static GstFlowReturn gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer) { GstDtsDec *dts; gint channels, i, num_blocks; gboolean need_renegotiation = FALSE; guint8 *data; gsize size; GstMapInfo map; gint chans; gint length = 0, flags, sample_rate, bit_rate, frame_length; GstFlowReturn result = GST_FLOW_OK; GstBuffer *outbuf; dts = GST_DTSDEC (bdec); /* no fancy draining */ if (G_UNLIKELY (!buffer)) return GST_FLOW_OK; /* parsed stuff already, so this should work out fine */ gst_buffer_map (buffer, &map, GST_MAP_READ); data = map.data; size = map.size; g_assert (size >= 7); bit_rate = dts->bit_rate; sample_rate = dts->sample_rate; flags = 0; length = dca_syncinfo (dts->state, data, &flags, &sample_rate, &bit_rate, &frame_length); g_assert (length == size); if (flags != dts->prev_flags) { dts->prev_flags = flags; dts->flag_update = TRUE; } /* go over stream properties, renegotiate or update streaminfo if needed */ if (dts->sample_rate != sample_rate) { need_renegotiation = TRUE; dts->sample_rate = sample_rate; } if (flags) { dts->stream_channels = flags & (DCA_CHANNEL_MASK | DCA_LFE); } if (bit_rate != dts->bit_rate) { dts->bit_rate = bit_rate; gst_dtsdec_update_streaminfo (dts); } /* If we haven't had an explicit number of channels chosen through properties * at this point, choose what to downmix to now, based on what the peer will * accept - this allows a52dec to do downmixing in preference to a * downstream element such as audioconvert. * FIXME: Add the property back in for forcing output channels. */ if (dts->request_channels != DCA_CHANNEL) { flags = dts->request_channels; } else if (dts->flag_update) { GstCaps *caps; dts->flag_update = FALSE; caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dts)); if (caps && gst_caps_get_size (caps) > 0) { GstCaps *copy = gst_caps_copy_nth (caps, 0); GstStructure *structure = gst_caps_get_structure (copy, 0); gint channels; const int dts_channels[6] = { DCA_MONO, DCA_STEREO, DCA_STEREO | DCA_LFE, DCA_2F2R, DCA_2F2R | DCA_LFE, DCA_3F2R | DCA_LFE, }; /* Prefer the original number of channels, but fixate to something * preferred (first in the caps) downstream if possible. */ gst_structure_fixate_field_nearest_int (structure, "channels", flags ? gst_dtsdec_channels (flags, NULL) : 6); gst_structure_get_int (structure, "channels", &channels); if (channels <= 6) flags = dts_channels[channels - 1]; else flags = dts_channels[5]; gst_caps_unref (copy); } else if (flags) { flags = dts->stream_channels; } else { flags = DCA_3F2R | DCA_LFE; } if (caps) gst_caps_unref (caps); } else { flags = dts->using_channels; } /* process */ flags |= DCA_ADJUST_LEVEL; dts->level = 1; if (dca_frame (dts->state, data, &flags, &dts->level, dts->bias)) { gst_buffer_unmap (buffer, &map); GST_AUDIO_DECODER_ERROR (dts, 1, STREAM, DECODE, (NULL), ("dts_frame error"), result); goto exit; } gst_buffer_unmap (buffer, &map); channels = flags & (DCA_CHANNEL_MASK | DCA_LFE); if (dts->using_channels != channels) { need_renegotiation = TRUE; dts->using_channels = channels; } /* negotiate if required */ if (need_renegotiation) { GST_DEBUG_OBJECT (dts, "dtsdec: sample_rate:%d stream_chans:0x%x using_chans:0x%x", dts->sample_rate, dts->stream_channels, dts->using_channels); if (!gst_dtsdec_renegotiate (dts)) goto failed_negotiation; } if (dts->dynamic_range_compression == FALSE) { dca_dynrng (dts->state, NULL, NULL); } flags &= (DCA_CHANNEL_MASK | DCA_LFE); chans = gst_dtsdec_channels (flags, NULL); if (!chans) goto invalid_flags; /* handle decoded data, one block is 256 samples */ num_blocks = dca_blocks_num (dts->state); outbuf = gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks); gst_buffer_map (outbuf, &map, GST_MAP_WRITE); data = map.data; size = map.size; { guint8 *ptr = data; for (i = 0; i < num_blocks; i++) { if (dca_block (dts->state)) { /* also marks discont */ GST_AUDIO_DECODER_ERROR (dts, 1, STREAM, DECODE, (NULL), ("error decoding block %d", i), result); if (result != GST_FLOW_OK) goto exit; } else { gint n, c; gint *reorder_map = dts->channel_reorder_map; for (n = 0; n < 256; n++) { for (c = 0; c < chans; c++) { ((sample_t *) ptr)[n * chans + reorder_map[c]] = dts->samples[c * 256 + n]; } } } ptr += 256 * chans * (SAMPLE_WIDTH / 8); } } gst_buffer_unmap (outbuf, &map); result = gst_audio_decoder_finish_frame (bdec, outbuf, 1); exit: return result; /* ERRORS */ failed_negotiation: { GST_ELEMENT_ERROR (dts, CORE, NEGOTIATION, (NULL), (NULL)); return GST_FLOW_ERROR; } invalid_flags: { GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL), ("Invalid channel flags: %d", flags)); return GST_FLOW_ERROR; } }
static GstFlowReturn gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data, guint length, gint flags, gint sample_rate, gint bit_rate) { gboolean need_renegotiation = FALSE; gint channels, num_blocks; GstBuffer *out; gint i, s, c, num_c; sample_t *samples; GstFlowReturn result = GST_FLOW_OK; /* go over stream properties, update caps/streaminfo if needed */ if (dts->sample_rate != sample_rate) { need_renegotiation = TRUE; dts->sample_rate = sample_rate; } dts->stream_channels = flags; if (bit_rate != dts->bit_rate) { dts->bit_rate = bit_rate; gst_dtsdec_update_streaminfo (dts); } if (dts->request_channels == DTS_CHANNEL) { GstCaps *caps; caps = gst_pad_get_allowed_caps (dts->srcpad); if (caps && gst_caps_get_size (caps) > 0) { GstCaps *copy = gst_caps_copy_nth (caps, 0); GstStructure *structure = gst_caps_get_structure (copy, 0); gint channels; const int dts_channels[6] = { DTS_MONO, DTS_STEREO, DTS_STEREO | DTS_LFE, DTS_2F2R, DTS_2F2R | DTS_LFE, DTS_3F2R | DTS_LFE, }; /* Prefer the original number of channels, but fixate to something * preferred (first in the caps) downstream if possible. */ gst_structure_fixate_field_nearest_int (structure, "channels", flags ? gst_dtsdec_channels (flags, NULL) : 6); gst_structure_get_int (structure, "channels", &channels); if (channels <= 6) dts->request_channels = dts_channels[channels - 1]; else dts->request_channels = dts_channels[5]; gst_caps_unref (copy); } else if (flags) { dts->request_channels = dts->stream_channels; } else { dts->request_channels = DTS_3F2R | DTS_LFE; } if (caps) gst_caps_unref (caps); } /* process */ flags = dts->request_channels | DTS_ADJUST_LEVEL; dts->level = 1; if (dts_frame (dts->state, data, &flags, &dts->level, dts->bias)) { GST_WARNING ("dts_frame error"); return GST_FLOW_OK; } channels = flags & (DTS_CHANNEL_MASK | DTS_LFE); if (dts->using_channels != channels) { need_renegotiation = TRUE; dts->using_channels = channels; } if (need_renegotiation == TRUE) { GST_DEBUG ("dtsdec: sample_rate:%d stream_chans:0x%x using_chans:0x%x", dts->sample_rate, dts->stream_channels, dts->using_channels); if (!gst_dtsdec_renegotiate (dts)) { GST_ELEMENT_ERROR (dts, CORE, NEGOTIATION, (NULL), (NULL)); return GST_FLOW_ERROR; } } if (dts->dynamic_range_compression == FALSE) { dts_dynrng (dts->state, NULL, NULL); } /* handle decoded data, one block is 256 samples */ num_blocks = dts_blocks_num (dts->state); for (i = 0; i < num_blocks; i++) { if (dts_block (dts->state)) { GST_WARNING ("dts_block error %d", i); continue; } samples = dts_samples (dts->state); num_c = gst_dtsdec_channels (dts->using_channels, NULL); result = gst_pad_alloc_buffer_and_set_caps (dts->srcpad, 0, (SAMPLE_WIDTH / 8) * 256 * num_c, GST_PAD_CAPS (dts->srcpad), &out); if (result != GST_FLOW_OK) break; GST_BUFFER_TIMESTAMP (out) = dts->current_ts; GST_BUFFER_DURATION (out) = GST_SECOND * 256 / dts->sample_rate; dts->current_ts += GST_BUFFER_DURATION (out); /* libdts returns buffers in 256-sample-blocks per channel, * we want interleaved. And we need to copy anyway... */ data = GST_BUFFER_DATA (out); for (s = 0; s < 256; s++) { for (c = 0; c < num_c; c++) { *(sample_t *) data = samples[s + c * 256]; data += (SAMPLE_WIDTH / 8); } } /* push on */ result = gst_pad_push (dts->srcpad, out); if (result != GST_FLOW_OK) break; } return result; }