mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 15:51:11 +00:00
faac: Port to the new raw audio caps
This commit is contained in:
Sebastian Dröge
parent
1cb6e68cd0
commit
ad63a0c6e3
2 changed files with 39 additions and 45 deletions
|
|
@ -38,7 +38,7 @@
|
|||
#include <stdlib.h>
|
||||
#include <string.h>
|
||||
|
||||
#include <gst/audio/multichannel.h>
|
||||
#include <gst/audio/audio.h>
|
||||
#include <gst/pbutils/codec-utils.h>
|
||||
|
||||
#include "gstfaac.h"
|
||||
|
|
@ -58,6 +58,7 @@
|
|||
#define SINK_CAPS \
|
||||
"audio/x-raw, " \
|
||||
"format = (string) "GST_AUDIO_NE (S16) ", " \
|
||||
"layout = (string) interleaved, " \
|
||||
"rate = (int) {" SAMPLE_RATES "}, " \
|
||||
"channels = (int) [ 1, 6 ] "
|
||||
|
||||
|
|
@ -277,7 +278,7 @@ gst_faac_stop (GstAudioEncoder * enc)
|
|||
}
|
||||
|
||||
static const GstAudioChannelPosition aac_channel_positions[][8] = {
|
||||
{GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
|
||||
{GST_AUDIO_CHANNEL_POSITION_MONO},
|
||||
{GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
||||
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
|
||||
{
|
||||
|
|
@ -302,7 +303,7 @@ static const GstAudioChannelPosition aac_channel_positions[][8] = {
|
|||
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
|
||||
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
|
||||
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
|
||||
GST_AUDIO_CHANNEL_POSITION_LFE}
|
||||
GST_AUDIO_CHANNEL_POSITION_LFE1}
|
||||
};
|
||||
|
||||
static GstCaps *
|
||||
|
|
@ -334,24 +335,19 @@ gst_faac_getcaps (GstAudioEncoder * enc, GstCaps * filter)
|
|||
gst_structure_set_value (s, "rate", &rates_arr);
|
||||
|
||||
for (i = 1; i <= 6; i++) {
|
||||
GValue chanpos = { 0 };
|
||||
GValue pos = { 0 };
|
||||
|
||||
guint64 channel_mask = 0;
|
||||
t = gst_structure_copy (s);
|
||||
|
||||
gst_structure_set (t, "channels", G_TYPE_INT, i, NULL);
|
||||
if (i == 1)
|
||||
continue;
|
||||
|
||||
g_value_init (&chanpos, GST_TYPE_ARRAY);
|
||||
g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
|
||||
for (c = 0; c < i; c++)
|
||||
channel_mask |=
|
||||
G_GUINT64_CONSTANT (1) << aac_channel_positions[i - 1][c];
|
||||
|
||||
for (c = 0; c < i; c++) {
|
||||
g_value_set_enum (&pos, aac_channel_positions[i - 1][c]);
|
||||
gst_value_array_append_value (&chanpos, &pos);
|
||||
}
|
||||
g_value_unset (&pos);
|
||||
|
||||
gst_structure_set_value (t, "channel-positions", &chanpos);
|
||||
g_value_unset (&chanpos);
|
||||
gst_structure_set (t, "channel-mask", GST_TYPE_BITMASK, channel_mask,
|
||||
NULL);
|
||||
gst_caps_append_structure (tmp, t);
|
||||
}
|
||||
gst_structure_free (s);
|
||||
|
|
@ -369,39 +365,30 @@ static gboolean
|
|||
gst_faac_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
|
||||
{
|
||||
GstFaac *faac = GST_FAAC (enc);
|
||||
gint channels, samplerate, width;
|
||||
gulong fmt = 0, bps = 0;
|
||||
gint width;
|
||||
gulong fmt = 0;
|
||||
gboolean result = FALSE;
|
||||
|
||||
/* base class takes care */
|
||||
channels = GST_AUDIO_INFO_CHANNELS (info);
|
||||
samplerate = GST_AUDIO_INFO_RATE (info);
|
||||
width = GST_AUDIO_INFO_WIDTH (info);
|
||||
|
||||
if (GST_AUDIO_INFO_IS_INTEGER (info)) {
|
||||
switch (width) {
|
||||
case 16:
|
||||
fmt = FAAC_INPUT_16BIT;
|
||||
bps = 2;
|
||||
break;
|
||||
case 24:
|
||||
case 32:
|
||||
fmt = FAAC_INPUT_32BIT;
|
||||
bps = 4;
|
||||
break;
|
||||
default:
|
||||
g_return_val_if_reached (FALSE);
|
||||
}
|
||||
} else {
|
||||
fmt = FAAC_INPUT_FLOAT;
|
||||
bps = 4;
|
||||
}
|
||||
|
||||
/* ok, record and set up */
|
||||
faac->format = fmt;
|
||||
faac->bps = bps;
|
||||
faac->channels = channels;
|
||||
faac->samplerate = samplerate;
|
||||
|
||||
/* finish up */
|
||||
result = gst_faac_configure_source_pad (faac);
|
||||
|
|
@ -482,18 +469,19 @@ gst_faac_open_encoder (GstFaac * faac)
|
|||
faacEncConfiguration *conf;
|
||||
guint maxbitrate;
|
||||
gulong samples, bytes;
|
||||
GstAudioInfo *info =
|
||||
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (faac));
|
||||
|
||||
g_return_val_if_fail (faac->samplerate != 0 && faac->channels != 0, FALSE);
|
||||
g_return_val_if_fail (info->rate != 0 && info->channels != 0, FALSE);
|
||||
|
||||
/* clean up in case of re-configure */
|
||||
gst_faac_close_encoder (faac);
|
||||
|
||||
if (!(handle = faacEncOpen (faac->samplerate, faac->channels,
|
||||
&samples, &bytes)))
|
||||
if (!(handle = faacEncOpen (info->rate, info->channels, &samples, &bytes)))
|
||||
goto setup_failed;
|
||||
|
||||
/* mind channel count */
|
||||
samples /= faac->channels;
|
||||
samples /= info->channels;
|
||||
|
||||
/* record */
|
||||
faac->handle = handle;
|
||||
|
|
@ -514,7 +502,7 @@ gst_faac_open_encoder (GstFaac * faac)
|
|||
if (faac->brtype == VBR) {
|
||||
conf->quantqual = faac->quality;
|
||||
} else if (faac->brtype == ABR) {
|
||||
conf->bitRate = faac->bitrate / faac->channels;
|
||||
conf->bitRate = faac->bitrate / info->channels;
|
||||
}
|
||||
|
||||
conf->inputFormat = faac->format;
|
||||
|
|
@ -524,13 +512,12 @@ gst_faac_open_encoder (GstFaac * faac)
|
|||
/* check, warn and correct if the max bitrate for the given samplerate is
|
||||
* exceeded. Maximum of 6144 bit for a channel */
|
||||
maxbitrate =
|
||||
(unsigned int) (6144.0 * (double) faac->samplerate / (double) 1024.0 +
|
||||
.5);
|
||||
(unsigned int) (6144.0 * (double) info->rate / (double) 1024.0 + .5);
|
||||
if (conf->bitRate > maxbitrate) {
|
||||
GST_ELEMENT_WARNING (faac, RESOURCE, SETTINGS, (NULL),
|
||||
("bitrate %lu exceeds maximum allowed bitrate of %u for samplerate %d. "
|
||||
"Setting bitrate to %u", conf->bitRate, maxbitrate,
|
||||
faac->samplerate, maxbitrate));
|
||||
info->rate, maxbitrate));
|
||||
conf->bitRate = maxbitrate;
|
||||
}
|
||||
|
||||
|
|
@ -543,7 +530,7 @@ gst_faac_open_encoder (GstFaac * faac)
|
|||
/* let's see what really happened,
|
||||
* note that this may not really match desired rate */
|
||||
GST_DEBUG_OBJECT (faac, "average bitrate: %lu kbps",
|
||||
(conf->bitRate + 500) / 1000 * faac->channels);
|
||||
(conf->bitRate + 500) / 1000 * info->channels);
|
||||
GST_DEBUG_OBJECT (faac, "quantization quality: %ld", conf->quantqual);
|
||||
GST_DEBUG_OBJECT (faac, "bandwidth: %d Hz", conf->bandWidth);
|
||||
|
||||
|
|
@ -562,6 +549,8 @@ gst_faac_configure_source_pad (GstFaac * faac)
|
|||
{
|
||||
GstCaps *srccaps;
|
||||
gboolean ret;
|
||||
GstAudioInfo *info =
|
||||
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (faac));
|
||||
|
||||
/* negotiate stream format */
|
||||
gst_faac_negotiate (faac);
|
||||
|
|
@ -572,8 +561,8 @@ gst_faac_configure_source_pad (GstFaac * faac)
|
|||
/* now create a caps for it all */
|
||||
srccaps = gst_caps_new_simple ("audio/mpeg",
|
||||
"mpegversion", G_TYPE_INT, faac->mpegversion,
|
||||
"channels", G_TYPE_INT, faac->channels,
|
||||
"rate", G_TYPE_INT, faac->samplerate,
|
||||
"channels", G_TYPE_INT, info->channels,
|
||||
"rate", G_TYPE_INT, info->rate,
|
||||
"stream-format", G_TYPE_STRING, (faac->outputformat ? "adts" : "raw"),
|
||||
NULL);
|
||||
|
||||
|
|
@ -661,11 +650,20 @@ gst_faac_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
|
|||
guint8 *data;
|
||||
guint8 *out_data;
|
||||
gsize out_size;
|
||||
GstAudioInfo *info =
|
||||
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (faac));
|
||||
|
||||
out_buf = gst_buffer_new_and_alloc (faac->bytes);
|
||||
out_data = gst_buffer_map (out_buf, &out_size, NULL, GST_MAP_WRITE);
|
||||
|
||||
if (G_LIKELY (in_buf)) {
|
||||
if (memcmp (info->position, aac_channel_positions[info->channels - 1],
|
||||
sizeof (GstAudioChannelPosition) * info->channels) != 0) {
|
||||
in_buf = gst_buffer_make_writable (in_buf);
|
||||
gst_audio_buffer_reorder_channels (in_buf, info->finfo->format,
|
||||
info->channels, info->position,
|
||||
aac_channel_positions[info->channels - 1]);
|
||||
}
|
||||
data = gst_buffer_map (in_buf, &size, NULL, GST_MAP_READ);
|
||||
} else {
|
||||
data = NULL;
|
||||
|
|
@ -673,7 +671,7 @@ gst_faac_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
|
|||
}
|
||||
|
||||
if (G_UNLIKELY ((ret_size = faacEncEncode (faac->handle, (gint32 *) data,
|
||||
size / faac->bps, out_data, out_size)) < 0))
|
||||
size / (info->finfo->width / 8), out_data, out_size)) < 0))
|
||||
goto encode_failed;
|
||||
|
||||
gst_buffer_unmap (in_buf, data, size);
|
||||
|
|
|
|||
|
|
@ -44,12 +44,8 @@ typedef struct _GstFaacClass GstFaacClass;
|
|||
struct _GstFaac {
|
||||
GstAudioEncoder element;
|
||||
|
||||
/* stream properties */
|
||||
gint samplerate,
|
||||
channels,
|
||||
format,
|
||||
bps;
|
||||
|
||||
/* input format */
|
||||
gint format;
|
||||
/* input frame size */
|
||||
gint samples;
|
||||
/* required output buffer size */
|
||||
|
|
|
|||
Loading…
Reference in a new issue