /* check for QoS, don't compute buffers that are known to be late */ need_skip = visual->earliest_time != -1 && qostime <= visual->earliest_time; GST_OBJECT_UNLOCK (visual); if (need_skip) { GST_WARNING_OBJECT (visual, "QoS: skip ts: %" GST_TIME_FORMAT ", earliest: %" GST_TIME_FORMAT, GST_TIME_ARGS (qostime), GST_TIME_ARGS (visual->earliest_time)); goto skip; } } /* Read 512 samples per channel */ data = (const guint16 *) gst_adapter_peek (visual->adapter, 512 * visual->bps); #if defined(VISUAL_API_VERSION) && VISUAL_API_VERSION >= 4000 && VISUAL_API_VERSION < 5000 { VisBuffer *lbuf, *rbuf; guint16 ldata[512], rdata[512]; VisAudioSampleRateType rate; lbuf = visual_buffer_new_with_buffer (ldata, sizeof (ldata), NULL); rbuf = visual_buffer_new_with_buffer (rdata, sizeof (rdata), NULL); if (visual->channels == 2) { for (i = 0; i < 512; i++) { ldata[i] = *data++; rdata[i] = *data++; } } else { for (i = 0; i < 512; i++) { ldata[i] = *data; rdata[i] = *data++; } } switch (visual->rate) { case 8000: rate = VISUAL_AUDIO_SAMPLE_RATE_8000; break; case 11250: rate = VISUAL_AUDIO_SAMPLE_RATE_11250; break; case 22500: rate = VISUAL_AUDIO_SAMPLE_RATE_22500; break; case 32000: rate = VISUAL_AUDIO_SAMPLE_RATE_32000; break; case 44100: rate = VISUAL_AUDIO_SAMPLE_RATE_44100; break; case 48000: rate = VISUAL_AUDIO_SAMPLE_RATE_48000; break; case 96000: rate = VISUAL_AUDIO_SAMPLE_RATE_96000; break; default: visual_object_unref (VISUAL_OBJECT (lbuf)); visual_object_unref (VISUAL_OBJECT (rbuf)); GST_ERROR_OBJECT (visual, "unsupported rate %d", visual->rate); ret = GST_FLOW_ERROR; goto beach; break; } visual_audio_samplepool_input_channel (visual->audio->samplepool, lbuf, rate, VISUAL_AUDIO_SAMPLE_FORMAT_S16, (char *) VISUAL_AUDIO_CHANNEL_LEFT); visual_audio_samplepool_input_channel (visual->audio->samplepool, rbuf, rate, VISUAL_AUDIO_SAMPLE_FORMAT_S16, (char *) VISUAL_AUDIO_CHANNEL_RIGHT); visual_object_unref (VISUAL_OBJECT (lbuf)); visual_object_unref (VISUAL_OBJECT (rbuf)); } #else if (visual->channels == 2) { for (i = 0; i < 512; i++) { visual->audio->plugpcm[0][i] = *data++; visual->audio->plugpcm[1][i] = *data++; } } else { for (i = 0; i < 512; i++) { visual->audio->plugpcm[0][i] = *data; visual->audio->plugpcm[1][i] = *data++; } } #endif /* alloc a buffer if we don't have one yet, this happens * when we pushed a buffer in this while loop before */ if (outbuf == NULL) { ret = get_buffer (visual, &outbuf); if (ret != GST_FLOW_OK) { goto beach; } } visual_video_set_buffer (visual->video, GST_BUFFER_DATA (outbuf)); visual_audio_analyze (visual->audio); visual_actor_run (visual->actor, visual->audio); visual_video_set_buffer (visual->video, NULL); GST_DEBUG_OBJECT (visual, "rendered one frame"); GST_BUFFER_TIMESTAMP (outbuf) = timestamp; GST_BUFFER_DURATION (outbuf) = visual->duration; ret = gst_pad_push (visual->srcpad, outbuf); outbuf = NULL; skip: GST_DEBUG_OBJECT (visual, "finished frame, flushing %u samples from input", visual->spf); /* Flush out the number of samples per frame */ gst_adapter_flush (visual->adapter, visual->spf * visual->bps); /* quit the loop if something was wrong */ if (ret != GST_FLOW_OK) break; } beach: if (outbuf != NULL) gst_buffer_unref (outbuf); gst_object_unref (visual); return ret; }
static void gst_frame_store_task (GstPad *pad) { GstFrameStore *fs; GstBuffer *buffer; GstEvent *event = NULL; fs = GST_FRAME_STORE (gst_pad_get_parent (pad)); GST_DEBUG("task"); g_mutex_lock (fs->lock); while(1) { if (fs->stepping == FALSE || (fs->frame_number != fs->pushed_frame_number)) { buffer = gst_frame_store_get_frame (fs, fs->frame_number); } if (buffer) break; g_cond_wait (fs->cond, fs->lock); } if (fs->need_newsegment) { GstClock *clock; GstClockTime now; GstClockTime stream_time; clock = GST_ELEMENT_CLOCK (fs); if (clock == NULL) { now = 0; stream_time = 0; } else { now = gst_clock_get_time (GST_ELEMENT_CLOCK (fs)); stream_time = now - GST_ELEMENT(fs)->base_time; } GST_ERROR("now %lld buffer %lld stream_time %lld", now, GST_BUFFER_TIMESTAMP(buffer), stream_time); stream_time = GST_SECOND*10; event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP(buffer), -1, stream_time); fs->need_newsegment = FALSE; } if (fs->stepping) { buffer = gst_buffer_make_metadata_writable (buffer); GST_BUFFER_TIMESTAMP(buffer) = -1; GST_BUFFER_DURATION(buffer) = -1; } fs->pushed_frame_number = fs->frame_number; if (!fs->stepping) { fs->frame_number++; } if (fs->frame_number + 1 >= fs->range_offset + fs->range_size) { gst_frame_store_advance (fs); } g_mutex_unlock (fs->lock); if (event) { gst_pad_push_event (fs->srcpad, event); } gst_pad_push (fs->srcpad, buffer); GST_DEBUG("task done"); gst_object_unref (fs); }
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 GstFlowReturn gst_decklink_video_sink_prepare (GstBaseSink * bsink, GstBuffer * buffer) { GstDecklinkVideoSink *self = GST_DECKLINK_VIDEO_SINK_CAST (bsink); GstVideoFrame vframe; IDeckLinkMutableVideoFrame *frame; guint8 *outdata, *indata; GstFlowReturn flow_ret; HRESULT ret; GstClockTime timestamp, duration; GstClockTime running_time, running_time_duration; gint i; GstClock *clock; GST_DEBUG_OBJECT (self, "Preparing buffer %p", buffer); // FIXME: Handle no timestamps if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) { return GST_FLOW_ERROR; } timestamp = GST_BUFFER_TIMESTAMP (buffer); duration = GST_BUFFER_DURATION (buffer); if (duration == GST_CLOCK_TIME_NONE) { duration = gst_util_uint64_scale_int (GST_SECOND, self->info.fps_d, self->info.fps_n); } running_time = gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment, GST_FORMAT_TIME, timestamp); running_time_duration = gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment, GST_FORMAT_TIME, timestamp + duration) - running_time; // FIXME: https://bugzilla.gnome.org/show_bug.cgi?id=742916 // We need to drop late buffers here immediately instead of // potentially overflowing the internal queue of the hardware clock = gst_element_get_clock (GST_ELEMENT_CAST (self)); if (clock) { GstClockTime clock_running_time, base_time, clock_time, latency, max_lateness; base_time = gst_element_get_base_time (GST_ELEMENT_CAST (self)); clock_time = gst_clock_get_time (clock); if (base_time != GST_CLOCK_TIME_NONE && clock_time != GST_CLOCK_TIME_NONE) { clock_running_time = clock_time - base_time; latency = gst_base_sink_get_latency (GST_BASE_SINK_CAST (self)); max_lateness = gst_base_sink_get_max_lateness (GST_BASE_SINK_CAST (self)); if (clock_running_time > running_time + running_time_duration + latency + max_lateness) { GST_DEBUG_OBJECT (self, "Late buffer: %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT, GST_TIME_ARGS (clock_running_time), GST_TIME_ARGS (running_time + running_time_duration)); if (self->last_render_time == GST_CLOCK_TIME_NONE || (self->last_render_time < clock_running_time && clock_running_time - self->last_render_time >= GST_SECOND)) { GST_DEBUG_OBJECT (self, "Rendering frame nonetheless because we had none for more than 1s"); running_time = clock_running_time; running_time_duration = 0; } else { GST_WARNING_OBJECT (self, "Dropping frame"); gst_object_unref (clock); return GST_FLOW_OK; } } } gst_object_unref (clock); } self->last_render_time = running_time; ret = self->output->output->CreateVideoFrame (self->info.width, self->info.height, self->info.stride[0], bmdFormat8BitYUV, bmdFrameFlagDefault, &frame); if (ret != S_OK) { GST_ELEMENT_ERROR (self, STREAM, FAILED, (NULL), ("Failed to create video frame: 0x%08x", ret)); return GST_FLOW_ERROR; } if (!gst_video_frame_map (&vframe, &self->info, buffer, GST_MAP_READ)) { GST_ERROR_OBJECT (self, "Failed to map video frame"); flow_ret = GST_FLOW_ERROR; goto out; } frame->GetBytes ((void **) &outdata); indata = (guint8 *) GST_VIDEO_FRAME_PLANE_DATA (&vframe, 0); for (i = 0; i < self->info.height; i++) { memcpy (outdata, indata, GST_VIDEO_FRAME_WIDTH (&vframe) * 2); indata += GST_VIDEO_FRAME_PLANE_STRIDE (&vframe, 0); outdata += frame->GetRowBytes (); } gst_video_frame_unmap (&vframe); convert_to_internal_clock (self, &running_time, &running_time_duration); GST_LOG_OBJECT (self, "Scheduling video frame %p at %" GST_TIME_FORMAT " with duration %" GST_TIME_FORMAT, frame, GST_TIME_ARGS (running_time), GST_TIME_ARGS (running_time_duration)); ret = self->output->output->ScheduleVideoFrame (frame, running_time, running_time_duration, GST_SECOND); if (ret != S_OK) { GST_ELEMENT_ERROR (self, STREAM, FAILED, (NULL), ("Failed to schedule frame: 0x%08x", ret)); flow_ret = GST_FLOW_ERROR; goto out; } flow_ret = GST_FLOW_OK; out: frame->Release (); return flow_ret; }
static GstFlowReturn celt_dec_chain_parse_data (GstCeltDec * dec, GstBuffer * buf, GstClockTime timestamp, GstClockTime duration) { GstFlowReturn res = GST_FLOW_OK; gint size; guint8 *data; GstBuffer *outbuf; gint16 *out_data; gint error = CELT_OK; if (timestamp != -1) { dec->segment.last_stop = timestamp; dec->granulepos = -1; } if (buf) { data = GST_BUFFER_DATA (buf); size = GST_BUFFER_SIZE (buf); GST_DEBUG_OBJECT (dec, "received buffer of size %u", size); if (!GST_BUFFER_TIMESTAMP_IS_VALID (buf) && GST_BUFFER_OFFSET_END_IS_VALID (buf)) { dec->granulepos = GST_BUFFER_OFFSET_END (buf); GST_DEBUG_OBJECT (dec, "Taking granulepos from upstream: %" G_GUINT64_FORMAT, dec->granulepos); } /* copy timestamp */ } else { /* concealment data, pass NULL as the bits parameters */ GST_DEBUG_OBJECT (dec, "creating concealment data"); data = NULL; size = 0; } res = gst_pad_alloc_buffer_and_set_caps (dec->srcpad, GST_BUFFER_OFFSET_NONE, dec->frame_size * dec->header.nb_channels * 2, GST_PAD_CAPS (dec->srcpad), &outbuf); if (res != GST_FLOW_OK) { GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res)); return res; } out_data = (gint16 *) GST_BUFFER_DATA (outbuf); GST_LOG_OBJECT (dec, "decoding frame"); error = celt_decode (dec->state, data, size, out_data); if (error != CELT_OK) { GST_WARNING_OBJECT (dec, "Decoding error: %d", error); return GST_FLOW_ERROR; } if (dec->granulepos == -1) { if (dec->segment.format != GST_FORMAT_TIME) { GST_WARNING_OBJECT (dec, "segment not initialized or not TIME format"); dec->granulepos = dec->frame_size; } else { dec->granulepos = gst_util_uint64_scale_int (dec->segment.last_stop, dec->header.sample_rate, GST_SECOND) + dec->frame_size; } GST_DEBUG_OBJECT (dec, "granulepos=%" G_GINT64_FORMAT, dec->granulepos); } GST_BUFFER_OFFSET (outbuf) = dec->granulepos - dec->frame_size; GST_BUFFER_OFFSET_END (outbuf) = dec->granulepos; GST_BUFFER_TIMESTAMP (outbuf) = gst_util_uint64_scale_int (dec->granulepos - dec->frame_size, GST_SECOND, dec->header.sample_rate); GST_BUFFER_DURATION (outbuf) = dec->frame_duration; dec->granulepos += dec->frame_size; dec->segment.last_stop += dec->frame_duration; GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), GST_TIME_ARGS (dec->frame_duration)); res = gst_pad_push (dec->srcpad, outbuf); if (res != GST_FLOW_OK) GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res)); return res; }
static GstFlowReturn gst_dvbsub_overlay_chain_video (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstDVBSubOverlay *overlay = GST_DVBSUB_OVERLAY (parent); GstFlowReturn ret = GST_FLOW_OK; gint64 start, stop; guint64 cstart, cstop; gboolean in_seg; GstClockTime vid_running_time, vid_running_time_end; if (GST_VIDEO_INFO_FORMAT (&overlay->info) == GST_VIDEO_FORMAT_UNKNOWN) return GST_FLOW_NOT_NEGOTIATED; if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) goto missing_timestamp; start = GST_BUFFER_TIMESTAMP (buffer); GST_LOG_OBJECT (overlay, "Video segment: %" GST_SEGMENT_FORMAT " --- Subtitle position: %" GST_TIME_FORMAT " --- BUFFER: ts=%" GST_TIME_FORMAT, &overlay->video_segment, GST_TIME_ARGS (overlay->subtitle_segment.position), GST_TIME_ARGS (start)); /* ignore buffers that are outside of the current segment */ if (!GST_BUFFER_DURATION_IS_VALID (buffer)) { stop = GST_CLOCK_TIME_NONE; } else { stop = start + GST_BUFFER_DURATION (buffer); } in_seg = gst_segment_clip (&overlay->video_segment, GST_FORMAT_TIME, start, stop, &cstart, &cstop); if (!in_seg) { GST_DEBUG_OBJECT (overlay, "Buffer outside configured segment -- dropping"); gst_buffer_unref (buffer); return GST_FLOW_OK; } buffer = gst_buffer_make_writable (buffer); GST_BUFFER_TIMESTAMP (buffer) = cstart; if (GST_BUFFER_DURATION_IS_VALID (buffer)) GST_BUFFER_DURATION (buffer) = cstop - cstart; vid_running_time = gst_segment_to_running_time (&overlay->video_segment, GST_FORMAT_TIME, cstart); if (GST_BUFFER_DURATION_IS_VALID (buffer)) vid_running_time_end = gst_segment_to_running_time (&overlay->video_segment, GST_FORMAT_TIME, cstop); else vid_running_time_end = vid_running_time; GST_DEBUG_OBJECT (overlay, "Video running time: %" GST_TIME_FORMAT, GST_TIME_ARGS (vid_running_time)); overlay->video_segment.position = GST_BUFFER_TIMESTAMP (buffer); g_mutex_lock (&overlay->dvbsub_mutex); if (!g_queue_is_empty (overlay->pending_subtitles)) { DVBSubtitles *tmp, *candidate = NULL; while (!g_queue_is_empty (overlay->pending_subtitles)) { tmp = g_queue_peek_head (overlay->pending_subtitles); if (tmp->pts > vid_running_time_end) { /* For a future video frame */ break; } else if (tmp->num_rects == 0) { /* Clear screen */ if (overlay->current_subtitle) dvb_subtitles_free (overlay->current_subtitle); overlay->current_subtitle = NULL; if (candidate) dvb_subtitles_free (candidate); candidate = NULL; g_queue_pop_head (overlay->pending_subtitles); dvb_subtitles_free (tmp); tmp = NULL; } else if (tmp->pts + tmp->page_time_out * GST_SECOND * ABS (overlay->subtitle_segment.rate) >= vid_running_time) { if (candidate) dvb_subtitles_free (candidate); candidate = tmp; g_queue_pop_head (overlay->pending_subtitles); } else { /* Too late */ dvb_subtitles_free (tmp); tmp = NULL; g_queue_pop_head (overlay->pending_subtitles); } } if (candidate) { GST_DEBUG_OBJECT (overlay, "Time to show the next subtitle page (%" GST_TIME_FORMAT " >= %" GST_TIME_FORMAT ") - it has %u regions", GST_TIME_ARGS (vid_running_time), GST_TIME_ARGS (candidate->pts), candidate->num_rects); dvb_subtitles_free (overlay->current_subtitle); overlay->current_subtitle = candidate; if (overlay->current_comp) gst_video_overlay_composition_unref (overlay->current_comp); overlay->current_comp = gst_dvbsub_overlay_subs_to_comp (overlay, overlay->current_subtitle); } } /* Check that we haven't hit the fallback timeout for current subtitle page */ if (overlay->current_subtitle && vid_running_time > (overlay->current_subtitle->pts + overlay->current_subtitle->page_time_out * GST_SECOND * ABS (overlay->subtitle_segment.rate))) { GST_INFO_OBJECT (overlay, "Subtitle page not redefined before fallback page_time_out of %u seconds (missed data?) - deleting current page", overlay->current_subtitle->page_time_out); dvb_subtitles_free (overlay->current_subtitle); overlay->current_subtitle = NULL; } /* Now render it */ if (g_atomic_int_get (&overlay->enable) && overlay->current_subtitle) { GstVideoFrame frame; g_assert (overlay->current_comp); if (overlay->attach_compo_to_buffer) { GST_DEBUG_OBJECT (overlay, "Attaching overlay image to video buffer"); gst_buffer_add_video_overlay_composition_meta (buffer, overlay->current_comp); } else { GST_DEBUG_OBJECT (overlay, "Blending overlay image to video buffer"); gst_video_frame_map (&frame, &overlay->info, buffer, GST_MAP_READWRITE); gst_video_overlay_composition_blend (overlay->current_comp, &frame); gst_video_frame_unmap (&frame); } } g_mutex_unlock (&overlay->dvbsub_mutex); ret = gst_pad_push (overlay->srcpad, buffer); return ret; missing_timestamp: { GST_WARNING_OBJECT (overlay, "video buffer without timestamp, discarding"); gst_buffer_unref (buffer); return GST_FLOW_OK; } }
static GstFlowReturn gst_decklink_video_sink_prepare (GstBaseSink * bsink, GstBuffer * buffer) { GstDecklinkVideoSink *self = GST_DECKLINK_VIDEO_SINK_CAST (bsink); GstVideoFrame vframe; IDeckLinkMutableVideoFrame *frame; guint8 *outdata, *indata; GstFlowReturn flow_ret; HRESULT ret; GstClockTime timestamp, duration; GstClockTime running_time, running_time_duration; GstClockTime latency, render_delay; GstClockTimeDiff ts_offset; gint i; GstDecklinkVideoFormat caps_format; BMDPixelFormat format; gint stride; GstVideoTimeCodeMeta *tc_meta; GST_DEBUG_OBJECT (self, "Preparing buffer %p", buffer); // FIXME: Handle no timestamps if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) { return GST_FLOW_ERROR; } caps_format = gst_decklink_type_from_video_format (self->info.finfo->format); format = gst_decklink_pixel_format_from_type (caps_format); timestamp = GST_BUFFER_TIMESTAMP (buffer); duration = GST_BUFFER_DURATION (buffer); if (duration == GST_CLOCK_TIME_NONE) { duration = gst_util_uint64_scale_int (GST_SECOND, self->info.fps_d, self->info.fps_n); } running_time = gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment, GST_FORMAT_TIME, timestamp); running_time_duration = gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment, GST_FORMAT_TIME, timestamp + duration) - running_time; /* See gst_base_sink_adjust_time() */ latency = gst_base_sink_get_latency (bsink); render_delay = gst_base_sink_get_render_delay (bsink); ts_offset = gst_base_sink_get_ts_offset (bsink); running_time += latency; if (ts_offset < 0) { ts_offset = -ts_offset; if ((GstClockTime) ts_offset < running_time) running_time -= ts_offset; else running_time = 0; } else { running_time += ts_offset; } if (running_time > render_delay) running_time -= render_delay; else running_time = 0; ret = self->output->output->CreateVideoFrame (self->info.width, self->info.height, self->info.stride[0], format, bmdFrameFlagDefault, &frame); if (ret != S_OK) { GST_ELEMENT_ERROR (self, STREAM, FAILED, (NULL), ("Failed to create video frame: 0x%08x", ret)); return GST_FLOW_ERROR; } if (!gst_video_frame_map (&vframe, &self->info, buffer, GST_MAP_READ)) { GST_ERROR_OBJECT (self, "Failed to map video frame"); flow_ret = GST_FLOW_ERROR; goto out; } frame->GetBytes ((void **) &outdata); indata = (guint8 *) GST_VIDEO_FRAME_PLANE_DATA (&vframe, 0); stride = MIN (GST_VIDEO_FRAME_PLANE_STRIDE (&vframe, 0), frame->GetRowBytes()); for (i = 0; i < self->info.height; i++) { memcpy (outdata, indata, stride); indata += GST_VIDEO_FRAME_PLANE_STRIDE (&vframe, 0); outdata += frame->GetRowBytes (); } gst_video_frame_unmap (&vframe); tc_meta = gst_buffer_get_video_time_code_meta (buffer); if (tc_meta) { BMDTimecodeFlags bflags = (BMDTimecodeFlags) 0; gchar *tc_str; if (((GstVideoTimeCodeFlags) (tc_meta->tc. config.flags)) & GST_VIDEO_TIME_CODE_FLAGS_DROP_FRAME) bflags = (BMDTimecodeFlags) (bflags | bmdTimecodeIsDropFrame); else bflags = (BMDTimecodeFlags) (bflags | bmdTimecodeFlagDefault); if (tc_meta->tc.field_count == 2) bflags = (BMDTimecodeFlags) (bflags | bmdTimecodeFieldMark); tc_str = gst_video_time_code_to_string (&tc_meta->tc); ret = frame->SetTimecodeFromComponents (self->timecode_format, (uint8_t) tc_meta->tc.hours, (uint8_t) tc_meta->tc.minutes, (uint8_t) tc_meta->tc.seconds, (uint8_t) tc_meta->tc.frames, bflags); if (ret != S_OK) { GST_ERROR_OBJECT (self, "Failed to set timecode %s to video frame: 0x%08x", tc_str, ret); flow_ret = GST_FLOW_ERROR; g_free (tc_str); goto out; } GST_DEBUG_OBJECT (self, "Set frame timecode to %s", tc_str); g_free (tc_str); } convert_to_internal_clock (self, &running_time, &running_time_duration); if (!self->output->started) { GST_LOG_OBJECT (self, "Showing video frame synchronously because PAUSED"); ret = self->output->output->DisplayVideoFrameSync (frame); if (ret != S_OK) { GST_ELEMENT_WARNING (self, STREAM, FAILED, (NULL), ("Failed to show video frame synchronously: 0x%08x", ret)); ret = S_OK; } } GST_LOG_OBJECT (self, "Scheduling video frame %p at %" GST_TIME_FORMAT " with duration %" GST_TIME_FORMAT, frame, GST_TIME_ARGS (running_time), GST_TIME_ARGS (running_time_duration)); ret = self->output->output->ScheduleVideoFrame (frame, running_time, running_time_duration, GST_SECOND); if (ret != S_OK) { GST_ELEMENT_ERROR (self, STREAM, FAILED, (NULL), ("Failed to schedule frame: 0x%08x", ret)); flow_ret = GST_FLOW_ERROR; goto out; } flow_ret = GST_FLOW_OK; out: frame->Release (); return flow_ret; }
static GstFlowReturn gst_gdiscreencapsrc_create (GstPushSrc * push_src, GstBuffer ** buf) { GstGDIScreenCapSrc *src = GST_GDISCREENCAPSRC (push_src); GstBuffer *new_buf; GstFlowReturn res; gint new_buf_size; GstClock *clock; GstClockTime time; GstClockTime base_time; if (G_UNLIKELY (!src->info.bmiHeader.biWidth || !src->info.bmiHeader.biHeight)) { GST_ELEMENT_ERROR (src, CORE, NEGOTIATION, (NULL), ("format wasn't negotiated before create function")); return GST_FLOW_NOT_NEGOTIATED; } else if (G_UNLIKELY (src->rate_numerator == 0 && src->frames == 1)) { GST_DEBUG_OBJECT (src, "eos: 0 framerate, frame %d", (gint) src->frames); return GST_FLOW_UNEXPECTED; } new_buf_size = GST_ROUND_UP_4 (src->info.bmiHeader.biWidth * 3) * (-src->info.bmiHeader.biHeight); GST_LOG_OBJECT (src, "creating buffer of %lu bytes with %dx%d image for frame %d", new_buf_size, src->info.bmiHeader.biWidth, -src->info.bmiHeader.biHeight, (gint) src->frames); res = gst_pad_alloc_buffer_and_set_caps (GST_BASE_SRC_PAD (src), GST_BUFFER_OFFSET_NONE, new_buf_size, GST_PAD_CAPS (GST_BASE_SRC_PAD (push_src)), &new_buf); if (res != GST_FLOW_OK) { GST_DEBUG_OBJECT (src, "could not allocate buffer, reason %s", gst_flow_get_name (res)); return res; } clock = gst_element_get_clock (GST_ELEMENT (src)); if (clock) { /* Calculate sync time. */ GstClockTime frame_time = gst_util_uint64_scale_int (src->frames * GST_SECOND, src->rate_denominator, src->rate_numerator); time = gst_clock_get_time (clock); base_time = gst_element_get_base_time (GST_ELEMENT (src)); GST_BUFFER_TIMESTAMP (new_buf) = MAX (time - base_time, frame_time); } else { GST_BUFFER_TIMESTAMP (new_buf) = GST_CLOCK_TIME_NONE; } /* Do screen capture and put it into buffer... */ gst_gdiscreencapsrc_screen_capture (src, new_buf); if (src->rate_numerator) { GST_BUFFER_DURATION (new_buf) = gst_util_uint64_scale_int (GST_SECOND, src->rate_denominator, src->rate_numerator); if (clock) { GST_BUFFER_DURATION (new_buf) = MAX (GST_BUFFER_DURATION (new_buf), gst_clock_get_time (clock) - time); } } else { /* NONE means forever */ GST_BUFFER_DURATION (new_buf) = GST_CLOCK_TIME_NONE; } GST_BUFFER_OFFSET (new_buf) = src->frames; src->frames++; GST_BUFFER_OFFSET_END (new_buf) = src->frames; gst_object_unref (clock); *buf = new_buf; return GST_FLOW_OK; }
static GstFlowReturn gst_rsvg_dec_chain (GstPad * pad, GstBuffer * buffer) { GstRsvgDec *rsvg = GST_RSVG_DEC (GST_PAD_PARENT (pad)); gboolean completed = FALSE; const guint8 *data; guint size; gboolean ret = GST_FLOW_OK; /* first_timestamp is used slightly differently where a framerate is given or not. If there is a frame rate, it will be used as a base. If there is not, it will be used to keep track of the timestamp of the first buffer, to be used as the timestamp of the output buffer. When a buffer is output, first timestamp will resync to the next buffer's timestamp. */ if (rsvg->first_timestamp == GST_CLOCK_TIME_NONE) { if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) rsvg->first_timestamp = GST_BUFFER_TIMESTAMP (buffer); else if (rsvg->fps_n != 0) rsvg->first_timestamp = 0; } gst_adapter_push (rsvg->adapter, buffer); size = gst_adapter_available (rsvg->adapter); /* "<svg></svg>" */ while (size >= 5 + 6 && ret == GST_FLOW_OK) { guint i; data = gst_adapter_peek (rsvg->adapter, size); for (i = size - 6; i >= 5; i--) { if (memcmp (data + i, "</svg>", 6) == 0) { completed = TRUE; size = i + 6; break; } } if (completed) { GstBuffer *outbuf = NULL; GST_LOG_OBJECT (rsvg, "have complete svg of %u bytes", size); data = gst_adapter_peek (rsvg->adapter, size); ret = gst_rsvg_decode_image (rsvg, data, size, &outbuf); if (ret != GST_FLOW_OK) break; if (rsvg->first_timestamp != GST_CLOCK_TIME_NONE) { GST_BUFFER_TIMESTAMP (outbuf) = rsvg->first_timestamp; GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE; if (GST_BUFFER_DURATION_IS_VALID (buffer)) { GstClockTime end = GST_BUFFER_TIMESTAMP_IS_VALID (buffer) ? GST_BUFFER_TIMESTAMP (buffer) : rsvg->first_timestamp; end += GST_BUFFER_DURATION (buffer); GST_BUFFER_DURATION (outbuf) = end - GST_BUFFER_TIMESTAMP (outbuf); } if (rsvg->fps_n == 0) { rsvg->first_timestamp = GST_CLOCK_TIME_NONE; } else { GST_BUFFER_DURATION (outbuf) = gst_util_uint64_scale (rsvg->frame_count, rsvg->fps_d, rsvg->fps_n * GST_SECOND); } } else if (rsvg->fps_n != 0) { GST_BUFFER_TIMESTAMP (outbuf) = rsvg->first_timestamp + gst_util_uint64_scale (rsvg->frame_count, rsvg->fps_d, rsvg->fps_n * GST_SECOND); GST_BUFFER_DURATION (outbuf) = gst_util_uint64_scale (rsvg->frame_count, rsvg->fps_d, rsvg->fps_n * GST_SECOND); } else { GST_BUFFER_TIMESTAMP (outbuf) = rsvg->first_timestamp; GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE; } rsvg->frame_count++; if (rsvg->need_newsegment) { gst_pad_push_event (rsvg->srcpad, gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0)); rsvg->need_newsegment = FALSE; } if (rsvg->pending_events) { GList *l; for (l = rsvg->pending_events; l; l = l->next) gst_pad_push_event (rsvg->srcpad, l->data); g_list_free (rsvg->pending_events); rsvg->pending_events = NULL; } if (rsvg->pending_tags) { gst_element_found_tags (GST_ELEMENT_CAST (rsvg), rsvg->pending_tags); rsvg->pending_tags = NULL; } GST_LOG_OBJECT (rsvg, "image rendered okay"); ret = gst_pad_push (rsvg->srcpad, outbuf); if (ret != GST_FLOW_OK) break; gst_adapter_flush (rsvg->adapter, size); size = gst_adapter_available (rsvg->adapter); continue; } else { break; } } return GST_FLOW_OK; }