static GstFlowReturn
gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
{
GstAmrnbEnc *amrnbenc;
GstFlowReturn ret;
GstBuffer *out;
GstMapInfo in_map, out_map;
gsize out_size;
amrnbenc = GST_AMRNBENC (enc);
g_return_val_if_fail (amrnbenc->handle, GST_FLOW_FLUSHING);
/* we don't deal with squeezing remnants, so simply discard those */
if (G_UNLIKELY (buffer == NULL)) {
GST_DEBUG_OBJECT (amrnbenc, "no data");
return GST_FLOW_OK;
}
gst_buffer_map (buffer, &in_map, GST_MAP_READ);
if (G_UNLIKELY (in_map.size < 320)) {
gst_buffer_unmap (buffer, &in_map);
GST_DEBUG_OBJECT (amrnbenc, "discarding trailing data of %" G_GSIZE_FORMAT
" bytes", in_map.size);
return gst_audio_encoder_finish_frame (enc, NULL, -1);
}
/* get output, max size is 32 */
out = gst_buffer_new_and_alloc (32);
/* AMR encoder actually writes into the source data buffers it gets */
/* should be able to handle that with what we are given */
gst_buffer_map (out, &out_map, GST_MAP_WRITE);
/* encode */
out_size =
Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
(short *) in_map.data, out_map.data, 0);
gst_buffer_unmap (out, &out_map);
gst_buffer_resize (out, 0, out_size);
gst_buffer_unmap (buffer, &in_map);
GST_LOG_OBJECT (amrnbenc, "output data size %" G_GSIZE_FORMAT, out_size);
if (out_size) {
ret = gst_audio_encoder_finish_frame (enc, out, 160);
} else {
/* should not happen (without dtx or so at least) */
GST_WARNING_OBJECT (amrnbenc, "no encoded data; discarding input");
gst_buffer_unref (out);
ret = gst_audio_encoder_finish_frame (enc, NULL, -1);
}
return ret;
}
static GstFlowReturn
gst_two_lame_flush_full (GstTwoLame * lame, gboolean push)
{
GstBuffer *buf;
GstMapInfo map;
gint size;
GstFlowReturn result = GST_FLOW_OK;
if (!lame->glopts)
return GST_FLOW_OK;
buf = gst_buffer_new_and_alloc (16384);
gst_buffer_map (buf, &map, GST_MAP_WRITE);
size = twolame_encode_flush (lame->glopts, map.data, 16384);
gst_buffer_unmap (buf, &map);
if (size > 0 && push) {
gst_buffer_set_size (buf, size);
GST_DEBUG_OBJECT (lame, "pushing final packet of %u bytes", size);
result = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (lame), buf, -1);
} else {
GST_DEBUG_OBJECT (lame, "no final packet (size=%d, push=%d)", size, push);
gst_buffer_unref (buf);
result = GST_FLOW_OK;
}
return result;
}
static GstFlowReturn
gst_audio_encoder_tester_handle_frame (GstAudioEncoder * enc,
GstBuffer * buffer)
{
guint8 *data;
GstMapInfo map;
guint64 input_num;
GstBuffer *output_buffer;
if (buffer == NULL)
return GST_FLOW_OK;
gst_buffer_map (buffer, &map, GST_MAP_READ);
input_num = *((guint64 *) map.data);
gst_buffer_unmap (buffer, &map);
data = g_malloc (sizeof (guint64));
*(guint64 *) data = input_num;
output_buffer = gst_buffer_new_wrapped (data, sizeof (guint64));
GST_BUFFER_PTS (output_buffer) = GST_BUFFER_PTS (buffer);
GST_BUFFER_DURATION (output_buffer) = GST_BUFFER_DURATION (buffer);
return gst_audio_encoder_finish_frame (enc, output_buffer, TEST_AUDIO_RATE);
}
static GstFlowReturn
gst_vorbis_enc_output_buffers (GstVorbisEnc * vorbisenc)
{
GstFlowReturn ret;
/* vorbis does some data preanalysis, then divides up blocks for
more involved (potentially parallel) processing. Get a single
block for encoding now */
while (vorbis_analysis_blockout (&vorbisenc->vd, &vorbisenc->vb) == 1) {
ogg_packet op;
GST_LOG_OBJECT (vorbisenc, "analysed to a block");
/* analysis */
vorbis_analysis (&vorbisenc->vb, NULL);
vorbis_bitrate_addblock (&vorbisenc->vb);
while (vorbis_bitrate_flushpacket (&vorbisenc->vd, &op)) {
GstBuffer *buf;
if (op.e_o_s) {
GstAudioEncoder *enc = GST_AUDIO_ENCODER (vorbisenc);
GstClockTime duration;
GST_DEBUG_OBJECT (vorbisenc, "Got EOS packet from libvorbis");
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
if (!GST_CLOCK_TIME_IS_VALID (enc->output_segment.stop)) {
GST_DEBUG_OBJECT (vorbisenc,
"Output segment has no end time, setting");
duration =
gst_util_uint64_scale (op.granulepos, GST_SECOND,
vorbisenc->frequency);
enc->output_segment.stop = enc->output_segment.start + duration;
GST_DEBUG_OBJECT (enc, "new output segment %" GST_SEGMENT_FORMAT,
&enc->output_segment);
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
gst_event_new_segment (&enc->output_segment));
}
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
}
GST_LOG_OBJECT (vorbisenc, "pushing out a data packet");
buf =
gst_audio_encoder_allocate_output_buffer (GST_AUDIO_ENCODER
(vorbisenc), op.bytes);
gst_buffer_fill (buf, 0, op.packet, op.bytes);
/* tracking granulepos should tell us samples accounted for */
ret =
gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER
(vorbisenc), buf, op.granulepos - vorbisenc->samples_out);
vorbisenc->samples_out = op.granulepos;
if (ret != GST_FLOW_OK)
return ret;
}
}
return GST_FLOW_OK;
}
static GstFlowReturn
gst_gsmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
{
GstGSMEnc *gsmenc;
gsm_signal *data;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
GstMapInfo map, omap;
gsmenc = GST_GSMENC (benc);
/* we don't deal with squeezing remnants, so simply discard those */
if (G_UNLIKELY (buffer == NULL)) {
GST_DEBUG_OBJECT (gsmenc, "no data");
goto done;
}
gst_buffer_map (buffer, &map, GST_MAP_READ);
if (G_UNLIKELY (map.size < 320)) {
GST_DEBUG_OBJECT (gsmenc, "discarding trailing data %d", (gint) map.size);
gst_buffer_unmap (buffer, &map);
ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
goto done;
}
outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
/* encode 160 16-bit samples into 33 bytes */
data = (gsm_signal *) map.data;
gsm_encode (gsmenc->state, data, (gsm_byte *) omap.data);
GST_LOG_OBJECT (gsmenc, "encoded to %d bytes", (gint) omap.size);
gst_buffer_unmap (buffer, &map);
gst_buffer_unmap (buffer, &omap);
ret = gst_audio_encoder_finish_frame (benc, outbuf, 160);
done:
return ret;
}
static GstFlowReturn
gst_mulawenc_handle_frame (GstAudioEncoder * audioenc, GstBuffer * buffer)
{
GstMuLawEnc *mulawenc;
GstMapInfo inmap, outmap;
gint16 *linear_data;
gsize linear_size;
guint8 *mulaw_data;
guint mulaw_size;
GstBuffer *outbuf;
GstFlowReturn ret;
if (!buffer) {
ret = GST_FLOW_OK;
goto done;
}
mulawenc = GST_MULAWENC (audioenc);
if (!mulawenc->rate || !mulawenc->channels)
goto not_negotiated;
gst_buffer_map (buffer, &inmap, GST_MAP_READ);
linear_data = (gint16 *) inmap.data;
linear_size = inmap.size;
mulaw_size = linear_size / 2;
outbuf = gst_audio_encoder_allocate_output_buffer (audioenc, mulaw_size);
g_assert (outbuf);
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
mulaw_data = outmap.data;
mulaw_encode (linear_data, mulaw_data, mulaw_size);
gst_buffer_unmap (outbuf, &outmap);
gst_buffer_unmap (buffer, &inmap);
ret = gst_audio_encoder_finish_frame (audioenc, outbuf, -1);
done:
return ret;
not_negotiated:
{
GST_DEBUG_OBJECT (mulawenc, "no format negotiated");
ret = GST_FLOW_NOT_NEGOTIATED;
goto done;
}
}
static GstFlowReturn
gst_vorbis_enc_output_buffers (GstVorbisEnc * vorbisenc)
{
GstFlowReturn ret;
/* vorbis does some data preanalysis, then divides up blocks for
more involved (potentially parallel) processing. Get a single
block for encoding now */
while (vorbis_analysis_blockout (&vorbisenc->vd, &vorbisenc->vb) == 1) {
ogg_packet op;
GST_LOG_OBJECT (vorbisenc, "analysed to a block");
/* analysis */
vorbis_analysis (&vorbisenc->vb, NULL);
vorbis_bitrate_addblock (&vorbisenc->vb);
while (vorbis_bitrate_flushpacket (&vorbisenc->vd, &op)) {
GstBuffer *buf;
GST_LOG_OBJECT (vorbisenc, "pushing out a data packet");
buf =
gst_audio_encoder_allocate_output_buffer (GST_AUDIO_ENCODER
(vorbisenc), op.bytes);
gst_buffer_fill (buf, 0, op.packet, op.bytes);
/* tracking granulepos should tell us samples accounted for */
ret =
gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER
(vorbisenc), buf, op.granulepos - vorbisenc->samples_out);
vorbisenc->samples_out = op.granulepos;
if (ret != GST_FLOW_OK)
return ret;
}
}
return GST_FLOW_OK;
}
static GstFlowReturn
gst_celt_enc_encode (GstCeltEnc * enc, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
gint frame_size = enc->frame_size;
gint bytes = frame_size * 2 * enc->channels;
gint bytes_per_packet;
gint16 *data, *data0 = NULL;
gint outsize, size;
GstBuffer *outbuf;
GstMapInfo map, omap;
if (G_LIKELY (buf)) {
gst_buffer_map (buf, &map, GST_MAP_READ);
data = (gint16 *) map.data;
size = map.size;
if (G_UNLIKELY (map.size % bytes)) {
GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
size = ((size / bytes) + 1) * bytes;
data0 = g_malloc0 (size);
memcpy (data0, data, size);
data = data0;
}
} else {
GST_DEBUG_OBJECT (enc, "nothing to drain");
goto done;
}
frame_size = size / (2 * enc->channels);
if (enc->cbr) {
bytes_per_packet = (enc->bitrate * frame_size / enc->rate + 4) / 8;
} else {
bytes_per_packet = (enc->max_bitrate * frame_size / enc->rate + 4) / 8;
}
outbuf = gst_buffer_new_and_alloc (bytes_per_packet);
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)", frame_size, bytes);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
#ifdef HAVE_CELT_0_8
outsize =
celt_encode (enc->state, data, frame_size, omap.data, bytes_per_packet);
#else
outsize = celt_encode (enc->state, data, NULL, omap.data, bytes_per_packet);
#endif
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unmap (buf, &map);
if (outsize < 0) {
GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
("encoding failed: %d", outsize));
ret = GST_FLOW_ERROR;
goto done;
}
GST_DEBUG_OBJECT (enc, "encoding %d bytes", bytes);
ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc),
outbuf, frame_size);
done:
g_free (data0);
return ret;
}
static GstFlowReturn
gst_ffmpegaudenc_encode_audio (GstFFMpegAudEnc * ffmpegaudenc,
GstBuffer * buffer, gint * have_data)
{
GstAudioEncoder *enc;
AVCodecContext *ctx;
gint res;
GstFlowReturn ret;
GstAudioInfo *info;
AVPacket *pkt;
AVFrame *frame = ffmpegaudenc->frame;
gboolean planar;
gint nsamples = -1;
enc = GST_AUDIO_ENCODER (ffmpegaudenc);
ctx = ffmpegaudenc->context;
pkt = g_slice_new0 (AVPacket);
if (buffer != NULL) {
BufferInfo *buffer_info = g_slice_new0 (BufferInfo);
guint8 *audio_in;
guint in_size;
buffer_info->buffer = buffer;
gst_buffer_map (buffer, &buffer_info->map, GST_MAP_READ);
audio_in = buffer_info->map.data;
in_size = buffer_info->map.size;
GST_LOG_OBJECT (ffmpegaudenc, "encoding buffer %p size:%u", audio_in,
in_size);
info = gst_audio_encoder_get_audio_info (enc);
planar = av_sample_fmt_is_planar (ffmpegaudenc->context->sample_fmt);
frame->format = ffmpegaudenc->context->sample_fmt;
frame->sample_rate = ffmpegaudenc->context->sample_rate;
frame->channels = ffmpegaudenc->context->channels;
frame->channel_layout = ffmpegaudenc->context->channel_layout;
if (planar && info->channels > 1) {
gint channels;
gint i, j;
nsamples = frame->nb_samples = in_size / info->bpf;
channels = info->channels;
frame->buf[0] =
av_buffer_create (NULL, 0, buffer_info_free, buffer_info, 0);
if (info->channels > AV_NUM_DATA_POINTERS) {
buffer_info->ext_data_array = frame->extended_data =
av_malloc_array (info->channels, sizeof (uint8_t *));
} else {
frame->extended_data = frame->data;
}
buffer_info->ext_data = frame->extended_data[0] = av_malloc (in_size);
frame->linesize[0] = in_size / channels;
for (i = 1; i < channels; i++)
frame->extended_data[i] =
frame->extended_data[i - 1] + frame->linesize[0];
switch (info->finfo->width) {
case 8: {
const guint8 *idata = (const guint8 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint8 *) frame->extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
case 16: {
const guint16 *idata = (const guint16 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint16 *) frame->extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
case 32: {
const guint32 *idata = (const guint32 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint32 *) frame->extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
case 64: {
const guint64 *idata = (const guint64 *) audio_in;
for (i = 0; i < nsamples; i++) {
for (j = 0; j < channels; j++) {
((guint64 *) frame->extended_data[j])[i] = idata[j];
}
idata += channels;
}
break;
}
default:
g_assert_not_reached ();
break;
}
gst_buffer_unmap (buffer, &buffer_info->map);
gst_buffer_unref (buffer);
buffer_info->buffer = NULL;
} else {
frame->data[0] = audio_in;
frame->extended_data = frame->data;
frame->linesize[0] = in_size;
frame->nb_samples = nsamples = in_size / info->bpf;
frame->buf[0] =
av_buffer_create (NULL, 0, buffer_info_free, buffer_info, 0);
}
/* we have a frame to feed the encoder */
res = avcodec_encode_audio2 (ctx, pkt, frame, have_data);
av_frame_unref (frame);
} else {
GST_LOG_OBJECT (ffmpegaudenc, "draining");
/* flushing the encoder */
res = avcodec_encode_audio2 (ctx, pkt, NULL, have_data);
}
if (res < 0) {
char error_str[128] = { 0, };
g_slice_free (AVPacket, pkt);
av_strerror (res, error_str, sizeof (error_str));
GST_ERROR_OBJECT (enc, "Failed to encode buffer: %d - %s", res, error_str);
return GST_FLOW_OK;
}
GST_LOG_OBJECT (ffmpegaudenc, "got output size %d", res);
if (*have_data) {
GstBuffer *outbuf;
const AVCodec *codec;
GST_LOG_OBJECT (ffmpegaudenc, "pushing size %d", pkt->size);
outbuf =
gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY, pkt->data,
pkt->size, 0, pkt->size, pkt, gst_ffmpegaudenc_free_avpacket);
codec = ffmpegaudenc->context->codec;
if ((codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE) || !buffer) {
/* FIXME: Not really correct, as -1 means "all the samples we got
given so far", which may not be true depending on the codec,
but we have no way to know AFAICT */
ret = gst_audio_encoder_finish_frame (enc, outbuf, -1);
} else {
ret = gst_audio_encoder_finish_frame (enc, outbuf, nsamples);
}
} else {
GST_LOG_OBJECT (ffmpegaudenc, "no output produced");
g_slice_free (AVPacket, pkt);
ret = GST_FLOW_OK;
}
return ret;
}
static GstFlowReturn
gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
{
guint8 *bdata, *data, *mdata = NULL;
gsize bsize, size;
gsize bytes;
gint ret = GST_FLOW_OK;
gint outsize;
GstBuffer *outbuf;
g_mutex_lock (enc->property_lock);
bytes = enc->frame_samples * enc->n_channels * 2;
if (G_LIKELY (buf)) {
bdata = GST_BUFFER_DATA (buf);
bsize = GST_BUFFER_SIZE (buf);
if (G_UNLIKELY (bsize % bytes)) {
GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
size = ((bsize / bytes) + 1) * bytes;
mdata = g_malloc0 (size);
memcpy (mdata, bdata, bsize);
bdata = NULL;
data = mdata;
} else {
data = bdata;
size = bsize;
}
} else {
GST_DEBUG_OBJECT (enc, "nothing to drain");
goto done;
}
g_assert (size == bytes);
ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
GST_BUFFER_OFFSET_NONE, enc->max_payload_size * enc->n_channels,
GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
if (GST_FLOW_OK != ret)
goto done;
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
enc->frame_samples, (int) bytes);
outsize =
opus_multistream_encode (enc->state, (const gint16 *) data,
enc->frame_samples, GST_BUFFER_DATA (outbuf),
enc->max_payload_size * enc->n_channels);
if (outsize < 0) {
GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize);
ret = GST_FLOW_ERROR;
goto done;
} else if (outsize > enc->max_payload_size) {
GST_WARNING_OBJECT (enc,
"Encoded size %d is higher than max payload size (%d bytes)",
outsize, enc->max_payload_size);
ret = GST_FLOW_ERROR;
goto done;
}
GST_DEBUG_OBJECT (enc, "Output packet is %u bytes", outsize);
GST_BUFFER_SIZE (outbuf) = outsize;
ret =
gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), outbuf,
enc->frame_samples);
done:
g_mutex_unlock (enc->property_lock);
if (mdata)
g_free (mdata);
return ret;
}
static GstFlowReturn
gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
{
guint8 *bdata = NULL, *data, *mdata = NULL;
gsize bsize, size;
gsize bytes = enc->frame_samples * enc->n_channels * 2;
gint ret = GST_FLOW_OK;
GstMapInfo map;
GstMapInfo omap;
gint outsize;
GstBuffer *outbuf;
g_mutex_lock (enc->property_lock);
if (G_LIKELY (buf)) {
gst_buffer_map (buf, &map, GST_MAP_READ);
bdata = map.data;
bsize = map.size;
if (G_UNLIKELY (bsize % bytes)) {
GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
size = ((bsize / bytes) + 1) * bytes;
mdata = g_malloc0 (size);
memcpy (mdata, bdata, bsize);
data = mdata;
} else {
data = bdata;
size = bsize;
}
} else {
GST_DEBUG_OBJECT (enc, "nothing to drain");
goto done;
}
g_assert (size == bytes);
outbuf = gst_buffer_new_and_alloc (enc->max_payload_size * enc->n_channels);
if (!outbuf)
goto done;
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
enc->frame_samples, (int) bytes);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
enc->frame_samples, (int) bytes);
outsize =
opus_multistream_encode (enc->state, (const gint16 *) data,
enc->frame_samples, omap.data, enc->max_payload_size * enc->n_channels);
gst_buffer_unmap (outbuf, &omap);
if (outsize < 0) {
GST_ERROR_OBJECT (enc, "Encoding failed: %d", outsize);
ret = GST_FLOW_ERROR;
goto done;
} else if (outsize > enc->max_payload_size) {
GST_WARNING_OBJECT (enc,
"Encoded size %d is higher than max payload size (%d bytes)",
outsize, enc->max_payload_size);
ret = GST_FLOW_ERROR;
goto done;
}
GST_DEBUG_OBJECT (enc, "Output packet is %u bytes", outsize);
gst_buffer_set_size (outbuf, outsize);
ret =
gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), outbuf,
enc->frame_samples);
done:
if (bdata)
gst_buffer_unmap (buf, &map);
g_mutex_unlock (enc->property_lock);
if (mdata)
g_free (mdata);
return ret;
}
static GstFlowReturn
gst_celt_enc_encode (GstCeltEnc * enc, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
gint frame_size = enc->frame_size;
gint bytes = frame_size * 2 * enc->channels;
gint bytes_per_packet;
gint16 *data, *data0 = NULL;
gint outsize, size;
GstBuffer *outbuf;
if (G_LIKELY (buf)) {
data = (gint16 *) GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
if (G_UNLIKELY (size % bytes)) {
GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
size = ((size / bytes) + 1) * bytes;
data0 = data = g_malloc0 (size);
memcpy (data, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
}
} else {
GST_DEBUG_OBJECT (enc, "nothing to drain");
goto done;
}
frame_size = size / (2 * enc->channels);
if (enc->cbr) {
bytes_per_packet = (enc->bitrate * frame_size / enc->rate + 4) / 8;
} else {
bytes_per_packet = (enc->max_bitrate * frame_size / enc->rate + 4) / 8;
}
ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
GST_BUFFER_OFFSET_NONE, bytes_per_packet,
GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
if (GST_FLOW_OK != ret)
goto done;
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)", frame_size, bytes);
#ifdef HAVE_CELT_0_8
outsize =
celt_encode (enc->state, data, frame_size,
GST_BUFFER_DATA (outbuf), bytes_per_packet);
#else
outsize =
celt_encode (enc->state, data, NULL,
GST_BUFFER_DATA (outbuf), bytes_per_packet);
#endif
if (outsize < 0) {
GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
("encoding failed: %d", outsize));
ret = GST_FLOW_ERROR;
goto done;
}
GST_DEBUG_OBJECT (enc, "encoding %d bytes", bytes);
ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc),
outbuf, frame_size);
done:
g_free (data0);
return ret;
}
static GstFlowReturn
gst_two_lame_handle_frame (GstAudioEncoder * enc, GstBuffer * buf)
{
GstTwoLame *twolame;
gint mp3_buffer_size, mp3_size;
GstBuffer *mp3_buf;
GstFlowReturn result;
gint num_samples;
GstMapInfo map, mp3_map;
twolame = GST_TWO_LAME (enc);
/* squeeze remaining and push */
if (G_UNLIKELY (buf == NULL))
return gst_two_lame_flush_full (twolame, TRUE);
gst_buffer_map (buf, &map, GST_MAP_READ);
if (twolame->float_input)
num_samples = map.size / 4;
else
num_samples = map.size / 2;
/* allocate space for output */
mp3_buffer_size = 1.25 * num_samples + 16384;
mp3_buf = gst_buffer_new_and_alloc (mp3_buffer_size);
gst_buffer_map (mp3_buf, &mp3_map, GST_MAP_WRITE);
if (twolame->num_channels == 1) {
if (twolame->float_input)
mp3_size = twolame_encode_buffer_float32 (twolame->glopts,
(float *) map.data,
(float *) map.data, num_samples, mp3_map.data, mp3_buffer_size);
else
mp3_size = twolame_encode_buffer (twolame->glopts,
(short int *) map.data,
(short int *) map.data, num_samples, mp3_map.data, mp3_buffer_size);
} else {
if (twolame->float_input)
mp3_size = twolame_encode_buffer_float32_interleaved (twolame->glopts,
(float *) map.data,
num_samples / twolame->num_channels, mp3_map.data, mp3_buffer_size);
else
mp3_size = twolame_encode_buffer_interleaved (twolame->glopts,
(short int *) map.data,
num_samples / twolame->num_channels, mp3_map.data, mp3_buffer_size);
}
GST_LOG_OBJECT (twolame, "encoded %" G_GSIZE_FORMAT " bytes of audio "
"to %d bytes of mp3", map.size, mp3_size);
gst_buffer_unmap (buf, &map);
gst_buffer_unmap (mp3_buf, &mp3_map);
if (mp3_size > 0) {
gst_buffer_set_size (mp3_buf, mp3_size);
result = gst_audio_encoder_finish_frame (enc, mp3_buf, -1);
} else {
if (mp3_size < 0) {
/* eat error ? */
g_warning ("error %d", mp3_size);
}
gst_buffer_unref (mp3_buf);
result = GST_FLOW_OK;
}
return result;
}
static GstFlowReturn
gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf)
{
gint frame_size = enc->frame_size;
gint bytes = frame_size * 2 * enc->channels, samples, size;
gint outsize, written, dtx_ret = 0;
guint8 *data;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
if (G_LIKELY (buf)) {
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
if (G_UNLIKELY (size % bytes)) {
GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
size = ((size / bytes) + 1) * bytes;
data = g_malloc0 (size);
memcpy (data, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
}
} else {
GST_DEBUG_OBJECT (enc, "nothing to drain");
goto done;
}
samples = size / (2 * enc->channels);
speex_bits_reset (&enc->bits);
/* FIXME what about dropped samples if DTS enabled ?? */
while (size) {
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)", frame_size, bytes);
if (enc->channels == 2) {
speex_encode_stereo_int ((gint16 *) data, frame_size, &enc->bits);
}
dtx_ret += speex_encode_int (enc->state, (gint16 *) data, &enc->bits);
data += bytes;
size -= bytes;
}
speex_bits_insert_terminator (&enc->bits);
outsize = speex_bits_nbytes (&enc->bits);
ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
GST_BUFFER_OFFSET_NONE, outsize,
GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
if ((GST_FLOW_OK != ret))
goto done;
written = speex_bits_write (&enc->bits,
(gchar *) GST_BUFFER_DATA (outbuf), outsize);
if (G_UNLIKELY (written < outsize)) {
GST_ERROR_OBJECT (enc, "short write: %d < %d bytes", written, outsize);
GST_BUFFER_SIZE (outbuf) = written;
} else if (G_UNLIKELY (written > outsize)) {
GST_ERROR_OBJECT (enc, "overrun: %d > %d bytes", written, outsize);
}
if (!dtx_ret)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc),
outbuf, samples);
done:
return ret;
}
static GstFlowReturn
gst_fdkaacenc_handle_frame (GstAudioEncoder * enc, GstBuffer * inbuf)
{
GstFdkAacEnc *self = GST_FDKAACENC (enc);
GstFlowReturn ret = GST_FLOW_OK;
GstAudioInfo *info;
GstMapInfo imap, omap;
GstBuffer *outbuf;
AACENC_BufDesc in_desc = { 0 };
AACENC_BufDesc out_desc = { 0 };
AACENC_InArgs in_args = { 0 };
AACENC_OutArgs out_args = { 0 };
gint in_id = IN_AUDIO_DATA, out_id = OUT_BITSTREAM_DATA;
gint in_sizes, out_sizes;
gint in_el_sizes, out_el_sizes;
AACENC_ERROR err;
info = gst_audio_encoder_get_audio_info (enc);
if (inbuf) {
if (self->need_reorder) {
inbuf = gst_buffer_copy (inbuf);
gst_buffer_map (inbuf, &imap, GST_MAP_READWRITE);
gst_audio_reorder_channels (imap.data, imap.size,
GST_AUDIO_INFO_FORMAT (info), GST_AUDIO_INFO_CHANNELS (info),
&GST_AUDIO_INFO_POSITION (info, 0), self->aac_positions);
} else {
gst_buffer_map (inbuf, &imap, GST_MAP_READ);
}
in_args.numInSamples = imap.size / GST_AUDIO_INFO_BPS (info);
in_sizes = imap.size;
in_el_sizes = GST_AUDIO_INFO_BPS (info);
in_desc.numBufs = 1;
} else {
in_args.numInSamples = -1;
in_sizes = 0;
in_el_sizes = 0;
in_desc.numBufs = 0;
}
in_desc.bufferIdentifiers = &in_id;
in_desc.bufs = (void *) &imap.data;
in_desc.bufSizes = &in_sizes;
in_desc.bufElSizes = &in_el_sizes;
outbuf = gst_audio_encoder_allocate_output_buffer (enc, self->outbuf_size);
if (!outbuf) {
ret = GST_FLOW_ERROR;
goto out;
}
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_sizes = omap.size;
out_el_sizes = 1;
out_desc.bufferIdentifiers = &out_id;
out_desc.numBufs = 1;
out_desc.bufs = (void *) &omap.data;
out_desc.bufSizes = &out_sizes;
out_desc.bufElSizes = &out_el_sizes;
err = aacEncEncode (self->enc, &in_desc, &out_desc, &in_args, &out_args);
if (err == AACENC_ENCODE_EOF && !inbuf)
goto out;
else if (err != AACENC_OK) {
GST_ERROR_OBJECT (self, "Failed to encode data: %d", err);
ret = GST_FLOW_ERROR;
goto out;
}
if (inbuf) {
gst_buffer_unmap (inbuf, &imap);
if (self->need_reorder)
gst_buffer_unref (inbuf);
inbuf = NULL;
}
if (!out_args.numOutBytes)
goto out;
gst_buffer_unmap (outbuf, &omap);
gst_buffer_set_size (outbuf, out_args.numOutBytes);
ret = gst_audio_encoder_finish_frame (enc, outbuf, self->samples_per_frame);
outbuf = NULL;
out:
if (outbuf) {
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unref (outbuf);
}
if (inbuf) {
gst_buffer_unmap (inbuf, &imap);
if (self->need_reorder)
gst_buffer_unref (inbuf);
}
return ret;
}
static GstFlowReturn
gst_siren_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstSirenEnc *enc;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
guint8 *in_data, *out_data;
guint i, size, num_frames;
gint out_size, in_size;
gint encode_ret;
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
enc = GST_SIREN_ENC (benc);
size = GST_BUFFER_SIZE (buf);
GST_LOG_OBJECT (enc, "Received buffer of size %d", GST_BUFFER_SIZE (buf));
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
g_return_val_if_fail (size % 640 == 0, GST_FLOW_ERROR);
/* we need to process 640 input bytes to produce 40 output bytes */
/* calculate the amount of frames we will handle */
num_frames = size / 640;
/* this is the input/output size */
in_size = num_frames * 640;
out_size = num_frames * 40;
GST_LOG_OBJECT (enc, "we have %u frames, %u in, %u out", num_frames, in_size,
out_size);
/* get a buffer */
ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (benc),
-1, out_size, GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (benc)), &out_buf);
if (ret != GST_FLOW_OK)
goto alloc_failed;
/* get the input data for all the frames */
in_data = GST_BUFFER_DATA (buf);
out_data = GST_BUFFER_DATA (out_buf);
for (i = 0; i < num_frames; i++) {
GST_LOG_OBJECT (enc, "Encoding frame %u/%u", i, num_frames);
/* encode 640 input bytes to 40 output bytes */
encode_ret = Siren7_EncodeFrame (enc->encoder, in_data, out_data);
if (encode_ret != 0)
goto encode_error;
/* move to next frame */
out_data += 40;
in_data += 640;
}
GST_LOG_OBJECT (enc, "Finished encoding");
/* we encode all we get, pass it along */
ret = gst_audio_encoder_finish_frame (benc, out_buf, -1);
done:
return ret;
/* ERRORS */
alloc_failed:
{
GST_DEBUG_OBJECT (enc, "failed to pad_alloc buffer: %d (%s)", ret,
gst_flow_get_name (ret));
goto done;
}
encode_error:
{
GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
("Error encoding frame: %d", encode_ret));
ret = GST_FLOW_ERROR;
gst_buffer_unref (out_buf);
goto done;
}
}
static GstFlowReturn
gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf)
{
gint frame_size = enc->frame_size;
gint bytes = frame_size * 2 * enc->channels, samples;
gint outsize, written, dtx_ret = 0;
GstMapInfo map;
guint8 *data, *data0 = NULL, *bdata;
gsize bsize, size;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
if (G_LIKELY (buf)) {
gst_buffer_map (buf, &map, GST_MAP_READ);
bdata = map.data;
bsize = map.size;
if (G_UNLIKELY (bsize % bytes)) {
GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
size = ((bsize / bytes) + 1) * bytes;
data0 = data = g_malloc0 (size);
memcpy (data, bdata, bsize);
gst_buffer_unmap (buf, &map);
bdata = NULL;
} else {
data = bdata;
size = bsize;
}
} else {
GST_DEBUG_OBJECT (enc, "nothing to drain");
goto done;
}
samples = size / (2 * enc->channels);
speex_bits_reset (&enc->bits);
/* FIXME what about dropped samples if DTS enabled ?? */
while (size) {
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)", frame_size, bytes);
if (enc->channels == 2) {
speex_encode_stereo_int ((gint16 *) data, frame_size, &enc->bits);
}
dtx_ret += speex_encode_int (enc->state, (gint16 *) data, &enc->bits);
data += bytes;
size -= bytes;
}
speex_bits_insert_terminator (&enc->bits);
outsize = speex_bits_nbytes (&enc->bits);
if (bdata)
gst_buffer_unmap (buf, &map);
#if 0
ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
GST_BUFFER_OFFSET_NONE, outsize,
GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
if ((GST_FLOW_OK != ret))
goto done;
#endif
outbuf = gst_buffer_new_allocate (NULL, outsize, NULL);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
written = speex_bits_write (&enc->bits, (gchar *) map.data, outsize);
if (G_UNLIKELY (written < outsize)) {
GST_ERROR_OBJECT (enc, "short write: %d < %d bytes", written, outsize);
} else if (G_UNLIKELY (written > outsize)) {
GST_ERROR_OBJECT (enc, "overrun: %d > %d bytes", written, outsize);
written = outsize;
}
gst_buffer_unmap (outbuf, &map);
gst_buffer_resize (outbuf, 0, written);
if (!dtx_ret)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc),
outbuf, samples);
done:
g_free (data0);
return ret;
}
static GstFlowReturn
gst_sbc_enc_handle_frame (GstAudioEncoder * audio_enc, GstBuffer * buffer)
{
GstSbcEnc *enc = GST_SBC_ENC (audio_enc);
GstMapInfo in_map, out_map;
GstBuffer *outbuf = NULL;
guint samples_per_frame, frames, i = 0;
/* no fancy draining */
if (buffer == NULL)
return GST_FLOW_OK;
if (G_UNLIKELY (enc->channels == 0 || enc->blocks == 0 || enc->subbands == 0))
return GST_FLOW_NOT_NEGOTIATED;
samples_per_frame = enc->channels * enc->blocks * enc->subbands;
if (!gst_buffer_map (buffer, &in_map, GST_MAP_READ))
goto map_failed;
frames = in_map.size / (samples_per_frame * sizeof (gint16));
GST_LOG_OBJECT (enc,
"encoding %" G_GSIZE_FORMAT " samples into %u SBC frames",
in_map.size / (enc->channels * sizeof (gint16)), frames);
if (frames > 0) {
gsize frame_len;
frame_len = sbc_get_frame_length (&enc->sbc);
outbuf = gst_audio_encoder_allocate_output_buffer (audio_enc,
frames * frame_len);
if (outbuf == NULL)
goto no_buffer;
gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE);
for (i = 0; i < frames; ++i) {
gssize ret, written = 0;
ret = sbc_encode (&enc->sbc, in_map.data + (i * samples_per_frame * 2),
samples_per_frame * 2, out_map.data + (i * frame_len), frame_len,
&written);
if (ret < 0 || written != frame_len) {
GST_WARNING_OBJECT (enc, "encoding error, ret = %" G_GSSIZE_FORMAT ", "
"written = %" G_GSSIZE_FORMAT, ret, written);
break;
}
}
gst_buffer_unmap (outbuf, &out_map);
if (i > 0)
gst_buffer_set_size (outbuf, i * frame_len);
else
gst_buffer_replace (&outbuf, NULL);
}
done:
gst_buffer_unmap (buffer, &in_map);
return gst_audio_encoder_finish_frame (audio_enc, outbuf,
i * (samples_per_frame / enc->channels));
/* ERRORS */
no_buffer:
{
GST_ERROR_OBJECT (enc, "could not allocate output buffer");
goto done;
}
map_failed:
{
GST_ERROR_OBJECT (enc, "could not map input buffer");
goto done;
}
}
static void
gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
{
GstOMXAudioEncClass *klass;
GstOMXPort *port = self->enc_out_port;
GstOMXBuffer *buf = NULL;
GstFlowReturn flow_ret = GST_FLOW_OK;
GstOMXAcquireBufferReturn acq_return;
OMX_ERRORTYPE err;
klass = GST_OMX_AUDIO_ENC_GET_CLASS (self);
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_ENCODER_SRC_PAD (self))
|| acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
GstAudioInfo *info =
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self));
GstCaps *caps;
GST_DEBUG_OBJECT (self, "Port settings have changed, updating caps");
/* Reallocate all buffers */
if (acq_return == GST_OMX_ACQUIRE_BUFFER_RECONFIGURE) {
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;
}
GST_AUDIO_ENCODER_STREAM_LOCK (self);
caps = klass->get_caps (self, self->enc_out_port, info);
if (!caps) {
if (buf)
gst_omx_port_release_buffer (self->enc_out_port, buf);
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed;
}
GST_DEBUG_OBJECT (self, "Setting output caps: %" GST_PTR_FORMAT, caps);
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
gst_caps_unref (caps);
if (buf)
gst_omx_port_release_buffer (self->enc_out_port, buf);
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed;
}
gst_caps_unref (caps);
GST_AUDIO_ENCODER_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_ENCODER_STREAM_LOCK (self);
goto eos;
}
GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %" G_GUINT64_FORMAT,
(guint) buf->omx_buf->nFlags, (guint64) buf->omx_buf->nTimeStamp);
/* This prevents a deadlock between the srcpad stream
* lock and the videocodec stream lock, if ::reset()
* is called at the wrong time
*/
if (gst_omx_port_is_flushing (self->enc_out_port)) {
GST_DEBUG_OBJECT (self, "Flushing");
gst_omx_port_release_buffer (self->enc_out_port, buf);
goto flushing;
}
GST_AUDIO_ENCODER_STREAM_LOCK (self);
if ((buf->omx_buf->nFlags & OMX_BUFFERFLAG_CODECCONFIG)
&& buf->omx_buf->nFilledLen > 0) {
GstCaps *caps;
GstBuffer *codec_data;
GstMapInfo map = GST_MAP_INFO_INIT;
GST_DEBUG_OBJECT (self, "Handling codec data");
caps =
gst_caps_copy (gst_pad_get_current_caps (GST_AUDIO_ENCODER_SRC_PAD
(self)));
codec_data = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
memcpy (map.data,
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
gst_buffer_unmap (codec_data, &map);
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL);
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
gst_caps_unref (caps);
if (buf)
gst_omx_port_release_buffer (self->enc_out_port, buf);
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
goto caps_failed;
}
gst_caps_unref (caps);
flow_ret = GST_FLOW_OK;
} else if (buf->omx_buf->nFilledLen > 0) {
GstBuffer *outbuf;
guint n_samples;
GST_DEBUG_OBJECT (self, "Handling output data");
n_samples =
klass->get_num_samples (self, self->enc_out_port,
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)), buf);
if (buf->omx_buf->nFilledLen > 0) {
GstMapInfo map = GST_MAP_INFO_INIT;
outbuf = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
memcpy (map.data,
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
buf->omx_buf->nFilledLen);
gst_buffer_unmap (outbuf, &map);
} else {
outbuf = gst_buffer_new ();
}
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_encoder_finish_frame (GST_AUDIO_ENCODER (self),
outbuf, n_samples);
}
GST_DEBUG_OBJECT (self, "Handled output data");
GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
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_ENCODER_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->enc),
gst_omx_component_get_last_error (self->enc)));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_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_ENCODER_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_ENCODER_SRC_PAD (self));
} else {
GST_DEBUG_OBJECT (self, "Component signalled EOS");
flow_ret = GST_FLOW_EOS;
}
g_mutex_unlock (&self->drain_lock);
GST_AUDIO_ENCODER_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_ENCODER_STREAM_UNLOCK (self);
return;
}
flow_error:
{
if (flow_ret == GST_FLOW_EOS) {
GST_DEBUG_OBJECT (self, "EOS");
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
} else if (flow_ret == GST_FLOW_NOT_LINKED || 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_ENCODER_SRC_PAD (self),
gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
}
self->started = FALSE;
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
return;
}
reconfigure_error:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL),
("Unable to reconfigure output port"));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_NOT_NEGOTIATED;
self->started = FALSE;
return;
}
caps_failed:
{
GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("Failed to set caps"));
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (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_ENCODER_SRC_PAD (self), gst_event_new_eos ());
gst_pad_pause_task (GST_AUDIO_ENCODER_SRC_PAD (self));
self->downstream_flow_ret = GST_FLOW_ERROR;
self->started = FALSE;
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
return;
}
}
static GstFlowReturn
gst_siren_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstSirenEnc *enc;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
guint8 *in_data, *out_data;
guint i, size, num_frames;
gint out_size;
#ifndef GST_DISABLE_GST_DEBUG
gint in_size;
#endif
gint encode_ret;
GstMapInfo inmap, outmap;
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
enc = GST_SIREN_ENC (benc);
size = gst_buffer_get_size (buf);
GST_LOG_OBJECT (enc, "Received buffer of size %d", size);
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
g_return_val_if_fail (size % 640 == 0, GST_FLOW_ERROR);
/* we need to process 640 input bytes to produce 40 output bytes */
/* calculate the amount of frames we will handle */
num_frames = size / 640;
/* this is the input/output size */
#ifndef GST_DISABLE_GST_DEBUG
in_size = num_frames * 640;
#endif
out_size = num_frames * 40;
GST_LOG_OBJECT (enc, "we have %u frames, %u in, %u out", num_frames, in_size,
out_size);
/* get a buffer */
out_buf = gst_audio_encoder_allocate_output_buffer (benc, out_size);
if (out_buf == NULL)
goto alloc_failed;
/* get the input data for all the frames */
gst_buffer_map (buf, &inmap, GST_MAP_READ);
gst_buffer_map (out_buf, &outmap, GST_MAP_READ);
in_data = inmap.data;
out_data = outmap.data;
for (i = 0; i < num_frames; i++) {
GST_LOG_OBJECT (enc, "Encoding frame %u/%u", i, num_frames);
/* encode 640 input bytes to 40 output bytes */
encode_ret = Siren7_EncodeFrame (enc->encoder, in_data, out_data);
if (encode_ret != 0)
goto encode_error;
/* move to next frame */
out_data += 40;
in_data += 640;
}
gst_buffer_unmap (buf, &inmap);
gst_buffer_unmap (out_buf, &outmap);
GST_LOG_OBJECT (enc, "Finished encoding");
/* we encode all we get, pass it along */
ret = gst_audio_encoder_finish_frame (benc, out_buf, -1);
done:
return ret;
/* ERRORS */
alloc_failed:
{
GST_DEBUG_OBJECT (enc, "failed to pad_alloc buffer: %d (%s)", ret,
gst_flow_get_name (ret));
goto done;
}
encode_error:
{
GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
("Error encoding frame: %d", encode_ret));
ret = GST_FLOW_ERROR;
gst_buffer_unref (out_buf);
goto done;
}
}
