static void
gst_dtsdec_init (GstDtsDec * dtsdec)
{
dtsdec->request_channels = DCA_CHANNEL;
dtsdec->dynamic_range_compression = FALSE;
/* retrieve and intercept base class chain.
* Quite HACKish, but that's dvd specs for you,
* since one buffer needs to be split into 2 frames */
dtsdec->base_chain = GST_PAD_CHAINFUNC (GST_AUDIO_DECODER_SINK_PAD (dtsdec));
gst_pad_set_chain_function (GST_AUDIO_DECODER_SINK_PAD (dtsdec),
GST_DEBUG_FUNCPTR (gst_dtsdec_chain));
}
static void
gst_a52dec_init (GstA52Dec * a52dec)
{
a52dec->request_channels = A52_CHANNEL;
a52dec->dynamic_range_compression = FALSE;
a52dec->state = NULL;
a52dec->samples = NULL;
/* retrieve and intercept base class chain.
* Quite HACKish, but that's dvd specs/caps for you,
* since one buffer needs to be split into 2 frames */
a52dec->base_chain = GST_PAD_CHAINFUNC (GST_AUDIO_DECODER_SINK_PAD (a52dec));
gst_pad_set_chain_function (GST_AUDIO_DECODER_SINK_PAD (a52dec),
GST_DEBUG_FUNCPTR (gst_a52dec_chain));
}
static void
gst_mulawdec_init (GstMuLawDec * mulawdec)
{
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (mulawdec), TRUE);
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(mulawdec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (mulawdec));
}
static void
gst_sbc_dec_init (GstSbcDec * dec)
{
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(dec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
dec->samples_per_frame = 0;
dec->frame_len = 0;
}
static void
gst_speex_dec_init (GstSpeexDec * dec)
{
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(dec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
dec->enh = DEFAULT_ENH;
gst_speex_dec_reset (dec);
}
static void
gst_wavpack_dec_init (GstWavpackDec * dec)
{
dec->context = NULL;
dec->stream_reader = gst_wavpack_stream_reader_new ();
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(dec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
gst_wavpack_dec_reset (dec);
}
static void
gst_opus_dec_init (GstOpusDec * dec)
{
dec->use_inband_fec = FALSE;
dec->apply_gain = DEFAULT_APPLY_GAIN;
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(dec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
gst_opus_dec_reset (dec);
}
static GstFlowReturn
vorbis_dec_handle_header_caps (GstVorbisDec * vd)
{
GstFlowReturn result = GST_FLOW_OK;
GstCaps *caps;
GstStructure *s = NULL;
const GValue *array = NULL;
caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (vd));
if (caps)
s = gst_caps_get_structure (caps, 0);
if (s)
array = gst_structure_get_value (s, "streamheader");
if (caps)
gst_caps_unref (caps);
if (array && (gst_value_array_get_size (array) >= MIN_NUM_HEADERS)) {
const GValue *value = NULL;
GstBuffer *buf = NULL;
gint i = 0;
while (result == GST_FLOW_OK && i < gst_value_array_get_size (array)) {
value = gst_value_array_get_value (array, i);
buf = gst_value_get_buffer (value);
if (!buf)
goto null_buffer;
result = vorbis_dec_handle_header_buffer (vd, buf);
i++;
}
} else
goto array_error;
done:
return (result != GST_FLOW_OK ? GST_FLOW_NOT_NEGOTIATED : GST_FLOW_OK);
/* ERRORS */
array_error:
{
GST_WARNING_OBJECT (vd, "streamheader array not found");
result = GST_FLOW_ERROR;
goto done;
}
null_buffer:
{
GST_WARNING_OBJECT (vd, "streamheader with null buffer received");
result = GST_FLOW_ERROR;
goto done;
}
}
static void
gst_wavpack_dec_post_tags (GstWavpackDec * dec)
{
GstTagList *list;
GstFormat format_time = GST_FORMAT_TIME, format_bytes = GST_FORMAT_BYTES;
gint64 duration, size;
/* try to estimate the average bitrate */
if (gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
format_bytes, &size) &&
gst_pad_peer_query_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
format_time, &duration) && size > 0 && duration > 0) {
guint64 bitrate;
list = gst_tag_list_new_empty ();
bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration);
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
(guint) bitrate, NULL);
gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (dec), list,
GST_TAG_MERGE_REPLACE);
}
}
static gboolean
gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
GstOpusDec *dec = GST_OPUS_DEC (bdec);
gboolean ret = TRUE;
GstStructure *s;
const GValue *streamheader;
GstCaps *old_caps;
GST_DEBUG_OBJECT (dec, "set_format: %" GST_PTR_FORMAT, caps);
if ((old_caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (bdec)))) {
if (gst_caps_is_equal (caps, old_caps)) {
gst_caps_unref (old_caps);
GST_DEBUG_OBJECT (dec, "caps didn't change");
goto done;
}
GST_DEBUG_OBJECT (dec, "caps have changed, resetting decoder");
gst_opus_dec_reset (dec);
gst_caps_unref (old_caps);
}
s = gst_caps_get_structure (caps, 0);
if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
gst_value_array_get_size (streamheader) >= 2) {
const GValue *header, *vorbiscomment;
GstBuffer *buf;
GstFlowReturn res = GST_FLOW_OK;
header = gst_value_array_get_value (streamheader, 0);
if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
buf = gst_value_get_buffer (header);
res = gst_opus_dec_parse_header (dec, buf);
if (res != GST_FLOW_OK)
goto done;
gst_buffer_replace (&dec->streamheader, buf);
}
vorbiscomment = gst_value_array_get_value (streamheader, 1);
if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
buf = gst_value_get_buffer (vorbiscomment);
res = gst_opus_dec_parse_comments (dec, buf);
if (res != GST_FLOW_OK)
goto done;
gst_buffer_replace (&dec->vorbiscomment, buf);
}
} else {
/* defaults if not in the caps */
dec->n_channels = 2;
dec->sample_rate = 48000;
gst_structure_get_int (s, "channels", &dec->n_channels);
gst_structure_get_int (s, "rate", &dec->sample_rate);
/* default stereo mapping */
dec->channel_mapping_family = 0;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
dec->n_streams = 1;
dec->n_stereo_streams = 1;
gst_opus_dec_negotiate (dec, NULL);
}
done:
return ret;
}
static GstFlowReturn
gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
GstBuffer * input_buffer)
{
GstMpg123AudioDec *mpg123_decoder;
int decode_error;
unsigned char *decoded_bytes;
size_t num_decoded_bytes;
GstFlowReturn retval;
mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
g_assert (mpg123_decoder->handle != NULL);
/* The actual decoding */
{
/* feed input data (if there is any) */
if (G_LIKELY (input_buffer != NULL)) {
GstMapInfo info;
if (gst_buffer_map (input_buffer, &info, GST_MAP_READ)) {
mpg123_feed (mpg123_decoder->handle, info.data, info.size);
gst_buffer_unmap (input_buffer, &info);
} else {
GST_ERROR_OBJECT (mpg123_decoder, "gst_memory_map() failed");
return GST_FLOW_ERROR;
}
}
/* Try to decode a frame */
decoded_bytes = NULL;
num_decoded_bytes = 0;
decode_error = mpg123_decode_frame (mpg123_decoder->handle,
&mpg123_decoder->frame_offset, &decoded_bytes, &num_decoded_bytes);
}
retval = GST_FLOW_OK;
switch (decode_error) {
case MPG123_NEW_FORMAT:
/* As mentioned in gst_mpg123_audio_dec_set_format(), the next audioinfo
* is not set immediately; instead, the code waits for mpg123 to take
* note of the new format, and then sets the audioinfo. This fixes glitches
* with mp3s containing several format headers (for example, first half
* using 44.1kHz, second half 32 kHz) */
GST_LOG_OBJECT (dec,
"mpg123 reported a new format -> setting next srccaps");
gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
num_decoded_bytes);
/* If there is a next audioinfo, use it, then set has_next_audioinfo to
* FALSE, to make sure gst_audio_decoder_set_output_format() isn't called
* again until set_format is called by the base class */
if (mpg123_decoder->has_next_audioinfo) {
if (!gst_audio_decoder_set_output_format (dec,
&(mpg123_decoder->next_audioinfo))) {
GST_WARNING_OBJECT (dec, "Unable to set output format");
retval = GST_FLOW_NOT_NEGOTIATED;
}
mpg123_decoder->has_next_audioinfo = FALSE;
}
break;
case MPG123_NEED_MORE:
case MPG123_OK:
retval = gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder,
decoded_bytes, num_decoded_bytes);
break;
case MPG123_DONE:
/* If this happens, then the upstream parser somehow missed the ending
* of the bitstream */
GST_LOG_OBJECT (dec, "mpg123 is done decoding");
gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
num_decoded_bytes);
retval = GST_FLOW_EOS;
break;
default:
{
/* Anything else is considered an error */
int errcode;
switch (decode_error) {
case MPG123_ERR:
errcode = mpg123_errcode (mpg123_decoder->handle);
break;
default:
errcode = decode_error;
}
switch (errcode) {
case MPG123_BAD_OUTFORMAT:{
GstCaps *input_caps =
gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (dec));
GST_ELEMENT_ERROR (dec, STREAM, FORMAT, (NULL),
("Output sample format could not be used when trying to decode frame. "
"This is typically caused when the input caps (often the sample "
"rate) do not match the actual format of the audio data. "
"Input caps: %" GST_PTR_FORMAT, input_caps)
);
gst_caps_unref (input_caps);
break;
}
default:{
char const *errmsg = mpg123_plain_strerror (errcode);
GST_ERROR_OBJECT (dec, "Reported error: %s", errmsg);
}
}
retval = GST_FLOW_ERROR;
}
}
return retval;
}
static gboolean
gst_opus_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
GstOpusDec *dec = GST_OPUS_DEC (bdec);
gboolean ret = TRUE;
GstStructure *s;
const GValue *streamheader;
GstCaps *old_caps;
GST_DEBUG_OBJECT (dec, "set_format: %" GST_PTR_FORMAT, caps);
if ((old_caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (bdec)))) {
if (gst_caps_is_equal (caps, old_caps)) {
gst_caps_unref (old_caps);
GST_DEBUG_OBJECT (dec, "caps didn't change");
goto done;
}
GST_DEBUG_OBJECT (dec, "caps have changed, resetting decoder");
gst_opus_dec_reset (dec);
gst_caps_unref (old_caps);
}
s = gst_caps_get_structure (caps, 0);
if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
gst_value_array_get_size (streamheader) >= 2) {
const GValue *header, *vorbiscomment;
GstBuffer *buf;
GstFlowReturn res = GST_FLOW_OK;
header = gst_value_array_get_value (streamheader, 0);
if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
buf = gst_value_get_buffer (header);
res = gst_opus_dec_parse_header (dec, buf);
if (res != GST_FLOW_OK) {
ret = FALSE;
goto done;
}
gst_buffer_replace (&dec->streamheader, buf);
}
vorbiscomment = gst_value_array_get_value (streamheader, 1);
if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
buf = gst_value_get_buffer (vorbiscomment);
res = gst_opus_dec_parse_comments (dec, buf);
if (res != GST_FLOW_OK) {
ret = FALSE;
goto done;
}
gst_buffer_replace (&dec->vorbiscomment, buf);
}
} else {
const GstAudioChannelPosition *posn = NULL;
if (!gst_codec_utils_opus_parse_caps (caps, &dec->sample_rate,
&dec->n_channels, &dec->channel_mapping_family, &dec->n_streams,
&dec->n_stereo_streams, dec->channel_mapping)) {
ret = FALSE;
goto done;
}
if (dec->channel_mapping_family == 1 && dec->n_channels <= 8)
posn = gst_opus_channel_positions[dec->n_channels - 1];
if (!gst_opus_dec_negotiate (dec, posn))
return FALSE;
}
done:
return ret;
}
static gboolean
gst_droidadec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
{
GstDroidADec *dec = GST_DROIDADEC (decoder);
GstStructure *str = gst_caps_get_structure (caps, 0);
const GValue *value = gst_structure_get_value (str, "codec_data");
GstBuffer *codec_data = value ? gst_value_get_buffer (value) : NULL;
/*
* destroying the droidmedia codec here will cause stagefright to call abort.
* That is why we create it after we are sure that everything is correct
*/
GST_DEBUG_OBJECT (dec, "set format %" GST_PTR_FORMAT, caps);
if (dec->codec) {
/* If we get a format change then we stop */
GstCaps *current =
gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (decoder));
gboolean equal = gst_caps_is_equal_fixed (caps, current);
gst_caps_unref (current);
GST_DEBUG_OBJECT (dec, "new format is similar to old format? %d", equal);
if (!equal) {
GST_ELEMENT_ERROR (dec, LIBRARY, SETTINGS, (NULL),
("codec already configured"));
}
return equal;
}
dec->codec_type =
gst_droid_codec_new_from_caps (caps, GST_DROID_CODEC_DECODER_AUDIO);
if (!dec->codec_type) {
GST_ELEMENT_ERROR (dec, LIBRARY, FAILED, (NULL),
("Unknown codec type for caps %" GST_PTR_FORMAT, caps));
return FALSE;
}
if (!gst_structure_get_int (str, "channels", &dec->channels) ||
!gst_structure_get_int (str, "rate", &dec->rate)) {
GST_ELEMENT_ERROR (dec, STREAM, FORMAT, (NULL),
("Failed to parse caps %" GST_PTR_FORMAT, caps));
return FALSE;
}
GST_INFO_OBJECT (dec, "configuring decoder. rate=%d, channels=%d", dec->rate,
dec->channels);
gst_buffer_replace (&dec->codec_data, codec_data);
/* handle_frame will create the codec */
dec->dirty = TRUE;
dec->spf = gst_droid_codec_get_samples_per_frane (caps);
GST_INFO_OBJECT (dec, "samples per frame: %d", dec->spf);
return TRUE;
}