static gboolean
adpcmdec_set_format (GstAudioDecoder * bdec, GstCaps * in_caps)
{
ADPCMDec *dec = (ADPCMDec *) (bdec);
GstStructure *structure = gst_caps_get_structure (in_caps, 0);
const gchar *layout;
GstAudioInfo info;
layout = gst_structure_get_string (structure, "layout");
if (!layout)
return FALSE;
if (g_str_equal (layout, "microsoft"))
dec->layout = LAYOUT_ADPCM_MICROSOFT;
else if (g_str_equal (layout, "dvi"))
dec->layout = LAYOUT_ADPCM_DVI;
else
return FALSE;
if (!gst_structure_get_int (structure, "block_align", &dec->blocksize))
dec->blocksize = -1; /* Not provided */
if (!gst_structure_get_int (structure, "rate", &dec->rate))
return FALSE;
if (!gst_structure_get_int (structure, "channels", &dec->channels))
return FALSE;
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, dec->rate,
dec->channels, NULL);
gst_audio_decoder_set_output_format (bdec, &info);
return TRUE;
}
void
gst_audio_parse_update_frame_size (GstAudioParse * ap)
{
gint framesize, width;
switch (ap->format) {
case GST_AUDIO_PARSE_FORMAT_ALAW:
case GST_AUDIO_PARSE_FORMAT_MULAW:
width = 8;
break;
case GST_AUDIO_PARSE_FORMAT_RAW:
default:
{
GstAudioInfo info;
gst_audio_info_init (&info);
/* rate, etc do not really matter here */
gst_audio_info_set_format (&info, ap->raw_format, 44100, ap->channels,
NULL);
width = GST_AUDIO_INFO_WIDTH (&info);
break;
}
}
framesize = (width / 8) * ap->channels;
gst_raw_parse_set_framesize (GST_RAW_PARSE (ap), framesize);
}
static gboolean
gst_dtsdec_renegotiate (GstDtsDec * dts)
{
gint channels;
gboolean result = FALSE;
GstAudioChannelPosition from[6], to[6];
GstAudioInfo info;
channels = gst_dtsdec_channels (dts->using_channels, from);
if (!channels)
goto done;
GST_INFO_OBJECT (dts, "dtsdec renegotiate, channels=%d, rate=%d",
channels, dts->sample_rate);
memcpy (to, from, sizeof (GstAudioChannelPosition) * channels);
gst_audio_channel_positions_to_valid_order (to, channels);
gst_audio_get_channel_reorder_map (channels, from, to,
dts->channel_reorder_map);
gst_audio_info_init (&info);
gst_audio_info_set_format (&info,
SAMPLE_TYPE, dts->sample_rate, channels, (channels > 1 ? to : NULL));
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dts), &info))
goto done;
result = TRUE;
done:
return result;
}
static void
gst_flac_dec_metadata_cb (const FLAC__StreamDecoder * decoder,
const FLAC__StreamMetadata * metadata, void *client_data)
{
GstFlacDec *flacdec = GST_FLAC_DEC (client_data);
GST_LOG_OBJECT (flacdec, "metadata type: %d", metadata->type);
switch (metadata->type) {
case FLAC__METADATA_TYPE_STREAMINFO:{
gint64 samples;
guint depth, width, gdepth;
samples = metadata->data.stream_info.total_samples;
flacdec->min_blocksize = metadata->data.stream_info.min_blocksize;
flacdec->max_blocksize = metadata->data.stream_info.max_blocksize;
flacdec->depth = depth = metadata->data.stream_info.bits_per_sample;
if (depth < 9) {
gdepth = width = 8;
} else if (depth < 17) {
gdepth = width = 16;
} else if (depth < 25) {
gdepth = 24;
width = 32;
} else {
gdepth = width = 32;
}
gst_audio_info_set_format (&flacdec->info,
gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, width, gdepth),
metadata->data.stream_info.sample_rate,
metadata->data.stream_info.channels, NULL);
memcpy (flacdec->info.position,
channel_positions[flacdec->info.channels - 1],
sizeof (GstAudioChannelPosition) * flacdec->info.channels);
gst_audio_channel_positions_to_valid_order (flacdec->info.position,
flacdec->info.channels);
/* Note: we create the inverse reordering map here */
gst_audio_get_channel_reorder_map (flacdec->info.channels,
flacdec->info.position, channel_positions[flacdec->info.channels - 1],
flacdec->channel_reorder_map);
GST_DEBUG_OBJECT (flacdec, "blocksize: min=%u, max=%u",
flacdec->min_blocksize, flacdec->max_blocksize);
GST_DEBUG_OBJECT (flacdec, "sample rate: %u, channels: %u",
flacdec->info.rate, flacdec->info.channels);
GST_DEBUG_OBJECT (flacdec, "depth: %u, width: %u", flacdec->depth,
flacdec->info.finfo->width);
GST_DEBUG_OBJECT (flacdec, "total samples = %" G_GINT64_FORMAT, samples);
break;
}
default:
break;
}
}
static GstFlowReturn
vorbis_handle_identification_packet (GstVorbisDec * vd)
{
GstAudioInfo info;
switch (vd->vi.channels) {
case 1:
case 2:
case 3:
case 4:
case 5:
case 6:
case 7:
case 8:
{
const GstAudioChannelPosition *pos;
pos = gst_vorbis_default_channel_positions[vd->vi.channels - 1];
gst_audio_info_set_format (&info, GST_VORBIS_AUDIO_FORMAT, vd->vi.rate,
vd->vi.channels, pos);
break;
}
default:{
GstAudioChannelPosition position[64];
gint i, max_pos = MAX (vd->vi.channels, 64);
GST_ELEMENT_WARNING (vd, STREAM, DECODE,
(NULL), ("Using NONE channel layout for more than 8 channels"));
for (i = 0; i < max_pos; i++)
position[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
gst_audio_info_set_format (&info, GST_VORBIS_AUDIO_FORMAT, vd->vi.rate,
vd->vi.channels, position);
break;
}
}
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (vd), &info);
vd->info = info;
/* select a copy_samples function, this way we can have specialized versions
* for mono/stereo and avoid the depth switch in tremor case */
vd->copy_samples = gst_vorbis_get_copy_sample_func (info.channels);
return GST_FLOW_OK;
}
static gboolean
gst_siren_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
GstAudioInfo info;
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16LE, 16000, 1, NULL);
return gst_audio_decoder_set_output_format (bdec, &info);
}
static gboolean
gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
{
GstGSMDec *gsmdec;
GstStructure *s;
gboolean ret = FALSE;
gint rate;
GstAudioInfo info;
gsmdec = GST_GSMDEC (dec);
s = gst_caps_get_structure (caps, 0);
if (s == NULL)
goto wrong_caps;
/* figure out if we deal with plain or MSGSM */
if (gst_structure_has_name (s, "audio/x-gsm"))
gsmdec->use_wav49 = 0;
else if (gst_structure_has_name (s, "audio/ms-gsm"))
gsmdec->use_wav49 = 1;
else
goto wrong_caps;
gsmdec->needed = 33;
if (!gst_structure_get_int (s, "rate", &rate)) {
GST_WARNING_OBJECT (gsmdec, "missing sample rate parameter from sink caps");
goto beach;
}
/* MSGSM needs different framing */
gsm_option (gsmdec->state, GSM_OPT_WAV49, &gsmdec->use_wav49);
/* Setting up src caps based on the input sample rate. */
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, rate, 1, NULL);
ret = gst_audio_decoder_set_output_format (dec, &info);
return ret;
/* ERRORS */
wrong_caps:
GST_ERROR_OBJECT (gsmdec, "invalid caps received");
beach:
return ret;
}
static gboolean
gst_amc_audio_dec_set_src_caps (GstAmcAudioDec * self, GstAmcFormat * format)
{
gint rate, channels;
guint32 channel_mask = 0;
GstAudioChannelPosition to[64];
GError *err = NULL;
if (!gst_amc_format_get_int (format, "sample-rate", &rate, &err) ||
!gst_amc_format_get_int (format, "channel-count", &channels, &err)) {
GST_ERROR_OBJECT (self, "Failed to get output format metadata: %s",
err->message);
g_clear_error (&err);
return FALSE;
}
if (rate == 0 || channels == 0) {
GST_ERROR_OBJECT (self, "Rate or channels not set");
return FALSE;
}
/* Not always present */
if (gst_amc_format_contains_key (format, "channel-mask", NULL))
gst_amc_format_get_int (format, "channel-mask", (gint *) & channel_mask,
NULL);
gst_amc_audio_channel_mask_to_positions (channel_mask, channels,
self->positions);
memcpy (to, self->positions, sizeof (to));
gst_audio_channel_positions_to_valid_order (to, channels);
self->needs_reorder =
(memcmp (self->positions, to,
sizeof (GstAudioChannelPosition) * channels) != 0);
if (self->needs_reorder)
gst_audio_get_channel_reorder_map (channels, self->positions, to,
self->reorder_map);
gst_audio_info_init (&self->info);
gst_audio_info_set_format (&self->info, GST_AUDIO_FORMAT_S16, rate, channels,
to);
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (self),
&self->info))
return FALSE;
self->input_caps_changed = FALSE;
return TRUE;
}
static gboolean
gst_sbc_dec_set_format (GstAudioDecoder * audio_dec, GstCaps * caps)
{
GstSbcDec *dec = GST_SBC_DEC (audio_dec);
const gchar *channel_mode;
GstAudioInfo info;
GstStructure *s;
gint channels, rate, subbands, blocks, bitpool;
s = gst_caps_get_structure (caps, 0);
gst_structure_get_int (s, "channels", &channels);
gst_structure_get_int (s, "rate", &rate);
/* save input format */
channel_mode = gst_structure_get_string (s, "channel-mode");
if (channel_mode == NULL ||
!gst_structure_get_int (s, "subbands", &subbands) ||
!gst_structure_get_int (s, "blocks", &blocks) ||
!gst_structure_get_int (s, "bitpool", &bitpool))
return FALSE;
if (strcmp (channel_mode, "mono") == 0) {
dec->frame_len = 4 + (subbands * 1) / 2 + ((blocks * 1 * bitpool) + 7) / 8;
} else if (strcmp (channel_mode, "dual") == 0) {
dec->frame_len = 4 + (subbands * 2) / 2 + ((blocks * 2 * bitpool) + 7) / 8;
} else if (strcmp (channel_mode, "stereo") == 0) {
dec->frame_len = 4 + (subbands * 2) / 2 + ((blocks * bitpool) + 7) / 8;
} else if (strcmp (channel_mode, "joint") == 0) {
dec->frame_len =
4 + (subbands * 2) / 2 + ((subbands + blocks * bitpool) + 7) / 8;
} else {
return FALSE;
}
dec->samples_per_frame = channels * blocks * subbands;
GST_INFO_OBJECT (dec, "frame len: %" G_GSIZE_FORMAT ", samples per frame "
"%" G_GSIZE_FORMAT, dec->frame_len, dec->samples_per_frame);
/* set up output format */
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, rate, channels, NULL);
gst_audio_decoder_set_output_format (audio_dec, &info);
return TRUE;
}
static void
gst_deinterleave_add_new_pads (GstDeinterleave * self, GstCaps * caps)
{
GstPad *pad;
guint i;
for (i = 0; i < GST_AUDIO_INFO_CHANNELS (&self->audio_info); i++) {
gchar *name = g_strdup_printf ("src_%u", i);
GstCaps *srccaps;
GstAudioInfo info;
GstAudioFormat format = GST_AUDIO_INFO_FORMAT (&self->audio_info);
gint rate = GST_AUDIO_INFO_RATE (&self->audio_info);
GstAudioChannelPosition position = GST_AUDIO_CHANNEL_POSITION_MONO;
CopyStickyEventsData data;
/* Set channel position if we know it */
if (self->keep_positions)
position = GST_AUDIO_INFO_POSITION (&self->audio_info, i);
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, format, rate, 1, &position);
srccaps = gst_audio_info_to_caps (&info);
pad = gst_pad_new_from_static_template (&src_template, name);
g_free (name);
gst_pad_use_fixed_caps (pad);
gst_pad_set_query_function (pad,
GST_DEBUG_FUNCPTR (gst_deinterleave_src_query));
gst_pad_set_active (pad, TRUE);
data.pad = pad;
data.caps = srccaps;
gst_pad_sticky_events_foreach (self->sink, copy_sticky_events, &data);
if (data.caps)
gst_pad_set_caps (pad, data.caps);
gst_element_add_pad (GST_ELEMENT (self), pad);
self->srcpads = g_list_prepend (self->srcpads, gst_object_ref (pad));
gst_caps_unref (srccaps);
}
gst_element_no_more_pads (GST_ELEMENT (self));
self->srcpads = g_list_reverse (self->srcpads);
}
static void
gst_opus_dec_negotiate (GstOpusDec * dec, const GstAudioChannelPosition * pos)
{
GstCaps *caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
GstStructure *s;
GstAudioInfo info;
if (caps) {
caps = gst_caps_truncate (caps);
caps = gst_caps_make_writable (caps);
s = gst_caps_get_structure (caps, 0);
gst_structure_fixate_field_nearest_int (s, "rate", 48000);
gst_structure_get_int (s, "rate", &dec->sample_rate);
gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels);
gst_structure_get_int (s, "channels", &dec->n_channels);
gst_caps_unref (caps);
} else {
dec->sample_rate = 48000;
}
GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels,
dec->sample_rate);
/* pass valid order to audio info */
if (pos) {
memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels);
}
/* set up source format */
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16,
dec->sample_rate, dec->n_channels, pos ? dec->opus_pos : NULL);
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
/* but we still need the opus order for later reordering */
if (pos) {
memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels);
} else {
dec->opus_pos[0] = GST_AUDIO_CHANNEL_POSITION_INVALID;
}
dec->info = info;
}
static void
gst_wavpack_dec_negotiate (GstWavpackDec * dec)
{
GstAudioInfo info;
GstAudioFormat fmt;
GstAudioChannelPosition pos[64] = { GST_AUDIO_CHANNEL_POSITION_INVALID, };
/* arrange for 1, 2 or 4-byte width == depth output */
dec->width = dec->depth;
switch (dec->depth) {
case 8:
fmt = GST_AUDIO_FORMAT_S8;
break;
case 16:
fmt = _GST_AUDIO_FORMAT_NE (S16);
break;
case 24:
case 32:
fmt = _GST_AUDIO_FORMAT_NE (S32);
dec->width = 32;
break;
default:
fmt = GST_AUDIO_FORMAT_UNKNOWN;
g_assert_not_reached ();
break;
}
g_assert (dec->channel_mask != 0);
if (!gst_wavpack_get_channel_positions (dec->channels,
dec->channel_mask, pos))
GST_WARNING_OBJECT (dec, "Failed to set channel layout");
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, fmt, dec->sample_rate, dec->channels, pos);
gst_audio_channel_positions_to_valid_order (info.position, info.channels);
gst_audio_get_channel_reorder_map (info.channels,
info.position, pos, dec->channel_reorder_map);
/* should always succeed */
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
}
static gboolean
gst_mulawdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
{
GstMuLawDec *mulawdec = GST_MULAWDEC (dec);
GstStructure *structure;
int rate, channels;
GstAudioInfo info;
structure = gst_caps_get_structure (caps, 0);
if (!structure) {
GST_ERROR ("failed to get structure from caps");
goto error_failed_get_structure;
}
if (!gst_structure_get_int (structure, "rate", &rate)) {
GST_ERROR ("failed to find field rate in input caps");
goto error_failed_find_rate;
}
if (!gst_structure_get_int (structure, "channels", &channels)) {
GST_ERROR ("failed to find field channels in input caps");
goto error_failed_find_channel;
}
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, rate, channels, NULL);
GST_DEBUG_OBJECT (mulawdec, "rate=%d, channels=%d", rate, channels);
return gst_audio_decoder_set_output_format (dec, &info);
error_failed_find_channel:
error_failed_find_rate:
error_failed_get_structure:
return FALSE;
}
static void
gst_droidadec_data_available (void *data, DroidMediaCodecData * encoded)
{
GstFlowReturn flow_ret;
GstDroidADec *dec = (GstDroidADec *) data;
GstAudioDecoder *decoder = GST_AUDIO_DECODER (dec);
GstBuffer *out;
GstMapInfo info;
GST_DEBUG_OBJECT (dec, "data available of size %d", encoded->data.size);
GST_AUDIO_DECODER_STREAM_LOCK (decoder);
if (G_UNLIKELY (dec->downstream_flow_ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (dec, "not handling data in error state: %s",
gst_flow_get_name (dec->downstream_flow_ret));
flow_ret = dec->downstream_flow_ret;
gst_audio_decoder_finish_frame (decoder, NULL, 1);
goto out;
}
if (G_UNLIKELY (gst_audio_decoder_get_audio_info (GST_AUDIO_DECODER
(dec))->finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
DroidMediaCodecMetaData md;
DroidMediaRect crop; /* TODO: get rid of that */
GstAudioInfo info;
memset (&md, 0x0, sizeof (md));
droid_media_codec_get_output_info (dec->codec, &md, &crop);
GST_INFO_OBJECT (dec, "output rate=%d, output channels=%d", md.sample_rate,
md.channels);
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, md.sample_rate,
md.channels, NULL);
if (!gst_audio_decoder_set_output_format (decoder, &info)) {
flow_ret = GST_FLOW_ERROR;
goto out;
}
dec->info = gst_audio_decoder_get_audio_info (GST_AUDIO_DECODER (dec));
}
out = gst_audio_decoder_allocate_output_buffer (decoder, encoded->data.size);
gst_buffer_map (out, &info, GST_MAP_READWRITE);
orc_memcpy (info.data, encoded->data.data, encoded->data.size);
gst_buffer_unmap (out, &info);
// GST_WARNING_OBJECT (dec, "bpf %d, bps %d", dec->info->bpf, GST_AUDIO_INFO_BPS(dec->info));
if (dec->spf == -1 || (encoded->data.size == dec->spf * dec->info->bpf
&& gst_adapter_available (dec->adapter) == 0)) {
/* fast path. no need for anything */
goto push;
}
gst_adapter_push (dec->adapter, out);
if (gst_adapter_available (dec->adapter) >= dec->spf * dec->info->bpf) {
out = gst_adapter_take_buffer (dec->adapter, dec->spf * dec->info->bpf);
} else {
flow_ret = GST_FLOW_OK;
goto out;
}
push:
GST_DEBUG_OBJECT (dec, "pushing %d bytes out", gst_buffer_get_size (out));
flow_ret = gst_audio_decoder_finish_frame (decoder, out, 1);
if (flow_ret == GST_FLOW_OK || flow_ret == GST_FLOW_FLUSHING) {
goto out;
} else if (flow_ret == GST_FLOW_EOS) {
GST_INFO_OBJECT (dec, "eos");
} else if (flow_ret < GST_FLOW_OK) {
GST_ELEMENT_ERROR (dec, STREAM, FAILED,
("Internal data stream error."), ("stream stopped, reason %s",
gst_flow_get_name (flow_ret)));
}
out:
dec->downstream_flow_ret = flow_ret;
GST_AUDIO_DECODER_STREAM_UNLOCK (decoder);
}
static gboolean
gst_ffmpegauddec_negotiate (GstFFMpegAudDec * ffmpegdec, gboolean force)
{
GstFFMpegAudDecClass *oclass;
gint depth;
GstAudioFormat format;
GstAudioChannelPosition pos[64] = { 0, };
oclass = (GstFFMpegAudDecClass *) (G_OBJECT_GET_CLASS (ffmpegdec));
depth = av_smp_format_depth (ffmpegdec->context->sample_fmt) * 8;
format = gst_ffmpeg_smpfmt_to_audioformat (ffmpegdec->context->sample_fmt);
if (format == GST_AUDIO_FORMAT_UNKNOWN)
goto no_caps;
if (!force && ffmpegdec->info.rate ==
ffmpegdec->context->sample_rate &&
ffmpegdec->info.channels == ffmpegdec->context->channels &&
ffmpegdec->info.finfo->depth == depth)
return TRUE;
GST_DEBUG_OBJECT (ffmpegdec,
"Renegotiating audio from %dHz@%dchannels (%d) to %dHz@%dchannels (%d)",
ffmpegdec->info.rate, ffmpegdec->info.channels,
ffmpegdec->info.finfo->depth,
ffmpegdec->context->sample_rate, ffmpegdec->context->channels, depth);
gst_ffmpeg_channel_layout_to_gst (ffmpegdec->context->channel_layout,
ffmpegdec->context->channels, pos);
memcpy (ffmpegdec->ffmpeg_layout, pos,
sizeof (GstAudioChannelPosition) * ffmpegdec->context->channels);
/* Get GStreamer channel layout */
gst_audio_channel_positions_to_valid_order (pos,
ffmpegdec->context->channels);
ffmpegdec->needs_reorder =
memcmp (pos, ffmpegdec->ffmpeg_layout,
sizeof (pos[0]) * ffmpegdec->context->channels) != 0;
gst_audio_info_set_format (&ffmpegdec->info, format,
ffmpegdec->context->sample_rate, ffmpegdec->context->channels, pos);
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (ffmpegdec),
&ffmpegdec->info))
goto caps_failed;
return TRUE;
/* ERRORS */
no_caps:
{
#ifdef HAVE_LIBAV_UNINSTALLED
/* using internal ffmpeg snapshot */
GST_ELEMENT_ERROR (ffmpegdec, CORE, NEGOTIATION,
("Could not find GStreamer caps mapping for libav codec '%s'.",
oclass->in_plugin->name), (NULL));
#else
/* using external ffmpeg */
GST_ELEMENT_ERROR (ffmpegdec, CORE, NEGOTIATION,
("Could not find GStreamer caps mapping for libav codec '%s', and "
"you are using an external libavcodec. This is most likely due to "
"a packaging problem and/or libavcodec having been upgraded to a "
"version that is not compatible with this version of "
"gstreamer-libav. Make sure your gstreamer-libav and libavcodec "
"packages come from the same source/repository.",
oclass->in_plugin->name), (NULL));
#endif
return FALSE;
}
caps_failed:
{
GST_ELEMENT_ERROR (ffmpegdec, CORE, NEGOTIATION, (NULL),
("Could not set caps for libav decoder (%s), not fixed?",
oclass->in_plugin->name));
return FALSE;
}
}
static gboolean
gst_opus_dec_negotiate (GstOpusDec * dec, const GstAudioChannelPosition * pos)
{
GstCaps *caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
GstStructure *s;
GstAudioInfo info;
if (caps) {
gint rate = dec->sample_rate, channels = dec->n_channels;
GstCaps *constraint, *inter;
constraint = gst_caps_from_string ("audio/x-raw");
if (dec->n_channels <= 2) { /* including 0 */
gst_caps_set_simple (constraint, "channels", GST_TYPE_INT_RANGE, 1, 2,
NULL);
} else {
gst_caps_set_simple (constraint, "channels", G_TYPE_INT, dec->n_channels,
NULL);
}
inter = gst_caps_intersect (caps, constraint);
gst_caps_unref (constraint);
if (gst_caps_is_empty (inter)) {
GST_DEBUG_OBJECT (dec, "Empty intersection, failed to negotiate");
gst_caps_unref (inter);
gst_caps_unref (caps);
return FALSE;
}
inter = gst_caps_truncate (inter);
s = gst_caps_get_structure (inter, 0);
rate = dec->sample_rate > 0 ? dec->sample_rate : 48000;
gst_structure_fixate_field_nearest_int (s, "rate", dec->sample_rate);
gst_structure_get_int (s, "rate", &rate);
channels = dec->n_channels > 0 ? dec->n_channels : 2;
gst_structure_fixate_field_nearest_int (s, "channels", dec->n_channels);
gst_structure_get_int (s, "channels", &channels);
gst_caps_unref (inter);
dec->sample_rate = rate;
dec->n_channels = channels;
gst_caps_unref (caps);
}
if (dec->n_channels == 0) {
GST_DEBUG_OBJECT (dec, "Using a default of 2 channels");
dec->n_channels = 2;
pos = NULL;
}
if (dec->sample_rate == 0) {
GST_DEBUG_OBJECT (dec, "Using a default of 48kHz sample rate");
dec->sample_rate = 48000;
}
GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels,
dec->sample_rate);
/* pass valid order to audio info */
if (pos) {
memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels);
}
/* set up source format */
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16,
dec->sample_rate, dec->n_channels, pos ? dec->opus_pos : NULL);
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
/* but we still need the opus order for later reordering */
if (pos) {
memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
} else {
dec->opus_pos[0] = GST_AUDIO_CHANNEL_POSITION_INVALID;
}
dec->info = info;
return TRUE;
}
static gboolean
gst_atdec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
OSStatus status;
AudioStreamBasicDescription input_format = { 0 };
AudioStreamBasicDescription output_format = { 0 };
GstAudioInfo output_info = { 0 };
AudioChannelLayout output_layout = { 0 };
GstCaps *output_caps;
GstATDec *atdec = GST_ATDEC (decoder);
GST_DEBUG_OBJECT (atdec, "set_format");
if (atdec->queue)
gst_atdec_destroy_queue (atdec, TRUE);
/* configure input_format from caps */
gst_caps_to_at_format (caps, &input_format);
/* negotiate output caps */
output_caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (atdec));
if (!output_caps)
output_caps =
gst_pad_get_pad_template_caps (GST_AUDIO_DECODER_SRC_PAD (atdec));
output_caps = gst_caps_fixate (output_caps);
gst_caps_set_simple (output_caps,
"rate", G_TYPE_INT, (int) input_format.mSampleRate,
"channels", G_TYPE_INT, input_format.mChannelsPerFrame, NULL);
/* configure output_format from caps */
gst_caps_to_at_format (output_caps, &output_format);
/* set the format we want to negotiate downstream */
gst_audio_info_from_caps (&output_info, output_caps);
gst_audio_info_set_format (&output_info,
output_format.mFormatFlags & kLinearPCMFormatFlagIsSignedInteger ?
GST_AUDIO_FORMAT_S16LE : GST_AUDIO_FORMAT_F32LE,
output_format.mSampleRate, output_format.mChannelsPerFrame, NULL);
gst_audio_decoder_set_output_format (decoder, &output_info);
gst_caps_unref (output_caps);
status = AudioQueueNewOutput (&input_format, gst_atdec_buffer_emptied,
atdec, NULL, NULL, 0, &atdec->queue);
if (status)
goto create_queue_error;
/* FIXME: figure out how to map this properly */
if (output_format.mChannelsPerFrame == 1)
output_layout.mChannelLayoutTag = kAudioChannelLayoutTag_Mono;
else
output_layout.mChannelLayoutTag = kAudioChannelLayoutTag_Stereo;
status = AudioQueueSetOfflineRenderFormat (atdec->queue,
&output_format, &output_layout);
if (status)
goto set_format_error;
status = AudioQueueStart (atdec->queue, NULL);
if (status)
goto start_error;
return TRUE;
create_queue_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("AudioQueueNewOutput returned error: %d", status));
return FALSE;
set_format_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("AudioQueueSetOfflineRenderFormat returned error: %d", status));
gst_atdec_destroy_queue (atdec, FALSE);
return FALSE;
start_error:
GST_ELEMENT_ERROR (atdec, STREAM, FORMAT, (NULL),
("AudioQueueStart returned error: %d", status));
gst_atdec_destroy_queue (atdec, FALSE);
return FALSE;
}
static GstCaps *
gst_audio_parse_get_caps (GstRawParse * rp)
{
GstAudioParse *ap = GST_AUDIO_PARSE (rp);
GstCaps *caps, *ncaps;
GstAudioInfo info;
gint fps_n, fps_d;
const GValue *val;
if (ap->use_sink_caps) {
gint rate;
GstCaps *caps = gst_pad_get_current_caps (rp->sinkpad);
gst_audio_info_from_caps (&info, caps);
rate = GST_AUDIO_INFO_RATE (&info);
gst_raw_parse_set_fps (GST_RAW_PARSE (ap), rate, 1);
return caps;
}
gst_raw_parse_get_fps (rp, &fps_n, &fps_d);
gst_audio_parse_setup_channel_positions (ap);
/* yes, even when format not raw */
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, ap->raw_format, fps_n, ap->channels,
ap->channel_order);
info.layout = ap->interleaved ? GST_AUDIO_LAYOUT_INTERLEAVED :
GST_AUDIO_LAYOUT_NON_INTERLEAVED;
caps = gst_audio_info_to_caps (&info);
switch (ap->format) {
case GST_AUDIO_PARSE_FORMAT_RAW:
break;
case GST_AUDIO_PARSE_FORMAT_ALAW:
ncaps = gst_caps_new_simple ("audio/x-alaw",
"rate", G_TYPE_INT, fps_n,
"channels", G_TYPE_INT, ap->channels, NULL);
/* pick mask stuff from faked raw format */
val = gst_structure_get_value (gst_caps_get_structure (caps, 0),
"channel-mask");
if (val)
gst_caps_set_value (ncaps, "channel-mask", val);
gst_caps_unref (caps);
caps = ncaps;
break;
case GST_AUDIO_PARSE_FORMAT_MULAW:
ncaps = gst_caps_new_simple ("audio/x-mulaw",
"rate", G_TYPE_INT, fps_n,
"channels", G_TYPE_INT, ap->channels, NULL);
/* pick mask stuff from faked raw format */
val = gst_structure_get_value (gst_caps_get_structure (caps, 0),
"channel-mask");
if (val)
gst_caps_set_value (ncaps, "channel-mask", val);
gst_caps_unref (caps);
caps = ncaps;
break;
default:
caps = gst_caps_new_empty ();
GST_ERROR_OBJECT (rp, "unexpected format %d", ap->format);
break;
}
return caps;
}
static void
gst_omx_audio_dec_loop (GstOMXAudioDec * self)
{
GstOMXPort *port = self->dec_out_port;
GstOMXBuffer *buf = NULL;
GstFlowReturn flow_ret = GST_FLOW_OK;
GstOMXAcquireBufferReturn acq_return;
OMX_ERRORTYPE err;
acq_return = gst_omx_port_acquire_buffer (port, &buf);
if (acq_return == GST_OMX_ACQUIRE_BUFFER_ERROR) {
goto component_error;
} else if (acq_return == GST_OMX_ACQUIRE_BUFFER_FLUSHING) {
goto flushing;
} else if (acq_return == GST_OMX_ACQUIRE_BUFFER_EOS) {
goto eos;
}
if (!gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (self)) ||
acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
OMX_PARAM_PORTDEFINITIONTYPE port_def;
OMX_AUDIO_PARAM_PCMMODETYPE pcm_param;
GstAudioChannelPosition omx_position[OMX_AUDIO_MAXCHANNELS];
GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self);
gint i;
GST_DEBUG_OBJECT (self, "Port settings have changed, updating caps");
/* Reallocate all buffers */
if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE
&& gst_omx_port_is_enabled (port)) {
err = gst_omx_port_set_enabled (port, FALSE);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_buffers_released (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_deallocate_buffers (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_enabled (port, 1 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
}
/* Just update caps */
GST_AUDIO_DECODER_STREAM_LOCK (self);
gst_omx_port_get_port_definition (port, &port_def);
g_assert (port_def.format.audio.eEncoding == OMX_AUDIO_CodingPCM);
GST_OMX_INIT_STRUCT (&pcm_param);
pcm_param.nPortIndex = self->dec_out_port->index;
err =
gst_omx_component_get_parameter (self->dec, OMX_IndexParamAudioPcm,
&pcm_param);
if (err != OMX_ErrorNone) {
GST_ERROR_OBJECT (self, "Failed to get PCM parameters: %s (0x%08x)",
gst_omx_error_to_string (err), err);
goto caps_failed;
}
g_assert (pcm_param.ePCMMode == OMX_AUDIO_PCMModeLinear);
g_assert (pcm_param.bInterleaved == OMX_TRUE);
gst_audio_info_init (&self->info);
for (i = 0; i < pcm_param.nChannels; i++) {
switch (pcm_param.eChannelMapping[i]) {
case OMX_AUDIO_ChannelLF:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
break;
case OMX_AUDIO_ChannelRF:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
break;
case OMX_AUDIO_ChannelCF:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
break;
case OMX_AUDIO_ChannelLS:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
break;
case OMX_AUDIO_ChannelRS:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
break;
case OMX_AUDIO_ChannelLFE:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_LFE1;
break;
case OMX_AUDIO_ChannelCS:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
break;
case OMX_AUDIO_ChannelLR:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
break;
case OMX_AUDIO_ChannelRR:
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
break;
case OMX_AUDIO_ChannelNone:
default:
/* This will break the outer loop too as the
* i == pcm_param.nChannels afterwards */
for (i = 0; i < pcm_param.nChannels; i++)
omx_position[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
break;
}
}
if (pcm_param.nChannels == 1
&& omx_position[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER)
omx_position[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
if (omx_position[0] == GST_AUDIO_CHANNEL_POSITION_NONE
&& klass->get_channel_positions) {
GST_WARNING_OBJECT (self,
"Failed to get a valid channel layout, trying fallback");
klass->get_channel_positions (self, self->dec_out_port, omx_position);
}
memcpy (self->position, omx_position, sizeof (omx_position));
gst_audio_channel_positions_to_valid_order (self->position,
pcm_param.nChannels);
self->needs_reorder =
(memcmp (self->position, omx_position,
sizeof (GstAudioChannelPosition) * pcm_param.nChannels) != 0);
if (self->needs_reorder)
gst_audio_get_channel_reorder_map (pcm_param.nChannels, self->position,
omx_position, self->reorder_map);
gst_audio_info_set_format (&self->info,
gst_audio_format_build_integer (pcm_param.eNumData ==
OMX_NumericalDataSigned,
pcm_param.eEndian ==
OMX_EndianLittle ? G_LITTLE_ENDIAN : G_BIG_ENDIAN,
pcm_param.nBitPerSample, pcm_param.nBitPerSample),
pcm_param.nSamplingRate, pcm_param.nChannels, self->position);
GST_DEBUG_OBJECT (self,
"Setting output state: format %s, rate %u, channels %u",
gst_audio_format_to_string (self->info.finfo->format),
(guint) pcm_param.nSamplingRate, (guint) pcm_param.nChannels);
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (self),
&self->info)
|| !gst_audio_decoder_negotiate (GST_AUDIO_DECODER (self))) {
if (buf)
gst_omx_port_release_buffer (port, buf);
goto caps_failed;
}
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
err = gst_omx_port_set_enabled (port, TRUE);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_allocate_buffers (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_wait_enabled (port, 5 * GST_SECOND);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_populate (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
err = gst_omx_port_mark_reconfigured (port);
if (err != OMX_ErrorNone)
goto reconfigure_error;
}
/* Now get a buffer */
if (acq_return != GST_OMX_ACQUIRE_BUFFER_OK) {
return;
}
}
g_assert (acq_return == GST_OMX_ACQUIRE_BUFFER_OK);
if (!buf) {
g_assert ((klass->cdata.hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER));
GST_AUDIO_DECODER_STREAM_LOCK (self);
goto eos;
}
/* This prevents a deadlock between the srcpad stream
* lock and the audiocodec stream lock, if ::reset()
* is called at the wrong time
*/
if (gst_omx_port_is_flushing (port)) {
GST_DEBUG_OBJECT (self, "Flushing");
gst_omx_port_release_buffer (port, buf);
goto flushing;
}
GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %" G_GUINT64_FORMAT,
(guint) buf->omx_buf->nFlags, (guint64) buf->omx_buf->nTimeStamp);
GST_AUDIO_DECODER_STREAM_LOCK (self);
if (buf->omx_buf->nFilledLen > 0) {
GstBuffer *outbuf;
gint nframes, spf;
GstMapInfo minfo;
GstOMXAudioDecClass *klass = GST_OMX_AUDIO_DEC_GET_CLASS (self);
GST_DEBUG_OBJECT (self, "Handling output data");
if (buf->omx_buf->nFilledLen % self->info.bpf != 0) {
gst_omx_port_release_buffer (port, buf);
goto invalid_buffer;
}
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (self),
buf->omx_buf->nFilledLen);
gst_buffer_map (outbuf, &minfo, GST_MAP_WRITE);
if (self->needs_reorder) {
gint i, n_samples, c, n_channels;
gint *reorder_map = self->reorder_map;
gint16 *dest, *source;
dest = (gint16 *) minfo.data;
source = (gint16 *) (buf->omx_buf->pBuffer + buf->omx_buf->nOffset);
n_samples = buf->omx_buf->nFilledLen / self->info.bpf;
n_channels = self->info.channels;
for (i = 0; i < n_samples; i++) {
for (c = 0; c < n_channels; c++) {
dest[i * n_channels + reorder_map[c]] = source[i * n_channels + c];
}
}
} else {
memcpy (minfo.data, buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
}
gst_buffer_unmap (outbuf, &minfo);
nframes = 1;
spf = klass->get_samples_per_frame (self, self->dec_out_port);
if (spf != -1) {
nframes = buf->omx_buf->nFilledLen / self->info.bpf;
if (nframes % spf != 0)
GST_WARNING_OBJECT (self, "Output buffer does not contain an integer "
"number of input frames (frames: %d, spf: %d)", nframes, spf);
nframes = (nframes + spf - 1) / spf;
}
GST_BUFFER_TIMESTAMP (outbuf) =
gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND,
OMX_TICKS_PER_SECOND);
if (buf->omx_buf->nTickCount != 0)
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND,
OMX_TICKS_PER_SECOND);
flow_ret =
gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (self), outbuf,
nframes);
}
GST_DEBUG_OBJECT (self, "Read frame from component");
GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
if (buf) {
err = gst_omx_port_release_buffer (port, buf);
if (err != OMX_ErrorNone)
goto release_error;
}
self->downstream_flow_ret = flow_ret;
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
component_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, FAILED, (NULL),
("OpenMAX component in error state %s (0x%08x)",
gst_omx_component_get_last_error_string (self->dec),
gst_omx_component_get_last_error (self->dec)));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
return;
}
flushing:
{
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_FLUSHING;
self->started = FALSE;
return;
}
eos:
{
g_mutex_lock (&self->drain_lock);
if (self->draining) {
GST_DEBUG_OBJECT (self, "Drained");
self->draining = FALSE;
g_cond_broadcast (&self->drain_cond);
flow_ret = GST_FLOW_OK;
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
} else {
GST_DEBUG_OBJECT (self, "Component signalled EOS");
flow_ret = GST_FLOW_EOS;
}
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_DECODER_STREAM_LOCK (self);
self->downstream_flow_ret = flow_ret;
/* Here we fallback and pause the task for the EOS case */
if (flow_ret != GST_FLOW_OK)
goto flow_error;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
flow_error:
{
if (flow_ret == GST_FLOW_EOS) {
GST_DEBUG_OBJECT (self, "EOS");
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->started = FALSE;
} else if (flow_ret < GST_FLOW_EOS) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
("Internal data stream error."), ("stream stopped, reason %s",
gst_flow_get_name (flow_ret)));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->started = FALSE;
} else if (flow_ret == GST_FLOW_FLUSHING) {
GST_DEBUG_OBJECT (self, "Flushing -- stopping task");
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->started = FALSE;
}
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
reconfigure_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Unable to reconfigure output port"));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
return;
}
invalid_buffer:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Invalid sized input buffer"));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
self->started = FALSE;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
caps_failed:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps"));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
self->started = FALSE;
return;
}
release_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Failed to relase output buffer to component: %s (0x%08x)",
gst_omx_error_to_string (err), err));
gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_DECODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
return;
}
}
/**
* gst_audio_info_from_caps:
* @info: a #GstAudioInfo
* @caps: a #GstCaps
*
* Parse @caps and update @info.
*
* Returns: TRUE if @caps could be parsed
*/
gboolean
gst_audio_info_from_caps (GstAudioInfo * info, const GstCaps * caps)
{
GstStructure *str;
const gchar *s;
GstAudioFormat format;
gint rate, channels;
guint64 channel_mask;
gint i;
GstAudioChannelPosition position[64];
GstAudioFlags flags;
GstAudioLayout layout;
g_return_val_if_fail (info != NULL, FALSE);
g_return_val_if_fail (caps != NULL, FALSE);
g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
GST_DEBUG ("parsing caps %" GST_PTR_FORMAT, caps);
flags = 0;
str = gst_caps_get_structure (caps, 0);
if (!gst_structure_has_name (str, "audio/x-raw"))
goto wrong_name;
if (!(s = gst_structure_get_string (str, "format")))
goto no_format;
format = gst_audio_format_from_string (s);
if (format == GST_AUDIO_FORMAT_UNKNOWN)
goto unknown_format;
if (!(s = gst_structure_get_string (str, "layout")))
goto no_layout;
if (g_str_equal (s, "interleaved"))
layout = GST_AUDIO_LAYOUT_INTERLEAVED;
else if (g_str_equal (s, "non-interleaved"))
layout = GST_AUDIO_LAYOUT_NON_INTERLEAVED;
else
goto unknown_layout;
if (!gst_structure_get_int (str, "rate", &rate))
goto no_rate;
if (!gst_structure_get_int (str, "channels", &channels))
goto no_channels;
if (!gst_structure_get (str, "channel-mask", GST_TYPE_BITMASK, &channel_mask,
NULL)) {
if (channels == 1) {
position[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
} else if (channels == 2) {
position[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
position[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
} else {
goto no_channel_mask;
}
} else if (channel_mask == 0) {
flags |= GST_AUDIO_FLAG_UNPOSITIONED;
for (i = 0; i < MIN (64, channels); i++)
position[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
} else {
if (!gst_audio_channel_positions_from_mask (channels, channel_mask,
position))
goto invalid_channel_mask;
}
gst_audio_info_set_format (info, format, rate, channels,
(channels > 64) ? NULL : position);
info->flags = flags;
info->layout = layout;
return TRUE;
/* ERROR */
wrong_name:
{
GST_ERROR ("wrong name, expected audio/x-raw");
return FALSE;
}
no_format:
{
GST_ERROR ("no format given");
return FALSE;
}
unknown_format:
{
GST_ERROR ("unknown format given");
return FALSE;
}
no_layout:
{
GST_ERROR ("no layout given");
return FALSE;
}
unknown_layout:
{
GST_ERROR ("unknown layout given");
return FALSE;
}
no_rate:
{
GST_ERROR ("no rate property given");
return FALSE;
}
no_channels:
{
GST_ERROR ("no channels property given");
return FALSE;
}
no_channel_mask:
{
GST_ERROR ("no channel-mask property given");
return FALSE;
}
invalid_channel_mask:
{
GST_ERROR ("Invalid channel mask 0x%016" G_GINT64_MODIFIER
"x for %d channels", channel_mask, channels);
return FALSE;
}
}
static GstFlowReturn gst_imx_audio_uniaudio_dec_handle_frame(GstAudioDecoder *dec, GstBuffer *buffer)
{
GstMapInfo in_map;
GstBuffer *out_buffer;
gsize avail_out_size;
GstImxAudioUniaudioDec *imx_audio_uniaudio_dec = GST_IMX_AUDIO_UNIAUDIO_DEC(dec);
int32 dec_ret;
uint32 offset = 0;
uint8 *in_buf = NULL;
uint32 in_size = 0;
gboolean dec_loop = TRUE, flow_error = FALSE;
/* With some formats such as Vorbis, the first few buffers are actually redundant,
* since they contain codec data that was already specified in codec_data or
* streamheader caps earlier. If this is the case, skip these buffers. */
if (imx_audio_uniaudio_dec->skip_header_counter < imx_audio_uniaudio_dec->num_vorbis_headers)
{
GST_TRACE_OBJECT(dec, "skipping header buffer #%u", imx_audio_uniaudio_dec->skip_header_counter);
++imx_audio_uniaudio_dec->skip_header_counter;
return gst_audio_decoder_finish_frame(dec, NULL, 1);
}
if (buffer != NULL)
{
gst_buffer_map(buffer, &in_map, GST_MAP_READ);
in_buf = in_map.data;
in_size = in_map.size;
}
while (dec_loop)
{
GstBuffer *tmp_buf;
uint8 *out_buf = NULL;
uint32 out_size = 0;
if (buffer != NULL)
GST_TRACE_OBJECT(dec, "feeding %" G_GUINT32_FORMAT " bytes to the decoder", (guint32)in_size);
else
GST_TRACE_OBJECT(dec, "draining decoder");
dec_ret = imx_audio_uniaudio_dec->codec->decode_frame(
imx_audio_uniaudio_dec->handle,
in_buf, in_size,
&offset,
&out_buf, &out_size
);
GST_TRACE_OBJECT(dec, "decode_frame: return 0x%x offset %" G_GUINT32_FORMAT " out_size %" G_GUINT32_FORMAT, (unsigned int)dec_ret, (guint32)offset, (guint32)out_size);
if ((out_buf != NULL) && (out_size > 0))
{
tmp_buf = gst_audio_decoder_allocate_output_buffer(dec, out_size);
tmp_buf = gst_buffer_make_writable(tmp_buf);
gst_buffer_fill(tmp_buf, 0, out_buf, out_size);
gst_adapter_push(imx_audio_uniaudio_dec->out_adapter, tmp_buf);
}
if (out_buf != NULL)
{
gst_imx_audio_uniaudio_dec_free(out_buf);
}
if ((buffer != NULL) && (offset == in_map.size))
{
dec_loop = FALSE;
}
switch (dec_ret)
{
case ACODEC_SUCCESS:
break;
case ACODEC_END_OF_STREAM:
dec_loop = FALSE;
break;
case ACODEC_NOT_ENOUGH_DATA:
break;
case ACODEC_CAPIBILITY_CHANGE:
break;
default:
{
dec_loop = FALSE;
flow_error = TRUE;
GST_ELEMENT_ERROR(dec, STREAM, DECODE, ("could not decode"), ("error message: %s", imx_audio_uniaudio_dec->codec->get_last_error(imx_audio_uniaudio_dec->handle)));
}
}
}
if (buffer != NULL)
gst_buffer_unmap(buffer, &in_map);
if (flow_error)
return GST_FLOW_ERROR;
if (!(imx_audio_uniaudio_dec->has_audioinfo_set))
{
UniACodecParameter parameter;
GstAudioFormat pcm_fmt;
GstAudioInfo audio_info;
imx_audio_uniaudio_dec->codec->get_parameter(imx_audio_uniaudio_dec->handle, UNIA_OUTPUT_PCM_FORMAT, ¶meter);
if ((parameter.outputFormat.width == 0) || (parameter.outputFormat.depth == 0))
{
GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "no output format available yet");
return gst_audio_decoder_finish_frame(dec, NULL, 1);
}
GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "output sample width: %" G_GUINT32_FORMAT " depth: %" G_GUINT32_FORMAT, (guint32)(parameter.outputFormat.width), (guint32)(parameter.outputFormat.depth));
pcm_fmt = gst_audio_format_build_integer(TRUE, G_BYTE_ORDER, parameter.outputFormat.width, parameter.outputFormat.depth);
GST_DEBUG_OBJECT(imx_audio_uniaudio_dec, "setting output format to: %s %d Hz %d channels", gst_audio_format_to_string(pcm_fmt), (gint)(parameter.outputFormat.samplerate), (gint)(parameter.outputFormat.channels));
gst_imx_audio_uniaudio_dec_clear_channel_positions(imx_audio_uniaudio_dec);
gst_imx_audio_uniaudio_dec_fill_channel_positions(imx_audio_uniaudio_dec, parameter.outputFormat.layout, parameter.outputFormat.channels);
imx_audio_uniaudio_dec->pcm_format = pcm_fmt;
imx_audio_uniaudio_dec->num_channels = parameter.outputFormat.channels;
gst_audio_info_set_format(
&audio_info,
pcm_fmt,
parameter.outputFormat.samplerate,
parameter.outputFormat.channels,
imx_audio_uniaudio_dec->reordered_channel_positions
);
gst_audio_decoder_set_output_format(dec, &audio_info);
imx_audio_uniaudio_dec->has_audioinfo_set = TRUE;
}
avail_out_size = gst_adapter_available(imx_audio_uniaudio_dec->out_adapter);
if (avail_out_size > 0)
{
out_buffer = gst_adapter_take_buffer(imx_audio_uniaudio_dec->out_adapter, avail_out_size);
if (imx_audio_uniaudio_dec->original_channel_positions != imx_audio_uniaudio_dec->reordered_channel_positions)
{
gst_audio_buffer_reorder_channels(
out_buffer,
imx_audio_uniaudio_dec->pcm_format,
imx_audio_uniaudio_dec->num_channels,
imx_audio_uniaudio_dec->original_channel_positions,
imx_audio_uniaudio_dec->reordered_channel_positions
);
}
return gst_audio_decoder_finish_frame(dec, out_buffer, 1);
}
else
{
return gst_audio_decoder_finish_frame(dec, NULL, 1);
}
}
static gboolean
gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec, GstCaps * input_caps)
{
/* Using the parsed information upstream, and the list of allowed caps
* downstream, this code tries to find a suitable audio info. It is important
* to keep in mind that the rate and number of channels should never deviate
* from the one the bitstream has, otherwise mpg123 has to mix channels and/or
* resample (and as its docs say, its internal resampler is very crude). The
* sample format, however, can be chosen freely, because the MPEG specs do not
* mandate any special format. Therefore, rate and number of channels are taken
* from upstream (which parsed the MPEG frames, so the input_caps contain
* exactly the rate and number of channels the bitstream actually has), while
* the sample format is chosen by trying out all caps that are allowed by
* downstream. This way, the output is adjusted to what the downstream prefers.
*
* Also, the new output audio info is not set immediately. Instead, it is
* considered the "next audioinfo". The code waits for mpg123 to notice the new
* format (= when mpg123_decode_frame() returns MPG123_AUDIO_DEC_NEW_FORMAT),
* and then sets the next audioinfo. Otherwise, the next audioinfo is set too
* soon, which may cause problems with mp3s containing several format headers.
* One example would be an mp3 with the first 30 seconds using 44.1 kHz, then
* the next 30 seconds using 32 kHz. Rare, but possible.
*
* STEPS:
*
* 1. get rate and channels from input_caps
* 2. get allowed caps from src pad
* 3. for each structure in allowed caps:
* 3.1. take format
* 3.2. if the combination of format with rate and channels is unsupported by
* mpg123, go to (3), or exit with error if there are no more structures
* to try
* 3.3. create next audioinfo out of rate,channels,format, and exit
*/
int rate, channels;
GstMpg123AudioDec *mpg123_decoder;
GstCaps *allowed_srccaps;
guint structure_nr;
gboolean match_found = FALSE;
mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
g_assert (mpg123_decoder->handle != NULL);
mpg123_decoder->has_next_audioinfo = FALSE;
/* Get rate and channels from input_caps */
{
GstStructure *structure;
gboolean err = FALSE;
/* Only the first structure is used (multiple
* input caps structures don't make sense */
structure = gst_caps_get_structure (input_caps, 0);
if (!gst_structure_get_int (structure, "rate", &rate)) {
err = TRUE;
GST_ERROR_OBJECT (dec, "Input caps do not have a rate value");
}
if (!gst_structure_get_int (structure, "channels", &channels)) {
err = TRUE;
GST_ERROR_OBJECT (dec, "Input caps do not have a channel value");
}
if (err)
return FALSE;
}
/* Get the caps that are allowed by downstream */
{
GstCaps *allowed_srccaps_unnorm =
gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
allowed_srccaps = gst_caps_normalize (allowed_srccaps_unnorm);
}
/* Go through all allowed caps, pick the first one that matches */
for (structure_nr = 0; structure_nr < gst_caps_get_size (allowed_srccaps);
++structure_nr) {
GstStructure *structure;
gchar const *format_str;
GstAudioFormat format;
int encoding;
structure = gst_caps_get_structure (allowed_srccaps, structure_nr);
format_str = gst_structure_get_string (structure, "format");
if (format_str == NULL) {
GST_DEBUG_OBJECT (dec, "Could not get format from src caps");
continue;
}
format = gst_audio_format_from_string (format_str);
if (format == GST_AUDIO_FORMAT_UNKNOWN) {
GST_DEBUG_OBJECT (dec, "Unknown format %s", format_str);
continue;
}
switch (format) {
case GST_AUDIO_FORMAT_S16:
encoding = MPG123_ENC_SIGNED_16;
break;
case GST_AUDIO_FORMAT_S24:
encoding = MPG123_ENC_SIGNED_24;
break;
case GST_AUDIO_FORMAT_S32:
encoding = MPG123_ENC_SIGNED_32;
break;
case GST_AUDIO_FORMAT_U16:
encoding = MPG123_ENC_UNSIGNED_16;
break;
case GST_AUDIO_FORMAT_U24:
encoding = MPG123_ENC_UNSIGNED_24;
break;
case GST_AUDIO_FORMAT_U32:
encoding = MPG123_ENC_UNSIGNED_32;
break;
case GST_AUDIO_FORMAT_F32:
encoding = MPG123_ENC_FLOAT_32;
break;
default:
GST_DEBUG_OBJECT (dec,
"Format %s in srccaps is not supported", format_str);
continue;
}
{
int err;
/* Cleanup old formats & set new one */
mpg123_format_none (mpg123_decoder->handle);
err = mpg123_format (mpg123_decoder->handle, rate, channels, encoding);
if (err != MPG123_OK) {
GST_DEBUG_OBJECT (dec,
"mpg123 cannot use caps %" GST_PTR_FORMAT
" because mpg123_format() failed: %s", structure,
mpg123_strerror (mpg123_decoder->handle));
continue;
}
}
gst_audio_info_init (&(mpg123_decoder->next_audioinfo));
gst_audio_info_set_format (&(mpg123_decoder->next_audioinfo), format, rate,
channels, NULL);
GST_LOG_OBJECT (dec, "The next audio format is: %s, %u Hz, %u channels",
format_str, rate, channels);
mpg123_decoder->has_next_audioinfo = TRUE;
match_found = TRUE;
break;
}
gst_caps_unref (allowed_srccaps);
return match_found;
}
static GstCaps *
gst_dshowaudiosrc_getcaps_from_streamcaps (GstDshowAudioSrc * src, IPin * pin,
IAMStreamConfig * streamcaps)
{
GstCaps *caps = NULL;
HRESULT hres = S_OK;
int icount = 0;
int isize = 0;
AUDIO_STREAM_CONFIG_CAPS ascc;
int i = 0;
if (!streamcaps)
return NULL;
streamcaps->GetNumberOfCapabilities (&icount, &isize);
if (isize != sizeof (ascc))
return NULL;
for (; i < icount; i++) {
GstCapturePinMediaType *pin_mediatype = g_new0 (GstCapturePinMediaType, 1);
pin->AddRef ();
pin_mediatype->capture_pin = pin;
hres = streamcaps->GetStreamCaps (i, &pin_mediatype->mediatype,
(BYTE *) & ascc);
if (hres == S_OK && pin_mediatype->mediatype) {
GstCaps *mediacaps = NULL;
if (!caps)
caps = gst_caps_new_empty ();
if (gst_dshow_check_mediatype (pin_mediatype->mediatype, MEDIASUBTYPE_PCM,
FORMAT_WaveFormatEx)) {
GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN;
WAVEFORMATEX *wavformat =
(WAVEFORMATEX *) pin_mediatype->mediatype->pbFormat;
switch (wavformat->wFormatTag) {
case WAVE_FORMAT_PCM:
format = gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, wavformat->wBitsPerSample, wavformat->wBitsPerSample);
break;
default:
break;
}
if (format != GST_AUDIO_FORMAT_UNKNOWN) {
GstAudioInfo info;
gst_audio_info_init(&info);
gst_audio_info_set_format(&info,
format,
wavformat->nSamplesPerSec,
wavformat->nChannels,
NULL);
mediacaps = gst_audio_info_to_caps(&info);
}
if (mediacaps) {
src->pins_mediatypes =
g_list_append (src->pins_mediatypes, pin_mediatype);
gst_caps_append (caps, mediacaps);
} else {
gst_dshow_free_pin_mediatype (pin_mediatype);
}
} else {
gst_dshow_free_pin_mediatype (pin_mediatype);
}
} else {
gst_dshow_free_pin_mediatype (pin_mediatype);
}
}
if (caps && gst_caps_is_empty (caps)) {
gst_caps_unref (caps);
caps = NULL;
}
return caps;
}
static GstFlowReturn
gst_speex_dec_parse_header (GstSpeexDec * dec, GstBuffer * buf)
{
SpeexHeader *header;
GstMapInfo map;
GstAudioInfo info;
static const GstAudioChannelPosition chan_pos[2][2] = {
{GST_AUDIO_CHANNEL_POSITION_MONO},
{GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}
};
/* get the header */
gst_buffer_map (buf, &map, GST_MAP_READ);
header = speex_packet_to_header ((gchar *) map.data, map.size);
gst_buffer_unmap (buf, &map);
if (!header)
goto no_header;
if (dec->header) {
GST_DEBUG_OBJECT (dec, "Replacing speex-header, resetting state");
gst_speex_dec_reset (dec);
}
dec->header = header;
if (dec->header->mode >= SPEEX_NB_MODES || dec->header->mode < 0)
goto mode_too_old;
dec->mode = speex_lib_get_mode (dec->header->mode);
/* initialize the decoder */
dec->state = speex_decoder_init (dec->mode);
if (!dec->state)
goto init_failed;
speex_decoder_ctl (dec->state, SPEEX_SET_ENH, &dec->enh);
speex_decoder_ctl (dec->state, SPEEX_GET_FRAME_SIZE, &dec->frame_size);
if (dec->header->nb_channels != 1) {
dec->stereo = speex_stereo_state_init ();
dec->callback.callback_id = SPEEX_INBAND_STEREO;
dec->callback.func = speex_std_stereo_request_handler;
dec->callback.data = dec->stereo;
speex_decoder_ctl (dec->state, SPEEX_SET_HANDLER, &dec->callback);
}
speex_decoder_ctl (dec->state, SPEEX_SET_SAMPLING_RATE, &dec->header->rate);
dec->frame_duration = gst_util_uint64_scale_int (dec->frame_size,
GST_SECOND, dec->header->rate);
speex_bits_init (&dec->bits);
/* set caps */
gst_audio_info_init (&info);
gst_audio_info_set_format (&info,
GST_AUDIO_FORMAT_S16,
dec->header->rate,
dec->header->nb_channels, chan_pos[dec->header->nb_channels - 1]);
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info))
goto nego_failed;
return GST_FLOW_OK;
/* ERRORS */
no_header:
{
GST_INFO_OBJECT (dec, "couldn't read header");
return GST_FLOW_NO_HEADER;
}
mode_too_old:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL),
("Mode number %d does not (yet/any longer) exist in this version",
dec->header->mode));
return GST_FLOW_ERROR;
}
init_failed:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("couldn't initialize decoder"));
return GST_FLOW_ERROR;
}
nego_failed:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("couldn't negotiate format"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
/* Converts an AudioStreamBasicDescription to preferred caps.
*
* These caps will indicate the AU element's canonical format, which won't
* make Core Audio resample nor convert.
*
* NOTE ON MULTI-CHANNEL AUDIO:
*
* If layout is not NULL, resulting caps will only include the subset
* of channels supported by GStreamer. If the Core Audio layout contained
* ANY positioned channels, then ONLY positioned channels will be included
* in the resulting caps. Otherwise, resulting caps will be unpositioned,
* and include only unpositioned channels.
* (Channels with unsupported AudioChannelLabel will be skipped either way.)
*
* Naturally, the number of channels indicated by 'channels' can be lower
* than the AU element's total number of channels.
*/
GstCaps *
gst_core_audio_asbd_to_caps (AudioStreamBasicDescription * asbd,
AudioChannelLayout * layout)
{
GstAudioInfo info;
GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN;
guint rate, channels, bps, endianness;
guint64 channel_mask;
gboolean sign, interleaved;
GstAudioChannelPosition pos[GST_OSX_AUDIO_MAX_CHANNEL];
if (asbd->mFormatID != kAudioFormatLinearPCM) {
GST_WARNING ("Only linear PCM is supported");
goto error;
}
if (!(asbd->mFormatFlags & kAudioFormatFlagIsPacked)) {
GST_WARNING ("Only packed formats supported");
goto error;
}
if (asbd->mFormatFlags & kLinearPCMFormatFlagsSampleFractionMask) {
GST_WARNING ("Fixed point audio is unsupported");
goto error;
}
rate = asbd->mSampleRate;
if (rate == kAudioStreamAnyRate) {
GST_WARNING ("No sample rate");
goto error;
}
bps = asbd->mBitsPerChannel;
endianness = asbd->mFormatFlags & kAudioFormatFlagIsBigEndian ?
G_BIG_ENDIAN : G_LITTLE_ENDIAN;
sign = asbd->mFormatID & kAudioFormatFlagIsSignedInteger ? TRUE : FALSE;
interleaved = asbd->mFormatFlags & kAudioFormatFlagIsNonInterleaved ?
TRUE : FALSE;
if (asbd->mFormatFlags & kAudioFormatFlagIsFloat) {
if (bps == 32) {
if (endianness == G_LITTLE_ENDIAN)
format = GST_AUDIO_FORMAT_F32LE;
else
format = GST_AUDIO_FORMAT_F32BE;
} else if (bps == 64) {
if (endianness == G_LITTLE_ENDIAN)
format = GST_AUDIO_FORMAT_F64LE;
else
format = GST_AUDIO_FORMAT_F64BE;
}
} else {
format = gst_audio_format_build_integer (sign, endianness, bps, bps);
}
if (format == GST_AUDIO_FORMAT_UNKNOWN) {
GST_WARNING ("Unsupported sample format");
goto error;
}
if (layout) {
if (!gst_core_audio_parse_channel_layout (layout, &channels, &channel_mask,
pos)) {
GST_WARNING
("Failed to parse channel layout, best effort channels layout mapping will be used");
layout = NULL;
}
}
if (layout) {
/* The AU can have arbitrary channel order, but we're using GstAudioInfo
* which supports only the GStreamer channel order.
* Also, we're eventually producing caps, which only have channel-mask
* (whose implied order is the GStreamer channel order). */
gst_audio_channel_positions_to_valid_order (pos, channels);
gst_audio_info_set_format (&info, format, rate, channels, pos);
} else {
channels = MIN (asbd->mChannelsPerFrame, GST_OSX_AUDIO_MAX_CHANNEL);
gst_audio_info_set_format (&info, format, rate, channels, NULL);
}
return gst_audio_info_to_caps (&info);
error:
return NULL;
}
static gboolean
gst_raw_audio_parse_config_to_caps (GstRawAudioParse * raw_audio_parse,
GstCaps ** caps, GstRawAudioParseConfig * config)
{
gboolean ret = TRUE;
GstAudioChannelPosition *channel_positions;
g_assert (caps != NULL);
if (config->bpf == 0) {
GST_ERROR_OBJECT (raw_audio_parse,
"cannot convert config to caps - config not filled with valid values");
*caps = NULL;
return FALSE;
}
channel_positions =
config->needs_channel_reordering ? &(config->
reordered_channel_positions[0]) : &(config->channel_positions[0]);
switch (config->format) {
case GST_RAW_AUDIO_PARSE_FORMAT_PCM:
{
GstAudioInfo info;
gst_audio_info_init (&info);
gst_audio_info_set_format (&info,
config->pcm_format,
config->sample_rate, config->num_channels, channel_positions);
*caps = gst_audio_info_to_caps (&info);
break;
}
case GST_RAW_AUDIO_PARSE_FORMAT_ALAW:
case GST_RAW_AUDIO_PARSE_FORMAT_MULAW:
{
guint64 channel_mask;
if (!gst_audio_channel_positions_to_mask (channel_positions,
config->num_channels, TRUE, &channel_mask)) {
GST_ERROR_OBJECT (raw_audio_parse, "invalid channel positions");
ret = FALSE;
break;
}
*caps = gst_caps_new_simple (
(config->format ==
GST_RAW_AUDIO_PARSE_FORMAT_ALAW) ? "audio/x-alaw" :
"audio/x-mulaw", "rate", G_TYPE_INT, config->sample_rate, "channels",
G_TYPE_INT, config->num_channels, "channel-mask", GST_TYPE_BITMASK,
channel_mask, NULL);
break;
}
default:
g_assert_not_reached ();
ret = FALSE;
}
if (!ret)
*caps = NULL;
return ret;
}
static FLAC__StreamDecoderWriteStatus
gst_flac_dec_write (GstFlacDec * flacdec, const FLAC__Frame * frame,
const FLAC__int32 * const buffer[])
{
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
guint depth = frame->header.bits_per_sample;
guint width, gdepth;
guint sample_rate = frame->header.sample_rate;
guint channels = frame->header.channels;
guint samples = frame->header.blocksize;
guint j, i;
GstMapInfo map;
gboolean caps_changed;
GST_LOG_OBJECT (flacdec, "samples in frame header: %d", samples);
if (depth == 0) {
if (flacdec->depth < 4 || flacdec->depth > 32) {
GST_ERROR_OBJECT (flacdec, "unsupported depth %d from STREAMINFO",
flacdec->depth);
ret = GST_FLOW_ERROR;
goto done;
}
depth = flacdec->depth;
}
switch (depth) {
case 8:
gdepth = width = 8;
break;
case 12:
case 16:
gdepth = width = 16;
break;
case 20:
case 24:
gdepth = 24;
width = 32;
break;
case 32:
gdepth = width = 32;
break;
default:
GST_ERROR_OBJECT (flacdec, "unsupported depth %d", depth);
ret = GST_FLOW_ERROR;
goto done;
}
if (sample_rate == 0) {
if (flacdec->info.rate != 0) {
sample_rate = flacdec->info.rate;
} else {
GST_ERROR_OBJECT (flacdec, "unknown sample rate");
ret = GST_FLOW_ERROR;
goto done;
}
}
caps_changed = (sample_rate != GST_AUDIO_INFO_RATE (&flacdec->info))
|| (width != GST_AUDIO_INFO_WIDTH (&flacdec->info))
|| (gdepth != GST_AUDIO_INFO_DEPTH (&flacdec->info))
|| (channels != GST_AUDIO_INFO_CHANNELS (&flacdec->info));
if (caps_changed
|| !gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (flacdec))) {
GST_DEBUG_OBJECT (flacdec, "Negotiating %d Hz @ %d channels", sample_rate,
channels);
gst_audio_info_set_format (&flacdec->info,
gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, width, gdepth),
sample_rate, channels, NULL);
memcpy (flacdec->info.position,
channel_positions[flacdec->info.channels - 1],
sizeof (GstAudioChannelPosition) * flacdec->info.channels);
gst_audio_channel_positions_to_valid_order (flacdec->info.position,
flacdec->info.channels);
/* Note: we create the inverse reordering map here */
gst_audio_get_channel_reorder_map (flacdec->info.channels,
flacdec->info.position, channel_positions[flacdec->info.channels - 1],
flacdec->channel_reorder_map);
flacdec->depth = depth;
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (flacdec),
&flacdec->info);
}
outbuf =
gst_buffer_new_allocate (NULL, samples * channels * (width / 8), NULL);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
if (width == 8) {
gint8 *outbuffer = (gint8 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint8) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint8) buffer[reorder_map[j]][i];
}
}
}
} else if (width == 16) {
gint16 *outbuffer = (gint16 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint16) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint16) buffer[reorder_map[j]][i];
}
}
}
} else if (width == 32) {
gint32 *outbuffer = (gint32 *) map.data;
gint *reorder_map = flacdec->channel_reorder_map;
if (gdepth != depth) {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ =
(gint32) (buffer[reorder_map[j]][i] << (gdepth - depth));
}
}
} else {
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*outbuffer++ = (gint32) buffer[reorder_map[j]][i];
}
}
}
} else {
g_assert_not_reached ();
}
gst_buffer_unmap (outbuf, &map);
GST_DEBUG_OBJECT (flacdec, "pushing %d samples", samples);
if (flacdec->error_count)
flacdec->error_count--;
ret = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (flacdec), outbuf, 1);
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
GST_DEBUG_OBJECT (flacdec, "finish_frame flow %s", gst_flow_get_name (ret));
}
done:
/* we act on the flow return value later in the handle_frame function, as we
* don't want to mess up the internal decoder state by returning ABORT when
* the error is in fact non-fatal (like a pad in flushing mode) and we want
* to continue later. So just pretend everything's dandy and act later. */
flacdec->last_flow = ret;
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
static gpointer
push_abuffers (gpointer data)
{
GstSegment segment;
GstPad *pad = data;
gint i, j, k;
GstClockTime timestamp = 0;
GstAudioInfo info;
GstCaps *caps;
guint buf_size = 1000;
if (audiodelay)
g_usleep (2000);
if (early_video)
timestamp = 50 * GST_MSECOND;
gst_pad_send_event (pad, gst_event_new_stream_start ("test"));
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S8, buf_size, channels,
NULL);
caps = gst_audio_info_to_caps (&info);
gst_pad_send_event (pad, gst_event_new_caps (caps));
gst_caps_unref (caps);
gst_segment_init (&segment, GST_FORMAT_TIME);
gst_pad_send_event (pad, gst_event_new_segment (&segment));
for (i = 0; i < n_abuffers; i++) {
GstBuffer *buf = gst_buffer_new_and_alloc (channels * buf_size);
if (per_channel) {
GstMapInfo map;
guint8 *in_data;
gst_buffer_map (buf, &map, GST_MAP_WRITE);
in_data = map.data;
for (j = 0; j < buf_size; j++) {
for (k = 0; k < channels; k++) {
in_data[j * channels + k] = fill_value_per_channel[k];
}
}
gst_buffer_unmap (buf, &map);
} else {
gst_buffer_memset (buf, 0, fill_value, channels * buf_size);
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
timestamp += 1 * GST_SECOND;
if (audio_drift)
timestamp += 50 * GST_MSECOND;
else if (i == 4 && audio_nondiscont)
timestamp += 30 * GST_MSECOND;
GST_BUFFER_DURATION (buf) = timestamp - GST_BUFFER_TIMESTAMP (buf);
fail_unless (gst_pad_chain (pad, buf) == GST_FLOW_OK);
}
gst_pad_send_event (pad, gst_event_new_eos ());
return NULL;
}
